diff --git a/include/linux/usb/audio-v3.h b/include/linux/usb/audio-v3.h
index a8959aa..38add1d 100644
--- a/include/linux/usb/audio-v3.h
+++ b/include/linux/usb/audio-v3.h
@@ -392,4 +392,30 @@ struct uac3_interrupt_data_msg {
 #define UAC3_AC_ACTIVE_INTERFACE_CONTROL	0x01
 #define UAC3_AC_POWER_DOMAIN_CONTROL		0x02
 
+/* BADD predefined Unit/Terminal values */
+#define UAC3_BADD_IT_ID1	1  /* Input Terminal ID1: bTerminalID = 1 */
+#define UAC3_BADD_FU_ID2	2  /* Feature Unit ID2: bUnitID = 2 */
+#define UAC3_BADD_OT_ID3	3  /* Output Terminal ID3: bTerminalID = 3 */
+#define UAC3_BADD_IT_ID4	4  /* Input Terminal ID4: bTerminalID = 4 */
+#define UAC3_BADD_FU_ID5	5  /* Feature Unit ID5: bUnitID = 5 */
+#define UAC3_BADD_OT_ID6	6  /* Output Terminal ID6: bTerminalID = 6 */
+#define UAC3_BADD_FU_ID7	7  /* Feature Unit ID7: bUnitID = 7 */
+#define UAC3_BADD_MU_ID8	8  /* Mixer Unit ID8: bUnitID = 8 */
+#define UAC3_BADD_CS_ID9	9  /* Clock Source Entity ID9: bClockID = 9 */
+#define UAC3_BADD_PD_ID10	10 /* Power Domain ID10: bPowerDomainID = 10 */
+#define UAC3_BADD_PD_ID11	11 /* Power Domain ID11: bPowerDomainID = 11 */
+
+/* BADD wMaxPacketSize of AS endpoints */
+#define UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_16		0x0060
+#define UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_16		0x0062
+#define UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_24		0x0090
+#define UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_24		0x0093
+#define UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_16		0x00C0
+#define UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_16		0x00C4
+#define UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_24		0x0120
+#define UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_24		0x0126
+
+/* BADD sample rate is always fixed to 48kHz */
+#define UAC3_BADD_SAMPLING_RATE				48000
+
 #endif /* __LINUX_USB_AUDIO_V3_H */
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 5ebcc51..8c1572d 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1610,7 +1610,7 @@ struct snd_emu10k1_fx8010_pcm {
 	struct snd_pcm_indirect pcm_rec;
 	unsigned int tram_pos;
 	unsigned int tram_shift;
-	struct snd_emu10k1_fx8010_irq *irq;
+	struct snd_emu10k1_fx8010_irq irq;
 };
 
 struct snd_emu10k1_fx8010 {
@@ -1902,7 +1902,7 @@ int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu,
 					    snd_fx8010_irq_handler_t *handler,
 					    unsigned char gpr_running,
 					    void *private_data,
-					    struct snd_emu10k1_fx8010_irq **r_irq);
+					    struct snd_emu10k1_fx8010_irq *irq);
 int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu,
 					      struct snd_emu10k1_fx8010_irq *irq);
 
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index 06536e0..c052afc 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -571,4 +571,9 @@ static inline unsigned int snd_array_index(struct snd_array *array, void *ptr)
 	return (unsigned long)(ptr - array->list) / array->elem_size;
 }
 
+/* a helper macro to iterate for each snd_array element */
+#define snd_array_for_each(array, idx, ptr) \
+	for ((idx) = 0, (ptr) = (array)->list; (idx) < (array)->used; \
+	     (ptr) = snd_array_elem(array, ++(idx)))
+
 #endif /* __SOUND_HDAUDIO_H */
diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h
index be5371f..7d6d65f 100644
--- a/include/uapi/sound/tlv.h
+++ b/include/uapi/sound/tlv.h
@@ -42,6 +42,10 @@
 #define SNDRV_CTL_TLVD_LENGTH(...) \
 	((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
 
+/* Accessor offsets for TLV data items */
+#define SNDRV_CTL_TLVO_TYPE		0
+#define SNDRV_CTL_TLVO_LEN		1
+
 #define SNDRV_CTL_TLVD_CONTAINER_ITEM(...) \
 	SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
 #define SNDRV_CTL_TLVD_DECLARE_CONTAINER(name, ...) \
@@ -61,6 +65,10 @@
 		SNDRV_CTL_TLVD_DB_SCALE_ITEM(min, step, mute) \
 	}
 
+/* Accessor offsets for min, mute and step items in dB scale type TLV */
+#define SNDRV_CTL_TLVO_DB_SCALE_MIN		2
+#define SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP	3
+
 /* dB scale specified with min/max values instead of step */
 #define SNDRV_CTL_TLVD_DB_MINMAX_ITEM(min_dB, max_dB) \
 	SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
@@ -75,6 +83,10 @@
 		SNDRV_CTL_TLVD_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
 	}
 
+/* Accessor offsets for min, max items in db-minmax types of TLV. */
+#define SNDRV_CTL_TLVO_DB_MINMAX_MIN	2
+#define SNDRV_CTL_TLVO_DB_MINMAX_MAX	3
+
 /* linear volume between min_dB and max_dB (.01dB unit) */
 #define SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
 	SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
@@ -83,6 +95,10 @@
 		SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
 	}
 
+/* Accessor offsets for min, max items in db-linear type of TLV. */
+#define SNDRV_CTL_TLVO_DB_LINEAR_MIN	2
+#define SNDRV_CTL_TLVO_DB_LINEAR_MAX	3
+
 /* dB range container:
  * Items in dB range container must be ordered by their values and by their
  * dB values. This implies that larger values must correspond with larger
diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c
index 6491afb..39d853b 100644
--- a/sound/core/pcm_compat.c
+++ b/sound/core/pcm_compat.c
@@ -45,10 +45,7 @@ static int snd_pcm_ioctl_rewind_compat(struct snd_pcm_substream *substream,
 
 	if (get_user(frames, src))
 		return -EFAULT;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		err = snd_pcm_playback_rewind(substream, frames);
-	else
-		err = snd_pcm_capture_rewind(substream, frames);
+	err = snd_pcm_rewind(substream, frames);
 	if (put_user(err, src))
 		return -EFAULT;
 	return err < 0 ? err : 0;
@@ -62,10 +59,7 @@ static int snd_pcm_ioctl_forward_compat(struct snd_pcm_substream *substream,
 
 	if (get_user(frames, src))
 		return -EFAULT;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		err = snd_pcm_playback_forward(substream, frames);
-	else
-		err = snd_pcm_capture_forward(substream, frames);
+	err = snd_pcm_forward(substream, frames);
 	if (put_user(err, src))
 		return -EFAULT;
 	return err < 0 ? err : 0;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index f4a1950..44b5ae8 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -191,10 +191,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream,
 {
 	snd_pcm_uframes_t avail;
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		avail = snd_pcm_playback_avail(runtime);
-	else
-		avail = snd_pcm_capture_avail(runtime);
+	avail = snd_pcm_avail(substream);
 	if (avail > runtime->avail_max)
 		runtime->avail_max = avail;
 	if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
@@ -1856,10 +1853,7 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 		 * This check must happen after been added to the waitqueue
 		 * and having current state be INTERRUPTIBLE.
 		 */
-		if (is_playback)
-			avail = snd_pcm_playback_avail(runtime);
-		else
-			avail = snd_pcm_capture_avail(runtime);
+		avail = snd_pcm_avail(substream);
 		if (avail >= runtime->twake)
 			break;
 		snd_pcm_stream_unlock_irq(substream);
@@ -2175,10 +2169,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
 	runtime->twake = runtime->control->avail_min ? : 1;
 	if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
 		snd_pcm_update_hw_ptr(substream);
-	if (is_playback)
-		avail = snd_pcm_playback_avail(runtime);
-	else
-		avail = snd_pcm_capture_avail(runtime);
+	avail = snd_pcm_avail(substream);
 	while (size > 0) {
 		snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
 		snd_pcm_uframes_t cont;
diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h
index 16f2547..7a499d0 100644
--- a/sound/core/pcm_local.h
+++ b/sound/core/pcm_local.h
@@ -36,6 +36,24 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream);
 void snd_pcm_playback_silence(struct snd_pcm_substream *substream,
 			      snd_pcm_uframes_t new_hw_ptr);
 
+static inline snd_pcm_uframes_t
+snd_pcm_avail(struct snd_pcm_substream *substream)
+{
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return snd_pcm_playback_avail(substream->runtime);
+	else
+		return snd_pcm_capture_avail(substream->runtime);
+}
+
+static inline snd_pcm_uframes_t
+snd_pcm_hw_avail(struct snd_pcm_substream *substream)
+{
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return snd_pcm_playback_hw_avail(substream->runtime);
+	else
+		return snd_pcm_capture_hw_avail(substream->runtime);
+}
+
 #ifdef CONFIG_SND_PCM_TIMER
 void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream);
 void snd_pcm_timer_init(struct snd_pcm_substream *substream);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 0e875d5..04c6301 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -99,6 +99,57 @@ static inline void down_write_nonblock(struct rw_semaphore *lock)
 		cond_resched();
 }
 
+#define PCM_LOCK_DEFAULT	0
+#define PCM_LOCK_IRQ	1
+#define PCM_LOCK_IRQSAVE	2
+
+static unsigned long __snd_pcm_stream_lock_mode(struct snd_pcm_substream *substream,
+						unsigned int mode)
+{
+	unsigned long flags = 0;
+	if (substream->pcm->nonatomic) {
+		down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING);
+		mutex_lock(&substream->self_group.mutex);
+	} else {
+		switch (mode) {
+		case PCM_LOCK_DEFAULT:
+			read_lock(&snd_pcm_link_rwlock);
+			break;
+		case PCM_LOCK_IRQ:
+			read_lock_irq(&snd_pcm_link_rwlock);
+			break;
+		case PCM_LOCK_IRQSAVE:
+			read_lock_irqsave(&snd_pcm_link_rwlock, flags);
+			break;
+		}
+		spin_lock(&substream->self_group.lock);
+	}
+	return flags;
+}
+
+static void __snd_pcm_stream_unlock_mode(struct snd_pcm_substream *substream,
+					 unsigned int mode, unsigned long flags)
+{
+	if (substream->pcm->nonatomic) {
+		mutex_unlock(&substream->self_group.mutex);
+		up_read(&snd_pcm_link_rwsem);
+	} else {
+		spin_unlock(&substream->self_group.lock);
+
+		switch (mode) {
+		case PCM_LOCK_DEFAULT:
+			read_unlock(&snd_pcm_link_rwlock);
+			break;
+		case PCM_LOCK_IRQ:
+			read_unlock_irq(&snd_pcm_link_rwlock);
+			break;
+		case PCM_LOCK_IRQSAVE:
+			read_unlock_irqrestore(&snd_pcm_link_rwlock, flags);
+			break;
+		}
+	}
+}
+
 /**
  * snd_pcm_stream_lock - Lock the PCM stream
  * @substream: PCM substream
@@ -109,13 +160,7 @@ static inline void down_write_nonblock(struct rw_semaphore *lock)
  */
 void snd_pcm_stream_lock(struct snd_pcm_substream *substream)
 {
-	if (substream->pcm->nonatomic) {
-		down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING);
-		mutex_lock(&substream->self_group.mutex);
-	} else {
-		read_lock(&snd_pcm_link_rwlock);
-		spin_lock(&substream->self_group.lock);
-	}
+	__snd_pcm_stream_lock_mode(substream, PCM_LOCK_DEFAULT);
 }
 EXPORT_SYMBOL_GPL(snd_pcm_stream_lock);
 
@@ -127,13 +172,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_lock);
  */
 void snd_pcm_stream_unlock(struct snd_pcm_substream *substream)
 {
-	if (substream->pcm->nonatomic) {
-		mutex_unlock(&substream->self_group.mutex);
-		up_read(&snd_pcm_link_rwsem);
-	} else {
-		spin_unlock(&substream->self_group.lock);
-		read_unlock(&snd_pcm_link_rwlock);
-	}
+	__snd_pcm_stream_unlock_mode(substream, PCM_LOCK_DEFAULT, 0);
 }
 EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock);
 
@@ -147,9 +186,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock);
  */
 void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream)
 {
-	if (!substream->pcm->nonatomic)
-		local_irq_disable();
-	snd_pcm_stream_lock(substream);
+	__snd_pcm_stream_lock_mode(substream, PCM_LOCK_IRQ);
 }
 EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq);
 
@@ -161,19 +198,13 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq);
  */
 void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream)
 {
-	snd_pcm_stream_unlock(substream);
-	if (!substream->pcm->nonatomic)
-		local_irq_enable();
+	__snd_pcm_stream_unlock_mode(substream, PCM_LOCK_IRQ, 0);
 }
 EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irq);
 
 unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream)
 {
-	unsigned long flags = 0;
-	if (!substream->pcm->nonatomic)
-		local_irq_save(flags);
-	snd_pcm_stream_lock(substream);
-	return flags;
+	return __snd_pcm_stream_lock_mode(substream, PCM_LOCK_IRQSAVE);
 }
 EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave);
 
@@ -187,9 +218,7 @@ EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave);
 void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream,
 				      unsigned long flags)
 {
-	snd_pcm_stream_unlock(substream);
-	if (!substream->pcm->nonatomic)
-		local_irq_restore(flags);
+	__snd_pcm_stream_unlock_mode(substream, PCM_LOCK_IRQSAVE, flags);
 }
 EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irqrestore);
 
@@ -857,6 +886,18 @@ static int snd_pcm_sw_params_user(struct snd_pcm_substream *substream,
 	return err;
 }
 
+static inline snd_pcm_uframes_t
+snd_pcm_calc_delay(struct snd_pcm_substream *substream)
+{
+	snd_pcm_uframes_t delay;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		delay = snd_pcm_playback_hw_avail(substream->runtime);
+	else
+		delay = snd_pcm_capture_avail(substream->runtime);
+	return delay + substream->runtime->delay;
+}
+
 int snd_pcm_status(struct snd_pcm_substream *substream,
 		   struct snd_pcm_status *status)
 {
@@ -908,21 +949,9 @@ int snd_pcm_status(struct snd_pcm_substream *substream,
  _tstamp_end:
 	status->appl_ptr = runtime->control->appl_ptr;
 	status->hw_ptr = runtime->status->hw_ptr;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		status->avail = snd_pcm_playback_avail(runtime);
-		if (runtime->status->state == SNDRV_PCM_STATE_RUNNING ||
-		    runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
-			status->delay = runtime->buffer_size - status->avail;
-			status->delay += runtime->delay;
-		} else
-			status->delay = 0;
-	} else {
-		status->avail = snd_pcm_capture_avail(runtime);
-		if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
-			status->delay = status->avail + runtime->delay;
-		else
-			status->delay = 0;
-	}
+	status->avail = snd_pcm_avail(substream);
+	status->delay = snd_pcm_running(substream) ?
+		snd_pcm_calc_delay(substream) : 0;
 	status->avail_max = runtime->avail_max;
 	status->overrange = runtime->overrange;
 	runtime->avail_max = 0;
@@ -2610,10 +2639,9 @@ static snd_pcm_sframes_t rewind_appl_ptr(struct snd_pcm_substream *substream,
 	return ret < 0 ? 0 : frames;
 }
 
-static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *substream,
-						 snd_pcm_uframes_t frames)
+static snd_pcm_sframes_t snd_pcm_rewind(struct snd_pcm_substream *substream,
+					snd_pcm_uframes_t frames)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_sframes_t ret;
 
 	if (frames == 0)
@@ -2623,33 +2651,14 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst
 	ret = do_pcm_hwsync(substream);
 	if (!ret)
 		ret = rewind_appl_ptr(substream, frames,
-				      snd_pcm_playback_hw_avail(runtime));
+				      snd_pcm_hw_avail(substream));
 	snd_pcm_stream_unlock_irq(substream);
 	return ret;
 }
 
-static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substream,
-						snd_pcm_uframes_t frames)
+static snd_pcm_sframes_t snd_pcm_forward(struct snd_pcm_substream *substream,
+					 snd_pcm_uframes_t frames)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	snd_pcm_sframes_t ret;
-
-	if (frames == 0)
-		return 0;
-
-	snd_pcm_stream_lock_irq(substream);
-	ret = do_pcm_hwsync(substream);
-	if (!ret)
-		ret = rewind_appl_ptr(substream, frames,
-				      snd_pcm_capture_hw_avail(runtime));
-	snd_pcm_stream_unlock_irq(substream);
-	return ret;
-}
-
-static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *substream,
-						  snd_pcm_uframes_t frames)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_sframes_t ret;
 
 	if (frames == 0)
@@ -2659,25 +2668,7 @@ static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *subs
 	ret = do_pcm_hwsync(substream);
 	if (!ret)
 		ret = forward_appl_ptr(substream, frames,
-				       snd_pcm_playback_avail(runtime));
-	snd_pcm_stream_unlock_irq(substream);
-	return ret;
-}
-
-static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *substream,
-						 snd_pcm_uframes_t frames)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	snd_pcm_sframes_t ret;
-
-	if (frames == 0)
-		return 0;
-
-	snd_pcm_stream_lock_irq(substream);
-	ret = do_pcm_hwsync(substream);
-	if (!ret)
-		ret = forward_appl_ptr(substream, frames,
-				       snd_pcm_capture_avail(runtime));
+				       snd_pcm_avail(substream));
 	snd_pcm_stream_unlock_irq(substream);
 	return ret;
 }
@@ -2695,19 +2686,13 @@ static int snd_pcm_hwsync(struct snd_pcm_substream *substream)
 static int snd_pcm_delay(struct snd_pcm_substream *substream,
 			 snd_pcm_sframes_t *delay)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	int err;
 	snd_pcm_sframes_t n = 0;
 
 	snd_pcm_stream_lock_irq(substream);
 	err = do_pcm_hwsync(substream);
-	if (!err) {
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			n = snd_pcm_playback_hw_avail(runtime);
-		else
-			n = snd_pcm_capture_avail(runtime);
-		n += runtime->delay;
-	}
+	if (!err)
+		n = snd_pcm_calc_delay(substream);
 	snd_pcm_stream_unlock_irq(substream);
 	if (!err)
 		*delay = n;
@@ -2834,10 +2819,7 @@ static int snd_pcm_rewind_ioctl(struct snd_pcm_substream *substream,
 		return -EFAULT;
 	if (put_user(0, _frames))
 		return -EFAULT;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		result = snd_pcm_playback_rewind(substream, frames);
-	else
-		result = snd_pcm_capture_rewind(substream, frames);
+	result = snd_pcm_rewind(substream, frames);
 	__put_user(result, _frames);
 	return result < 0 ? result : 0;
 }
@@ -2852,10 +2834,7 @@ static int snd_pcm_forward_ioctl(struct snd_pcm_substream *substream,
 		return -EFAULT;
 	if (put_user(0, _frames))
 		return -EFAULT;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		result = snd_pcm_playback_forward(substream, frames);
-	else
-		result = snd_pcm_capture_forward(substream, frames);
+	result = snd_pcm_forward(substream, frames);
 	__put_user(result, _frames);
 	return result < 0 ? result : 0;
 }
@@ -2998,7 +2977,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream,
 		/* provided only for OSS; capture-only and no value returned */
 		if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
 			return -EINVAL;
-		result = snd_pcm_capture_forward(substream, *frames);
+		result = snd_pcm_forward(substream, *frames);
 		return result < 0 ? result : 0;
 	}
 	case SNDRV_PCM_IOCTL_HW_PARAMS:
@@ -3140,82 +3119,46 @@ static ssize_t snd_pcm_writev(struct kiocb *iocb, struct iov_iter *from)
 	return result;
 }
 
-static __poll_t snd_pcm_playback_poll(struct file *file, poll_table * wait)
+static __poll_t snd_pcm_poll(struct file *file, poll_table *wait)
 {
 	struct snd_pcm_file *pcm_file;
 	struct snd_pcm_substream *substream;
 	struct snd_pcm_runtime *runtime;
-        __poll_t mask;
+	__poll_t mask, ok;
 	snd_pcm_uframes_t avail;
 
 	pcm_file = file->private_data;
 
 	substream = pcm_file->substream;
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		ok = EPOLLOUT | EPOLLWRNORM;
+	else
+		ok = EPOLLIN | EPOLLRDNORM;
 	if (PCM_RUNTIME_CHECK(substream))
-		return EPOLLOUT | EPOLLWRNORM | EPOLLERR;
-	runtime = substream->runtime;
+		return ok | EPOLLERR;
 
+	runtime = substream->runtime;
 	poll_wait(file, &runtime->sleep, wait);
 
+	mask = 0;
 	snd_pcm_stream_lock_irq(substream);
-	avail = snd_pcm_playback_avail(runtime);
+	avail = snd_pcm_avail(substream);
 	switch (runtime->status->state) {
 	case SNDRV_PCM_STATE_RUNNING:
 	case SNDRV_PCM_STATE_PREPARED:
 	case SNDRV_PCM_STATE_PAUSED:
-		if (avail >= runtime->control->avail_min) {
-			mask = EPOLLOUT | EPOLLWRNORM;
-			break;
-		}
-		/* Fall through */
-	case SNDRV_PCM_STATE_DRAINING:
-		mask = 0;
-		break;
-	default:
-		mask = EPOLLOUT | EPOLLWRNORM | EPOLLERR;
-		break;
-	}
-	snd_pcm_stream_unlock_irq(substream);
-	return mask;
-}
-
-static __poll_t snd_pcm_capture_poll(struct file *file, poll_table * wait)
-{
-	struct snd_pcm_file *pcm_file;
-	struct snd_pcm_substream *substream;
-	struct snd_pcm_runtime *runtime;
-        __poll_t mask;
-	snd_pcm_uframes_t avail;
-
-	pcm_file = file->private_data;
-
-	substream = pcm_file->substream;
-	if (PCM_RUNTIME_CHECK(substream))
-		return EPOLLIN | EPOLLRDNORM | EPOLLERR;
-	runtime = substream->runtime;
-
-	poll_wait(file, &runtime->sleep, wait);
-
-	snd_pcm_stream_lock_irq(substream);
-	avail = snd_pcm_capture_avail(runtime);
-	switch (runtime->status->state) {
-	case SNDRV_PCM_STATE_RUNNING:
-	case SNDRV_PCM_STATE_PREPARED:
-	case SNDRV_PCM_STATE_PAUSED:
-		if (avail >= runtime->control->avail_min) {
-			mask = EPOLLIN | EPOLLRDNORM;
-			break;
-		}
-		mask = 0;
+		if (avail >= runtime->control->avail_min)
+			mask = ok;
 		break;
 	case SNDRV_PCM_STATE_DRAINING:
-		if (avail > 0) {
-			mask = EPOLLIN | EPOLLRDNORM;
-			break;
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+			mask = ok;
+			if (!avail)
+				mask |= EPOLLERR;
 		}
-		/* Fall through */
+		break;
 	default:
-		mask = EPOLLIN | EPOLLRDNORM | EPOLLERR;
+		mask = ok | EPOLLERR;
 		break;
 	}
 	snd_pcm_stream_unlock_irq(substream);
@@ -3707,7 +3650,7 @@ const struct file_operations snd_pcm_f_ops[2] = {
 		.open =			snd_pcm_playback_open,
 		.release =		snd_pcm_release,
 		.llseek =		no_llseek,
-		.poll =			snd_pcm_playback_poll,
+		.poll =			snd_pcm_poll,
 		.unlocked_ioctl =	snd_pcm_ioctl,
 		.compat_ioctl = 	snd_pcm_ioctl_compat,
 		.mmap =			snd_pcm_mmap,
@@ -3721,7 +3664,7 @@ const struct file_operations snd_pcm_f_ops[2] = {
 		.open =			snd_pcm_capture_open,
 		.release =		snd_pcm_release,
 		.llseek =		no_llseek,
-		.poll =			snd_pcm_capture_poll,
+		.poll =			snd_pcm_poll,
 		.unlocked_ioctl =	snd_pcm_ioctl,
 		.compat_ioctl = 	snd_pcm_ioctl_compat,
 		.mmap =			snd_pcm_mmap,
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 9e96186..58fa3f9 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -421,13 +421,15 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
 	kctl->private_free = master_free;
 
 	/* additional (constant) TLV read */
-	if (tlv &&
-	    (tlv[0] == SNDRV_CTL_TLVT_DB_SCALE ||
-	     tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX ||
-	     tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX_MUTE)) {
-		kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
-		memcpy(master->tlv, tlv, sizeof(master->tlv));
-		kctl->tlv.p = master->tlv;
+	if (tlv) {
+		unsigned int type = tlv[SNDRV_CTL_TLVO_TYPE];
+		if (type == SNDRV_CTL_TLVT_DB_SCALE ||
+		    type == SNDRV_CTL_TLVT_DB_MINMAX ||
+		    type == SNDRV_CTL_TLVT_DB_MINMAX_MUTE) {
+			kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+			memcpy(master->tlv, tlv, sizeof(master->tlv));
+			kctl->tlv.p = master->tlv;
+		}
 	}
 
 	return kctl;
diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile
index 55b4be9..7b7997a 100644
--- a/sound/firewire/dice/Makefile
+++ b/sound/firewire/dice/Makefile
@@ -1,3 +1,4 @@
 snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \
-		 dice-pcm.o dice-hwdep.o dice.o
+		 dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \
+		 dice-alesis.o dice-extension.o
 obj-$(CONFIG_SND_DICE) += snd-dice.o
diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c
new file mode 100644
index 0000000..b2efb1c
--- /dev/null
+++ b/sound/firewire/dice/dice-alesis.c
@@ -0,0 +1,52 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * dice-alesis.c - a part of driver for DICE based devices
+ *
+ * Copyright (c) 2018 Takashi Sakamoto
+ */
+
+#include "dice.h"
+
+static const unsigned int
+alesis_io14_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
+	{6, 6, 4},	/* Tx0 = Analog + S/PDIF. */
+	{8, 4, 0},	/* Tx1 = ADAT1. */
+};
+
+static const unsigned int
+alesis_io26_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
+	{10, 10, 8},	/* Tx0 = Analog + S/PDIF. */
+	{16, 8, 0},	/* Tx1 = ADAT1 + ADAT2. */
+};
+
+int snd_dice_detect_alesis_formats(struct snd_dice *dice)
+{
+	__be32 reg;
+	u32 data;
+	int i;
+	int err;
+
+	err = snd_dice_transaction_read_tx(dice, TX_NUMBER_AUDIO, &reg,
+					   sizeof(reg));
+	if (err < 0)
+		return err;
+	data = be32_to_cpu(reg);
+
+	if (data == 4 || data == 6) {
+		memcpy(dice->tx_pcm_chs, alesis_io14_tx_pcm_chs,
+				MAX_STREAMS * SND_DICE_RATE_MODE_COUNT *
+				sizeof(unsigned int));
+	} else {
+		memcpy(dice->rx_pcm_chs, alesis_io26_tx_pcm_chs,
+				MAX_STREAMS * SND_DICE_RATE_MODE_COUNT *
+				sizeof(unsigned int));
+	}
+
+	for (i = 0; i < SND_DICE_RATE_MODE_COUNT; ++i)
+		dice->rx_pcm_chs[0][i] = 8;
+
+	dice->tx_midi_ports[0] = 1;
+	dice->rx_midi_ports[0] = 1;
+
+	return 0;
+}
diff --git a/sound/firewire/dice/dice-extension.c b/sound/firewire/dice/dice-extension.c
new file mode 100644
index 0000000..a63fcbc
--- /dev/null
+++ b/sound/firewire/dice/dice-extension.c
@@ -0,0 +1,172 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * dice-extension.c - a part of driver for DICE based devices
+ *
+ * Copyright (c) 2018 Takashi Sakamoto
+ */
+
+#include "dice.h"
+
+/* For TCD2210/2220, TCAT defines extension of application protocol. */
+
+#define DICE_EXT_APP_SPACE		0xffffe0200000uLL
+
+#define DICE_EXT_APP_CAPS_OFFSET	0x00
+#define DICE_EXT_APP_CAPS_SIZE		0x04
+#define DICE_EXT_APP_CMD_OFFSET		0x08
+#define DICE_EXT_APP_CMD_SIZE		0x0c
+#define DICE_EXT_APP_MIXER_OFFSET	0x10
+#define DICE_EXT_APP_MIXER_SIZE		0x14
+#define DICE_EXT_APP_PEAK_OFFSET	0x18
+#define DICE_EXT_APP_PEAK_SIZE		0x1c
+#define DICE_EXT_APP_ROUTER_OFFSET	0x20
+#define DICE_EXT_APP_ROUTER_SIZE	0x24
+#define DICE_EXT_APP_STREAM_OFFSET	0x28
+#define DICE_EXT_APP_STREAM_SIZE	0x2c
+#define DICE_EXT_APP_CURRENT_OFFSET	0x30
+#define DICE_EXT_APP_CURRENT_SIZE	0x34
+#define DICE_EXT_APP_STANDALONE_OFFSET	0x38
+#define DICE_EXT_APP_STANDALONE_SIZE	0x3c
+#define DICE_EXT_APP_APPLICATION_OFFSET	0x40
+#define DICE_EXT_APP_APPLICATION_SIZE	0x44
+
+#define EXT_APP_STREAM_TX_NUMBER	0x0000
+#define EXT_APP_STREAM_RX_NUMBER	0x0004
+#define EXT_APP_STREAM_ENTRIES		0x0008
+#define EXT_APP_STREAM_ENTRY_SIZE	0x010c
+#define  EXT_APP_NUMBER_AUDIO		0x0000
+#define  EXT_APP_NUMBER_MIDI		0x0004
+#define  EXT_APP_NAMES			0x0008
+#define   EXT_APP_NAMES_SIZE		256
+#define  EXT_APP_AC3			0x0108
+
+#define EXT_APP_CONFIG_LOW_ROUTER	0x0000
+#define EXT_APP_CONFIG_LOW_STREAM	0x1000
+#define EXT_APP_CONFIG_MIDDLE_ROUTER	0x2000
+#define EXT_APP_CONFIG_MIDDLE_STREAM	0x3000
+#define EXT_APP_CONFIG_HIGH_ROUTER	0x4000
+#define EXT_APP_CONFIG_HIGH_STREAM	0x5000
+
+static inline int read_transaction(struct snd_dice *dice, u64 section_addr,
+				   u32 offset, void *buf, size_t len)
+{
+	return snd_fw_transaction(dice->unit,
+				  len == 4 ? TCODE_READ_QUADLET_REQUEST :
+					     TCODE_READ_BLOCK_REQUEST,
+				  section_addr + offset, buf, len, 0);
+}
+
+static int read_stream_entries(struct snd_dice *dice, u64 section_addr,
+			       u32 base_offset, unsigned int stream_count,
+			       unsigned int mode,
+			       unsigned int pcm_channels[MAX_STREAMS][3],
+			       unsigned int midi_ports[MAX_STREAMS])
+{
+	u32 entry_offset;
+	__be32 reg[2];
+	int err;
+	int i;
+
+	for (i = 0; i < stream_count; ++i) {
+		entry_offset = base_offset + i * EXT_APP_STREAM_ENTRY_SIZE;
+		err = read_transaction(dice, section_addr,
+				    entry_offset + EXT_APP_NUMBER_AUDIO,
+				    reg, sizeof(reg));
+		if (err < 0)
+			return err;
+		pcm_channels[i][mode] = be32_to_cpu(reg[0]);
+		midi_ports[i] = max(midi_ports[i], be32_to_cpu(reg[1]));
+	}
+
+	return 0;
+}
+
+static int detect_stream_formats(struct snd_dice *dice, u64 section_addr)
+{
+	u32 base_offset;
+	__be32 reg[2];
+	unsigned int stream_count;
+	int mode;
+	int err = 0;
+
+	for (mode = 0; mode < SND_DICE_RATE_MODE_COUNT; ++mode) {
+		unsigned int cap;
+
+		/*
+		 * Some models report stream formats at highest mode, however
+		 * they don't support the mode. Check clock capabilities.
+		 */
+		if (mode == 2) {
+			cap = CLOCK_CAP_RATE_176400 | CLOCK_CAP_RATE_192000;
+		} else if (mode == 1) {
+			cap = CLOCK_CAP_RATE_88200 | CLOCK_CAP_RATE_96000;
+		} else {
+			cap = CLOCK_CAP_RATE_32000 | CLOCK_CAP_RATE_44100 |
+			      CLOCK_CAP_RATE_48000;
+		}
+		if (!(cap & dice->clock_caps))
+			continue;
+
+		base_offset = 0x2000 * mode + 0x1000;
+
+		err = read_transaction(dice, section_addr,
+				       base_offset + EXT_APP_STREAM_TX_NUMBER,
+				       &reg, sizeof(reg));
+		if (err < 0)
+			break;
+
+		base_offset += EXT_APP_STREAM_ENTRIES;
+		stream_count = be32_to_cpu(reg[0]);
+		err = read_stream_entries(dice, section_addr, base_offset,
+					  stream_count, mode,
+					  dice->tx_pcm_chs,
+					  dice->tx_midi_ports);
+		if (err < 0)
+			break;
+
+		base_offset += stream_count * EXT_APP_STREAM_ENTRY_SIZE;
+		stream_count = be32_to_cpu(reg[1]);
+		err = read_stream_entries(dice, section_addr, base_offset,
+					  stream_count,
+					  mode, dice->rx_pcm_chs,
+					  dice->rx_midi_ports);
+		if (err < 0)
+			break;
+	}
+
+	return err;
+}
+
+int snd_dice_detect_extension_formats(struct snd_dice *dice)
+{
+	__be32 *pointers;
+	unsigned int i;
+	u64 section_addr;
+	int err;
+
+	pointers = kmalloc_array(9, sizeof(__be32) * 2, GFP_KERNEL);
+	if (pointers == NULL)
+		return -ENOMEM;
+
+	err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST,
+				 DICE_EXT_APP_SPACE, pointers,
+				 9 * sizeof(__be32) * 2, 0);
+	if (err < 0)
+		goto end;
+
+	/* Check two of them for offset have the same value or not. */
+	for (i = 0; i < 9; ++i) {
+		int j;
+
+		for (j = i + 1; j < 9; ++j) {
+			if (pointers[i * 2] == pointers[j * 2])
+				goto end;
+		}
+	}
+
+	section_addr = DICE_EXT_APP_SPACE + be32_to_cpu(pointers[12]) * 4;
+	err = detect_stream_formats(dice, section_addr);
+end:
+	kfree(pointers);
+	return err;
+}
diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h
index 15a484b..9cad3d6 100644
--- a/sound/firewire/dice/dice-interface.h
+++ b/sound/firewire/dice/dice-interface.h
@@ -175,13 +175,18 @@
 #define GLOBAL_SAMPLE_RATE		0x05c
 
 /*
+ * Some old firmware versions do not have the following global registers.
+ * Windows drivers produced by TCAT lost backward compatibility in its
+ * early release because they can handle firmware only which supports the
+ * following registers.
+ */
+
+/*
  * The version of the DICE driver specification that this device conforms to;
  * read-only.
  */
 #define GLOBAL_VERSION			0x060
 
-/* Some old firmware versions do not have the following global registers: */
-
 /*
  * Supported sample rates and clock sources; read-only.
  */
diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c
index 8ff6da3..84eca8a 100644
--- a/sound/firewire/dice/dice-midi.c
+++ b/sound/firewire/dice/dice-midi.c
@@ -101,27 +101,18 @@ int snd_dice_create_midi(struct snd_dice *dice)
 		.close		= midi_close,
 		.trigger	= midi_playback_trigger,
 	};
-	__be32 reg;
 	struct snd_rawmidi *rmidi;
 	struct snd_rawmidi_str *str;
 	unsigned int midi_in_ports, midi_out_ports;
+	int i;
 	int err;
 
-	/*
-	 * Use the number of MIDI conformant data channel at current sampling
-	 * transfer frequency.
-	 */
-	err = snd_dice_transaction_read_tx(dice, TX_NUMBER_MIDI,
-					   &reg, sizeof(reg));
-	if (err < 0)
-		return err;
-	midi_in_ports = be32_to_cpu(reg);
-
-	err = snd_dice_transaction_read_rx(dice, RX_NUMBER_MIDI,
-					   &reg, sizeof(reg));
-	if (err < 0)
-		return err;
-	midi_out_ports = be32_to_cpu(reg);
+	midi_in_ports = 0;
+	midi_out_ports = 0;
+	for (i = 0; i < MAX_STREAMS; ++i) {
+		midi_in_ports = max(midi_in_ports, dice->tx_midi_ports[i]);
+		midi_out_ports = max(midi_out_ports, dice->rx_midi_ports[i]);
+	}
 
 	if (midi_in_ports + midi_out_ports == 0)
 		return 0;
diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c
index 7cb9e97..80351b2 100644
--- a/sound/firewire/dice/dice-pcm.c
+++ b/sound/firewire/dice/dice-pcm.c
@@ -9,43 +9,115 @@
 
 #include "dice.h"
 
+static int dice_rate_constraint(struct snd_pcm_hw_params *params,
+				struct snd_pcm_hw_rule *rule)
+{
+	struct snd_pcm_substream *substream = rule->private;
+	struct snd_dice *dice = substream->private_data;
+	unsigned int index = substream->pcm->device;
+
+	const struct snd_interval *c =
+		hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct snd_interval *r =
+		hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval rates = {
+		.min = UINT_MAX, .max = 0, .integer = 1
+	};
+	unsigned int *pcm_channels;
+	enum snd_dice_rate_mode mode;
+	unsigned int i, rate;
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		pcm_channels = dice->tx_pcm_chs[index];
+	else
+		pcm_channels = dice->rx_pcm_chs[index];
+
+	for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) {
+		rate = snd_dice_rates[i];
+		if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0)
+			continue;
+
+		if (!snd_interval_test(c, pcm_channels[mode]))
+			continue;
+
+		rates.min = min(rates.min, rate);
+		rates.max = max(rates.max, rate);
+	}
+
+	return snd_interval_refine(r, &rates);
+}
+
+static int dice_channels_constraint(struct snd_pcm_hw_params *params,
+				    struct snd_pcm_hw_rule *rule)
+{
+	struct snd_pcm_substream *substream = rule->private;
+	struct snd_dice *dice = substream->private_data;
+	unsigned int index = substream->pcm->device;
+
+	const struct snd_interval *r =
+		hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *c =
+		hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct snd_interval channels = {
+		.min = UINT_MAX, .max = 0, .integer = 1
+	};
+	unsigned int *pcm_channels;
+	enum snd_dice_rate_mode mode;
+	unsigned int i, rate;
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		pcm_channels = dice->tx_pcm_chs[index];
+	else
+		pcm_channels = dice->rx_pcm_chs[index];
+
+	for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) {
+		rate = snd_dice_rates[i];
+		if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0)
+			continue;
+
+		if (!snd_interval_test(r, rate))
+			continue;
+
+		channels.min = min(channels.min, pcm_channels[mode]);
+		channels.max = max(channels.max, pcm_channels[mode]);
+	}
+
+	return snd_interval_refine(c, &channels);
+}
+
 static int limit_channels_and_rates(struct snd_dice *dice,
 				    struct snd_pcm_runtime *runtime,
 				    enum amdtp_stream_direction dir,
-				    unsigned int index, unsigned int size)
+				    unsigned int index)
 {
 	struct snd_pcm_hardware *hw = &runtime->hw;
-	struct amdtp_stream *stream;
-	unsigned int rate;
-	__be32 reg;
-	int err;
+	unsigned int *pcm_channels;
+	unsigned int i;
 
-	/*
-	 * Retrieve current Multi Bit Linear Audio data channel and limit to
-	 * it.
-	 */
-	if (dir == AMDTP_IN_STREAM) {
-		stream = &dice->tx_stream[index];
-		err = snd_dice_transaction_read_tx(dice,
-				size * index + TX_NUMBER_AUDIO,
-				&reg, sizeof(reg));
-	} else {
-		stream = &dice->rx_stream[index];
-		err = snd_dice_transaction_read_rx(dice,
-				size * index + RX_NUMBER_AUDIO,
-				&reg, sizeof(reg));
+	if (dir == AMDTP_IN_STREAM)
+		pcm_channels = dice->tx_pcm_chs[index];
+	else
+		pcm_channels = dice->rx_pcm_chs[index];
+
+	hw->channels_min = UINT_MAX;
+	hw->channels_max = 0;
+
+	for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) {
+		enum snd_dice_rate_mode mode;
+		unsigned int rate, channels;
+
+		rate = snd_dice_rates[i];
+		if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0)
+			continue;
+		hw->rates |= snd_pcm_rate_to_rate_bit(rate);
+
+		channels = pcm_channels[mode];
+		if (channels == 0)
+			continue;
+		hw->channels_min = min(hw->channels_min, channels);
+		hw->channels_max = max(hw->channels_max, channels);
 	}
-	if (err < 0)
-		return err;
 
-	hw->channels_min = hw->channels_max = be32_to_cpu(reg);
-
-	/* Retrieve current sampling transfer frequency and limit to it. */
-	err = snd_dice_transaction_get_rate(dice, &rate);
-	if (err < 0)
-		return err;
-
-	hw->rates = snd_pcm_rate_to_rate_bit(rate);
 	snd_pcm_limit_hw_rates(runtime);
 
 	return 0;
@@ -56,36 +128,34 @@ static int init_hw_info(struct snd_dice *dice,
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_pcm_hardware *hw = &runtime->hw;
+	unsigned int index = substream->pcm->device;
 	enum amdtp_stream_direction dir;
 	struct amdtp_stream *stream;
-	__be32 reg[2];
-	unsigned int count, size;
 	int err;
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		hw->formats = AM824_IN_PCM_FORMAT_BITS;
 		dir = AMDTP_IN_STREAM;
-		stream = &dice->tx_stream[substream->pcm->device];
-		err = snd_dice_transaction_read_tx(dice, TX_NUMBER, reg,
-						   sizeof(reg));
+		stream = &dice->tx_stream[index];
 	} else {
 		hw->formats = AM824_OUT_PCM_FORMAT_BITS;
 		dir = AMDTP_OUT_STREAM;
-		stream = &dice->rx_stream[substream->pcm->device];
-		err = snd_dice_transaction_read_rx(dice, RX_NUMBER, reg,
-						   sizeof(reg));
+		stream = &dice->rx_stream[index];
 	}
 
+	err = limit_channels_and_rates(dice, substream->runtime, dir,
+				       index);
 	if (err < 0)
 		return err;
 
-	count = min_t(unsigned int, be32_to_cpu(reg[0]), MAX_STREAMS);
-	if (substream->pcm->device >= count)
-		return -ENXIO;
-
-	size = be32_to_cpu(reg[1]) * 4;
-	err = limit_channels_and_rates(dice, substream->runtime, dir,
-				       substream->pcm->device, size);
+	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+				  dice_rate_constraint, substream,
+				  SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+	if (err < 0)
+		return err;
+	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				  dice_channels_constraint, substream,
+				  SNDRV_PCM_HW_PARAM_RATE, -1);
 	if (err < 0)
 		return err;
 
@@ -95,6 +165,8 @@ static int init_hw_info(struct snd_dice *dice,
 static int pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_dice *dice = substream->private_data;
+	unsigned int source;
+	bool internal;
 	int err;
 
 	err = snd_dice_stream_lock_try(dice);
@@ -105,6 +177,43 @@ static int pcm_open(struct snd_pcm_substream *substream)
 	if (err < 0)
 		goto err_locked;
 
+	err = snd_dice_transaction_get_clock_source(dice, &source);
+	if (err < 0)
+		goto err_locked;
+	switch (source) {
+	case CLOCK_SOURCE_AES1:
+	case CLOCK_SOURCE_AES2:
+	case CLOCK_SOURCE_AES3:
+	case CLOCK_SOURCE_AES4:
+	case CLOCK_SOURCE_AES_ANY:
+	case CLOCK_SOURCE_ADAT:
+	case CLOCK_SOURCE_TDIF:
+	case CLOCK_SOURCE_WC:
+		internal = false;
+		break;
+	default:
+		internal = true;
+		break;
+	}
+
+	/*
+	 * When source of clock is not internal or any PCM streams are running,
+	 * available sampling rate is limited at current sampling rate.
+	 */
+	if (!internal ||
+	    amdtp_stream_pcm_running(&dice->tx_stream[0]) ||
+	    amdtp_stream_pcm_running(&dice->tx_stream[1]) ||
+	    amdtp_stream_pcm_running(&dice->rx_stream[0]) ||
+	    amdtp_stream_pcm_running(&dice->rx_stream[1])) {
+		unsigned int rate;
+
+		err = snd_dice_transaction_get_rate(dice, &rate);
+		if (err < 0)
+			goto err_locked;
+		substream->runtime->hw.rate_min = rate;
+		substream->runtime->hw.rate_max = rate;
+	}
+
 	snd_pcm_set_sync(substream);
 end:
 	return err;
@@ -318,37 +427,19 @@ int snd_dice_create_pcm(struct snd_dice *dice)
 		.page      = snd_pcm_lib_get_vmalloc_page,
 		.mmap      = snd_pcm_lib_mmap_vmalloc,
 	};
-	__be32 reg;
 	struct snd_pcm *pcm;
-	unsigned int i, max_capture, max_playback, capture, playback;
+	unsigned int capture, playback;
+	int i, j;
 	int err;
 
-	/* Check whether PCM substreams are required. */
-	if (dice->force_two_pcms) {
-		max_capture = max_playback = 2;
-	} else {
-		max_capture = max_playback = 0;
-		err = snd_dice_transaction_read_tx(dice, TX_NUMBER, &reg,
-						   sizeof(reg));
-		if (err < 0)
-			return err;
-		max_capture = min_t(unsigned int, be32_to_cpu(reg), MAX_STREAMS);
-
-		err = snd_dice_transaction_read_rx(dice, RX_NUMBER, &reg,
-						   sizeof(reg));
-		if (err < 0)
-			return err;
-		max_playback = min_t(unsigned int, be32_to_cpu(reg), MAX_STREAMS);
-	}
-
 	for (i = 0; i < MAX_STREAMS; i++) {
 		capture = playback = 0;
-		if (i < max_capture)
-			capture = 1;
-		if (i < max_playback)
-			playback = 1;
-		if (capture == 0 && playback == 0)
-			break;
+		for (j = 0; j < SND_DICE_RATE_MODE_COUNT; ++j) {
+			if (dice->tx_pcm_chs[i][j] > 0)
+				capture = 1;
+			if (dice->rx_pcm_chs[i][j] > 0)
+				playback = 1;
+		}
 
 		err = snd_pcm_new(dice->card, "DICE", i, playback, capture,
 				  &pcm);
diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c
index f5c1d1b..faf8c85 100644
--- a/sound/firewire/dice/dice-proc.c
+++ b/sound/firewire/dice/dice-proc.c
@@ -148,12 +148,12 @@ static void dice_proc_read(struct snd_info_entry *entry,
 				   >> CLOCK_RATE_SHIFT));
 	snd_iprintf(buffer, "  ext status: %08x\n", buf.global.extended_status);
 	snd_iprintf(buffer, "  sample rate: %u\n", buf.global.sample_rate);
-	snd_iprintf(buffer, "  version: %u.%u.%u.%u\n",
-		    (buf.global.version >> 24) & 0xff,
-		    (buf.global.version >> 16) & 0xff,
-		    (buf.global.version >>  8) & 0xff,
-		    (buf.global.version >>  0) & 0xff);
 	if (quadlets >= 90) {
+		snd_iprintf(buffer, "  version: %u.%u.%u.%u\n",
+			    (buf.global.version >> 24) & 0xff,
+			    (buf.global.version >> 16) & 0xff,
+			    (buf.global.version >>  8) & 0xff,
+			    (buf.global.version >>  0) & 0xff);
 		snd_iprintf(buffer, "  clock caps:");
 		for (i = 0; i <= 6; ++i)
 			if (buf.global.clock_caps & (1 << i))
@@ -243,10 +243,74 @@ static void dice_proc_read(struct snd_info_entry *entry,
 	}
 }
 
-void snd_dice_create_proc(struct snd_dice *dice)
+static void dice_proc_read_formation(struct snd_info_entry *entry,
+				     struct snd_info_buffer *buffer)
+{
+	static const char *const rate_labels[] = {
+		[SND_DICE_RATE_MODE_LOW]	= "low",
+		[SND_DICE_RATE_MODE_MIDDLE]	= "middle",
+		[SND_DICE_RATE_MODE_HIGH]	= "high",
+	};
+	struct snd_dice *dice = entry->private_data;
+	int i, j;
+
+	snd_iprintf(buffer, "Output stream from unit:\n");
+	for (i = 0; i < SND_DICE_RATE_MODE_COUNT; ++i)
+		snd_iprintf(buffer, "\t%s", rate_labels[i]);
+	snd_iprintf(buffer, "\tMIDI\n");
+	for (i = 0; i < MAX_STREAMS; ++i) {
+		snd_iprintf(buffer, "Tx %u:", i);
+		for (j = 0; j < SND_DICE_RATE_MODE_COUNT; ++j)
+			snd_iprintf(buffer, "\t%u", dice->tx_pcm_chs[i][j]);
+		snd_iprintf(buffer, "\t%u\n", dice->tx_midi_ports[i]);
+	}
+
+	snd_iprintf(buffer, "Input stream to unit:\n");
+	for (i = 0; i < SND_DICE_RATE_MODE_COUNT; ++i)
+		snd_iprintf(buffer, "\t%s", rate_labels[i]);
+	snd_iprintf(buffer, "\n");
+	for (i = 0; i < MAX_STREAMS; ++i) {
+		snd_iprintf(buffer, "Rx %u:", i);
+		for (j = 0; j < SND_DICE_RATE_MODE_COUNT; ++j)
+			snd_iprintf(buffer, "\t%u", dice->rx_pcm_chs[i][j]);
+		snd_iprintf(buffer, "\t%u\n", dice->rx_midi_ports[i]);
+	}
+}
+
+static void add_node(struct snd_dice *dice, struct snd_info_entry *root,
+		     const char *name,
+		     void (*op)(struct snd_info_entry *entry,
+				struct snd_info_buffer *buffer))
 {
 	struct snd_info_entry *entry;
 
-	if (!snd_card_proc_new(dice->card, "dice", &entry))
-		snd_info_set_text_ops(entry, dice, dice_proc_read);
+	entry = snd_info_create_card_entry(dice->card, name, root);
+	if (!entry)
+		return;
+
+	snd_info_set_text_ops(entry, dice, op);
+	if (snd_info_register(entry) < 0)
+		snd_info_free_entry(entry);
+}
+
+void snd_dice_create_proc(struct snd_dice *dice)
+{
+	struct snd_info_entry *root;
+
+	/*
+	 * All nodes are automatically removed at snd_card_disconnect(),
+	 * by following to link list.
+	 */
+	root = snd_info_create_card_entry(dice->card, "firewire",
+					  dice->card->proc_root);
+	if (!root)
+		return;
+	root->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+	if (snd_info_register(root) < 0) {
+		snd_info_free_entry(root);
+		return;
+	}
+
+	add_node(dice, root, "dice", dice_proc_read);
+	add_node(dice, root, "formation", dice_proc_read_formation);
 }
diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index 928a255..c3c892c 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -30,13 +30,43 @@ const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT] = {
 	[6] = 192000,
 };
 
+int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate,
+				  enum snd_dice_rate_mode *mode)
+{
+	/* Corresponding to each entry in snd_dice_rates. */
+	static const enum snd_dice_rate_mode modes[] = {
+		[0] = SND_DICE_RATE_MODE_LOW,
+		[1] = SND_DICE_RATE_MODE_LOW,
+		[2] = SND_DICE_RATE_MODE_LOW,
+		[3] = SND_DICE_RATE_MODE_MIDDLE,
+		[4] = SND_DICE_RATE_MODE_MIDDLE,
+		[5] = SND_DICE_RATE_MODE_HIGH,
+		[6] = SND_DICE_RATE_MODE_HIGH,
+	};
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(snd_dice_rates); i++) {
+		if (!(dice->clock_caps & BIT(i)))
+			continue;
+		if (snd_dice_rates[i] != rate)
+			continue;
+
+		*mode = modes[i];
+		return 0;
+	}
+
+	return -EINVAL;
+}
+
 /*
  * This operation has an effect to synchronize GLOBAL_STATUS/GLOBAL_SAMPLE_RATE
  * to GLOBAL_STATUS. Especially, just after powering on, these are different.
  */
-static int ensure_phase_lock(struct snd_dice *dice)
+static int ensure_phase_lock(struct snd_dice *dice, unsigned int rate)
 {
 	__be32 reg, nominal;
+	u32 data;
+	int i;
 	int err;
 
 	err = snd_dice_transaction_read_global(dice, GLOBAL_CLOCK_SELECT,
@@ -44,9 +74,21 @@ static int ensure_phase_lock(struct snd_dice *dice)
 	if (err < 0)
 		return err;
 
+	data = be32_to_cpu(reg);
+
+	data &= ~CLOCK_RATE_MASK;
+	for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) {
+		if (snd_dice_rates[i] == rate)
+			break;
+	}
+	if (i == ARRAY_SIZE(snd_dice_rates))
+		return -EINVAL;
+	data |= i << CLOCK_RATE_SHIFT;
+
 	if (completion_done(&dice->clock_accepted))
 		reinit_completion(&dice->clock_accepted);
 
+	reg = cpu_to_be32(data);
 	err = snd_dice_transaction_write_global(dice, GLOBAL_CLOCK_SELECT,
 						&reg, sizeof(reg));
 	if (err < 0)
@@ -192,6 +234,7 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir,
 			 unsigned int rate, struct reg_params *params)
 {
 	__be32 reg[2];
+	enum snd_dice_rate_mode mode;
 	unsigned int i, pcm_chs, midi_ports;
 	struct amdtp_stream *streams;
 	struct fw_iso_resources *resources;
@@ -206,12 +249,23 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir,
 		resources = dice->rx_resources;
 	}
 
+	err = snd_dice_stream_get_rate_mode(dice, rate, &mode);
+	if (err < 0)
+		return err;
+
 	for (i = 0; i < params->count; i++) {
+		unsigned int pcm_cache;
+		unsigned int midi_cache;
+
 		if (dir == AMDTP_IN_STREAM) {
+			pcm_cache = dice->tx_pcm_chs[i][mode];
+			midi_cache = dice->tx_midi_ports[i];
 			err = snd_dice_transaction_read_tx(dice,
 					params->size * i + TX_NUMBER_AUDIO,
 					reg, sizeof(reg));
 		} else {
+			pcm_cache = dice->rx_pcm_chs[i][mode];
+			midi_cache = dice->rx_midi_ports[i];
 			err = snd_dice_transaction_read_rx(dice,
 					params->size * i + RX_NUMBER_AUDIO,
 					reg, sizeof(reg));
@@ -221,6 +275,14 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir,
 		pcm_chs = be32_to_cpu(reg[0]);
 		midi_ports = be32_to_cpu(reg[1]);
 
+		/* These are important for developer of this driver. */
+		if (pcm_chs != pcm_cache || midi_ports != midi_cache) {
+			dev_info(&dice->unit->device,
+				 "cache mismatch: pcm: %u:%u, midi: %u:%u\n",
+				 pcm_chs, pcm_cache, midi_ports, midi_cache);
+			return -EPROTO;
+		}
+
 		err = keep_resources(dice, dir, i, rate, pcm_chs, midi_ports);
 		if (err < 0)
 			return err;
@@ -256,6 +318,68 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir,
 	return err;
 }
 
+static int start_duplex_streams(struct snd_dice *dice, unsigned int rate)
+{
+	struct reg_params tx_params, rx_params;
+	int i;
+	int err;
+
+	err = get_register_params(dice, &tx_params, &rx_params);
+	if (err < 0)
+		return err;
+
+	/* Stop transmission. */
+	stop_streams(dice, AMDTP_IN_STREAM, &tx_params);
+	stop_streams(dice, AMDTP_OUT_STREAM, &rx_params);
+	snd_dice_transaction_clear_enable(dice);
+	release_resources(dice);
+
+	err = ensure_phase_lock(dice, rate);
+	if (err < 0) {
+		dev_err(&dice->unit->device, "fail to ensure phase lock\n");
+		return err;
+	}
+
+	/* Likely to have changed stream formats. */
+	err = get_register_params(dice, &tx_params, &rx_params);
+	if (err < 0)
+		return err;
+
+	/* Start both streams. */
+	err = start_streams(dice, AMDTP_IN_STREAM, rate, &tx_params);
+	if (err < 0)
+		goto error;
+	err = start_streams(dice, AMDTP_OUT_STREAM, rate, &rx_params);
+	if (err < 0)
+		goto error;
+
+	err = snd_dice_transaction_set_enable(dice);
+	if (err < 0) {
+		dev_err(&dice->unit->device, "fail to enable interface\n");
+		goto error;
+	}
+
+	for (i = 0; i < MAX_STREAMS; i++) {
+		if ((i < tx_params.count &&
+		    !amdtp_stream_wait_callback(&dice->tx_stream[i],
+						CALLBACK_TIMEOUT)) ||
+		    (i < rx_params.count &&
+		     !amdtp_stream_wait_callback(&dice->rx_stream[i],
+						 CALLBACK_TIMEOUT))) {
+			err = -ETIMEDOUT;
+			goto error;
+		}
+	}
+
+	return 0;
+error:
+	stop_streams(dice, AMDTP_IN_STREAM, &tx_params);
+	stop_streams(dice, AMDTP_OUT_STREAM, &rx_params);
+	snd_dice_transaction_clear_enable(dice);
+	release_resources(dice);
+	return err;
+}
+
 /*
  * MEMO: After this function, there're two states of streams:
  *  - None streams are running.
@@ -265,17 +389,13 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate)
 {
 	unsigned int curr_rate;
 	unsigned int i;
-	struct reg_params tx_params, rx_params;
-	bool need_to_start;
+	enum snd_dice_rate_mode mode;
 	int err;
 
 	if (dice->substreams_counter == 0)
 		return -EIO;
 
-	err = get_register_params(dice, &tx_params, &rx_params);
-	if (err < 0)
-		return err;
-
+	/* Check sampling transmission frequency. */
 	err = snd_dice_transaction_get_rate(dice, &curr_rate);
 	if (err < 0) {
 		dev_err(&dice->unit->device,
@@ -285,72 +405,36 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate)
 	if (rate == 0)
 		rate = curr_rate;
 	if (rate != curr_rate)
-		return -EINVAL;
+		goto restart;
 
-	/* Judge to need to restart streams. */
-	for (i = 0; i < MAX_STREAMS; i++) {
-		if (i < tx_params.count) {
-			if (amdtp_streaming_error(&dice->tx_stream[i]) ||
-			    !amdtp_stream_running(&dice->tx_stream[i]))
-				break;
-		}
-		if (i < rx_params.count) {
-			if (amdtp_streaming_error(&dice->rx_stream[i]) ||
-			    !amdtp_stream_running(&dice->rx_stream[i]))
-				break;
-		}
+	/* Check error of packet streaming. */
+	for (i = 0; i < MAX_STREAMS; ++i) {
+		if (amdtp_streaming_error(&dice->tx_stream[i]))
+			break;
+		if (amdtp_streaming_error(&dice->rx_stream[i]))
+			break;
 	}
-	need_to_start = (i < MAX_STREAMS);
+	if (i < MAX_STREAMS)
+		goto restart;
 
-	if (need_to_start) {
-		/* Stop transmission. */
-		snd_dice_transaction_clear_enable(dice);
-		stop_streams(dice, AMDTP_IN_STREAM, &tx_params);
-		stop_streams(dice, AMDTP_OUT_STREAM, &rx_params);
-		release_resources(dice);
-
-		err = ensure_phase_lock(dice);
-		if (err < 0) {
-			dev_err(&dice->unit->device,
-				"fail to ensure phase lock\n");
-			return err;
-		}
-
-		/* Start both streams. */
-		err = start_streams(dice, AMDTP_IN_STREAM, rate, &tx_params);
-		if (err < 0)
-			goto error;
-		err = start_streams(dice, AMDTP_OUT_STREAM, rate, &rx_params);
-		if (err < 0)
-			goto error;
-
-		err = snd_dice_transaction_set_enable(dice);
-		if (err < 0) {
-			dev_err(&dice->unit->device,
-				"fail to enable interface\n");
-			goto error;
-		}
-
-		for (i = 0; i < MAX_STREAMS; i++) {
-			if ((i < tx_params.count &&
-			    !amdtp_stream_wait_callback(&dice->tx_stream[i],
-							CALLBACK_TIMEOUT)) ||
-			    (i < rx_params.count &&
-			     !amdtp_stream_wait_callback(&dice->rx_stream[i],
-							 CALLBACK_TIMEOUT))) {
-				err = -ETIMEDOUT;
-				goto error;
-			}
-		}
+	/* Check required streams are running or not. */
+	err = snd_dice_stream_get_rate_mode(dice, rate, &mode);
+	if (err < 0)
+		return err;
+	for (i = 0; i < MAX_STREAMS; ++i) {
+		if (dice->tx_pcm_chs[i][mode] > 0 &&
+		    !amdtp_stream_running(&dice->tx_stream[i]))
+			break;
+		if (dice->rx_pcm_chs[i][mode] > 0 &&
+		    !amdtp_stream_running(&dice->rx_stream[i]))
+			break;
 	}
+	if (i < MAX_STREAMS)
+		goto restart;
 
-	return err;
-error:
-	snd_dice_transaction_clear_enable(dice);
-	stop_streams(dice, AMDTP_IN_STREAM, &tx_params);
-	stop_streams(dice, AMDTP_OUT_STREAM, &rx_params);
-	release_resources(dice);
-	return err;
+	return 0;
+restart:
+	return start_duplex_streams(dice, rate);
 }
 
 /*
@@ -484,6 +568,69 @@ void snd_dice_stream_update_duplex(struct snd_dice *dice)
 	}
 }
 
+int snd_dice_stream_detect_current_formats(struct snd_dice *dice)
+{
+	unsigned int rate;
+	enum snd_dice_rate_mode mode;
+	__be32 reg[2];
+	struct reg_params tx_params, rx_params;
+	int i;
+	int err;
+
+	/* If extended protocol is available, detect detail spec. */
+	err = snd_dice_detect_extension_formats(dice);
+	if (err >= 0)
+		return err;
+
+	/*
+	 * Available stream format is restricted at current mode of sampling
+	 * clock.
+	 */
+	err = snd_dice_transaction_get_rate(dice, &rate);
+	if (err < 0)
+		return err;
+
+	err = snd_dice_stream_get_rate_mode(dice, rate, &mode);
+	if (err < 0)
+		return err;
+
+	/*
+	 * Just after owning the unit (GLOBAL_OWNER), the unit can return
+	 * invalid stream formats. Selecting clock parameters have an effect
+	 * for the unit to refine it.
+	 */
+	err = ensure_phase_lock(dice, rate);
+	if (err < 0)
+		return err;
+
+	err = get_register_params(dice, &tx_params, &rx_params);
+	if (err < 0)
+		return err;
+
+	for (i = 0; i < tx_params.count; ++i) {
+		err = snd_dice_transaction_read_tx(dice,
+				tx_params.size * i + TX_NUMBER_AUDIO,
+				reg, sizeof(reg));
+		if (err < 0)
+			return err;
+		dice->tx_pcm_chs[i][mode] = be32_to_cpu(reg[0]);
+		dice->tx_midi_ports[i] = max_t(unsigned int,
+				be32_to_cpu(reg[1]), dice->tx_midi_ports[i]);
+	}
+	for (i = 0; i < rx_params.count; ++i) {
+		err = snd_dice_transaction_read_rx(dice,
+				rx_params.size * i + RX_NUMBER_AUDIO,
+				reg, sizeof(reg));
+		if (err < 0)
+			return err;
+		dice->rx_pcm_chs[i][mode] = be32_to_cpu(reg[0]);
+		dice->rx_midi_ports[i] = max_t(unsigned int,
+				be32_to_cpu(reg[1]), dice->rx_midi_ports[i]);
+	}
+
+	return 0;
+}
+
 static void dice_lock_changed(struct snd_dice *dice)
 {
 	dice->dev_lock_changed = true;
diff --git a/sound/firewire/dice/dice-tcelectronic.c b/sound/firewire/dice/dice-tcelectronic.c
new file mode 100644
index 0000000..af8203b
--- /dev/null
+++ b/sound/firewire/dice/dice-tcelectronic.c
@@ -0,0 +1,100 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * dice-tc_electronic.c - a part of driver for DICE based devices
+ *
+ * Copyright (c) 2018 Takashi Sakamoto
+ */
+
+#include "dice.h"
+
+struct dice_tc_spec {
+	unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT];
+	unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT];
+	bool has_midi;
+};
+
+static const struct dice_tc_spec desktop_konnekt6 = {
+	.tx_pcm_chs = {{6, 6, 2}, {0, 0, 0} },
+	.rx_pcm_chs = {{6, 6, 4}, {0, 0, 0} },
+	.has_midi = false,
+};
+
+static const struct dice_tc_spec impact_twin = {
+	.tx_pcm_chs = {{14, 10, 6}, {0, 0, 0} },
+	.rx_pcm_chs = {{14, 10, 6}, {0, 0, 0} },
+	.has_midi = true,
+};
+
+static const struct dice_tc_spec konnekt_8 = {
+	.tx_pcm_chs = {{4, 4, 3}, {0, 0, 0} },
+	.rx_pcm_chs = {{4, 4, 3}, {0, 0, 0} },
+	.has_midi = true,
+};
+
+static const struct dice_tc_spec konnekt_24d = {
+	.tx_pcm_chs = {{16, 16, 6}, {0, 0, 0} },
+	.rx_pcm_chs = {{16, 16, 6}, {0, 0, 0} },
+	.has_midi = true,
+};
+
+static const struct dice_tc_spec konnekt_live = {
+	.tx_pcm_chs = {{16, 16, 16}, {0, 0, 0} },
+	.rx_pcm_chs = {{16, 16, 16}, {0, 0, 0} },
+	.has_midi = true,
+};
+
+static const struct dice_tc_spec studio_konnekt_48 = {
+	.tx_pcm_chs = {{16, 16, 16}, {16, 16, 0} },
+	.rx_pcm_chs = {{16, 16, 16}, {16, 16, 0} },
+	.has_midi = true,
+};
+
+int snd_dice_detect_tcelectronic_formats(struct snd_dice *dice)
+{
+	static const struct {
+		u32 model_id;
+		const struct dice_tc_spec *spec;
+	} *entry, entries[] = {
+		{0x00000020, &konnekt_24d},
+		{0x00000021, &konnekt_8},
+		{0x00000022, &studio_konnekt_48},
+		{0x00000023, &konnekt_live},
+		{0x00000024, &desktop_konnekt6},
+		{0x00000027, &impact_twin},
+	};
+	struct fw_csr_iterator it;
+	int key, val, model_id;
+	int i;
+
+	model_id = 0;
+	fw_csr_iterator_init(&it, dice->unit->directory);
+	while (fw_csr_iterator_next(&it, &key, &val)) {
+		if (key == CSR_MODEL) {
+			model_id = val;
+			break;
+		}
+	}
+
+	entry = NULL;
+	for (i = 0; i < ARRAY_SIZE(entries); ++i) {
+		entry = entries + i;
+		if (entry->model_id == model_id)
+			break;
+	}
+	if (!entry)
+		return -ENODEV;
+
+	memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs,
+	       MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int));
+	memcpy(dice->rx_pcm_chs, entry->spec->rx_pcm_chs,
+	       MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int));
+
+	for (i = 0; i < MAX_STREAMS; ++i) {
+		if (entry->spec->has_midi) {
+			dice->tx_midi_ports[i] = 1;
+			dice->rx_midi_ports[i] = 1;
+		}
+	}
+
+	return 0;
+}
diff --git a/sound/firewire/dice/dice-transaction.c b/sound/firewire/dice/dice-transaction.c
index 0f03503..b7e138b 100644
--- a/sound/firewire/dice/dice-transaction.c
+++ b/sound/firewire/dice/dice-transaction.c
@@ -265,7 +265,7 @@ int snd_dice_transaction_reinit(struct snd_dice *dice)
 static int get_subaddrs(struct snd_dice *dice)
 {
 	static const int min_values[10] = {
-		10, 0x64 / 4,
+		10, 0x60 / 4,
 		10, 0x18 / 4,
 		10, 0x18 / 4,
 		0, 0,
@@ -301,33 +301,40 @@ static int get_subaddrs(struct snd_dice *dice)
 		}
 	}
 
-	/*
-	 * Check that the implemented DICE driver specification major version
-	 * number matches.
-	 */
-	err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST,
-				 DICE_PRIVATE_SPACE +
-				 be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION,
-				 &version, sizeof(version), 0);
-	if (err < 0)
-		goto end;
+	if (be32_to_cpu(pointers[1]) > 0x18) {
+		/*
+		 * Check that the implemented DICE driver specification major
+		 * version number matches.
+		 */
+		err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST,
+				DICE_PRIVATE_SPACE +
+				be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION,
+				&version, sizeof(version), 0);
+		if (err < 0)
+			goto end;
 
-	if ((version & cpu_to_be32(0xff000000)) != cpu_to_be32(0x01000000)) {
-		dev_err(&dice->unit->device,
-			"unknown DICE version: 0x%08x\n", be32_to_cpu(version));
-		err = -ENODEV;
-		goto end;
+		if ((version & cpu_to_be32(0xff000000)) !=
+						cpu_to_be32(0x01000000)) {
+			dev_err(&dice->unit->device,
+				"unknown DICE version: 0x%08x\n",
+				be32_to_cpu(version));
+			err = -ENODEV;
+			goto end;
+		}
+
+		/* Set up later. */
+		dice->clock_caps = 1;
 	}
 
 	dice->global_offset = be32_to_cpu(pointers[0]) * 4;
 	dice->tx_offset = be32_to_cpu(pointers[2]) * 4;
 	dice->rx_offset = be32_to_cpu(pointers[4]) * 4;
-	dice->sync_offset = be32_to_cpu(pointers[6]) * 4;
-	dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4;
 
-	/* Set up later. */
-	if (be32_to_cpu(pointers[1]) * 4 >= GLOBAL_CLOCK_CAPABILITIES + 4)
-		dice->clock_caps = 1;
+	/* Old firmware doesn't support these fields. */
+	if (pointers[7])
+		dice->sync_offset = be32_to_cpu(pointers[6]) * 4;
+	if (pointers[9])
+		dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4;
 end:
 	kfree(pointers);
 	return err;
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 96bb01b..40f7a32 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -15,40 +15,14 @@ MODULE_LICENSE("GPL v2");
 #define OUI_LOUD		0x000ff2
 #define OUI_FOCUSRITE		0x00130e
 #define OUI_TCELECTRONIC	0x000166
+#define OUI_ALESIS		0x000595
+#define OUI_MAUDIO		0x000d6c
 
 #define DICE_CATEGORY_ID	0x04
 #define WEISS_CATEGORY_ID	0x00
 #define LOUD_CATEGORY_ID	0x10
 
-/*
- * Some models support several isochronous channels, while these streams are not
- * always available. In this case, add the model name to this list.
- */
-static bool force_two_pcm_support(struct fw_unit *unit)
-{
-	static const char *const models[] = {
-		/* TC Electronic models. */
-		"StudioKonnekt48",
-		/* Focusrite models. */
-		"SAFFIRE_PRO_40",
-		"LIQUID_SAFFIRE_56",
-		"SAFFIRE_PRO_40_1",
-	};
-	char model[32];
-	unsigned int i;
-	int err;
-
-	err = fw_csr_string(unit->directory, CSR_MODEL, model, sizeof(model));
-	if (err < 0)
-		return false;
-
-	for (i = 0; i < ARRAY_SIZE(models); i++) {
-		if (strcmp(models[i], model) == 0)
-			break;
-	}
-
-	return i < ARRAY_SIZE(models);
-}
+#define MODEL_ALESIS_IO_BOTH	0x000001
 
 static int check_dice_category(struct fw_unit *unit)
 {
@@ -75,11 +49,6 @@ static int check_dice_category(struct fw_unit *unit)
 		}
 	}
 
-	if (vendor == OUI_FOCUSRITE || vendor == OUI_TCELECTRONIC) {
-		if (force_two_pcm_support(unit))
-			return 0;
-	}
-
 	if (vendor == OUI_WEISS)
 		category = WEISS_CATEGORY_ID;
 	else if (vendor == OUI_LOUD)
@@ -186,9 +155,6 @@ static void do_registration(struct work_struct *work)
 	if (err < 0)
 		return;
 
-	if (force_two_pcm_support(dice->unit))
-		dice->force_two_pcms = true;
-
 	err = snd_dice_transaction_init(dice);
 	if (err < 0)
 		goto error;
@@ -199,6 +165,10 @@ static void do_registration(struct work_struct *work)
 
 	dice_card_strings(dice);
 
+	err = dice->detect_formats(dice);
+	if (err < 0)
+		goto error;
+
 	err = snd_dice_stream_init_duplex(dice);
 	if (err < 0)
 		goto error;
@@ -239,14 +209,17 @@ static void do_registration(struct work_struct *work)
 		 "Sound card registration failed: %d\n", err);
 }
 
-static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id)
+static int dice_probe(struct fw_unit *unit,
+		      const struct ieee1394_device_id *entry)
 {
 	struct snd_dice *dice;
 	int err;
 
-	err = check_dice_category(unit);
-	if (err < 0)
-		return -ENODEV;
+	if (!entry->driver_data) {
+		err = check_dice_category(unit);
+		if (err < 0)
+			return -ENODEV;
+	}
 
 	/* Allocate this independent of sound card instance. */
 	dice = kzalloc(sizeof(struct snd_dice), GFP_KERNEL);
@@ -256,6 +229,13 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id)
 	dice->unit = fw_unit_get(unit);
 	dev_set_drvdata(&unit->device, dice);
 
+	if (!entry->driver_data) {
+		dice->detect_formats = snd_dice_stream_detect_current_formats;
+	} else {
+		dice->detect_formats =
+				(snd_dice_detect_formats_t)entry->driver_data;
+	}
+
 	spin_lock_init(&dice->lock);
 	mutex_init(&dice->mutex);
 	init_completion(&dice->clock_accepted);
@@ -313,17 +293,82 @@ static void dice_bus_reset(struct fw_unit *unit)
 #define DICE_INTERFACE	0x000001
 
 static const struct ieee1394_device_id dice_id_table[] = {
+	/* M-Audio Profire 2626 has a different value in version field. */
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_MAUDIO,
+		.model_id	= 0x000010,
+		.driver_data = (kernel_ulong_t)snd_dice_detect_extension_formats,
+	},
+	/* M-Audio Profire 610 has a different value in version field. */
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_MAUDIO,
+		.model_id	= 0x000011,
+		.driver_data = (kernel_ulong_t)snd_dice_detect_extension_formats,
+	},
+	/* TC Electronic Konnekt 24D. */
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_TCELECTRONIC,
+		.model_id	= 0x000020,
+		.driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats,
+	},
+	/* TC Electronic Konnekt 8. */
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_TCELECTRONIC,
+		.model_id	= 0x000021,
+		.driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats,
+	},
+	/* TC Electronic Studio Konnekt 48. */
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_TCELECTRONIC,
+		.model_id	= 0x000022,
+		.driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats,
+	},
+	/* TC Electronic Konnekt Live. */
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_TCELECTRONIC,
+		.model_id	= 0x000023,
+		.driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats,
+	},
+	/* TC Electronic Desktop Konnekt 6. */
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_TCELECTRONIC,
+		.model_id	= 0x000024,
+		.driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats,
+	},
+	/* TC Electronic Impact Twin. */
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_TCELECTRONIC,
+		.model_id	= 0x000027,
+		.driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats,
+	},
+	/* Alesis iO14/iO26. */
+	{
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID,
+		.vendor_id	= OUI_ALESIS,
+		.model_id	= MODEL_ALESIS_IO_BOTH,
+		.driver_data = (kernel_ulong_t)snd_dice_detect_alesis_formats,
+	},
 	{
 		.match_flags = IEEE1394_MATCH_VERSION,
 		.version     = DICE_INTERFACE,
 	},
-	/* M-Audio Profire 610/2626 has a different value in version field. */
-	{
-		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
-				  IEEE1394_MATCH_SPECIFIER_ID,
-		.vendor_id	= 0x000d6c,
-		.specifier_id	= 0x000d6c,
-	},
 	{ }
 };
 MODULE_DEVICE_TABLE(ieee1394, dice_id_table);
diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h
index da00e75..505b79f 100644
--- a/sound/firewire/dice/dice.h
+++ b/sound/firewire/dice/dice.h
@@ -63,6 +63,16 @@
  */
 #define MAX_STREAMS	2
 
+enum snd_dice_rate_mode {
+	SND_DICE_RATE_MODE_LOW = 0,
+	SND_DICE_RATE_MODE_MIDDLE,
+	SND_DICE_RATE_MODE_HIGH,
+	SND_DICE_RATE_MODE_COUNT,
+};
+
+struct snd_dice;
+typedef int (*snd_dice_detect_formats_t)(struct snd_dice *dice);
+
 struct snd_dice {
 	struct snd_card *card;
 	struct fw_unit *unit;
@@ -80,6 +90,11 @@ struct snd_dice {
 	unsigned int rsrv_offset;
 
 	unsigned int clock_caps;
+	unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT];
+	unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT];
+	unsigned int tx_midi_ports[MAX_STREAMS];
+	unsigned int rx_midi_ports[MAX_STREAMS];
+	snd_dice_detect_formats_t detect_formats;
 
 	struct fw_address_handler notification_handler;
 	int owner_generation;
@@ -98,8 +113,6 @@ struct snd_dice {
 	bool global_enabled;
 	struct completion clock_accepted;
 	unsigned int substreams_counter;
-
-	bool force_two_pcms;
 };
 
 enum snd_dice_addr_type {
@@ -190,11 +203,14 @@ void snd_dice_transaction_destroy(struct snd_dice *dice);
 #define SND_DICE_RATES_COUNT	7
 extern const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT];
 
+int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate,
+				  enum snd_dice_rate_mode *mode);
 int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate);
 void snd_dice_stream_stop_duplex(struct snd_dice *dice);
 int snd_dice_stream_init_duplex(struct snd_dice *dice);
 void snd_dice_stream_destroy_duplex(struct snd_dice *dice);
 void snd_dice_stream_update_duplex(struct snd_dice *dice);
+int snd_dice_stream_detect_current_formats(struct snd_dice *dice);
 
 int snd_dice_stream_lock_try(struct snd_dice *dice);
 void snd_dice_stream_lock_release(struct snd_dice *dice);
@@ -207,4 +223,8 @@ void snd_dice_create_proc(struct snd_dice *dice);
 
 int snd_dice_create_midi(struct snd_dice *dice);
 
+int snd_dice_detect_tcelectronic_formats(struct snd_dice *dice);
+int snd_dice_detect_alesis_formats(struct snd_dice *dice);
+int snd_dice_detect_extension_formats(struct snd_dice *dice);
+
 #endif
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 46092fa..3919e18 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -569,18 +569,20 @@ static int isight_create_mixer(struct isight *isight)
 		return err;
 	isight->gain_max = be32_to_cpu(value);
 
-	isight->gain_tlv[0] = SNDRV_CTL_TLVT_DB_MINMAX;
-	isight->gain_tlv[1] = 2 * sizeof(unsigned int);
+	isight->gain_tlv[SNDRV_CTL_TLVO_TYPE] = SNDRV_CTL_TLVT_DB_MINMAX;
+	isight->gain_tlv[SNDRV_CTL_TLVO_LEN] = 2 * sizeof(unsigned int);
 
 	err = reg_read(isight, REG_GAIN_DB_START, &value);
 	if (err < 0)
 		return err;
-	isight->gain_tlv[2] = (s32)be32_to_cpu(value) * 100;
+	isight->gain_tlv[SNDRV_CTL_TLVO_DB_MINMAX_MIN] =
+						(s32)be32_to_cpu(value) * 100;
 
 	err = reg_read(isight, REG_GAIN_DB_END, &value);
 	if (err < 0)
 		return err;
-	isight->gain_tlv[3] = (s32)be32_to_cpu(value) * 100;
+	isight->gain_tlv[SNDRV_CTL_TLVO_DB_MINMAX_MAX] =
+						(s32)be32_to_cpu(value) * 100;
 
 	ctl = snd_ctl_new1(&gain_control, isight);
 	if (ctl)
diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c
index 47a358f..419e285 100644
--- a/sound/hda/hdac_regmap.c
+++ b/sound/hda/hdac_regmap.c
@@ -65,10 +65,10 @@ static bool hda_writeable_reg(struct device *dev, unsigned int reg)
 {
 	struct hdac_device *codec = dev_to_hdac_dev(dev);
 	unsigned int verb = get_verb(reg);
+	const unsigned int *v;
 	int i;
 
-	for (i = 0; i < codec->vendor_verbs.used; i++) {
-		unsigned int *v = snd_array_elem(&codec->vendor_verbs, i);
+	snd_array_for_each(&codec->vendor_verbs, i, v) {
 		if (verb == *v)
 			return true;
 	}
diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c
index d09e456..de6ef1b 100644
--- a/sound/isa/cmi8328.c
+++ b/sound/isa/cmi8328.c
@@ -192,7 +192,7 @@ static int snd_cmi8328_mixer(struct snd_wss *chip)
 }
 
 /* find index of an item in "-1"-ended array */
-int array_find(int array[], int item)
+static int array_find(int array[], int item)
 {
 	int i;
 
@@ -203,7 +203,7 @@ int array_find(int array[], int item)
 	return -1;
 }
 /* the same for long */
-int array_find_l(long array[], long item)
+static int array_find_l(long array[], long item)
 {
 	int i;
 
diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c
index c09d9b9..a985e91 100644
--- a/sound/isa/sc6000.c
+++ b/sound/isa/sc6000.c
@@ -592,7 +592,7 @@ static int snd_sc6000_probe(struct device *devptr, unsigned int dev)
 	*vport = devm_ioport_map(devptr, port[dev], 0x10);
 	if (*vport == NULL) {
 		snd_printk(KERN_ERR PFX
-			   "I/O port cannot be iomaped.\n");
+			   "I/O port cannot be iomapped.\n");
 		err = -EBUSY;
 		goto err_unmap1;
 	}
@@ -607,7 +607,7 @@ static int snd_sc6000_probe(struct device *devptr, unsigned int dev)
 	vmss_port = devm_ioport_map(devptr, mss_port[dev], 4);
 	if (!vmss_port) {
 		snd_printk(KERN_ERR PFX
-			   "MSS port I/O cannot be iomaped.\n");
+			   "MSS port I/O cannot be iomapped.\n");
 		err = -EBUSY;
 		goto err_unmap2;
 	}
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index 0bf2c04..d9c54c0 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -258,7 +258,7 @@ snd_ad1889_ac97_ready(struct snd_ad1889 *chip)
 	
 	while (!(ad1889_readw(chip, AD_AC97_ACIC) & AD_AC97_ACIC_ACRDY) 
 			&& --retry)
-		mdelay(1);
+		usleep_range(1000, 2000);
 	if (!retry) {
 		dev_err(chip->card->dev, "[%s] Link is not ready.\n",
 			__func__);
@@ -872,7 +872,7 @@ snd_ad1889_init(struct snd_ad1889 *chip)
 	ad1889_writew(chip, AD_DS_CCS, AD_DS_CCS_CLKEN); /* turn on clock */
 	ad1889_readw(chip, AD_DS_CCS);	/* flush posted write */
 
-	mdelay(10);
+	usleep_range(10000, 11000);
 
 	/* enable Master and Target abort interrupts */
 	ad1889_writel(chip, AD_DMA_DISR, AD_DMA_DISR_PMAE | AD_DMA_DISR_PTAE);
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 26a6578..452cc79 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -1139,7 +1139,7 @@ static int save_mixer_state(struct cmipci *cm)
 		struct snd_ctl_elem_value *val;
 		unsigned int i;
 
-		val = kmalloc(sizeof(*val), GFP_ATOMIC);
+		val = kmalloc(sizeof(*val), GFP_KERNEL);
 		if (!val)
 			return -ENOMEM;
 		for (i = 0; i < CM_SAVED_MIXERS; i++) {
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 655fbea..4910d3f 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -58,7 +58,7 @@ MODULE_PARM_DESC(id, "ID string for the CS46xx soundcard.");
 module_param_array(enable, bool, NULL, 0444);
 MODULE_PARM_DESC(enable, "Enable CS46xx soundcard.");
 module_param_array(external_amp, bool, NULL, 0444);
-MODULE_PARM_DESC(external_amp, "Force to enable external amplifer.");
+MODULE_PARM_DESC(external_amp, "Force to enable external amplifier.");
 module_param_array(thinkpad, bool, NULL, 0444);
 MODULE_PARM_DESC(thinkpad, "Force to enable Thinkpad's CLKRUN control.");
 module_param_array(mmap_valid, bool, NULL, 0444);
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index a2b56b1..608ff48 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -421,14 +421,10 @@ int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu,
 					    snd_fx8010_irq_handler_t *handler,
 					    unsigned char gpr_running,
 					    void *private_data,
-					    struct snd_emu10k1_fx8010_irq **r_irq)
+					    struct snd_emu10k1_fx8010_irq *irq)
 {
-	struct snd_emu10k1_fx8010_irq *irq;
 	unsigned long flags;
 	
-	irq = kmalloc(sizeof(*irq), GFP_ATOMIC);
-	if (irq == NULL)
-		return -ENOMEM;
 	irq->handler = handler;
 	irq->gpr_running = gpr_running;
 	irq->private_data = private_data;
@@ -443,8 +439,6 @@ int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu,
 		emu->fx8010.irq_handlers = irq;
 	}
 	spin_unlock_irqrestore(&emu->fx8010.irq_lock, flags);
-	if (r_irq)
-		*r_irq = irq;
 	return 0;
 }
 
@@ -468,7 +462,6 @@ int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu,
 			tmp->next = tmp->next->next;
 	}
 	spin_unlock_irqrestore(&emu->fx8010.irq_lock, flags);
-	kfree(irq);
 	return 0;
 }
 
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index cefe613..d39458a 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1724,7 +1724,7 @@ static int snd_emu10k1_fx8010_playback_trigger(struct snd_pcm_substream *substre
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-		snd_emu10k1_fx8010_unregister_irq_handler(emu, pcm->irq); pcm->irq = NULL;
+		snd_emu10k1_fx8010_unregister_irq_handler(emu, &pcm->irq);
 		snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_trigger, 0, 0);
 		pcm->tram_pos = INITIAL_TRAM_POS(pcm->buffer_size);
 		pcm->tram_shift = 0;
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index f7a492c..4235907 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -127,11 +127,15 @@
 
 config SND_HDA_CODEC_HDMI
 	tristate "Build HDMI/DisplayPort HD-audio codec support"
+	select SND_DYNAMIC_MINORS
 	help
 	  Say Y or M here to include HDMI and DisplayPort HD-audio codec
 	  support in snd-hda-intel driver.  This includes all AMD/ATI,
 	  Intel and Nvidia HDMI/DisplayPort codecs.
 
+	  Note that this option mandatorily enables CONFIG_SND_DYNAMIC_MINORS
+	  to assure the multiple streams for DP-MST support.
+
 comment "Set to Y if you want auto-loading the codec driver"
 	depends on SND_HDA=y && SND_HDA_CODEC_HDMI=m
 
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index d3ea731..b9a6b66 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -793,11 +793,11 @@ EXPORT_SYMBOL_GPL(snd_hda_add_verbs);
  */
 void snd_hda_apply_verbs(struct hda_codec *codec)
 {
+	const struct hda_verb **v;
 	int i;
-	for (i = 0; i < codec->verbs.used; i++) {
-		struct hda_verb **v = snd_array_elem(&codec->verbs, i);
+
+	snd_array_for_each(&codec->verbs, i, v)
 		snd_hda_sequence_write(codec, *v);
-	}
 }
 EXPORT_SYMBOL_GPL(snd_hda_apply_verbs);
 
@@ -890,10 +890,10 @@ EXPORT_SYMBOL_GPL(snd_hda_apply_fixup);
 static bool pin_config_match(struct hda_codec *codec,
 			     const struct hda_pintbl *pins)
 {
+	const struct hda_pincfg *pin;
 	int i;
 
-	for (i = 0; i < codec->init_pins.used; i++) {
-		struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+	snd_array_for_each(&codec->init_pins, i, pin) {
 		hda_nid_t nid = pin->nid;
 		u32 cfg = pin->cfg;
 		const struct hda_pintbl *t_pins;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 5bc3a74..08151f3 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -481,9 +481,10 @@ static struct hda_pincfg *look_up_pincfg(struct hda_codec *codec,
 					 struct snd_array *array,
 					 hda_nid_t nid)
 {
+	struct hda_pincfg *pin;
 	int i;
-	for (i = 0; i < array->used; i++) {
-		struct hda_pincfg *pin = snd_array_elem(array, i);
+
+	snd_array_for_each(array, i, pin) {
 		if (pin->nid == nid)
 			return pin;
 	}
@@ -618,14 +619,15 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_get_pin_target);
  */
 void snd_hda_shutup_pins(struct hda_codec *codec)
 {
+	const struct hda_pincfg *pin;
 	int i;
+
 	/* don't shut up pins when unloading the driver; otherwise it breaks
 	 * the default pin setup at the next load of the driver
 	 */
 	if (codec->bus->shutdown)
 		return;
-	for (i = 0; i < codec->init_pins.used; i++) {
-		struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+	snd_array_for_each(&codec->init_pins, i, pin) {
 		/* use read here for syncing after issuing each verb */
 		snd_hda_codec_read(codec, pin->nid, 0,
 				   AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
@@ -638,13 +640,14 @@ EXPORT_SYMBOL_GPL(snd_hda_shutup_pins);
 /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
 static void restore_shutup_pins(struct hda_codec *codec)
 {
+	const struct hda_pincfg *pin;
 	int i;
+
 	if (!codec->pins_shutup)
 		return;
 	if (codec->bus->shutdown)
 		return;
-	for (i = 0; i < codec->init_pins.used; i++) {
-		struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+	snd_array_for_each(&codec->init_pins, i, pin) {
 		snd_hda_codec_write(codec, pin->nid, 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL,
 				    pin->ctrl);
@@ -697,8 +700,7 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid)
 	struct hda_cvt_setup *p;
 	int i;
 
-	for (i = 0; i < codec->cvt_setups.used; i++) {
-		p = snd_array_elem(&codec->cvt_setups, i);
+	snd_array_for_each(&codec->cvt_setups, i, p) {
 		if (p->nid == nid)
 			return p;
 	}
@@ -1076,8 +1078,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 	/* make other inactive cvts with the same stream-tag dirty */
 	type = get_wcaps_type(get_wcaps(codec, nid));
 	list_for_each_codec(c, codec->bus) {
-		for (i = 0; i < c->cvt_setups.used; i++) {
-			p = snd_array_elem(&c->cvt_setups, i);
+		snd_array_for_each(&c->cvt_setups, i, p) {
 			if (!p->active && p->stream_tag == stream_tag &&
 			    get_wcaps_type(get_wcaps(c, p->nid)) == type)
 				p->dirty = 1;
@@ -1140,12 +1141,11 @@ static void really_cleanup_stream(struct hda_codec *codec,
 static void purify_inactive_streams(struct hda_codec *codec)
 {
 	struct hda_codec *c;
+	struct hda_cvt_setup *p;
 	int i;
 
 	list_for_each_codec(c, codec->bus) {
-		for (i = 0; i < c->cvt_setups.used; i++) {
-			struct hda_cvt_setup *p;
-			p = snd_array_elem(&c->cvt_setups, i);
+		snd_array_for_each(&c->cvt_setups, i, p) {
 			if (p->dirty)
 				really_cleanup_stream(c, p);
 		}
@@ -1156,10 +1156,10 @@ static void purify_inactive_streams(struct hda_codec *codec)
 /* clean up all streams; called from suspend */
 static void hda_cleanup_all_streams(struct hda_codec *codec)
 {
+	struct hda_cvt_setup *p;
 	int i;
 
-	for (i = 0; i < codec->cvt_setups.used; i++) {
-		struct hda_cvt_setup *p = snd_array_elem(&codec->cvt_setups, i);
+	snd_array_for_each(&codec->cvt_setups, i, p) {
 		if (p->stream_tag)
 			really_cleanup_stream(codec, p);
 	}
@@ -1493,10 +1493,10 @@ static void get_ctl_amp_tlv(struct snd_kcontrol *kcontrol, unsigned int *tlv)
 	val1 = ((int)val1) * ((int)val2);
 	if (min_mute || (caps & AC_AMPCAP_MIN_MUTE))
 		val2 |= TLV_DB_SCALE_MUTE;
-	tlv[0] = SNDRV_CTL_TLVT_DB_SCALE;
-	tlv[1] = 2 * sizeof(unsigned int);
-	tlv[2] = val1;
-	tlv[3] = val2;
+	tlv[SNDRV_CTL_TLVO_TYPE] = SNDRV_CTL_TLVT_DB_SCALE;
+	tlv[SNDRV_CTL_TLVO_LEN] = 2 * sizeof(unsigned int);
+	tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] = val1;
+	tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] = val2;
 }
 
 /**
@@ -1544,10 +1544,10 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
 	nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
 	step = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT;
 	step = (step + 1) * 25;
-	tlv[0] = SNDRV_CTL_TLVT_DB_SCALE;
-	tlv[1] = 2 * sizeof(unsigned int);
-	tlv[2] = -nums * step;
-	tlv[3] = step;
+	tlv[SNDRV_CTL_TLVO_TYPE] = SNDRV_CTL_TLVT_DB_SCALE;
+	tlv[SNDRV_CTL_TLVO_LEN] = 2 * sizeof(unsigned int);
+	tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] = -nums * step;
+	tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] = step;
 }
 EXPORT_SYMBOL_GPL(snd_hda_set_vmaster_tlv);
 
@@ -1845,10 +1845,10 @@ static int init_slave_0dB(struct snd_kcontrol *slave,
 	} else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ)
 		tlv = kctl->tlv.p;
 
-	if (!tlv || tlv[0] != SNDRV_CTL_TLVT_DB_SCALE)
+	if (!tlv || tlv[SNDRV_CTL_TLVO_TYPE] != SNDRV_CTL_TLVT_DB_SCALE)
 		return 0;
 
-	step = tlv[3];
+	step = tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP];
 	step &= ~TLV_DB_SCALE_MUTE;
 	if (!step)
 		return 0;
@@ -1860,7 +1860,7 @@ static int init_slave_0dB(struct snd_kcontrol *slave,
 	}
 
 	arg->step = step;
-	val = -tlv[2] / step;
+	val = -tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] / step;
 	if (val > 0) {
 		put_kctl_with_value(slave, val);
 		return val;
@@ -2175,6 +2175,8 @@ static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol,
 	int idx = kcontrol->private_value;
 	struct hda_spdif_out *spdif;
 
+	if (WARN_ON(codec->spdif_out.used <= idx))
+		return -EINVAL;
 	mutex_lock(&codec->spdif_mutex);
 	spdif = snd_array_elem(&codec->spdif_out, idx);
 	ucontrol->value.iec958.status[0] = spdif->status & 0xff;
@@ -2282,6 +2284,8 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol,
 	unsigned short val;
 	int change;
 
+	if (WARN_ON(codec->spdif_out.used <= idx))
+		return -EINVAL;
 	mutex_lock(&codec->spdif_mutex);
 	spdif = snd_array_elem(&codec->spdif_out, idx);
 	nid = spdif->nid;
@@ -2308,6 +2312,8 @@ static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol,
 	int idx = kcontrol->private_value;
 	struct hda_spdif_out *spdif;
 
+	if (WARN_ON(codec->spdif_out.used <= idx))
+		return -EINVAL;
 	mutex_lock(&codec->spdif_mutex);
 	spdif = snd_array_elem(&codec->spdif_out, idx);
 	ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE;
@@ -2336,6 +2342,8 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol,
 	unsigned short val;
 	int change;
 
+	if (WARN_ON(codec->spdif_out.used <= idx))
+		return -EINVAL;
 	mutex_lock(&codec->spdif_mutex);
 	spdif = snd_array_elem(&codec->spdif_out, idx);
 	nid = spdif->nid;
@@ -2461,10 +2469,10 @@ EXPORT_SYMBOL_GPL(snd_hda_create_dig_out_ctls);
 struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
 					       hda_nid_t nid)
 {
+	struct hda_spdif_out *spdif;
 	int i;
-	for (i = 0; i < codec->spdif_out.used; i++) {
-		struct hda_spdif_out *spdif =
-				snd_array_elem(&codec->spdif_out, i);
+
+	snd_array_for_each(&codec->spdif_out, i, spdif) {
 		if (spdif->nid == nid)
 			return spdif;
 	}
@@ -2483,6 +2491,8 @@ void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
 {
 	struct hda_spdif_out *spdif;
 
+	if (WARN_ON(codec->spdif_out.used <= idx))
+		return;
 	mutex_lock(&codec->spdif_mutex);
 	spdif = snd_array_elem(&codec->spdif_out, idx);
 	spdif->nid = (u16)-1;
@@ -2503,6 +2513,8 @@ void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid)
 	struct hda_spdif_out *spdif;
 	unsigned short val;
 
+	if (WARN_ON(codec->spdif_out.used <= idx))
+		return;
 	mutex_lock(&codec->spdif_mutex);
 	spdif = snd_array_elem(&codec->spdif_out, idx);
 	if (spdif->nid != nid) {
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 5cc6509..db773e2 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -264,10 +264,10 @@ static struct nid_path *get_nid_path(struct hda_codec *codec,
 				     int anchor_nid)
 {
 	struct hda_gen_spec *spec = codec->spec;
+	struct nid_path *path;
 	int i;
 
-	for (i = 0; i < spec->paths.used; i++) {
-		struct nid_path *path = snd_array_elem(&spec->paths, i);
+	snd_array_for_each(&spec->paths, i, path) {
 		if (path->depth <= 0)
 			continue;
 		if ((!from_nid || path->path[0] == from_nid) &&
@@ -325,10 +325,10 @@ EXPORT_SYMBOL_GPL(snd_hda_get_path_from_idx);
 static bool is_dac_already_used(struct hda_codec *codec, hda_nid_t nid)
 {
 	struct hda_gen_spec *spec = codec->spec;
+	const struct nid_path *path;
 	int i;
 
-	for (i = 0; i < spec->paths.used; i++) {
-		struct nid_path *path = snd_array_elem(&spec->paths, i);
+	snd_array_for_each(&spec->paths, i, path) {
 		if (path->path[0] == nid)
 			return true;
 	}
@@ -351,11 +351,11 @@ static bool is_reachable_path(struct hda_codec *codec,
 static bool is_ctl_used(struct hda_codec *codec, unsigned int val, int type)
 {
 	struct hda_gen_spec *spec = codec->spec;
+	const struct nid_path *path;
 	int i;
 
 	val &= AMP_VAL_COMPARE_MASK;
-	for (i = 0; i < spec->paths.used; i++) {
-		struct nid_path *path = snd_array_elem(&spec->paths, i);
+	snd_array_for_each(&spec->paths, i, path) {
 		if ((path->ctls[type] & AMP_VAL_COMPARE_MASK) == val)
 			return true;
 	}
@@ -638,13 +638,13 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid,
 {
 	struct hda_gen_spec *spec = codec->spec;
 	int type = get_wcaps_type(get_wcaps(codec, nid));
+	const struct nid_path *path;
 	int i, n;
 
 	if (nid == codec->core.afg)
 		return true;
 
-	for (n = 0; n < spec->paths.used; n++) {
-		struct nid_path *path = snd_array_elem(&spec->paths, n);
+	snd_array_for_each(&spec->paths, n, path) {
 		if (!path->active)
 			continue;
 		if (codec->power_save_node) {
@@ -2065,7 +2065,7 @@ static int parse_output_paths(struct hda_codec *codec)
 			snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
 						HDA_OUTPUT, spec->vmaster_tlv);
 			if (spec->dac_min_mute)
-				spec->vmaster_tlv[3] |= TLV_DB_SCALE_MUTE;
+				spec->vmaster_tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] |= TLV_DB_SCALE_MUTE;
 		}
 	}
 
@@ -2696,10 +2696,10 @@ static const struct snd_kcontrol_new out_jack_mode_enum = {
 static bool find_kctl_name(struct hda_codec *codec, const char *name, int idx)
 {
 	struct hda_gen_spec *spec = codec->spec;
+	const struct snd_kcontrol_new *kctl;
 	int i;
 
-	for (i = 0; i < spec->kctls.used; i++) {
-		struct snd_kcontrol_new *kctl = snd_array_elem(&spec->kctls, i);
+	snd_array_for_each(&spec->kctls, i, kctl) {
 		if (!strcmp(kctl->name, name) && kctl->index == idx)
 			return true;
 	}
@@ -4021,8 +4021,7 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid,
 	struct nid_path *path;
 	int n;
 
-	for (n = 0; n < spec->paths.used; n++) {
-		path = snd_array_elem(&spec->paths, n);
+	snd_array_for_each(&spec->paths, n, path) {
 		if (!path->depth)
 			continue;
 		if (path->path[0] == nid ||
@@ -5831,10 +5830,10 @@ static void init_digital(struct hda_codec *codec)
  */
 static void clear_unsol_on_unused_pins(struct hda_codec *codec)
 {
+	const struct hda_pincfg *pin;
 	int i;
 
-	for (i = 0; i < codec->init_pins.used; i++) {
-		struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+	snd_array_for_each(&codec->init_pins, i, pin) {
 		hda_nid_t nid = pin->nid;
 		if (is_jack_detectable(codec, nid) &&
 		    !snd_hda_jack_tbl_get(codec, nid))
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index 9b7efec..6ec79c5 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -80,10 +80,10 @@ static ssize_t pin_configs_show(struct hda_codec *codec,
 				struct snd_array *list,
 				char *buf)
 {
+	const struct hda_pincfg *pin;
 	int i, len = 0;
 	mutex_lock(&codec->user_mutex);
-	for (i = 0; i < list->used; i++) {
-		struct hda_pincfg *pin = snd_array_elem(list, i);
+	snd_array_for_each(list, i, pin) {
 		len += sprintf(buf + len, "0x%02x 0x%08x\n",
 			       pin->nid, pin->cfg);
 	}
@@ -217,10 +217,10 @@ static ssize_t init_verbs_show(struct device *dev,
 			       char *buf)
 {
 	struct hda_codec *codec = dev_get_drvdata(dev);
+	const struct hda_verb *v;
 	int i, len = 0;
 	mutex_lock(&codec->user_mutex);
-	for (i = 0; i < codec->init_verbs.used; i++) {
-		struct hda_verb *v = snd_array_elem(&codec->init_verbs, i);
+	snd_array_for_each(&codec->init_verbs, i, v) {
 		len += snprintf(buf + len, PAGE_SIZE - len,
 				"0x%02x 0x%03x 0x%04x\n",
 				v->nid, v->verb, v->param);
@@ -267,10 +267,10 @@ static ssize_t hints_show(struct device *dev,
 			  char *buf)
 {
 	struct hda_codec *codec = dev_get_drvdata(dev);
+	const struct hda_hint *hint;
 	int i, len = 0;
 	mutex_lock(&codec->user_mutex);
-	for (i = 0; i < codec->hints.used; i++) {
-		struct hda_hint *hint = snd_array_elem(&codec->hints, i);
+	snd_array_for_each(&codec->hints, i, hint) {
 		len += snprintf(buf + len, PAGE_SIZE - len,
 				"%s = %s\n", hint->key, hint->val);
 	}
@@ -280,10 +280,10 @@ static ssize_t hints_show(struct device *dev,
 
 static struct hda_hint *get_hint(struct hda_codec *codec, const char *key)
 {
+	struct hda_hint *hint;
 	int i;
 
-	for (i = 0; i < codec->hints.used; i++) {
-		struct hda_hint *hint = snd_array_elem(&codec->hints, i);
+	snd_array_for_each(&codec->hints, i, hint) {
 		if (!strcmp(hint->key, key))
 			return hint;
 	}
@@ -783,13 +783,13 @@ void snd_hda_sysfs_init(struct hda_codec *codec)
 void snd_hda_sysfs_clear(struct hda_codec *codec)
 {
 #ifdef CONFIG_SND_HDA_RECONFIG
+	struct hda_hint *hint;
 	int i;
 
 	/* clear init verbs */
 	snd_array_free(&codec->init_verbs);
 	/* clear hints */
-	for (i = 0; i < codec->hints.used; i++) {
-		struct hda_hint *hint = snd_array_elem(&codec->hints, i);
+	snd_array_for_each(&codec->hints, i, hint) {
 		kfree(hint->key); /* we don't need to free hint->val */
 	}
 	snd_array_free(&codec->hints);
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 768ea86..1cc4f51 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -28,6 +28,9 @@
 #include <linux/module.h>
 #include <linux/firmware.h>
 #include <linux/kernel.h>
+#include <linux/types.h>
+#include <linux/io.h>
+#include <linux/pci.h>
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
@@ -39,9 +42,15 @@
 /* Enable this to see controls for tuning purpose. */
 /*#define ENABLE_TUNING_CONTROLS*/
 
+#ifdef ENABLE_TUNING_CONTROLS
+#include <sound/tlv.h>
+#endif
+
 #define FLOAT_ZERO	0x00000000
 #define FLOAT_ONE	0x3f800000
 #define FLOAT_TWO	0x40000000
+#define FLOAT_THREE     0x40400000
+#define FLOAT_EIGHT     0x41000000
 #define FLOAT_MINUS_5	0xc0a00000
 
 #define UNSOL_TAG_DSP	0x16
@@ -72,16 +81,22 @@
 #define SCP_GET    1
 
 #define EFX_FILE   "ctefx.bin"
+#define SBZ_EFX_FILE   "ctefx-sbz.bin"
+#define R3DI_EFX_FILE  "ctefx-r3di.bin"
 
 #ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP
 MODULE_FIRMWARE(EFX_FILE);
+MODULE_FIRMWARE(SBZ_EFX_FILE);
+MODULE_FIRMWARE(R3DI_EFX_FILE);
 #endif
 
-static char *dirstr[2] = { "Playback", "Capture" };
+static const char *dirstr[2] = { "Playback", "Capture" };
 
+#define NUM_OF_OUTPUTS 3
 enum {
 	SPEAKER_OUT,
-	HEADPHONE_OUT
+	HEADPHONE_OUT,
+	SURROUND_OUT
 };
 
 enum {
@@ -89,6 +104,15 @@ enum {
 	LINE_MIC_IN
 };
 
+/* Strings for Input Source Enum Control */
+static const char *in_src_str[3] = {"Rear Mic", "Line", "Front Mic" };
+#define IN_SRC_NUM_OF_INPUTS 3
+enum {
+	REAR_MIC,
+	REAR_LINE_IN,
+	FRONT_MIC,
+};
+
 enum {
 #define VNODE_START_NID    0x80
 	VNID_SPK = VNODE_START_NID,			/* Speaker vnid */
@@ -122,13 +146,28 @@ enum {
 	VOICEFX = IN_EFFECT_END_NID,
 	PLAY_ENHANCEMENT,
 	CRYSTAL_VOICE,
-	EFFECT_END_NID
+	EFFECT_END_NID,
+	OUTPUT_SOURCE_ENUM,
+	INPUT_SOURCE_ENUM,
+	XBASS_XOVER,
+	EQ_PRESET_ENUM,
+	SMART_VOLUME_ENUM,
+	MIC_BOOST_ENUM
 #define EFFECTS_COUNT  (EFFECT_END_NID - EFFECT_START_NID)
 };
 
 /* Effects values size*/
 #define EFFECT_VALS_MAX_COUNT 12
 
+/*
+ * Default values for the effect slider controls, they are in order of their
+ * effect NID's. Surround, Crystalizer, Dialog Plus, Smart Volume, and then
+ * X-bass.
+ */
+static const unsigned int effect_slider_defaults[] = {67, 65, 50, 74, 50};
+/* Amount of effect level sliders for ca0132_alt controls. */
+#define EFFECT_LEVEL_SLIDERS 5
+
 /* Latency introduced by DSP blocks in milliseconds. */
 #define DSP_CAPTURE_INIT_LATENCY        0
 #define DSP_CRYSTAL_VOICE_LATENCY       124
@@ -472,6 +511,161 @@ static struct ct_voicefx_preset ca0132_voicefx_presets[] = {
 	}
 };
 
+/* ca0132 EQ presets, taken from Windows Sound Blaster Z Driver */
+
+#define EQ_PRESET_MAX_PARAM_COUNT 11
+
+struct ct_eq {
+	char *name;
+	hda_nid_t nid;
+	int mid;
+	int reqs[EQ_PRESET_MAX_PARAM_COUNT]; /*effect module request*/
+};
+
+struct ct_eq_preset {
+	char *name; /*preset name*/
+	unsigned int vals[EQ_PRESET_MAX_PARAM_COUNT];
+};
+
+static struct ct_eq ca0132_alt_eq_enum = {
+	.name = "FX: Equalizer Preset Switch",
+	.nid = EQ_PRESET_ENUM,
+	.mid = 0x96,
+	.reqs = {10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20}
+};
+
+
+static struct ct_eq_preset ca0132_alt_eq_presets[] = {
+	{ .name = "Flat",
+	 .vals = { 0x00000000, 0x00000000, 0x00000000,
+		   0x00000000, 0x00000000, 0x00000000,
+		   0x00000000, 0x00000000, 0x00000000,
+		   0x00000000, 0x00000000	     }
+	},
+	{ .name = "Acoustic",
+	 .vals = { 0x00000000, 0x00000000, 0x3F8CCCCD,
+		   0x40000000, 0x00000000, 0x00000000,
+		   0x00000000, 0x00000000, 0x40000000,
+		   0x40000000, 0x40000000	     }
+	},
+	{ .name = "Classical",
+	 .vals = { 0x00000000, 0x00000000, 0x40C00000,
+		   0x40C00000, 0x40466666, 0x00000000,
+		   0x00000000, 0x00000000, 0x00000000,
+		   0x40466666, 0x40466666	     }
+	},
+	{ .name = "Country",
+	 .vals = { 0x00000000, 0xBF99999A, 0x00000000,
+		   0x3FA66666, 0x3FA66666, 0x3F8CCCCD,
+		   0x00000000, 0x00000000, 0x40000000,
+		   0x40466666, 0x40800000	     }
+	},
+	{ .name = "Dance",
+	 .vals = { 0x00000000, 0xBF99999A, 0x40000000,
+		   0x40466666, 0x40866666, 0xBF99999A,
+		   0xBF99999A, 0x00000000, 0x00000000,
+		   0x40800000, 0x40800000	     }
+	},
+	{ .name = "Jazz",
+	 .vals = { 0x00000000, 0x00000000, 0x00000000,
+		   0x3F8CCCCD, 0x40800000, 0x40800000,
+		   0x40800000, 0x00000000, 0x3F8CCCCD,
+		   0x40466666, 0x40466666	     }
+	},
+	{ .name = "New Age",
+	 .vals = { 0x00000000, 0x00000000, 0x40000000,
+		   0x40000000, 0x00000000, 0x00000000,
+		   0x00000000, 0x3F8CCCCD, 0x40000000,
+		   0x40000000, 0x40000000	     }
+	},
+	{ .name = "Pop",
+	 .vals = { 0x00000000, 0xBFCCCCCD, 0x00000000,
+		   0x40000000, 0x40000000, 0x00000000,
+		   0xBF99999A, 0xBF99999A, 0x00000000,
+		   0x40466666, 0x40C00000	     }
+	},
+	{ .name = "Rock",
+	 .vals = { 0x00000000, 0xBF99999A, 0xBF99999A,
+		   0x3F8CCCCD, 0x40000000, 0xBF99999A,
+		   0xBF99999A, 0x00000000, 0x00000000,
+		   0x40800000, 0x40800000	     }
+	},
+	{ .name = "Vocal",
+	 .vals = { 0x00000000, 0xC0000000, 0xBF99999A,
+		   0xBF99999A, 0x00000000, 0x40466666,
+		   0x40800000, 0x40466666, 0x00000000,
+		   0x00000000, 0x3F8CCCCD	     }
+	}
+};
+
+/* 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];
+};
+
+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 }
+	},
+	{ .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 = "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 }
+	}
+};
+
+/*
+ * DSP volume setting structs. Req 1 is left volume, req 2 is right volume,
+ * and I don't know what the third req is, but it's always zero. I assume it's
+ * some sort of update or set command to tell the DSP there's new volume info.
+ */
+#define DSP_VOL_OUT 0
+#define DSP_VOL_IN  1
+
+struct ct_dsp_volume_ctl {
+	hda_nid_t vnid;
+	int mid; /* module ID*/
+	unsigned int reqs[3]; /* scp req ID */
+};
+
+static struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = {
+	{ .vnid = VNID_SPK,
+	  .mid = 0x32,
+	  .reqs = {3, 4, 2}
+	},
+	{ .vnid = VNID_MIC,
+	  .mid = 0x37,
+	  .reqs = {2, 3, 1}
+	}
+};
+
 enum hda_cmd_vendor_io {
 	/* for DspIO node */
 	VENDOR_DSPIO_SCP_WRITE_DATA_LOW      = 0x000,
@@ -703,6 +897,7 @@ struct ca0132_spec {
 	const struct hda_verb *base_init_verbs;
 	const struct hda_verb *base_exit_verbs;
 	const struct hda_verb *chip_init_verbs;
+	const struct hda_verb *sbz_init_verbs;
 	struct hda_verb *spec_init_verbs;
 	struct auto_pin_cfg autocfg;
 
@@ -719,6 +914,7 @@ struct ca0132_spec {
 	hda_nid_t shared_mic_nid;
 	hda_nid_t shared_out_nid;
 	hda_nid_t unsol_tag_hp;
+	hda_nid_t unsol_tag_front_hp; /* for desktop ca0132 codecs */
 	hda_nid_t unsol_tag_amic1;
 
 	/* chip access */
@@ -734,6 +930,9 @@ struct ca0132_spec {
 	unsigned int scp_resp_header;
 	unsigned int scp_resp_data[4];
 	unsigned int scp_resp_count;
+	bool alt_firmware_present;
+	bool startup_check_entered;
+	bool dsp_reload;
 
 	/* mixer and effects related */
 	unsigned char dmic_ctl;
@@ -746,6 +945,17 @@ struct ca0132_spec {
 	long effects_switch[EFFECTS_COUNT];
 	long voicefx_val;
 	long cur_mic_boost;
+	/* ca0132_alt control related values */
+	unsigned char in_enum_val;
+	unsigned char out_enum_val;
+	unsigned char mic_boost_enum_val;
+	unsigned char smart_volume_setting;
+	long fx_ctl_val[EFFECT_LEVEL_SLIDERS];
+	long xbass_xover_freq;
+	long eq_preset_val;
+	unsigned int tlv[4];
+	struct hda_vmaster_mute_hook vmaster_mute;
+
 
 	struct hda_codec *codec;
 	struct delayed_work unsol_hp_work;
@@ -754,6 +964,25 @@ struct ca0132_spec {
 #ifdef ENABLE_TUNING_CONTROLS
 	long cur_ctl_vals[TUNING_CTLS_COUNT];
 #endif
+	/*
+	 * Sound Blaster Z PCI region 2 iomem, used for input and output
+	 * switching, and other unknown commands.
+	 */
+	void __iomem *mem_base;
+
+	/*
+	 * Whether or not to use the alt functions like alt_select_out,
+	 * alt_select_in, etc. Only used on desktop codecs for now, because of
+	 * surround sound support.
+	 */
+	bool use_alt_functions;
+
+	/*
+	 * Whether or not to use alt controls:	volume effect sliders, EQ
+	 * presets, smart volume presets, and new control names with FX prefix.
+	 * Renames PlayEnhancement and CrystalVoice too.
+	 */
+	bool use_alt_controls;
 };
 
 /*
@@ -762,6 +991,8 @@ struct ca0132_spec {
 enum {
 	QUIRK_NONE,
 	QUIRK_ALIENWARE,
+	QUIRK_SBZ,
+	QUIRK_R3DI,
 };
 
 static const struct hda_pintbl alienware_pincfgs[] = {
@@ -778,10 +1009,44 @@ static const struct hda_pintbl alienware_pincfgs[] = {
 	{}
 };
 
+/* Sound Blaster Z pin configs taken from Windows Driver */
+static const struct hda_pintbl sbz_pincfgs[] = {
+	{ 0x0b, 0x01017010 }, /* Port G -- Lineout FRONT L/R */
+	{ 0x0c, 0x014510f0 }, /* SPDIF Out 1 */
+	{ 0x0d, 0x014510f0 }, /* Digital Out */
+	{ 0x0e, 0x01c510f0 }, /* SPDIF In */
+	{ 0x0f, 0x0221701f }, /* Port A -- BackPanel HP */
+	{ 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */
+	{ 0x11, 0x01017014 }, /* Port B -- LineMicIn2 / Rear L/R */
+	{ 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */
+	{ 0x13, 0x908700f0 }, /* What U Hear In*/
+	{ 0x18, 0x50d000f0 }, /* N/A */
+	{}
+};
+
+/* Recon3D integrated pin configs taken from Windows Driver */
+static const struct hda_pintbl r3di_pincfgs[] = {
+	{ 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */
+	{ 0x0c, 0x014510f0 }, /* SPDIF Out 1 */
+	{ 0x0d, 0x014510f0 }, /* Digital Out */
+	{ 0x0e, 0x41c520f0 }, /* SPDIF In */
+	{ 0x0f, 0x0221401f }, /* Port A -- BackPanel HP */
+	{ 0x10, 0x01016011 }, /* Port D -- Center/LFE or FP Hp */
+	{ 0x11, 0x01011014 }, /* Port B -- LineMicIn2 / Rear L/R */
+	{ 0x12, 0x02a090f0 }, /* Port C -- LineIn1 */
+	{ 0x13, 0x908700f0 }, /* What U Hear In*/
+	{ 0x18, 0x500000f0 }, /* N/A */
+	{}
+};
+
 static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE),
 	SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE),
 	SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE),
+	SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ),
+	SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ),
+	SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI),
+	SND_PCI_QUIRK(0x1458, 0xA036, "Recon3Di", QUIRK_R3DI),
 	{}
 };
 
@@ -965,6 +1230,29 @@ static int chipio_write(struct hda_codec *codec,
 }
 
 /*
+ * Write given value to the given address through the chip I/O widget.
+ * not protected by the Mutex
+ */
+static int chipio_write_no_mutex(struct hda_codec *codec,
+		unsigned int chip_addx, const unsigned int data)
+{
+	int err;
+
+
+	/* write the address, and if successful proceed to write data */
+	err = chipio_write_address(codec, chip_addx);
+	if (err < 0)
+		goto exit;
+
+	err = chipio_write_data(codec, data);
+	if (err < 0)
+		goto exit;
+
+exit:
+	return err;
+}
+
+/*
  * Write multiple values to the given address through the chip I/O widget.
  * protected by the Mutex
  */
@@ -1058,6 +1346,81 @@ static void chipio_set_control_param(struct hda_codec *codec,
 }
 
 /*
+ * Set chip parameters through the chip I/O widget. NO MUTEX.
+ */
+static void chipio_set_control_param_no_mutex(struct hda_codec *codec,
+		enum control_param_id param_id, int param_val)
+{
+	int val;
+
+	if ((param_id < 32) && (param_val < 8)) {
+		val = (param_val << 5) | (param_id);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_PARAM_SET, val);
+	} else {
+		if (chipio_send(codec, VENDOR_CHIPIO_STATUS, 0) == 0) {
+			snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+					    VENDOR_CHIPIO_PARAM_EX_ID_SET,
+					    param_id);
+			snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+					    VENDOR_CHIPIO_PARAM_EX_VALUE_SET,
+					    param_val);
+		}
+	}
+}
+/*
+ * Connect stream to a source point, and then connect
+ * that source point to a destination point.
+ */
+static void chipio_set_stream_source_dest(struct hda_codec *codec,
+				int streamid, int source_point, int dest_point)
+{
+	chipio_set_control_param_no_mutex(codec,
+			CONTROL_PARAM_STREAM_ID, streamid);
+	chipio_set_control_param_no_mutex(codec,
+			CONTROL_PARAM_STREAM_SOURCE_CONN_POINT, source_point);
+	chipio_set_control_param_no_mutex(codec,
+			CONTROL_PARAM_STREAM_DEST_CONN_POINT, dest_point);
+}
+
+/*
+ * Set number of channels in the selected stream.
+ */
+static void chipio_set_stream_channels(struct hda_codec *codec,
+				int streamid, unsigned int channels)
+{
+	chipio_set_control_param_no_mutex(codec,
+			CONTROL_PARAM_STREAM_ID, streamid);
+	chipio_set_control_param_no_mutex(codec,
+			CONTROL_PARAM_STREAMS_CHANNELS, channels);
+}
+
+/*
+ * Enable/Disable audio stream.
+ */
+static void chipio_set_stream_control(struct hda_codec *codec,
+				int streamid, int enable)
+{
+	chipio_set_control_param_no_mutex(codec,
+			CONTROL_PARAM_STREAM_ID, streamid);
+	chipio_set_control_param_no_mutex(codec,
+			CONTROL_PARAM_STREAM_CONTROL, enable);
+}
+
+
+/*
+ * Set sampling rate of the connection point. NO MUTEX.
+ */
+static void chipio_set_conn_rate_no_mutex(struct hda_codec *codec,
+				int connid, enum ca0132_sample_rate rate)
+{
+	chipio_set_control_param_no_mutex(codec,
+			CONTROL_PARAM_CONN_POINT_ID, connid);
+	chipio_set_control_param_no_mutex(codec,
+			CONTROL_PARAM_CONN_POINT_SAMPLE_RATE, rate);
+}
+
+/*
  * Set sampling rate of the connection point.
  */
 static void chipio_set_conn_rate(struct hda_codec *codec,
@@ -1420,8 +1783,8 @@ static int dspio_send_scp_message(struct hda_codec *codec,
  * Returns zero or a negative error code.
  */
 static int dspio_scp(struct hda_codec *codec,
-		int mod_id, int req, int dir, void *data, unsigned int len,
-		void *reply, unsigned int *reply_len)
+		int mod_id, int src_id, int req, int dir, const void *data,
+		unsigned int len, void *reply, unsigned int *reply_len)
 {
 	int status = 0;
 	struct scp_msg scp_send, scp_reply;
@@ -1445,7 +1808,7 @@ static int dspio_scp(struct hda_codec *codec,
 		return -EINVAL;
 	}
 
-	scp_send.hdr = make_scp_header(mod_id, 0x20, (dir == SCP_GET), req,
+	scp_send.hdr = make_scp_header(mod_id, src_id, (dir == SCP_GET), req,
 				       0, 0, 0, len/sizeof(unsigned int));
 	if (data != NULL && len > 0) {
 		len = min((unsigned int)(sizeof(scp_send.data)), len);
@@ -1502,15 +1865,24 @@ static int dspio_scp(struct hda_codec *codec,
  * Set DSP parameters
  */
 static int dspio_set_param(struct hda_codec *codec, int mod_id,
-			int req, void *data, unsigned int len)
+			int src_id, int req, const void *data, unsigned int len)
 {
-	return dspio_scp(codec, mod_id, req, SCP_SET, data, len, NULL, NULL);
+	return dspio_scp(codec, mod_id, src_id, req, SCP_SET, data, len, NULL,
+			NULL);
 }
 
 static int dspio_set_uint_param(struct hda_codec *codec, int mod_id,
-			int req, unsigned int data)
+			int req, const unsigned int data)
 {
-	return dspio_set_param(codec, mod_id, req, &data, sizeof(unsigned int));
+	return dspio_set_param(codec, mod_id, 0x20, req, &data,
+			sizeof(unsigned int));
+}
+
+static int dspio_set_uint_param_no_source(struct hda_codec *codec, int mod_id,
+			int req, const unsigned int data)
+{
+	return dspio_set_param(codec, mod_id, 0x00, req, &data,
+			sizeof(unsigned int));
 }
 
 /*
@@ -1522,8 +1894,9 @@ static int dspio_alloc_dma_chan(struct hda_codec *codec, unsigned int *dma_chan)
 	unsigned int size = sizeof(dma_chan);
 
 	codec_dbg(codec, "     dspio_alloc_dma_chan() -- begin\n");
-	status = dspio_scp(codec, MASTERCONTROL, MASTERCONTROL_ALLOC_DMA_CHAN,
-			SCP_GET, NULL, 0, dma_chan, &size);
+	status = dspio_scp(codec, MASTERCONTROL, 0x20,
+			MASTERCONTROL_ALLOC_DMA_CHAN, SCP_GET, NULL, 0,
+			dma_chan, &size);
 
 	if (status < 0) {
 		codec_dbg(codec, "dspio_alloc_dma_chan: SCP Failed\n");
@@ -1552,8 +1925,9 @@ static int dspio_free_dma_chan(struct hda_codec *codec, unsigned int dma_chan)
 	codec_dbg(codec, "     dspio_free_dma_chan() -- begin\n");
 	codec_dbg(codec, "dspio_free_dma_chan: chan=%d\n", dma_chan);
 
-	status = dspio_scp(codec, MASTERCONTROL, MASTERCONTROL_ALLOC_DMA_CHAN,
-			   SCP_SET, &dma_chan, sizeof(dma_chan), NULL, &dummy);
+	status = dspio_scp(codec, MASTERCONTROL, 0x20,
+			MASTERCONTROL_ALLOC_DMA_CHAN, SCP_SET, &dma_chan,
+			sizeof(dma_chan), NULL, &dummy);
 
 	if (status < 0) {
 		codec_dbg(codec, "dspio_free_dma_chan: SCP Failed\n");
@@ -2575,14 +2949,16 @@ static int dspxfr_image(struct hda_codec *codec,
  */
 static void dspload_post_setup(struct hda_codec *codec)
 {
+	struct ca0132_spec *spec = codec->spec;
 	codec_dbg(codec, "---- dspload_post_setup ------\n");
+	if (!spec->use_alt_functions) {
+		/*set DSP speaker to 2.0 configuration*/
+		chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080);
+		chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000);
 
-	/*set DSP speaker to 2.0 configuration*/
-	chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080);
-	chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000);
-
-	/*update write pointer*/
-	chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x29), 0x00000002);
+		/*update write pointer*/
+		chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x29), 0x00000002);
+	}
 }
 
 /**
@@ -2690,6 +3066,170 @@ static bool dspload_wait_loaded(struct hda_codec *codec)
 }
 
 /*
+ * Setup GPIO for the other variants of Core3D.
+ */
+
+/*
+ * Sets up the GPIO pins so that they are discoverable. If this isn't done,
+ * the card shows as having no GPIO pins.
+ */
+static void ca0132_gpio_init(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	switch (spec->quirk) {
+	case QUIRK_SBZ:
+		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);
+		break;
+	case QUIRK_R3DI:
+		snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
+		snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5B);
+		break;
+	}
+
+}
+
+/* Sets the GPIO for audio output. */
+static void ca0132_gpio_setup(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	switch (spec->quirk) {
+	case QUIRK_SBZ:
+		snd_hda_codec_write(codec, 0x01, 0,
+				AC_VERB_SET_GPIO_DIRECTION, 0x07);
+		snd_hda_codec_write(codec, 0x01, 0,
+				AC_VERB_SET_GPIO_MASK, 0x07);
+		snd_hda_codec_write(codec, 0x01, 0,
+				AC_VERB_SET_GPIO_DATA, 0x04);
+		snd_hda_codec_write(codec, 0x01, 0,
+				AC_VERB_SET_GPIO_DATA, 0x06);
+		break;
+	case QUIRK_R3DI:
+		snd_hda_codec_write(codec, 0x01, 0,
+				AC_VERB_SET_GPIO_DIRECTION, 0x1E);
+		snd_hda_codec_write(codec, 0x01, 0,
+				AC_VERB_SET_GPIO_MASK, 0x1F);
+		snd_hda_codec_write(codec, 0x01, 0,
+				AC_VERB_SET_GPIO_DATA, 0x0C);
+		break;
+	}
+}
+
+/*
+ * GPIO control functions for the Recon3D integrated.
+ */
+
+enum r3di_gpio_bit {
+	/* Bit 1 - Switch between front/rear mic. 0 = rear, 1 = front */
+	R3DI_MIC_SELECT_BIT = 1,
+	/* Bit 2 - Switch between headphone/line out. 0 = Headphone, 1 = Line */
+	R3DI_OUT_SELECT_BIT = 2,
+	/*
+	 * I dunno what this actually does, but it stays on until the dsp
+	 * is downloaded.
+	 */
+	R3DI_GPIO_DSP_DOWNLOADING = 3,
+	/*
+	 * Same as above, no clue what it does, but it comes on after the dsp
+	 * is downloaded.
+	 */
+	R3DI_GPIO_DSP_DOWNLOADED = 4
+};
+
+enum r3di_mic_select {
+	/* Set GPIO bit 1 to 0 for rear mic */
+	R3DI_REAR_MIC = 0,
+	/* Set GPIO bit 1 to 1 for front microphone*/
+	R3DI_FRONT_MIC = 1
+};
+
+enum r3di_out_select {
+	/* Set GPIO bit 2 to 0 for headphone */
+	R3DI_HEADPHONE_OUT = 0,
+	/* Set GPIO bit 2 to 1 for speaker */
+	R3DI_LINE_OUT = 1
+};
+enum r3di_dsp_status {
+	/* Set GPIO bit 3 to 1 until DSP is downloaded */
+	R3DI_DSP_DOWNLOADING = 0,
+	/* Set GPIO bit 4 to 1 once DSP is downloaded */
+	R3DI_DSP_DOWNLOADED = 1
+};
+
+
+static void r3di_gpio_mic_set(struct hda_codec *codec,
+		enum r3di_mic_select cur_mic)
+{
+	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_mic) {
+	case R3DI_REAR_MIC:
+		cur_gpio &= ~(1 << R3DI_MIC_SELECT_BIT);
+		break;
+	case R3DI_FRONT_MIC:
+		cur_gpio |= (1 << R3DI_MIC_SELECT_BIT);
+		break;
+	}
+	snd_hda_codec_write(codec, codec->core.afg, 0,
+			    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)
+{
+	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 (dsp_status) {
+	case R3DI_DSP_DOWNLOADING:
+		cur_gpio |= (1 << R3DI_GPIO_DSP_DOWNLOADING);
+		snd_hda_codec_write(codec, codec->core.afg, 0,
+				AC_VERB_SET_GPIO_DATA, cur_gpio);
+		break;
+	case R3DI_DSP_DOWNLOADED:
+		/* Set DOWNLOADING bit to 0. */
+		cur_gpio &= ~(1 << R3DI_GPIO_DSP_DOWNLOADING);
+
+		snd_hda_codec_write(codec, codec->core.afg, 0,
+				AC_VERB_SET_GPIO_DATA, cur_gpio);
+
+		cur_gpio |= (1 << R3DI_GPIO_DSP_DOWNLOADED);
+		break;
+	}
+
+	snd_hda_codec_write(codec, codec->core.afg, 0,
+			    AC_VERB_SET_GPIO_DATA, cur_gpio);
+}
+
+/*
  * PCM callbacks
  */
 static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
@@ -2852,6 +3392,24 @@ static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info,
 	  .tlv = { .c = ca0132_volume_tlv }, \
 	  .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) }
 
+/*
+ * Creates a mixer control that uses defaults of HDA_CODEC_VOL except for the
+ * volume put, which is used for setting the DSP volume. This was done because
+ * the ca0132 functions were taking too much time and causing lag.
+ */
+#define CA0132_ALT_CODEC_VOL_MONO(xname, nid, channel, dir) \
+	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+	  .name = xname, \
+	  .subdevice = HDA_SUBDEV_AMP_FLAG, \
+	  .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+			SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+			SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
+	  .info = snd_hda_mixer_amp_volume_info, \
+	  .get = snd_hda_mixer_amp_volume_get, \
+	  .put = ca0132_alt_volume_put, \
+	  .tlv = { .c = snd_hda_mixer_amp_tlv }, \
+	  .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) }
+
 #define CA0132_CODEC_MUTE_MONO(xname, nid, channel, dir) \
 	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
 	  .name = xname, \
@@ -2864,9 +3422,88 @@ static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info,
 /* stereo */
 #define CA0132_CODEC_VOL(xname, nid, dir) \
 	CA0132_CODEC_VOL_MONO(xname, nid, 3, dir)
+#define CA0132_ALT_CODEC_VOL(xname, nid, dir) \
+	CA0132_ALT_CODEC_VOL_MONO(xname, nid, 3, dir)
 #define CA0132_CODEC_MUTE(xname, nid, dir) \
 	CA0132_CODEC_MUTE_MONO(xname, nid, 3, dir)
 
+/* lookup tables */
+/*
+ * Lookup table with decibel values for the DSP. When volume is changed in
+ * Windows, the DSP is also sent the dB value in floating point. In Windows,
+ * these values have decimal points, probably because the Windows driver
+ * actually uses floating point. We can't here, so I made a lookup table of
+ * values -90 to 9. -90 is the lowest decibel value for both the ADC's and the
+ * DAC's, and 9 is the maximum.
+ */
+static const unsigned int float_vol_db_lookup[] = {
+0xC2B40000, 0xC2B20000, 0xC2B00000, 0xC2AE0000, 0xC2AC0000, 0xC2AA0000,
+0xC2A80000, 0xC2A60000, 0xC2A40000, 0xC2A20000, 0xC2A00000, 0xC29E0000,
+0xC29C0000, 0xC29A0000, 0xC2980000, 0xC2960000, 0xC2940000, 0xC2920000,
+0xC2900000, 0xC28E0000, 0xC28C0000, 0xC28A0000, 0xC2880000, 0xC2860000,
+0xC2840000, 0xC2820000, 0xC2800000, 0xC27C0000, 0xC2780000, 0xC2740000,
+0xC2700000, 0xC26C0000, 0xC2680000, 0xC2640000, 0xC2600000, 0xC25C0000,
+0xC2580000, 0xC2540000, 0xC2500000, 0xC24C0000, 0xC2480000, 0xC2440000,
+0xC2400000, 0xC23C0000, 0xC2380000, 0xC2340000, 0xC2300000, 0xC22C0000,
+0xC2280000, 0xC2240000, 0xC2200000, 0xC21C0000, 0xC2180000, 0xC2140000,
+0xC2100000, 0xC20C0000, 0xC2080000, 0xC2040000, 0xC2000000, 0xC1F80000,
+0xC1F00000, 0xC1E80000, 0xC1E00000, 0xC1D80000, 0xC1D00000, 0xC1C80000,
+0xC1C00000, 0xC1B80000, 0xC1B00000, 0xC1A80000, 0xC1A00000, 0xC1980000,
+0xC1900000, 0xC1880000, 0xC1800000, 0xC1700000, 0xC1600000, 0xC1500000,
+0xC1400000, 0xC1300000, 0xC1200000, 0xC1100000, 0xC1000000, 0xC0E00000,
+0xC0C00000, 0xC0A00000, 0xC0800000, 0xC0400000, 0xC0000000, 0xBF800000,
+0x00000000, 0x3F800000, 0x40000000, 0x40400000, 0x40800000, 0x40A00000,
+0x40C00000, 0x40E00000, 0x41000000, 0x41100000
+};
+
+/*
+ * This table counts from float 0 to 1 in increments of .01, which is
+ * useful for a few different sliders.
+ */
+static const unsigned int float_zero_to_one_lookup[] = {
+0x00000000, 0x3C23D70A, 0x3CA3D70A, 0x3CF5C28F, 0x3D23D70A, 0x3D4CCCCD,
+0x3D75C28F, 0x3D8F5C29, 0x3DA3D70A, 0x3DB851EC, 0x3DCCCCCD, 0x3DE147AE,
+0x3DF5C28F, 0x3E051EB8, 0x3E0F5C29, 0x3E19999A, 0x3E23D70A, 0x3E2E147B,
+0x3E3851EC, 0x3E428F5C, 0x3E4CCCCD, 0x3E570A3D, 0x3E6147AE, 0x3E6B851F,
+0x3E75C28F, 0x3E800000, 0x3E851EB8, 0x3E8A3D71, 0x3E8F5C29, 0x3E947AE1,
+0x3E99999A, 0x3E9EB852, 0x3EA3D70A, 0x3EA8F5C3, 0x3EAE147B, 0x3EB33333,
+0x3EB851EC, 0x3EBD70A4, 0x3EC28F5C, 0x3EC7AE14, 0x3ECCCCCD, 0x3ED1EB85,
+0x3ED70A3D, 0x3EDC28F6, 0x3EE147AE, 0x3EE66666, 0x3EEB851F, 0x3EF0A3D7,
+0x3EF5C28F, 0x3EFAE148, 0x3F000000, 0x3F028F5C, 0x3F051EB8, 0x3F07AE14,
+0x3F0A3D71, 0x3F0CCCCD, 0x3F0F5C29, 0x3F11EB85, 0x3F147AE1, 0x3F170A3D,
+0x3F19999A, 0x3F1C28F6, 0x3F1EB852, 0x3F2147AE, 0x3F23D70A, 0x3F266666,
+0x3F28F5C3, 0x3F2B851F, 0x3F2E147B, 0x3F30A3D7, 0x3F333333, 0x3F35C28F,
+0x3F3851EC, 0x3F3AE148, 0x3F3D70A4, 0x3F400000, 0x3F428F5C, 0x3F451EB8,
+0x3F47AE14, 0x3F4A3D71, 0x3F4CCCCD, 0x3F4F5C29, 0x3F51EB85, 0x3F547AE1,
+0x3F570A3D, 0x3F59999A, 0x3F5C28F6, 0x3F5EB852, 0x3F6147AE, 0x3F63D70A,
+0x3F666666, 0x3F68F5C3, 0x3F6B851F, 0x3F6E147B, 0x3F70A3D7, 0x3F733333,
+0x3F75C28F, 0x3F7851EC, 0x3F7AE148, 0x3F7D70A4, 0x3F800000
+};
+
+/*
+ * This table counts from float 10 to 1000, which is the range of the x-bass
+ * crossover slider in Windows.
+ */
+static const unsigned int float_xbass_xover_lookup[] = {
+0x41200000, 0x41A00000, 0x41F00000, 0x42200000, 0x42480000, 0x42700000,
+0x428C0000, 0x42A00000, 0x42B40000, 0x42C80000, 0x42DC0000, 0x42F00000,
+0x43020000, 0x430C0000, 0x43160000, 0x43200000, 0x432A0000, 0x43340000,
+0x433E0000, 0x43480000, 0x43520000, 0x435C0000, 0x43660000, 0x43700000,
+0x437A0000, 0x43820000, 0x43870000, 0x438C0000, 0x43910000, 0x43960000,
+0x439B0000, 0x43A00000, 0x43A50000, 0x43AA0000, 0x43AF0000, 0x43B40000,
+0x43B90000, 0x43BE0000, 0x43C30000, 0x43C80000, 0x43CD0000, 0x43D20000,
+0x43D70000, 0x43DC0000, 0x43E10000, 0x43E60000, 0x43EB0000, 0x43F00000,
+0x43F50000, 0x43FA0000, 0x43FF0000, 0x44020000, 0x44048000, 0x44070000,
+0x44098000, 0x440C0000, 0x440E8000, 0x44110000, 0x44138000, 0x44160000,
+0x44188000, 0x441B0000, 0x441D8000, 0x44200000, 0x44228000, 0x44250000,
+0x44278000, 0x442A0000, 0x442C8000, 0x442F0000, 0x44318000, 0x44340000,
+0x44368000, 0x44390000, 0x443B8000, 0x443E0000, 0x44408000, 0x44430000,
+0x44458000, 0x44480000, 0x444A8000, 0x444D0000, 0x444F8000, 0x44520000,
+0x44548000, 0x44570000, 0x44598000, 0x445C0000, 0x445E8000, 0x44610000,
+0x44638000, 0x44660000, 0x44688000, 0x446B0000, 0x446D8000, 0x44700000,
+0x44728000, 0x44750000, 0x44778000, 0x447A0000
+};
+
 /* The following are for tuning of products */
 #ifdef ENABLE_TUNING_CONTROLS
 
@@ -2942,7 +3579,7 @@ static int tuning_ctl_set(struct hda_codec *codec, hda_nid_t nid,
 			break;
 
 	snd_hda_power_up(codec);
-	dspio_set_param(codec, ca0132_tuning_ctls[i].mid,
+	dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20,
 			ca0132_tuning_ctls[i].req,
 			&(lookup[idx]), sizeof(unsigned int));
 	snd_hda_power_down(codec);
@@ -3068,8 +3705,8 @@ static int equalizer_ctl_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static const DECLARE_TLV_DB_SCALE(voice_focus_db_scale, 2000, 100, 0);
-static const DECLARE_TLV_DB_SCALE(eq_db_scale, -2400, 100, 0);
+static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(voice_focus_db_scale, 2000, 100, 0);
+static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(eq_db_scale, -2400, 100, 0);
 
 static int add_tuning_control(struct hda_codec *codec,
 				hda_nid_t pnid, hda_nid_t nid,
@@ -3207,7 +3844,7 @@ static int ca0132_select_out(struct hda_codec *codec)
 				    pin_ctl & ~PIN_HP);
 		/* enable speaker node */
 		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
-					     AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 		snd_hda_set_pin_ctl(codec, spec->out_pins[0],
 				    pin_ctl | PIN_OUT);
 	} else {
@@ -3251,13 +3888,209 @@ static int ca0132_select_out(struct hda_codec *codec)
 	return err < 0 ? err : 0;
 }
 
+/*
+ * This function behaves similarly to the ca0132_select_out funciton above,
+ * except with a few differences. It adds the ability to select the current
+ * 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.
+ */
+static int ca0132_alt_select_out(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int pin_ctl;
+	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];
+
+	codec_dbg(codec, "%s\n", __func__);
+
+	snd_hda_power_up_pm(codec);
+
+	auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID];
+
+	/*
+	 * If headphone rear or front is plugged in, set to headphone.
+	 * If neither is plugged in, set to rear line out. Only if
+	 * hp/speaker auto detect is enabled.
+	 */
+	if (auto_jack) {
+		jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_hp) ||
+			   snd_hda_jack_detect(codec, spec->unsol_tag_front_hp);
+
+		if (jack_present)
+			spec->cur_out_type = HEADPHONE_OUT;
+		else
+			spec->cur_out_type = SPEAKER_OUT;
+	} 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);
+	if (err < 0)
+		goto exit;
+
+	switch (spec->cur_out_type) {
+	case SPEAKER_OUT:
+		codec_dbg(codec, "%s speaker\n", __func__);
+		/*speaker out config*/
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
+			writew(0x0007, spec->mem_base + 0x320);
+			writew(0x0104, spec->mem_base + 0x320);
+			writew(0x0101, spec->mem_base + 0x320);
+			chipio_set_control_param(codec, 0x0D, 0x18);
+			break;
+		case QUIRK_R3DI:
+			chipio_set_control_param(codec, 0x0D, 0x24);
+			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
+			break;
+		}
+
+		/* 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);
+		else
+			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+		break;
+	case HEADPHONE_OUT:
+		codec_dbg(codec, "%s hp\n", __func__);
+		/* Headphone out config*/
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
+			writew(0x0107, spec->mem_base + 0x320);
+			writew(0x0104, spec->mem_base + 0x320);
+			writew(0x0001, spec->mem_base + 0x320);
+			chipio_set_control_param(codec, 0x0D, 0x12);
+			break;
+		case QUIRK_R3DI:
+			chipio_set_control_param(codec, 0x0D, 0x21);
+			r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT);
+			break;
+		}
+
+		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);
+
+		/* 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);
+
+		if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
+			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__);
+		/* Surround out config*/
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
+			writew(0x0007, spec->mem_base + 0x320);
+			writew(0x0104, spec->mem_base + 0x320);
+			writew(0x0101, spec->mem_base + 0x320);
+			chipio_set_control_param(codec, 0x0D, 0x18);
+			break;
+		case QUIRK_R3DI:
+			chipio_set_control_param(codec, 0x0D, 0x24);
+			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
+			break;
+		}
+		/* 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);
+
+		if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
+			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
+		else
+			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+		break;
+	}
+
+	/* run through the output dsp commands for line-out */
+	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]);
+
+		if (err < 0)
+			goto exit;
+	}
+
+exit:
+	snd_hda_power_down_pm(codec);
+
+	return err < 0 ? err : 0;
+}
+
 static void ca0132_unsol_hp_delayed(struct work_struct *work)
 {
 	struct ca0132_spec *spec = container_of(
 		to_delayed_work(work), struct ca0132_spec, unsol_hp_work);
 	struct hda_jack_tbl *jack;
 
-	ca0132_select_out(spec->codec);
+	if (spec->use_alt_functions)
+		ca0132_alt_select_out(spec->codec);
+	else
+		ca0132_select_out(spec->codec);
+
 	jack = snd_hda_jack_tbl_get(spec->codec, spec->unsol_tag_hp);
 	if (jack) {
 		jack->block_report = 0;
@@ -3268,6 +4101,10 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work)
 static void ca0132_set_dmic(struct hda_codec *codec, int enable);
 static int ca0132_mic_boost_set(struct hda_codec *codec, long val);
 static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
+static void resume_mic1(struct hda_codec *codec, unsigned int oldval);
+static int stop_mic1(struct hda_codec *codec);
+static int ca0132_cvoice_switch_set(struct hda_codec *codec);
+static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val);
 
 /*
  * Select the active VIP source
@@ -3310,6 +4147,71 @@ static int ca0132_set_vipsource(struct hda_codec *codec, int val)
 	return 1;
 }
 
+static int ca0132_alt_set_vipsource(struct hda_codec *codec, int val)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp;
+
+	if (spec->dsp_state != DSP_DOWNLOADED)
+		return 0;
+
+	codec_dbg(codec, "%s\n", __func__);
+
+	chipio_set_stream_control(codec, 0x03, 0);
+	chipio_set_stream_control(codec, 0x04, 0);
+
+	/* if CrystalVoice is off, vipsource should be 0 */
+	if (!spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] ||
+	    (val == 0) || spec->in_enum_val == REAR_LINE_IN) {
+		codec_dbg(codec, "%s: off.", __func__);
+		chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0);
+
+		tmp = FLOAT_ZERO;
+		dspio_set_uint_param(codec, 0x80, 0x05, tmp);
+
+		chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+		chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+		if (spec->quirk == QUIRK_R3DI)
+			chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+
+
+		if (spec->in_enum_val == REAR_LINE_IN)
+			tmp = FLOAT_ZERO;
+		else {
+			if (spec->quirk == QUIRK_SBZ)
+				tmp = FLOAT_THREE;
+			else
+				tmp = FLOAT_ONE;
+		}
+
+		dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+	} else {
+		codec_dbg(codec, "%s: on.", __func__);
+		chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_16_000);
+		chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_16_000);
+		if (spec->quirk == QUIRK_R3DI)
+			chipio_set_conn_rate(codec, 0x0F, SR_16_000);
+
+		if (spec->effects_switch[VOICE_FOCUS - EFFECT_START_NID])
+			tmp = FLOAT_TWO;
+		else
+			tmp = FLOAT_ONE;
+		dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+		tmp = FLOAT_ONE;
+		dspio_set_uint_param(codec, 0x80, 0x05, tmp);
+
+		msleep(20);
+		chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, val);
+	}
+
+	chipio_set_stream_control(codec, 0x03, 1);
+	chipio_set_stream_control(codec, 0x04, 1);
+
+	return 1;
+}
+
 /*
  * Select the active microphone.
  * If autodetect is enabled, mic will be selected based on jack detection.
@@ -3363,6 +4265,125 @@ static int ca0132_select_mic(struct hda_codec *codec)
 }
 
 /*
+ * Select the active input.
+ * Mic detection isn't used, because it's kind of pointless on the SBZ.
+ * The front mic has no jack-detection, so the only way to switch to it
+ * is to do it manually in alsamixer.
+ */
+static int ca0132_alt_select_in(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp;
+
+	codec_dbg(codec, "%s\n", __func__);
+
+	snd_hda_power_up_pm(codec);
+
+	chipio_set_stream_control(codec, 0x03, 0);
+	chipio_set_stream_control(codec, 0x04, 0);
+
+	spec->cur_mic_type = spec->in_enum_val;
+
+	switch (spec->cur_mic_type) {
+	case REAR_MIC:
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
+			writew(0x0000, spec->mem_base + 0x320);
+			tmp = FLOAT_THREE;
+			break;
+		case QUIRK_R3DI:
+			r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
+			tmp = FLOAT_ONE;
+			break;
+		default:
+			tmp = FLOAT_ONE;
+			break;
+		}
+
+		chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+		chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+		if (spec->quirk == QUIRK_R3DI)
+			chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+
+		dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+		chipio_set_stream_control(codec, 0x03, 1);
+		chipio_set_stream_control(codec, 0x04, 1);
+
+		if (spec->quirk == QUIRK_SBZ) {
+			chipio_write(codec, 0x18B098, 0x0000000C);
+			chipio_write(codec, 0x18B09C, 0x0000000C);
+		}
+		ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
+		break;
+	case REAR_LINE_IN:
+		ca0132_mic_boost_set(codec, 0);
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
+			writew(0x0000, spec->mem_base + 0x320);
+			break;
+		case QUIRK_R3DI:
+			r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
+			break;
+		}
+
+		chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+		chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+		if (spec->quirk == QUIRK_R3DI)
+			chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+
+		tmp = FLOAT_ZERO;
+		dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+		if (spec->quirk == QUIRK_SBZ) {
+			chipio_write(codec, 0x18B098, 0x00000000);
+			chipio_write(codec, 0x18B09C, 0x00000000);
+		}
+
+		chipio_set_stream_control(codec, 0x03, 1);
+		chipio_set_stream_control(codec, 0x04, 1);
+		break;
+	case FRONT_MIC:
+		switch (spec->quirk) {
+		case QUIRK_SBZ:
+			writew(0x0100, spec->mem_base + 0x320);
+			writew(0x0005, spec->mem_base + 0x320);
+			tmp = FLOAT_THREE;
+			break;
+		case QUIRK_R3DI:
+			r3di_gpio_mic_set(codec, R3DI_FRONT_MIC);
+			tmp = FLOAT_ONE;
+			break;
+		default:
+			tmp = FLOAT_ONE;
+			break;
+		}
+
+		chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+		chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+		if (spec->quirk == QUIRK_R3DI)
+			chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+
+		dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+		chipio_set_stream_control(codec, 0x03, 1);
+		chipio_set_stream_control(codec, 0x04, 1);
+
+		if (spec->quirk == QUIRK_SBZ) {
+			chipio_write(codec, 0x18B098, 0x0000000C);
+			chipio_write(codec, 0x18B09C, 0x000000CC);
+		}
+		ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
+		break;
+	}
+	ca0132_cvoice_switch_set(codec);
+
+	snd_hda_power_down_pm(codec);
+	return 0;
+
+}
+
+/*
  * Check if VNODE settings take effect immediately.
  */
 static bool ca0132_is_vnode_effective(struct hda_codec *codec,
@@ -3418,7 +4439,7 @@ 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;
+	unsigned int on, tmp;
 	int num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
 	int err = 0;
 	int idx = nid - EFFECT_START_NID;
@@ -3442,6 +4463,46 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
 		/* Voice Focus applies to 2-ch Mic, Digital Mic */
 		if ((nid == VOICE_FOCUS) && (spec->cur_mic_type != DIGITAL_MIC))
 			val = 0;
+
+		/* If Voice Focus on SBZ, set to two channel. */
+		if ((nid == VOICE_FOCUS) && (spec->quirk == QUIRK_SBZ)
+				&& (spec->cur_mic_type != REAR_LINE_IN)) {
+			if (spec->effects_switch[CRYSTAL_VOICE -
+						 EFFECT_START_NID]) {
+
+				if (spec->effects_switch[VOICE_FOCUS -
+							 EFFECT_START_NID]) {
+					tmp = FLOAT_TWO;
+					val = 1;
+				} else
+					tmp = FLOAT_ONE;
+
+				dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+			}
+		}
+		/*
+		 * For SBZ noise reduction, there's an extra command
+		 * to module ID 0x47. No clue why.
+		 */
+		if ((nid == NOISE_REDUCTION) && (spec->quirk == QUIRK_SBZ)
+				&& (spec->cur_mic_type != REAR_LINE_IN)) {
+			if (spec->effects_switch[CRYSTAL_VOICE -
+						 EFFECT_START_NID]) {
+				if (spec->effects_switch[NOISE_REDUCTION -
+							 EFFECT_START_NID])
+					tmp = FLOAT_ONE;
+				else
+					tmp = FLOAT_ZERO;
+			} else
+				tmp = FLOAT_ZERO;
+
+			dspio_set_uint_param(codec, 0x47, 0x00, tmp);
+		}
+
+		/* If rear line in disable effects. */
+		if (spec->use_alt_functions &&
+				spec->in_enum_val == REAR_LINE_IN)
+			val = 0;
 	}
 
 	codec_dbg(codec, "ca0132_effect_set: nid=0x%x, val=%ld\n",
@@ -3469,6 +4530,9 @@ static int ca0132_pe_switch_set(struct hda_codec *codec)
 	codec_dbg(codec, "ca0132_pe_switch_set: val=%ld\n",
 		    spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]);
 
+	if (spec->use_alt_functions)
+		ca0132_alt_select_out(codec);
+
 	i = OUT_EFFECT_START_NID - EFFECT_START_NID;
 	nid = OUT_EFFECT_START_NID;
 	/* PE affects all out effects */
@@ -3526,7 +4590,10 @@ static int ca0132_cvoice_switch_set(struct hda_codec *codec)
 
 	/* set correct vipsource */
 	oldval = stop_mic1(codec);
-	ret |= ca0132_set_vipsource(codec, 1);
+	if (spec->use_alt_functions)
+		ret |= ca0132_alt_set_vipsource(codec, 1);
+	else
+		ret |= ca0132_set_vipsource(codec, 1);
 	resume_mic1(codec, oldval);
 	return ret;
 }
@@ -3546,6 +4613,16 @@ static int ca0132_mic_boost_set(struct hda_codec *codec, long val)
 	return ret;
 }
 
+static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val)
+{
+	struct ca0132_spec *spec = codec->spec;
+	int ret = 0;
+
+	ret = snd_hda_codec_amp_update(codec, spec->input_pins[0], 0,
+				HDA_INPUT, 0, HDA_AMP_VOLMASK, val);
+	return ret;
+}
+
 static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
 {
@@ -3560,8 +4637,12 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
 	if (nid == VNID_HP_SEL) {
 		auto_jack =
 			spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID];
-		if (!auto_jack)
-			ca0132_select_out(codec);
+		if (!auto_jack) {
+			if (spec->use_alt_functions)
+				ca0132_alt_select_out(codec);
+			else
+				ca0132_select_out(codec);
+		}
 		return 1;
 	}
 
@@ -3574,7 +4655,10 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
 	}
 
 	if (nid == VNID_HP_ASEL) {
-		ca0132_select_out(codec);
+		if (spec->use_alt_functions)
+			ca0132_alt_select_out(codec);
+		else
+			ca0132_select_out(codec);
 		return 1;
 	}
 
@@ -3602,6 +4686,432 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
 	return ret;
 }
 /* End of control change helpers. */
+/*
+ * 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
+ * Chromebook. I figured they were probably tuned specifically for it, and left
+ * out for a reason.
+ */
+
+/* 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)
+{
+	int i = 0;
+	unsigned int y;
+	/*
+	 * For X_BASS, req 2 is actually crossover freq instead of
+	 * effect level
+	 */
+	if (nid == X_BASS)
+		y = 2;
+	else
+		y = 1;
+
+	snd_hda_power_up(codec);
+	if (nid == XBASS_XOVER) {
+		for (i = 0; i < OUT_EFFECTS_COUNT; i++)
+			if (ca0132_effects[i].nid == X_BASS)
+				break;
+
+		dspio_set_param(codec, ca0132_effects[i].mid, 0x20,
+				ca0132_effects[i].reqs[1],
+				&(lookup[idx - 1]), sizeof(unsigned int));
+	} else {
+		/* Find the actual effect structure */
+		for (i = 0; i < OUT_EFFECTS_COUNT; i++)
+			if (nid == ca0132_effects[i].nid)
+				break;
+
+		dspio_set_param(codec, ca0132_effects[i].mid, 0x20,
+				ca0132_effects[i].reqs[y],
+				&(lookup[idx]), sizeof(unsigned int));
+	}
+
+	snd_hda_power_down(codec);
+
+	return 0;
+}
+
+static int ca0132_alt_xbass_xover_slider_ctl_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;
+	long *valp = ucontrol->value.integer.value;
+
+	*valp = spec->xbass_xover_freq;
+	return 0;
+}
+
+static int ca0132_alt_slider_ctl_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;
+	hda_nid_t nid = get_amp_nid(kcontrol);
+	long *valp = ucontrol->value.integer.value;
+	int idx = nid - OUT_EFFECT_START_NID;
+
+	*valp = spec->fx_ctl_val[idx];
+	return 0;
+}
+
+/*
+ * The X-bass crossover starts at 10hz, so the min is 1. The
+ * frequency is set in multiples of 10.
+ */
+static int ca0132_alt_xbass_xover_slider_info(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 1;
+	uinfo->value.integer.max = 100;
+	uinfo->value.integer.step = 1;
+
+	return 0;
+}
+
+static int ca0132_alt_effect_slider_info(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_info *uinfo)
+{
+	int chs = get_amp_channels(kcontrol);
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = chs == 3 ? 2 : 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 100;
+	uinfo->value.integer.step = 1;
+
+	return 0;
+}
+
+static int ca0132_alt_xbass_xover_slider_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;
+	hda_nid_t nid = get_amp_nid(kcontrol);
+	long *valp = ucontrol->value.integer.value;
+	int idx;
+
+	/* any change? */
+	if (spec->xbass_xover_freq == *valp)
+		return 0;
+
+	spec->xbass_xover_freq = *valp;
+
+	idx = *valp;
+	ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx);
+
+	return 0;
+}
+
+static int ca0132_alt_effect_slider_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;
+	hda_nid_t nid = get_amp_nid(kcontrol);
+	long *valp = ucontrol->value.integer.value;
+	int idx;
+
+	idx = nid - EFFECT_START_NID;
+	/* any change? */
+	if (spec->fx_ctl_val[idx] == *valp)
+		return 0;
+
+	spec->fx_ctl_val[idx] = *valp;
+
+	idx = *valp;
+	ca0132_alt_slider_ctl_set(codec, nid, float_zero_to_one_lookup, idx);
+
+	return 0;
+}
+
+
+/*
+ * Mic Boost Enum for alternative ca0132 codecs. I didn't like that the original
+ * only has off or full 30 dB, and didn't like making a volume slider that has
+ * traditional 0-100 in alsamixer that goes in big steps. I like enum better.
+ */
+#define MIC_BOOST_NUM_OF_STEPS 4
+#define MIC_BOOST_ENUM_MAX_STRLEN 10
+
+static int ca0132_alt_mic_boost_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	char *sfx = "dB";
+	char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = MIC_BOOST_NUM_OF_STEPS;
+	if (uinfo->value.enumerated.item >= MIC_BOOST_NUM_OF_STEPS)
+		uinfo->value.enumerated.item = MIC_BOOST_NUM_OF_STEPS - 1;
+	sprintf(namestr, "%d %s", (uinfo->value.enumerated.item * 10), sfx);
+	strcpy(uinfo->value.enumerated.name, namestr);
+	return 0;
+}
+
+static int ca0132_alt_mic_boost_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->mic_boost_enum_val;
+	return 0;
+}
+
+static int ca0132_alt_mic_boost_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 = MIC_BOOST_NUM_OF_STEPS;
+
+	if (sel >= items)
+		return 0;
+
+	codec_dbg(codec, "ca0132_alt_mic_boost: boost=%d\n",
+		    sel);
+
+	spec->mic_boost_enum_val = sel;
+
+	if (spec->in_enum_val != REAR_LINE_IN)
+		ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
+
+	return 1;
+}
+
+
+/*
+ * Input Select Control for alternative ca0132 codecs. This exists because
+ * front microphone has no auto-detect, and we need a way to set the rear
+ * as line-in
+ */
+static int ca0132_alt_input_source_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = IN_SRC_NUM_OF_INPUTS;
+	if (uinfo->value.enumerated.item >= IN_SRC_NUM_OF_INPUTS)
+		uinfo->value.enumerated.item = IN_SRC_NUM_OF_INPUTS - 1;
+	strcpy(uinfo->value.enumerated.name,
+			in_src_str[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int ca0132_alt_input_source_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->in_enum_val;
+	return 0;
+}
+
+static int ca0132_alt_input_source_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 = IN_SRC_NUM_OF_INPUTS;
+
+	if (sel >= items)
+		return 0;
+
+	codec_dbg(codec, "ca0132_alt_input_select: sel=%d, preset=%s\n",
+		    sel, in_src_str[sel]);
+
+	spec->in_enum_val = sel;
+
+	ca0132_alt_select_in(codec);
+
+	return 1;
+}
+
+/* Sound Blaster Z Output Select Control */
+static int ca0132_alt_output_select_get_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = NUM_OF_OUTPUTS;
+	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);
+	return 0;
+}
+
+static int ca0132_alt_output_select_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->out_enum_val;
+	return 0;
+}
+
+static int ca0132_alt_output_select_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 = NUM_OF_OUTPUTS;
+	unsigned int auto_jack;
+
+	if (sel >= items)
+		return 0;
+
+	codec_dbg(codec, "ca0132_alt_output_select: sel=%d, preset=%s\n",
+		    sel, alt_out_presets[sel].name);
+
+	spec->out_enum_val = sel;
+
+	auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID];
+
+	if (!auto_jack)
+		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
+ * and night, disregard the slider value and have uneditable values.
+ */
+#define NUM_OF_SVM_SETTINGS 3
+static const char *out_svm_set_enum_str[3] = {"Normal", "Loud", "Night" };
+
+static int ca0132_alt_svm_setting_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = NUM_OF_SVM_SETTINGS;
+	if (uinfo->value.enumerated.item >= NUM_OF_SVM_SETTINGS)
+		uinfo->value.enumerated.item = NUM_OF_SVM_SETTINGS - 1;
+	strcpy(uinfo->value.enumerated.name,
+			out_svm_set_enum_str[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int ca0132_alt_svm_setting_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->smart_volume_setting;
+	return 0;
+}
+
+static int ca0132_alt_svm_setting_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 = NUM_OF_SVM_SETTINGS;
+	unsigned int idx = SMART_VOLUME - EFFECT_START_NID;
+	unsigned int tmp;
+
+	if (sel >= items)
+		return 0;
+
+	codec_dbg(codec, "ca0132_alt_svm_setting: sel=%d, preset=%s\n",
+		    sel, out_svm_set_enum_str[sel]);
+
+	spec->smart_volume_setting = sel;
+
+	switch (sel) {
+	case 0:
+		tmp = FLOAT_ZERO;
+		break;
+	case 1:
+		tmp = FLOAT_ONE;
+		break;
+	case 2:
+		tmp = FLOAT_TWO;
+		break;
+	default:
+		tmp = FLOAT_ZERO;
+		break;
+	}
+	/* Req 2 is the Smart Volume Setting req. */
+	dspio_set_uint_param(codec, ca0132_effects[idx].mid,
+			ca0132_effects[idx].reqs[2], tmp);
+	return 1;
+}
+
+/* Sound Blaster Z EQ preset controls */
+static int ca0132_alt_eq_preset_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	unsigned int items = ARRAY_SIZE(ca0132_alt_eq_presets);
+
+	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,
+		ca0132_alt_eq_presets[uinfo->value.enumerated.item].name);
+	return 0;
+}
+
+static int ca0132_alt_eq_preset_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->eq_preset_val;
+	return 0;
+}
+
+static int ca0132_alt_eq_preset_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 i, err = 0;
+	int sel = ucontrol->value.enumerated.item[0];
+	unsigned int items = ARRAY_SIZE(ca0132_alt_eq_presets);
+
+	if (sel >= items)
+		return 0;
+
+	codec_dbg(codec, "%s: sel=%d, preset=%s\n", __func__, sel,
+			ca0132_alt_eq_presets[sel].name);
+	/*
+	 * Idx 0 is default.
+	 * Default needs to qualify with CrystalVoice state.
+	 */
+	for (i = 0; i < EQ_PRESET_MAX_PARAM_COUNT; i++) {
+		err = dspio_set_uint_param(codec, ca0132_alt_eq_enum.mid,
+				ca0132_alt_eq_enum.reqs[i],
+				ca0132_alt_eq_presets[sel].vals[i]);
+		if (err < 0)
+			break;
+	}
+
+	if (err >= 0)
+		spec->eq_preset_val = sel;
+
+	return 1;
+}
 
 static int ca0132_voicefx_info(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_info *uinfo)
@@ -3753,10 +5263,15 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
 	/* mic boost */
 	if (nid == spec->input_pins[0]) {
 		spec->cur_mic_boost = *valp;
+		if (spec->use_alt_functions) {
+			if (spec->in_enum_val != REAR_LINE_IN)
+				changed = ca0132_mic_boost_set(codec, *valp);
+		} else {
+			/* Mic boost does not apply to Digital Mic */
+			if (spec->cur_mic_type != DIGITAL_MIC)
+				changed = ca0132_mic_boost_set(codec, *valp);
+		}
 
-		/* Mic boost does not apply to Digital Mic */
-		if (spec->cur_mic_type != DIGITAL_MIC)
-			changed = ca0132_mic_boost_set(codec, *valp);
 		goto exit;
 	}
 
@@ -3768,6 +5283,41 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
 /*
  * Volume related
  */
+/*
+ * Sets the internal DSP decibel level to match the DAC for output, and the
+ * ADC for input. Currently only the SBZ sets dsp capture volume level, and
+ * all alternative codecs set DSP playback volume.
+ */
+static void ca0132_alt_dsp_volume_put(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int dsp_dir;
+	unsigned int lookup_val;
+
+	if (nid == VNID_SPK)
+		dsp_dir = DSP_VOL_OUT;
+	else
+		dsp_dir = DSP_VOL_IN;
+
+	lookup_val = spec->vnode_lvol[nid - VNODE_START_NID];
+
+	dspio_set_uint_param(codec,
+		ca0132_alt_vol_ctls[dsp_dir].mid,
+		ca0132_alt_vol_ctls[dsp_dir].reqs[0],
+		float_vol_db_lookup[lookup_val]);
+
+	lookup_val = spec->vnode_rvol[nid - VNODE_START_NID];
+
+	dspio_set_uint_param(codec,
+		ca0132_alt_vol_ctls[dsp_dir].mid,
+		ca0132_alt_vol_ctls[dsp_dir].reqs[1],
+		float_vol_db_lookup[lookup_val]);
+
+	dspio_set_uint_param(codec,
+		ca0132_alt_vol_ctls[dsp_dir].mid,
+		ca0132_alt_vol_ctls[dsp_dir].reqs[2], FLOAT_ZERO);
+}
+
 static int ca0132_volume_info(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_info *uinfo)
 {
@@ -3869,6 +5419,51 @@ static int ca0132_volume_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
+/*
+ * This function is the same as the one above, because using an if statement
+ * inside of the above volume control for the DSP volume would cause too much
+ * lag. This is a lot more smooth.
+ */
+static int ca0132_alt_volume_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;
+	hda_nid_t nid = get_amp_nid(kcontrol);
+	int ch = get_amp_channels(kcontrol);
+	long *valp = ucontrol->value.integer.value;
+	hda_nid_t vnid = 0;
+	int changed = 1;
+
+	switch (nid) {
+	case 0x02:
+		vnid = VNID_SPK;
+		break;
+	case 0x07:
+		vnid = VNID_MIC;
+		break;
+	}
+
+	/* store the left and right volume */
+	if (ch & 1) {
+		spec->vnode_lvol[vnid - VNODE_START_NID] = *valp;
+		valp++;
+	}
+	if (ch & 2) {
+		spec->vnode_rvol[vnid - VNODE_START_NID] = *valp;
+		valp++;
+	}
+
+	snd_hda_power_up(codec);
+	ca0132_alt_dsp_volume_put(codec, vnid);
+	mutex_lock(&codec->control_mutex);
+	changed = snd_hda_mixer_amp_volume_put(kcontrol, ucontrol);
+	mutex_unlock(&codec->control_mutex);
+	snd_hda_power_down(codec);
+
+	return changed;
+}
+
 static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 			     unsigned int size, unsigned int __user *tlv)
 {
@@ -3907,14 +5502,61 @@ static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 	return err;
 }
 
+/* Add volume slider control for effect level */
+static int ca0132_alt_add_effect_slider(struct hda_codec *codec, hda_nid_t nid,
+					const char *pfx, int dir)
+{
+	char *fx = "FX:";
+	char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	int type = dir ? HDA_INPUT : HDA_OUTPUT;
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_VOLUME_MONO(namestr, nid, 1, 0, type);
+
+	sprintf(namestr, "%s %s %s Volume", fx, pfx, dirstr[dir]);
+
+	knew.tlv.c = 0;
+	knew.tlv.p = 0;
+
+	switch (nid) {
+	case XBASS_XOVER:
+		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;
+		break;
+	default:
+		knew.info = ca0132_alt_effect_slider_info;
+		knew.get = ca0132_alt_slider_ctl_get;
+		knew.put = ca0132_alt_effect_slider_put;
+		knew.private_value =
+			HDA_COMPOSE_AMP_VAL(nid, 1, 0, type);
+		break;
+	}
+
+	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
+}
+
+/*
+ * Added FX: prefix for the alternative codecs, because otherwise the surround
+ * effect would conflict with the Surround sound volume control. Also seems more
+ * clear as to what the switches do. Left alone for others.
+ */
 static int add_fx_switch(struct hda_codec *codec, hda_nid_t nid,
 			 const char *pfx, int dir)
 {
+	struct ca0132_spec *spec = codec->spec;
+	char *fx = "FX:";
 	char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
 	int type = dir ? HDA_INPUT : HDA_OUTPUT;
 	struct snd_kcontrol_new knew =
 		CA0132_CODEC_MUTE_MONO(namestr, nid, 1, type);
-	sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
+	/* If using alt_controls, add FX: prefix. But, don't add FX:
+	 * prefix to OutFX or InFX enable controls.
+	 */
+	if ((spec->use_alt_controls) && (nid <= IN_EFFECT_END_NID))
+		sprintf(namestr, "%s %s %s Switch", fx, pfx, dirstr[dir]);
+	else
+		sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
+
 	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
 }
 
@@ -3929,6 +5571,136 @@ static int add_voicefx(struct hda_codec *codec)
 	return snd_hda_ctl_add(codec, VOICEFX, snd_ctl_new1(&knew, codec));
 }
 
+/* Create the EQ Preset control */
+static int add_ca0132_alt_eq_presets(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO(ca0132_alt_eq_enum.name,
+				    EQ_PRESET_ENUM, 1, 0, HDA_OUTPUT);
+	knew.info = ca0132_alt_eq_preset_info;
+	knew.get = ca0132_alt_eq_preset_get;
+	knew.put = ca0132_alt_eq_preset_put;
+	return snd_hda_ctl_add(codec, EQ_PRESET_ENUM,
+				snd_ctl_new1(&knew, codec));
+}
+
+/*
+ * Add enumerated control for the three different settings of the smart volume
+ * output effect. Normal just uses the slider value, and loud and night are
+ * their own things that ignore that value.
+ */
+static int ca0132_alt_add_svm_enum(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO("FX: Smart Volume Setting",
+				    SMART_VOLUME_ENUM, 1, 0, HDA_OUTPUT);
+	knew.info = ca0132_alt_svm_setting_info;
+	knew.get = ca0132_alt_svm_setting_get;
+	knew.put = ca0132_alt_svm_setting_put;
+	return snd_hda_ctl_add(codec, SMART_VOLUME_ENUM,
+				snd_ctl_new1(&knew, codec));
+
+}
+
+/*
+ * Create an Output Select enumerated control for codecs with surround
+ * out capabilities.
+ */
+static int ca0132_alt_add_output_enum(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO("Output Select",
+				    OUTPUT_SOURCE_ENUM, 1, 0, HDA_OUTPUT);
+	knew.info = ca0132_alt_output_select_get_info;
+	knew.get = ca0132_alt_output_select_get;
+	knew.put = ca0132_alt_output_select_put;
+	return snd_hda_ctl_add(codec, OUTPUT_SOURCE_ENUM,
+				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.
+ */
+static int ca0132_alt_add_input_enum(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO("Input Source",
+				    INPUT_SOURCE_ENUM, 1, 0, HDA_INPUT);
+	knew.info = ca0132_alt_input_source_info;
+	knew.get = ca0132_alt_input_source_get;
+	knew.put = ca0132_alt_input_source_put;
+	return snd_hda_ctl_add(codec, INPUT_SOURCE_ENUM,
+				snd_ctl_new1(&knew, codec));
+}
+
+/*
+ * Add mic boost enumerated control. Switches through 0dB to 30dB. This adds
+ * more control than the original mic boost, which is either full 30dB or off.
+ */
+static int ca0132_alt_add_mic_boost_enum(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO("Mic Boost Capture Switch",
+				    MIC_BOOST_ENUM, 1, 0, HDA_INPUT);
+	knew.info = ca0132_alt_mic_boost_info;
+	knew.get = ca0132_alt_mic_boost_get;
+	knew.put = ca0132_alt_mic_boost_put;
+	return snd_hda_ctl_add(codec, MIC_BOOST_ENUM,
+				snd_ctl_new1(&knew, codec));
+
+}
+
+/*
+ * Need to create slave controls for the alternate codecs that have surround
+ * capabilities.
+ */
+static const char * const ca0132_alt_slave_pfxs[] = {
+	"Front", "Surround", "Center", "LFE", NULL,
+};
+
+/*
+ * Also need special channel map, because the default one is incorrect.
+ * I think this has to do with the pin for rear surround being 0x11,
+ * and the center/lfe being 0x10. Usually the pin order is the opposite.
+ */
+const struct snd_pcm_chmap_elem ca0132_alt_chmaps[] = {
+	{ .channels = 2,
+	  .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } },
+	{ .channels = 4,
+	  .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR,
+		   SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } },
+	{ .channels = 6,
+	  .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR,
+		   SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE,
+		   SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } },
+	{ }
+};
+
+/* Add the correct chmap for streams with 6 channels. */
+static void ca0132_alt_add_chmap_ctls(struct hda_codec *codec)
+{
+	int err = 0;
+	struct hda_pcm *pcm;
+
+	list_for_each_entry(pcm, &codec->pcm_list_head, list) {
+		struct hda_pcm_stream *hinfo =
+			&pcm->stream[SNDRV_PCM_STREAM_PLAYBACK];
+		struct snd_pcm_chmap *chmap;
+		const struct snd_pcm_chmap_elem *elem;
+
+		elem = ca0132_alt_chmaps;
+		if (hinfo->channels_max == 6) {
+			err = snd_pcm_add_chmap_ctls(pcm->pcm,
+					SNDRV_PCM_STREAM_PLAYBACK,
+					elem, hinfo->channels_max, 0, &chmap);
+			if (err < 0)
+				codec_dbg(codec, "snd_pcm_add_chmap_ctls failed!");
+		}
+	}
+}
+
 /*
  * When changing Node IDs for Mixer Controls below, make sure to update
  * Node IDs in ca0132_config() as well.
@@ -3955,10 +5727,55 @@ static struct snd_kcontrol_new ca0132_mixer[] = {
 	{ } /* end */
 };
 
+/*
+ * SBZ specific control mixer. Removes auto-detect for mic, and adds surround
+ * controls. Also sets both the Front Playback and Capture Volume controls to
+ * alt so they set the DSP's decibel level.
+ */
+static struct snd_kcontrol_new sbz_mixer[] = {
+	CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT),
+	CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x03, 1, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x03, 1, 0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x03, 2, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x03, 2, 0, HDA_OUTPUT),
+	CA0132_ALT_CODEC_VOL("Capture Volume", 0x07, HDA_INPUT),
+	CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT),
+	HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT),
+	CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch",
+				VNID_HP_ASEL, 1, HDA_OUTPUT),
+	{ } /* end */
+};
+
+/*
+ * Same as the Sound Blaster Z, except doesn't use the alt volume for capture
+ * because it doesn't set decibel levels for the DSP for capture.
+ */
+static struct snd_kcontrol_new r3di_mixer[] = {
+	CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT),
+	CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x03, 1, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x03, 1, 0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x03, 2, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x03, 2, 0, HDA_OUTPUT),
+	CA0132_CODEC_VOL("Capture Volume", VNID_MIC, HDA_INPUT),
+	CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT),
+	HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT),
+	CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch",
+				VNID_HP_ASEL, 1, HDA_OUTPUT),
+	{ } /* end */
+};
+
 static int ca0132_build_controls(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
-	int i, num_fx;
+	int i, num_fx, num_sliders;
 	int err = 0;
 
 	/* Add Mixer controls */
@@ -3967,29 +5784,94 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
+	/* Setup vmaster with surround slaves for desktop ca0132 devices */
+	if (spec->use_alt_functions) {
+		snd_hda_set_vmaster_tlv(codec, spec->dacs[0], HDA_OUTPUT,
+					spec->tlv);
+		snd_hda_add_vmaster(codec, "Master Playback Volume",
+					spec->tlv, ca0132_alt_slave_pfxs,
+					"Playback Volume");
+		err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
+					    NULL, ca0132_alt_slave_pfxs,
+					    "Playback Switch",
+					    true, &spec->vmaster_mute.sw_kctl);
+
+	}
 
 	/* Add in and out effects controls.
 	 * VoiceFX, PE and CrystalVoice are added separately.
 	 */
 	num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
 	for (i = 0; i < num_fx; i++) {
+		/* SBZ breaks if Echo Cancellation is used */
+		if (spec->quirk == QUIRK_SBZ) {
+			if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID +
+						OUT_EFFECTS_COUNT))
+				continue;
+		}
+
 		err = add_fx_switch(codec, ca0132_effects[i].nid,
 				    ca0132_effects[i].name,
 				    ca0132_effects[i].direct);
 		if (err < 0)
 			return err;
 	}
+	/*
+	 * If codec has use_alt_controls set to true, add effect level sliders,
+	 * EQ presets, and Smart Volume presets. Also, change names to add FX
+	 * prefix, and change PlayEnhancement and CrystalVoice to match.
+	 */
+	if (spec->use_alt_controls) {
+		ca0132_alt_add_svm_enum(codec);
+		add_ca0132_alt_eq_presets(codec);
+		err = add_fx_switch(codec, PLAY_ENHANCEMENT,
+					"Enable OutFX", 0);
+		if (err < 0)
+			return err;
 
-	err = add_fx_switch(codec, PLAY_ENHANCEMENT, "PlayEnhancement", 0);
-	if (err < 0)
-		return err;
+		err = add_fx_switch(codec, CRYSTAL_VOICE,
+					"Enable InFX", 1);
+		if (err < 0)
+			return err;
 
-	err = add_fx_switch(codec, CRYSTAL_VOICE, "CrystalVoice", 1);
-	if (err < 0)
-		return err;
+		num_sliders = OUT_EFFECTS_COUNT - 1;
+		for (i = 0; i < num_sliders; i++) {
+			err = ca0132_alt_add_effect_slider(codec,
+					    ca0132_effects[i].nid,
+					    ca0132_effects[i].name,
+					    ca0132_effects[i].direct);
+			if (err < 0)
+				return err;
+		}
 
+		err = ca0132_alt_add_effect_slider(codec, XBASS_XOVER,
+					"X-Bass Crossover", EFX_DIR_OUT);
+
+		if (err < 0)
+			return err;
+	} else {
+		err = add_fx_switch(codec, PLAY_ENHANCEMENT,
+					"PlayEnhancement", 0);
+		if (err < 0)
+			return err;
+
+		err = add_fx_switch(codec, CRYSTAL_VOICE,
+					"CrystalVoice", 1);
+		if (err < 0)
+			return err;
+	}
 	add_voicefx(codec);
 
+	/*
+	 * If the codec uses alt_functions, you need the enumerated controls
+	 * to select the new outputs and inputs, plus add the new mic boost
+	 * setting control.
+	 */
+	if (spec->use_alt_functions) {
+		ca0132_alt_add_output_enum(codec);
+		ca0132_alt_add_input_enum(codec);
+		ca0132_alt_add_mic_boost_enum(codec);
+	}
 #ifdef ENABLE_TUNING_CONTROLS
 	add_tuning_ctls(codec);
 #endif
@@ -4014,6 +5896,10 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
+
+	if (spec->use_alt_functions)
+		ca0132_alt_add_chmap_ctls(codec);
+
 	return 0;
 }
 
@@ -4068,6 +5954,11 @@ static int ca0132_build_pcms(struct hda_codec *codec)
 	info = snd_hda_codec_pcm_new(codec, "CA0132 Analog");
 	if (!info)
 		return -ENOMEM;
+	if (spec->use_alt_functions) {
+		info->own_chmap = true;
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap
+			= ca0132_alt_chmaps;
+	}
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback;
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0];
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
@@ -4076,12 +5967,16 @@ static int ca0132_build_pcms(struct hda_codec *codec)
 	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
 	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0];
 
-	info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2");
-	if (!info)
-		return -ENOMEM;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1];
+	/* With the DSP enabled, desktops don't use this ADC. */
+	if (spec->use_alt_functions) {
+		info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2");
+		if (!info)
+			return -ENOMEM;
+		info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+			ca0132_pcm_analog_capture;
+		info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
+		info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1];
+	}
 
 	info = snd_hda_codec_pcm_new(codec, "CA0132 What U Hear");
 	if (!info)
@@ -4288,6 +6183,196 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
 }
 
 /*
+ * Recon3Di r3di_setup_defaults sub functions.
+ */
+
+static void r3di_dsp_scp_startup(struct hda_codec *codec)
+{
+	unsigned int tmp;
+
+	tmp = 0x00000000;
+	dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
+
+	tmp = 0x00000001;
+	dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
+
+	tmp = 0x00000004;
+	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+	tmp = 0x00000005;
+	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+	tmp = 0x00000000;
+	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+}
+
+static void r3di_dsp_initial_mic_setup(struct hda_codec *codec)
+{
+	unsigned int tmp;
+
+	/* Mic 1 Setup */
+	chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+	chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+	/* This ConnPointID is unique to Recon3Di. Haven't seen it elsewhere */
+	chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+	tmp = FLOAT_ONE;
+	dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+	/* Mic 2 Setup, even though it isn't connected on SBZ */
+	chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000);
+	chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000);
+	chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x80, 0x01, tmp);
+}
+
+/*
+ * Initialize Sound Blaster Z analog microphones.
+ */
+static void sbz_init_analog_mics(struct hda_codec *codec)
+{
+	unsigned int tmp;
+
+	/* Mic 1 Setup */
+	chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+	chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+	tmp = FLOAT_THREE;
+	dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+	/* Mic 2 Setup, even though it isn't connected on SBZ */
+	chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000);
+	chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000);
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x80, 0x01, tmp);
+
+}
+
+/*
+ * Sets the source of stream 0x14 to connpointID 0x48, and the destination
+ * connpointID to 0x91. If this isn't done, the destination is 0x71, and
+ * you get no sound. I'm guessing this has to do with the Sound Blaster Z
+ * having an updated DAC, which changes the destination to that DAC.
+ */
+static void sbz_connect_streams(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	codec_dbg(codec, "Connect Streams entered, mutex locked and loaded.\n");
+
+	chipio_set_stream_channels(codec, 0x0C, 6);
+	chipio_set_stream_control(codec, 0x0C, 1);
+
+	/* This value is 0x43 for 96khz, and 0x83 for 192khz. */
+	chipio_write_no_mutex(codec, 0x18a020, 0x00000043);
+
+	/* Setup stream 0x14 with it's source and destination points */
+	chipio_set_stream_source_dest(codec, 0x14, 0x48, 0x91);
+	chipio_set_conn_rate_no_mutex(codec, 0x48, SR_96_000);
+	chipio_set_conn_rate_no_mutex(codec, 0x91, SR_96_000);
+	chipio_set_stream_channels(codec, 0x14, 2);
+	chipio_set_stream_control(codec, 0x14, 1);
+
+	codec_dbg(codec, "Connect Streams exited, mutex released.\n");
+
+	mutex_unlock(&spec->chipio_mutex);
+
+}
+
+/*
+ * Write data through ChipIO to setup proper stream destinations.
+ * Not sure how it exactly works, but it seems to direct data
+ * to different destinations. Example is f8 to c0, e0 to c0.
+ * All I know is, if you don't set these, you get no sound.
+ */
+static void sbz_chipio_startup_data(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	mutex_lock(&spec->chipio_mutex);
+	codec_dbg(codec, "Startup Data entered, mutex locked and loaded.\n");
+
+	/* These control audio output */
+	chipio_write_no_mutex(codec, 0x190060, 0x0001f8c0);
+	chipio_write_no_mutex(codec, 0x190064, 0x0001f9c1);
+	chipio_write_no_mutex(codec, 0x190068, 0x0001fac6);
+	chipio_write_no_mutex(codec, 0x19006c, 0x0001fbc7);
+	/* Signal to update I think */
+	chipio_write_no_mutex(codec, 0x19042c, 0x00000001);
+
+	chipio_set_stream_channels(codec, 0x0C, 6);
+	chipio_set_stream_control(codec, 0x0C, 1);
+	/* No clue what these control */
+	chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0);
+	chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1);
+	chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2);
+	chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3);
+	chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4);
+	chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5);
+	chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6);
+	chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7);
+	chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8);
+	chipio_write_no_mutex(codec, 0x190054, 0x0001edc9);
+	chipio_write_no_mutex(codec, 0x190058, 0x0001eaca);
+	chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb);
+
+	chipio_write_no_mutex(codec, 0x19042c, 0x00000001);
+
+	codec_dbg(codec, "Startup Data exited, mutex released.\n");
+	mutex_unlock(&spec->chipio_mutex);
+}
+
+/*
+ * Sound Blaster Z uses these after DSP is loaded. Weird SCP commands
+ * without a 0x20 source like normal.
+ */
+static void sbz_dsp_scp_startup(struct hda_codec *codec)
+{
+	unsigned int tmp;
+
+	tmp = 0x00000003;
+	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+	tmp = 0x00000000;
+	dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
+
+	tmp = 0x00000001;
+	dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
+
+	tmp = 0x00000004;
+	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+	tmp = 0x00000005;
+	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+	tmp = 0x00000000;
+	dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+}
+
+static void sbz_dsp_initial_mic_setup(struct hda_codec *codec)
+{
+	unsigned int tmp;
+
+	chipio_set_stream_control(codec, 0x03, 0);
+	chipio_set_stream_control(codec, 0x04, 0);
+
+	chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+	chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+
+	tmp = FLOAT_THREE;
+	dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+	chipio_set_stream_control(codec, 0x03, 1);
+	chipio_set_stream_control(codec, 0x04, 1);
+
+	chipio_write(codec, 0x18b098, 0x0000000c);
+	chipio_write(codec, 0x18b09C, 0x0000000c);
+}
+
+/*
  * Setup default parameters for DSP
  */
 static void ca0132_setup_defaults(struct hda_codec *codec)
@@ -4332,16 +6417,159 @@ static void ca0132_setup_defaults(struct hda_codec *codec)
 }
 
 /*
+ * Setup default parameters for Recon3Di DSP.
+ */
+
+static void r3di_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;
+
+	r3di_dsp_scp_startup(codec);
+
+	r3di_dsp_initial_mic_setup(codec);
+
+	/*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);
+
+	r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED);
+
+	/* Setup effect defaults */
+	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]);
+		}
+	}
+
+}
+
+/*
+ * Setup default parameters for the Sound Blaster Z DSP. A lot more going on
+ * than the Chromebook setup.
+ */
+static void sbz_setup_defaults(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp, stream_format;
+	int num_fx;
+	int idx, i;
+
+	if (spec->dsp_state != DSP_DOWNLOADED)
+		return;
+
+	sbz_dsp_scp_startup(codec);
+
+	sbz_init_analog_mics(codec);
+
+	sbz_connect_streams(codec);
+
+	sbz_chipio_startup_data(codec);
+
+	chipio_set_stream_control(codec, 0x03, 1);
+	chipio_set_stream_control(codec, 0x04, 1);
+
+	/*
+	 * Sets internal input loopback to off, used to have a switch to
+	 * enable input loopback, but turned out to be way too buggy.
+	 */
+	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);
+
+	sbz_dsp_initial_mic_setup(codec);
+
+
+	/* 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]);
+		}
+	}
+
+	/*
+	 * Have to make a stream to bind the sound output to, otherwise
+	 * you'll get dead audio. Before I did this, it would bind to an
+	 * audio input, and would never work
+	 */
+	stream_format = snd_hdac_calc_stream_format(48000, 2,
+			SNDRV_PCM_FORMAT_S32_LE, 32, 0);
+
+	snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
+					0, stream_format);
+
+	snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
+
+	snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
+					0, stream_format);
+
+	snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
+}
+
+/*
  * Initialization of flags in chip
  */
 static void ca0132_init_flags(struct hda_codec *codec)
 {
-	chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0);
-	chipio_set_control_flag(codec, CONTROL_FLAG_PORT_A_COMMON_MODE, 0);
-	chipio_set_control_flag(codec, CONTROL_FLAG_PORT_D_COMMON_MODE, 0);
-	chipio_set_control_flag(codec, CONTROL_FLAG_PORT_A_10KOHM_LOAD, 0);
-	chipio_set_control_flag(codec, CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0);
-	chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_HIGH_PASS, 1);
+	struct ca0132_spec *spec = codec->spec;
+
+	if (spec->use_alt_functions) {
+		chipio_set_control_flag(codec, CONTROL_FLAG_DSP_96KHZ, 1);
+		chipio_set_control_flag(codec, CONTROL_FLAG_DAC_96KHZ, 1);
+		chipio_set_control_flag(codec, CONTROL_FLAG_ADC_B_96KHZ, 1);
+		chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_96KHZ, 1);
+		chipio_set_control_flag(codec, CONTROL_FLAG_SRC_RATE_96KHZ, 1);
+		chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0);
+		chipio_set_control_flag(codec, CONTROL_FLAG_SPDIF2OUT, 0);
+		chipio_set_control_flag(codec,
+				CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0);
+		chipio_set_control_flag(codec,
+				CONTROL_FLAG_PORT_A_10KOHM_LOAD, 1);
+	} else {
+		chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0);
+		chipio_set_control_flag(codec,
+				CONTROL_FLAG_PORT_A_COMMON_MODE, 0);
+		chipio_set_control_flag(codec,
+				CONTROL_FLAG_PORT_D_COMMON_MODE, 0);
+		chipio_set_control_flag(codec,
+				CONTROL_FLAG_PORT_A_10KOHM_LOAD, 0);
+		chipio_set_control_flag(codec,
+				CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0);
+		chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_HIGH_PASS, 1);
+	}
 }
 
 /*
@@ -4349,6 +6577,16 @@ static void ca0132_init_flags(struct hda_codec *codec)
  */
 static void ca0132_init_params(struct hda_codec *codec)
 {
+	struct ca0132_spec *spec = codec->spec;
+
+	if (spec->use_alt_functions) {
+		chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000);
+		chipio_set_conn_rate(codec, 0x0B, SR_48_000);
+		chipio_set_control_param(codec, CONTROL_PARAM_SPDIF1_SOURCE, 0);
+		chipio_set_control_param(codec, 0, 0);
+		chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0);
+	}
+
 	chipio_set_control_param(codec, CONTROL_PARAM_PORTA_160OHM_GAIN, 6);
 	chipio_set_control_param(codec, CONTROL_PARAM_PORTD_160OHM_GAIN, 6);
 }
@@ -4370,11 +6608,49 @@ static void ca0132_set_dsp_msr(struct hda_codec *codec, bool is96k)
 static bool ca0132_download_dsp_images(struct hda_codec *codec)
 {
 	bool dsp_loaded = false;
+	struct ca0132_spec *spec = codec->spec;
 	const struct dsp_image_seg *dsp_os_image;
 	const struct firmware *fw_entry;
-
-	if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0)
-		return false;
+	/*
+	 * Alternate firmwares for different variants. The Recon3Di apparently
+	 * can use the default firmware, but I'll leave the option in case
+	 * it needs it again.
+	 */
+	switch (spec->quirk) {
+	case QUIRK_SBZ:
+		if (request_firmware(&fw_entry, SBZ_EFX_FILE,
+					codec->card->dev) != 0) {
+			codec_dbg(codec, "SBZ alt firmware not detected. ");
+			spec->alt_firmware_present = false;
+		} else {
+			codec_dbg(codec, "Sound Blaster Z firmware selected.");
+			spec->alt_firmware_present = true;
+		}
+		break;
+	case QUIRK_R3DI:
+		if (request_firmware(&fw_entry, R3DI_EFX_FILE,
+					codec->card->dev) != 0) {
+			codec_dbg(codec, "Recon3Di alt firmware not detected.");
+			spec->alt_firmware_present = false;
+		} else {
+			codec_dbg(codec, "Recon3Di firmware selected.");
+			spec->alt_firmware_present = true;
+		}
+		break;
+	default:
+		spec->alt_firmware_present = false;
+		break;
+	}
+	/*
+	 * Use default ctefx.bin if no alt firmware is detected, or if none
+	 * exists for your particular codec.
+	 */
+	if (!spec->alt_firmware_present) {
+		codec_dbg(codec, "Default firmware selected.");
+		if (request_firmware(&fw_entry, EFX_FILE,
+					codec->card->dev) != 0)
+			return false;
+	}
 
 	dsp_os_image = (struct dsp_image_seg *)(fw_entry->data);
 	if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) {
@@ -4402,13 +6678,17 @@ static void ca0132_download_dsp(struct hda_codec *codec)
 		return; /* don't retry failures */
 
 	chipio_enable_clocks(codec);
-	spec->dsp_state = DSP_DOWNLOADING;
-	if (!ca0132_download_dsp_images(codec))
-		spec->dsp_state = DSP_DOWNLOAD_FAILED;
-	else
-		spec->dsp_state = DSP_DOWNLOADED;
+	if (spec->dsp_state != DSP_DOWNLOADED) {
+		spec->dsp_state = DSP_DOWNLOADING;
 
-	if (spec->dsp_state == DSP_DOWNLOADED)
+		if (!ca0132_download_dsp_images(codec))
+			spec->dsp_state = DSP_DOWNLOAD_FAILED;
+		else
+			spec->dsp_state = DSP_DOWNLOADED;
+	}
+
+	/* For codecs using alt functions, this is already done earlier */
+	if (spec->dsp_state == DSP_DOWNLOADED && (!spec->use_alt_functions))
 		ca0132_set_dsp_msr(codec, true);
 }
 
@@ -4454,6 +6734,10 @@ static void ca0132_init_unsol(struct hda_codec *codec)
 					    amic_callback);
 	snd_hda_jack_detect_enable_callback(codec, UNSOL_TAG_DSP,
 					    ca0132_process_dsp_response);
+	/* Front headphone jack detection */
+	if (spec->use_alt_functions)
+		snd_hda_jack_detect_enable_callback(codec,
+			spec->unsol_tag_front_hp, hp_callback);
 }
 
 /*
@@ -4476,7 +6760,8 @@ static struct hda_verb ca0132_base_exit_verbs[] = {
 	{}
 };
 
-/* Other verbs tables.  Sends after DSP download. */
+/* Other verbs tables. Sends after DSP download. */
+
 static struct hda_verb ca0132_init_verbs0[] = {
 	/* chip init verbs */
 	{0x15, 0x70D, 0xF0},
@@ -4506,8 +6791,27 @@ static struct hda_verb ca0132_init_verbs0[] = {
 	{0x15, 0x546, 0xC9},
 	{0x15, 0x53B, 0xCE},
 	{0x15, 0x5E8, 0xC9},
-	{0x15, 0x717, 0x0D},
-	{0x15, 0x718, 0x20},
+	{}
+};
+
+/* Extra init verbs for SBZ */
+static struct hda_verb sbz_init_verbs[] = {
+	{0x15, 0x70D, 0x20},
+	{0x15, 0x70E, 0x19},
+	{0x15, 0x707, 0x00},
+	{0x15, 0x539, 0xCE},
+	{0x15, 0x546, 0xC9},
+	{0x15, 0x70D, 0xB7},
+	{0x15, 0x70E, 0x09},
+	{0x15, 0x707, 0x10},
+	{0x15, 0x70D, 0xAF},
+	{0x15, 0x70E, 0x09},
+	{0x15, 0x707, 0x01},
+	{0x15, 0x707, 0x05},
+	{0x15, 0x70D, 0x73},
+	{0x15, 0x70E, 0x09},
+	{0x15, 0x707, 0x14},
+	{0x15, 0x6FF, 0xC4},
 	{}
 };
 
@@ -4521,7 +6825,11 @@ static void ca0132_init_chip(struct hda_codec *codec)
 	mutex_init(&spec->chipio_mutex);
 
 	spec->cur_out_type = SPEAKER_OUT;
-	spec->cur_mic_type = DIGITAL_MIC;
+	if (!spec->use_alt_functions)
+		spec->cur_mic_type = DIGITAL_MIC;
+	else
+		spec->cur_mic_type = REAR_MIC;
+
 	spec->cur_mic_boost = 0;
 
 	for (i = 0; i < VNODES_COUNT; i++) {
@@ -4539,6 +6847,15 @@ static void ca0132_init_chip(struct hda_codec *codec)
 		on = (unsigned int)ca0132_effects[i].reqs[0];
 		spec->effects_switch[i] = on ? 1 : 0;
 	}
+	/*
+	 * Sets defaults for the effect slider controls, only for alternative
+	 * ca0132 codecs. Also sets x-bass crossover frequency to 80hz.
+	 */
+	if (spec->use_alt_controls) {
+		spec->xbass_xover_freq = 8;
+		for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++)
+			spec->fx_ctl_val[i] = effect_slider_defaults[i];
+	}
 
 	spec->voicefx_val = 0;
 	spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID] = 1;
@@ -4549,6 +6866,120 @@ static void ca0132_init_chip(struct hda_codec *codec)
 #endif
 }
 
+/*
+ * Recon3Di exit specific commands.
+ */
+/* prevents popping noise on shutdown */
+static void r3di_gpio_shutdown(struct hda_codec *codec)
+{
+	snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x00);
+}
+
+/*
+ * Sound Blaster Z exit specific commands.
+ */
+static void sbz_region2_exit(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int i;
+
+	for (i = 0; i < 4; i++)
+		writeb(0x0, spec->mem_base + 0x100);
+	for (i = 0; i < 8; i++)
+		writeb(0xb3, spec->mem_base + 0x304);
+	/*
+	 * I believe these are GPIO, with the right most hex digit being the
+	 * gpio pin, and the second digit being on or off. We see this more in
+	 * the input/output select functions.
+	 */
+	writew(0x0000, spec->mem_base + 0x320);
+	writew(0x0001, spec->mem_base + 0x320);
+	writew(0x0104, spec->mem_base + 0x320);
+	writew(0x0005, spec->mem_base + 0x320);
+	writew(0x0007, spec->mem_base + 0x320);
+}
+
+static void sbz_set_pin_ctl_default(struct hda_codec *codec)
+{
+	hda_nid_t pins[5] = {0x0B, 0x0C, 0x0E, 0x12, 0x13};
+	unsigned int i;
+
+	snd_hda_codec_write(codec, 0x11, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40);
+
+	for (i = 0; i < 5; i++)
+		snd_hda_codec_write(codec, pins[i], 0,
+				AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00);
+}
+
+static void sbz_clear_unsolicited(struct hda_codec *codec)
+{
+	hda_nid_t pins[7] = {0x0B, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13};
+	unsigned int i;
+
+	for (i = 0; i < 7; i++) {
+		snd_hda_codec_write(codec, pins[i], 0,
+				AC_VERB_SET_UNSOLICITED_ENABLE, 0x00);
+	}
+}
+
+/* On shutdown, sends commands in sets of three */
+static void sbz_gpio_shutdown_commands(struct hda_codec *codec, int dir,
+							int mask, int data)
+{
+	if (dir >= 0)
+		snd_hda_codec_write(codec, 0x01, 0,
+				AC_VERB_SET_GPIO_DIRECTION, dir);
+	if (mask >= 0)
+		snd_hda_codec_write(codec, 0x01, 0,
+				AC_VERB_SET_GPIO_MASK, mask);
+
+	if (data >= 0)
+		snd_hda_codec_write(codec, 0x01, 0,
+				AC_VERB_SET_GPIO_DATA, data);
+}
+
+static void sbz_exit_chip(struct hda_codec *codec)
+{
+	chipio_set_stream_control(codec, 0x03, 0);
+	chipio_set_stream_control(codec, 0x04, 0);
+
+	/* Mess with GPIO */
+	sbz_gpio_shutdown_commands(codec, 0x07, 0x07, -1);
+	sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x05);
+	sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x01);
+
+	chipio_set_stream_control(codec, 0x14, 0);
+	chipio_set_stream_control(codec, 0x0C, 0);
+
+	chipio_set_conn_rate(codec, 0x41, SR_192_000);
+	chipio_set_conn_rate(codec, 0x91, SR_192_000);
+
+	chipio_write(codec, 0x18a020, 0x00000083);
+
+	sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x03);
+	sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x07);
+	sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x06);
+
+	chipio_set_stream_control(codec, 0x0C, 0);
+
+	chipio_set_control_param(codec, 0x0D, 0x24);
+
+	sbz_clear_unsolicited(codec);
+	sbz_set_pin_ctl_default(codec);
+
+	snd_hda_codec_write(codec, 0x0B, 0,
+		AC_VERB_SET_EAPD_BTLENABLE, 0x00);
+
+	if (dspload_is_loaded(codec))
+		dsp_reset(codec);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+		VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE, 0x00);
+
+	sbz_region2_exit(codec);
+}
+
 static void ca0132_exit_chip(struct hda_codec *codec)
 {
 	/* put any chip cleanup stuffs here. */
@@ -4557,28 +6988,264 @@ static void ca0132_exit_chip(struct hda_codec *codec)
 		dsp_reset(codec);
 }
 
+/*
+ * This fixes a problem that was hard to reproduce. Very rarely, I would
+ * boot up, and there would be no sound, but the DSP indicated it had loaded
+ * properly. I did a few memory dumps to see if anything was different, and
+ * there were a few areas of memory uninitialized with a1a2a3a4. This function
+ * checks if those areas are uninitialized, and if they are, it'll attempt to
+ * reload the card 3 times. Usually it fixes by the second.
+ */
+static void sbz_dsp_startup_check(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int dsp_data_check[4];
+	unsigned int cur_address = 0x390;
+	unsigned int i;
+	unsigned int failure = 0;
+	unsigned int reload = 3;
+
+	if (spec->startup_check_entered)
+		return;
+
+	spec->startup_check_entered = true;
+
+	for (i = 0; i < 4; i++) {
+		chipio_read(codec, cur_address, &dsp_data_check[i]);
+		cur_address += 0x4;
+	}
+	for (i = 0; i < 4; i++) {
+		if (dsp_data_check[i] == 0xa1a2a3a4)
+			failure = 1;
+	}
+
+	codec_dbg(codec, "Startup Check: %d ", failure);
+	if (failure)
+		codec_info(codec, "DSP not initialized properly. Attempting to fix.");
+	/*
+	 * While the failure condition is true, and we haven't reached our
+	 * three reload limit, continue trying to reload the driver and
+	 * fix the issue.
+	 */
+	while (failure && (reload != 0)) {
+		codec_info(codec, "Reloading... Tries left: %d", reload);
+		sbz_exit_chip(codec);
+		spec->dsp_state = DSP_DOWNLOAD_INIT;
+		codec->patch_ops.init(codec);
+		failure = 0;
+		for (i = 0; i < 4; i++) {
+			chipio_read(codec, cur_address, &dsp_data_check[i]);
+			cur_address += 0x4;
+		}
+		for (i = 0; i < 4; i++) {
+			if (dsp_data_check[i] == 0xa1a2a3a4)
+				failure = 1;
+		}
+		reload--;
+	}
+
+	if (!failure && reload < 3)
+		codec_info(codec, "DSP fixed.");
+
+	if (!failure)
+		return;
+
+	codec_info(codec, "DSP failed to initialize properly. Either try a full shutdown or a suspend to clear the internal memory.");
+}
+
+/*
+ * This is for the extra volume verbs 0x797 (left) and 0x798 (right). These add
+ * extra precision for decibel values. If you had the dB value in floating point
+ * you would take the value after the decimal point, multiply by 64, and divide
+ * by 2. So for 8.59, it's (59 * 64) / 100. Useful if someone wanted to
+ * implement fixed point or floating point dB volumes. For now, I'll set them
+ * to 0 just incase a value has lingered from a boot into Windows.
+ */
+static void ca0132_alt_vol_setup(struct hda_codec *codec)
+{
+	snd_hda_codec_write(codec, 0x02, 0, 0x797, 0x00);
+	snd_hda_codec_write(codec, 0x02, 0, 0x798, 0x00);
+	snd_hda_codec_write(codec, 0x03, 0, 0x797, 0x00);
+	snd_hda_codec_write(codec, 0x03, 0, 0x798, 0x00);
+	snd_hda_codec_write(codec, 0x04, 0, 0x797, 0x00);
+	snd_hda_codec_write(codec, 0x04, 0, 0x798, 0x00);
+	snd_hda_codec_write(codec, 0x07, 0, 0x797, 0x00);
+	snd_hda_codec_write(codec, 0x07, 0, 0x798, 0x00);
+}
+
+/*
+ * Extra commands that don't really fit anywhere else.
+ */
+static void sbz_pre_dsp_setup(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	writel(0x00820680, spec->mem_base + 0x01C);
+	writel(0x00820680, spec->mem_base + 0x01C);
+
+	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc);
+	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd);
+	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe);
+	snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff);
+
+	chipio_write(codec, 0x18b0a4, 0x000000c2);
+
+	snd_hda_codec_write(codec, 0x11, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44);
+}
+
+/*
+ * Extra commands that don't really fit anywhere else.
+ */
+static void r3di_pre_dsp_setup(struct hda_codec *codec)
+{
+	chipio_write(codec, 0x18b0a4, 0x000000c2);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x1E);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x1C);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_DATA_WRITE, 0x5B);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x20);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x19);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_DATA_WRITE, 0x00);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_DATA_WRITE, 0x40);
+
+	snd_hda_codec_write(codec, 0x11, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, 0x04);
+}
+
+
+/*
+ * These are sent before the DSP is downloaded. Not sure
+ * what they do, or if they're necessary. Could possibly
+ * be removed. Figure they're better to leave in.
+ */
+static void sbz_region2_startup(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	writel(0x00000000, spec->mem_base + 0x400);
+	writel(0x00000000, spec->mem_base + 0x408);
+	writel(0x00000000, spec->mem_base + 0x40C);
+	writel(0x00880680, spec->mem_base + 0x01C);
+	writel(0x00000083, spec->mem_base + 0xC0C);
+	writel(0x00000030, spec->mem_base + 0xC00);
+	writel(0x00000000, spec->mem_base + 0xC04);
+	writel(0x00000003, spec->mem_base + 0xC0C);
+	writel(0x00000003, spec->mem_base + 0xC0C);
+	writel(0x00000003, spec->mem_base + 0xC0C);
+	writel(0x00000003, spec->mem_base + 0xC0C);
+	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);
+}
+
+/*
+ * Extra init functions for alternative ca0132 codecs. Done
+ * here so they don't clutter up the main ca0132_init function
+ * anymore than they have to.
+ */
+static void ca0132_alt_init(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	ca0132_alt_vol_setup(codec);
+
+	switch (spec->quirk) {
+	case QUIRK_SBZ:
+		codec_dbg(codec, "SBZ alt_init");
+		ca0132_gpio_init(codec);
+		sbz_pre_dsp_setup(codec);
+		snd_hda_sequence_write(codec, spec->chip_init_verbs);
+		snd_hda_sequence_write(codec, spec->sbz_init_verbs);
+		break;
+	case QUIRK_R3DI:
+		codec_dbg(codec, "R3DI alt_init");
+		ca0132_gpio_init(codec);
+		ca0132_gpio_setup(codec);
+		r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADING);
+		r3di_pre_dsp_setup(codec);
+		snd_hda_sequence_write(codec, spec->chip_init_verbs);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x6FF, 0xC4);
+		break;
+	}
+}
+
 static int ca0132_init(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	int i;
+	bool dsp_loaded;
+
+	/*
+	 * If the DSP is already downloaded, and init has been entered again,
+	 * there's only two reasons for it. One, the codec has awaken from a
+	 * suspended state, and in that case dspload_is_loaded will return
+	 * false, and the init will be ran again. The other reason it gets
+	 * re entered is on startup for some reason it triggers a suspend and
+	 * resume state. In this case, it will check if the DSP is downloaded,
+	 * and not run the init function again. For codecs using alt_functions,
+	 * it will check if the DSP is loaded properly.
+	 */
+	if (spec->dsp_state == DSP_DOWNLOADED) {
+		dsp_loaded = dspload_is_loaded(codec);
+		if (!dsp_loaded) {
+			spec->dsp_reload = true;
+			spec->dsp_state = DSP_DOWNLOAD_INIT;
+		} else {
+			if (spec->quirk == QUIRK_SBZ)
+				sbz_dsp_startup_check(codec);
+			return 0;
+		}
+	}
 
 	if (spec->dsp_state != DSP_DOWNLOAD_FAILED)
 		spec->dsp_state = DSP_DOWNLOAD_INIT;
 	spec->curr_chip_addx = INVALID_CHIP_ADDRESS;
 
+	if (spec->quirk == QUIRK_SBZ)
+		sbz_region2_startup(codec);
+
 	snd_hda_power_up_pm(codec);
 
 	ca0132_init_unsol(codec);
-
 	ca0132_init_params(codec);
 	ca0132_init_flags(codec);
+
 	snd_hda_sequence_write(codec, spec->base_init_verbs);
+
+	if (spec->quirk != QUIRK_NONE)
+		ca0132_alt_init(codec);
+
 	ca0132_download_dsp(codec);
+
 	ca0132_refresh_widget_caps(codec);
-	ca0132_setup_defaults(codec);
-	ca0132_init_analog_mic2(codec);
-	ca0132_init_dmic(codec);
+
+	if (spec->quirk == QUIRK_SBZ)
+		writew(0x0107, spec->mem_base + 0x320);
+
+	switch (spec->quirk) {
+	case QUIRK_R3DI:
+		r3di_setup_defaults(codec);
+		break;
+	case QUIRK_NONE:
+	case QUIRK_ALIENWARE:
+		ca0132_setup_defaults(codec);
+		ca0132_init_analog_mic2(codec);
+		ca0132_init_dmic(codec);
+		break;
+	}
 
 	for (i = 0; i < spec->num_outputs; i++)
 		init_output(codec, spec->out_pins[i], spec->dacs[0]);
@@ -4590,14 +7257,45 @@ static int ca0132_init(struct hda_codec *codec)
 
 	init_input(codec, cfg->dig_in_pin, spec->dig_in);
 
-	snd_hda_sequence_write(codec, spec->chip_init_verbs);
-	snd_hda_sequence_write(codec, spec->spec_init_verbs);
+	if (!spec->use_alt_functions) {
+		snd_hda_sequence_write(codec, spec->chip_init_verbs);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PARAM_EX_ID_SET, 0x0D);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PARAM_EX_VALUE_SET, 0x20);
+	}
 
-	ca0132_select_out(codec);
-	ca0132_select_mic(codec);
+	if (spec->quirk == QUIRK_SBZ)
+		ca0132_gpio_setup(codec);
+
+	snd_hda_sequence_write(codec, spec->spec_init_verbs);
+	switch (spec->quirk) {
+	case QUIRK_SBZ:
+		sbz_setup_defaults(codec);
+		ca0132_alt_select_out(codec);
+		ca0132_alt_select_in(codec);
+		break;
+	case QUIRK_R3DI:
+		ca0132_alt_select_out(codec);
+		ca0132_alt_select_in(codec);
+		break;
+	default:
+		ca0132_select_out(codec);
+		ca0132_select_mic(codec);
+		break;
+	}
 
 	snd_hda_jack_report_sync(codec);
 
+	/*
+	 * Re set the PlayEnhancement switch on a resume event, because the
+	 * controls will not be reloaded.
+	 */
+	if (spec->dsp_reload) {
+		spec->dsp_reload = false;
+		ca0132_pe_switch_set(codec);
+	}
+
 	snd_hda_power_down_pm(codec);
 
 	return 0;
@@ -4609,19 +7307,39 @@ static void ca0132_free(struct hda_codec *codec)
 
 	cancel_delayed_work_sync(&spec->unsol_hp_work);
 	snd_hda_power_up(codec);
-	snd_hda_sequence_write(codec, spec->base_exit_verbs);
-	ca0132_exit_chip(codec);
+	switch (spec->quirk) {
+	case QUIRK_SBZ:
+		sbz_exit_chip(codec);
+		break;
+	case QUIRK_R3DI:
+		r3di_gpio_shutdown(codec);
+		snd_hda_sequence_write(codec, spec->base_exit_verbs);
+		ca0132_exit_chip(codec);
+		break;
+	default:
+		snd_hda_sequence_write(codec, spec->base_exit_verbs);
+		ca0132_exit_chip(codec);
+		break;
+	}
 	snd_hda_power_down(codec);
+	if (spec->mem_base)
+		iounmap(spec->mem_base);
 	kfree(spec->spec_init_verbs);
 	kfree(codec->spec);
 }
 
+static void ca0132_reboot_notify(struct hda_codec *codec)
+{
+	codec->patch_ops.free(codec);
+}
+
 static const struct hda_codec_ops ca0132_patch_ops = {
 	.build_controls = ca0132_build_controls,
 	.build_pcms = ca0132_build_pcms,
 	.init = ca0132_init,
 	.free = ca0132_free,
 	.unsol_event = snd_hda_jack_unsol_event,
+	.reboot_notify = ca0132_reboot_notify,
 };
 
 static void ca0132_config(struct hda_codec *codec)
@@ -4635,9 +7353,14 @@ static void ca0132_config(struct hda_codec *codec)
 
 	spec->multiout.dac_nids = spec->dacs;
 	spec->multiout.num_dacs = 3;
-	spec->multiout.max_channels = 2;
 
-	if (spec->quirk == QUIRK_ALIENWARE) {
+	if (!spec->use_alt_functions)
+		spec->multiout.max_channels = 2;
+	else
+		spec->multiout.max_channels = 6;
+
+	switch (spec->quirk) {
+	case QUIRK_ALIENWARE:
 		codec_dbg(codec, "ca0132_config: QUIRK_ALIENWARE applied.\n");
 		snd_hda_apply_pincfgs(codec, alienware_pincfgs);
 
@@ -4657,7 +7380,71 @@ static void ca0132_config(struct hda_codec *codec)
 		spec->input_pins[2] = 0x13;
 		spec->shared_mic_nid = 0x7;
 		spec->unsol_tag_amic1 = 0x11;
-	} else {
+		break;
+	case QUIRK_SBZ:
+		codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__);
+		snd_hda_apply_pincfgs(codec, sbz_pincfgs);
+
+		spec->num_outputs = 2;
+		spec->out_pins[0] = 0x0B; /* Line out */
+		spec->out_pins[1] = 0x0F; /* Rear headphone out */
+		spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/
+		spec->out_pins[3] = 0x11; /* Rear surround */
+		spec->shared_out_nid = 0x2;
+		spec->unsol_tag_hp = spec->out_pins[1];
+		spec->unsol_tag_front_hp = spec->out_pins[2];
+
+		spec->adcs[0] = 0x7; /* Rear Mic / Line-in */
+		spec->adcs[1] = 0x8; /* Front Mic, but only if no DSP */
+		spec->adcs[2] = 0xa; /* what u hear */
+
+		spec->num_inputs = 2;
+		spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
+		spec->input_pins[1] = 0x13; /* What U Hear */
+		spec->shared_mic_nid = 0x7;
+		spec->unsol_tag_amic1 = spec->input_pins[0];
+
+		/* SPDIF I/O */
+		spec->dig_out = 0x05;
+		spec->multiout.dig_out_nid = spec->dig_out;
+		cfg->dig_out_pins[0] = 0x0c;
+		cfg->dig_outs = 1;
+		cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF;
+		spec->dig_in = 0x09;
+		cfg->dig_in_pin = 0x0e;
+		cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
+		break;
+	case QUIRK_R3DI:
+		codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__);
+		snd_hda_apply_pincfgs(codec, r3di_pincfgs);
+
+		spec->num_outputs = 2;
+		spec->out_pins[0] = 0x0B; /* Line out */
+		spec->out_pins[1] = 0x0F; /* Rear headphone out */
+		spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/
+		spec->out_pins[3] = 0x11; /* Rear surround */
+		spec->shared_out_nid = 0x2;
+		spec->unsol_tag_hp = spec->out_pins[1];
+		spec->unsol_tag_front_hp = spec->out_pins[2];
+
+		spec->adcs[0] = 0x07; /* Rear Mic / Line-in */
+		spec->adcs[1] = 0x08; /* Front Mic, but only if no DSP */
+		spec->adcs[2] = 0x0a; /* what u hear */
+
+		spec->num_inputs = 2;
+		spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
+		spec->input_pins[1] = 0x13; /* What U Hear */
+		spec->shared_mic_nid = 0x7;
+		spec->unsol_tag_amic1 = spec->input_pins[0];
+
+		/* SPDIF I/O */
+		spec->dig_out = 0x05;
+		spec->multiout.dig_out_nid = spec->dig_out;
+		cfg->dig_out_pins[0] = 0x0c;
+		cfg->dig_outs = 1;
+		cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF;
+		break;
+	default:
 		spec->num_outputs = 2;
 		spec->out_pins[0] = 0x0b; /* speaker out */
 		spec->out_pins[1] = 0x10; /* headphone out */
@@ -4684,6 +7471,7 @@ static void ca0132_config(struct hda_codec *codec)
 		spec->dig_in = 0x09;
 		cfg->dig_in_pin = 0x0e;
 		cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
+		break;
 	}
 }
 
@@ -4694,6 +7482,8 @@ static int ca0132_prepare_verbs(struct hda_codec *codec)
 	struct ca0132_spec *spec = codec->spec;
 
 	spec->chip_init_verbs = ca0132_init_verbs0;
+	if (spec->quirk == QUIRK_SBZ)
+		spec->sbz_init_verbs = sbz_init_verbs;
 	spec->spec_init_verbs = kzalloc(sizeof(struct hda_verb) * NUM_SPEC_VERBS, GFP_KERNEL);
 	if (!spec->spec_init_verbs)
 		return -ENOMEM;
@@ -4757,9 +7547,46 @@ static int patch_ca0132(struct hda_codec *codec)
 	else
 		spec->quirk = QUIRK_NONE;
 
+	/* Setup BAR Region 2 for Sound Blaster Z */
+	if (spec->quirk == QUIRK_SBZ) {
+		spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20);
+		if (spec->mem_base == NULL) {
+			codec_warn(codec, "pci_iomap failed!");
+			codec_info(codec, "perhaps this is not an SBZ?");
+			spec->quirk = QUIRK_NONE;
+		}
+	}
+
 	spec->dsp_state = DSP_DOWNLOAD_INIT;
 	spec->num_mixers = 1;
-	spec->mixers[0] = ca0132_mixer;
+
+	/* Set which mixers each quirk uses. */
+	switch (spec->quirk) {
+	case QUIRK_SBZ:
+		spec->mixers[0] = sbz_mixer;
+		snd_hda_codec_set_name(codec, "Sound Blaster Z");
+		break;
+	case QUIRK_R3DI:
+		spec->mixers[0] = r3di_mixer;
+		snd_hda_codec_set_name(codec, "Recon3Di");
+		break;
+	default:
+		spec->mixers[0] = ca0132_mixer;
+		break;
+	}
+
+	/* Setup whether or not to use alt functions/controls */
+	switch (spec->quirk) {
+	case QUIRK_SBZ:
+	case QUIRK_R3DI:
+		spec->use_alt_controls = true;
+		spec->use_alt_functions = true;
+		break;
+	default:
+		spec->use_alt_controls = false;
+		spec->use_alt_functions = false;
+		break;
+	}
 
 	spec->base_init_verbs = ca0132_base_init_verbs;
 	spec->base_exit_verbs = ca0132_base_exit_verbs;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 5b4dbce..093d2a9 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -588,6 +588,7 @@ static void cxt_fixup_olpc_xo(struct hda_codec *codec,
 				    const struct hda_fixup *fix, int action)
 {
 	struct conexant_spec *spec = codec->spec;
+	struct snd_kcontrol_new *kctl;
 	int i;
 
 	if (action != HDA_FIXUP_ACT_PROBE)
@@ -606,9 +607,7 @@ static void cxt_fixup_olpc_xo(struct hda_codec *codec,
 	snd_hda_codec_set_pin_target(codec, 0x1a, PIN_VREF50);
 
 	/* override mic boost control */
-	for (i = 0; i < spec->gen.kctls.used; i++) {
-		struct snd_kcontrol_new *kctl =
-			snd_array_elem(&spec->gen.kctls, i);
+	snd_array_for_each(&spec->gen.kctls, i, kctl) {
 		if (!strcmp(kctl->name, "Mic Boost Volume")) {
 			kctl->put = olpc_xo_mic_boost_put;
 			break;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 01a6643..3af9384 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2830,6 +2830,7 @@ static int find_ext_mic_pin(struct hda_codec *codec);
 
 static void alc286_shutup(struct hda_codec *codec)
 {
+	const struct hda_pincfg *pin;
 	int i;
 	int mic_pin = find_ext_mic_pin(codec);
 	/* don't shut up pins when unloading the driver; otherwise it breaks
@@ -2837,8 +2838,7 @@ static void alc286_shutup(struct hda_codec *codec)
 	 */
 	if (codec->bus->shutdown)
 		return;
-	for (i = 0; i < codec->init_pins.used; i++) {
-		struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+	snd_array_for_each(&codec->init_pins, i, pin) {
 		/* use read here for syncing after issuing each verb */
 		if (pin->nid != mic_pin)
 			snd_hda_codec_read(codec, pin->nid, 0,
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index abc7bd5..f0e7135 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -2542,7 +2542,7 @@ static int snd_dbri_create(struct snd_card *card,
 	dbri->irq = irq;
 
 	dbri->dma = dma_zalloc_coherent(&op->dev, sizeof(struct dbri_dma),
-					&dbri->dma_dvma, GFP_ATOMIC);
+					&dbri->dma_dvma, GFP_KERNEL);
 	if (!dbri->dma)
 		return -ENOMEM;
 
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 4a1c6bb..f6c3c1c 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -221,32 +221,13 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
 	struct usb_device *dev = chip->dev;
 	struct usb_host_interface *host_iface;
 	struct usb_interface_descriptor *altsd;
-	void *control_header;
 	int i, protocol;
-	int rest_bytes;
 
 	/* find audiocontrol interface */
 	host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0];
-	control_header = snd_usb_find_csint_desc(host_iface->extra,
-						 host_iface->extralen,
-						 NULL, UAC_HEADER);
 	altsd = get_iface_desc(host_iface);
 	protocol = altsd->bInterfaceProtocol;
 
-	if (!control_header) {
-		dev_err(&dev->dev, "cannot find UAC_HEADER\n");
-		return -EINVAL;
-	}
-
-	rest_bytes = (void *)(host_iface->extra + host_iface->extralen) -
-		control_header;
-
-	/* just to be sure -- this shouldn't hit at all */
-	if (rest_bytes <= 0) {
-		dev_err(&dev->dev, "invalid control header\n");
-		return -EINVAL;
-	}
-
 	switch (protocol) {
 	default:
 		dev_warn(&dev->dev,
@@ -255,7 +236,25 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
 		/* fall through */
 
 	case UAC_VERSION_1: {
-		struct uac1_ac_header_descriptor *h1 = control_header;
+		struct uac1_ac_header_descriptor *h1;
+		int rest_bytes;
+
+		h1 = snd_usb_find_csint_desc(host_iface->extra,
+							 host_iface->extralen,
+							 NULL, UAC_HEADER);
+		if (!h1) {
+			dev_err(&dev->dev, "cannot find UAC_HEADER\n");
+			return -EINVAL;
+		}
+
+		rest_bytes = (void *)(host_iface->extra +
+				host_iface->extralen) - (void *)h1;
+
+		/* just to be sure -- this shouldn't hit at all */
+		if (rest_bytes <= 0) {
+			dev_err(&dev->dev, "invalid control header\n");
+			return -EINVAL;
+		}
 
 		if (rest_bytes < sizeof(*h1)) {
 			dev_err(&dev->dev, "too short v1 buffer descriptor\n");
@@ -308,6 +307,20 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
 			return -EINVAL;
 		}
 
+		if (protocol == UAC_VERSION_3) {
+			int badd = assoc->bFunctionSubClass;
+
+			if (badd != UAC3_FUNCTION_SUBCLASS_FULL_ADC_3_0 &&
+			    (badd < UAC3_FUNCTION_SUBCLASS_GENERIC_IO ||
+			     badd > UAC3_FUNCTION_SUBCLASS_SPEAKERPHONE)) {
+				dev_err(&dev->dev,
+					"Unsupported UAC3 BADD profile\n");
+				return -EINVAL;
+			}
+
+			chip->badd_profile = badd;
+		}
+
 		for (i = 0; i < assoc->bInterfaceCount; i++) {
 			int intf = assoc->bFirstInterface + i;
 
@@ -349,6 +362,90 @@ static int snd_usb_audio_dev_free(struct snd_device *device)
 	return snd_usb_audio_free(chip);
 }
 
+static void usb_audio_make_shortname(struct usb_device *dev,
+				     struct snd_usb_audio *chip,
+				     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));
+		return;
+	}
+
+	/* retrieve the device string as shortname */
+	if (!dev->descriptor.iProduct ||
+	    usb_string(dev, dev->descriptor.iProduct,
+		       card->shortname, sizeof(card->shortname)) <= 0) {
+		/* no name available from anywhere, so use ID */
+		sprintf(card->shortname, "USB Device %#04x:%#04x",
+			USB_ID_VENDOR(chip->usb_id),
+			USB_ID_PRODUCT(chip->usb_id));
+	}
+
+	strim(card->shortname);
+}
+
+static void usb_audio_make_longname(struct usb_device *dev,
+				    struct snd_usb_audio *chip,
+				    const struct snd_usb_audio_quirk *quirk)
+{
+	struct snd_card *card = chip->card;
+	int len;
+
+	/* 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));
+		return;
+	}
+
+	if (quirk && quirk->vendor_name && *quirk->vendor_name) {
+		len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname));
+	} else {
+		/* retrieve the vendor and device strings as longname */
+		if (dev->descriptor.iManufacturer)
+			len = usb_string(dev, dev->descriptor.iManufacturer,
+					 card->longname, sizeof(card->longname));
+		else
+			len = 0;
+		/* we don't really care if there isn't any vendor string */
+	}
+	if (len > 0) {
+		strim(card->longname);
+		if (*card->longname)
+			strlcat(card->longname, " ", sizeof(card->longname));
+	}
+
+	strlcat(card->longname, card->shortname, sizeof(card->longname));
+
+	len = strlcat(card->longname, " at ", sizeof(card->longname));
+
+	if (len < sizeof(card->longname))
+		usb_make_path(dev, card->longname + len, sizeof(card->longname) - len);
+
+	switch (snd_usb_get_speed(dev)) {
+	case USB_SPEED_LOW:
+		strlcat(card->longname, ", low speed", sizeof(card->longname));
+		break;
+	case USB_SPEED_FULL:
+		strlcat(card->longname, ", full speed", sizeof(card->longname));
+		break;
+	case USB_SPEED_HIGH:
+		strlcat(card->longname, ", high speed", sizeof(card->longname));
+		break;
+	case USB_SPEED_SUPER:
+		strlcat(card->longname, ", super speed", sizeof(card->longname));
+		break;
+	case USB_SPEED_SUPER_PLUS:
+		strlcat(card->longname, ", super speed plus", sizeof(card->longname));
+		break;
+	default:
+		break;
+	}
+}
+
 /*
  * create a chip instance and set its names.
  */
@@ -360,7 +457,7 @@ static int snd_usb_audio_create(struct usb_interface *intf,
 {
 	struct snd_card *card;
 	struct snd_usb_audio *chip;
-	int err, len;
+	int err;
 	char component[14];
 	static struct snd_device_ops ops = {
 		.dev_free =	snd_usb_audio_dev_free,
@@ -422,64 +519,8 @@ static int snd_usb_audio_create(struct usb_interface *intf,
 		USB_ID_VENDOR(chip->usb_id), USB_ID_PRODUCT(chip->usb_id));
 	snd_component_add(card, component);
 
-	/* retrieve the device string as shortname */
-	if (quirk && quirk->product_name && *quirk->product_name) {
-		strlcpy(card->shortname, quirk->product_name, sizeof(card->shortname));
-	} else {
-		if (!dev->descriptor.iProduct ||
-		    usb_string(dev, dev->descriptor.iProduct,
-		    card->shortname, sizeof(card->shortname)) <= 0) {
-			/* no name available from anywhere, so use ID */
-			sprintf(card->shortname, "USB Device %#04x:%#04x",
-				USB_ID_VENDOR(chip->usb_id),
-				USB_ID_PRODUCT(chip->usb_id));
-		}
-	}
-	strim(card->shortname);
-
-	/* retrieve the vendor and device strings as longname */
-	if (quirk && quirk->vendor_name && *quirk->vendor_name) {
-		len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname));
-	} else {
-		if (dev->descriptor.iManufacturer)
-			len = usb_string(dev, dev->descriptor.iManufacturer,
-					 card->longname, sizeof(card->longname));
-		else
-			len = 0;
-		/* we don't really care if there isn't any vendor string */
-	}
-	if (len > 0) {
-		strim(card->longname);
-		if (*card->longname)
-			strlcat(card->longname, " ", sizeof(card->longname));
-	}
-
-	strlcat(card->longname, card->shortname, sizeof(card->longname));
-
-	len = strlcat(card->longname, " at ", sizeof(card->longname));
-
-	if (len < sizeof(card->longname))
-		usb_make_path(dev, card->longname + len, sizeof(card->longname) - len);
-
-	switch (snd_usb_get_speed(dev)) {
-	case USB_SPEED_LOW:
-		strlcat(card->longname, ", low speed", sizeof(card->longname));
-		break;
-	case USB_SPEED_FULL:
-		strlcat(card->longname, ", full speed", sizeof(card->longname));
-		break;
-	case USB_SPEED_HIGH:
-		strlcat(card->longname, ", high speed", sizeof(card->longname));
-		break;
-	case USB_SPEED_SUPER:
-		strlcat(card->longname, ", super speed", sizeof(card->longname));
-		break;
-	case USB_SPEED_SUPER_PLUS:
-		strlcat(card->longname, ", super speed plus", sizeof(card->longname));
-		break;
-	default:
-		break;
-	}
+	usb_audio_make_shortname(dev, chip, quirk);
+	usb_audio_make_longname(dev, chip, quirk);
 
 	snd_usb_audio_create_proc(chip);
 
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 0b030d8..17673f3 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -587,8 +587,15 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
 	default:
 		return set_sample_rate_v1(chip, iface, alts, fmt, rate);
 
-	case UAC_VERSION_2:
 	case UAC_VERSION_3:
+		if (chip->badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) {
+			if (rate != UAC3_BADD_SAMPLING_RATE)
+				return -ENXIO;
+			else
+				return 0;
+		}
+	/* fall through */
+	case UAC_VERSION_2:
 		return set_sample_rate_v2v3(chip, iface, alts, fmt, rate);
 	}
 }
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index bb5ab7a..173979e 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -112,14 +112,12 @@ enum {
 #include "mixer_maps.c"
 
 static const struct usbmix_name_map *
-find_map(struct mixer_build *state, int unitid, int control)
+find_map(const struct usbmix_name_map *p, int unitid, int control)
 {
-	const struct usbmix_name_map *p = state->map;
-
 	if (!p)
 		return NULL;
 
-	for (p = state->map; p->id; p++) {
+	for (; p->id; p++) {
 		if (p->id == unitid &&
 		    (!control || !p->control || control == p->control))
 			return p;
@@ -201,10 +199,10 @@ static void *find_audio_control_unit(struct mixer_build *state,
 /*
  * copy a string with the given id
  */
-static int snd_usb_copy_string_desc(struct mixer_build *state,
+static int snd_usb_copy_string_desc(struct snd_usb_audio *chip,
 				    int index, char *buf, int maxlen)
 {
-	int len = usb_string(state->chip->dev, index, buf, maxlen - 1);
+	int len = usb_string(chip->dev, index, buf, maxlen - 1);
 
 	if (len < 0)
 		return 0;
@@ -658,14 +656,14 @@ static struct iterm_name_combo {
 	{ 0 },
 };
 
-static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm,
+static int get_term_name(struct snd_usb_audio *chip, struct usb_audio_term *iterm,
 			 unsigned char *name, int maxlen, int term_only)
 {
 	struct iterm_name_combo *names;
 	int len;
 
 	if (iterm->name) {
-		len = snd_usb_copy_string_desc(state, iterm->name,
+		len = snd_usb_copy_string_desc(chip, iterm->name,
 						name, maxlen);
 		if (len)
 			return len;
@@ -1341,16 +1339,16 @@ static struct usb_feature_control_info *get_feature_control_info(int control)
 	return NULL;
 }
 
-static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
-			      unsigned int ctl_mask, int control,
-			      struct usb_audio_term *iterm, int unitid,
-			      int readonly_mask)
+static void __build_feature_ctl(struct usb_mixer_interface *mixer,
+				const struct usbmix_name_map *imap,
+				unsigned int ctl_mask, int control,
+				struct usb_audio_term *iterm,
+				struct usb_audio_term *oterm,
+				int unitid, int nameid, int readonly_mask)
 {
-	struct uac_feature_unit_descriptor *desc = raw_desc;
 	struct usb_feature_control_info *ctl_info;
 	unsigned int len = 0;
 	int mapped_name = 0;
-	int nameid = uac_feature_unit_iFeature(desc);
 	struct snd_kcontrol *kctl;
 	struct usb_mixer_elem_info *cval;
 	const struct usbmix_name_map *map;
@@ -1361,14 +1359,14 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		return;
 	}
 
-	map = find_map(state, unitid, control);
+	map = find_map(imap, unitid, control);
 	if (check_ignored_ctl(map))
 		return;
 
 	cval = kzalloc(sizeof(*cval), GFP_KERNEL);
 	if (!cval)
 		return;
-	snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid);
+	snd_usb_mixer_elem_init_std(&cval->head, mixer, unitid);
 	cval->control = control;
 	cval->cmask = ctl_mask;
 
@@ -1377,7 +1375,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		kfree(cval);
 		return;
 	}
-	if (state->mixer->protocol == UAC_VERSION_1)
+	if (mixer->protocol == UAC_VERSION_1)
 		cval->val_type = ctl_info->type;
 	else /* UAC_VERSION_2 */
 		cval->val_type = ctl_info->type_uac2 >= 0 ?
@@ -1406,7 +1404,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval);
 
 	if (!kctl) {
-		usb_audio_err(state->chip, "cannot malloc kcontrol\n");
+		usb_audio_err(mixer->chip, "cannot malloc kcontrol\n");
 		kfree(cval);
 		return;
 	}
@@ -1415,7 +1413,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 	len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name));
 	mapped_name = len != 0;
 	if (!len && nameid)
-		len = snd_usb_copy_string_desc(state, nameid,
+		len = snd_usb_copy_string_desc(mixer->chip, nameid,
 				kctl->id.name, sizeof(kctl->id.name));
 
 	switch (control) {
@@ -1430,10 +1428,12 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		 * - otherwise, anonymous name.
 		 */
 		if (!len) {
-			len = get_term_name(state, iterm, kctl->id.name,
-					    sizeof(kctl->id.name), 1);
-			if (!len)
-				len = get_term_name(state, &state->oterm,
+			if (iterm)
+				len = get_term_name(mixer->chip, iterm,
+						    kctl->id.name,
+						    sizeof(kctl->id.name), 1);
+			if (!len && oterm)
+				len = get_term_name(mixer->chip, oterm,
 						    kctl->id.name,
 						    sizeof(kctl->id.name), 1);
 			if (!len)
@@ -1442,15 +1442,15 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		}
 
 		if (!mapped_name)
-			check_no_speaker_on_headset(kctl, state->mixer->chip->card);
+			check_no_speaker_on_headset(kctl, mixer->chip->card);
 
 		/*
 		 * determine the stream direction:
 		 * if the connected output is USB stream, then it's likely a
 		 * capture stream.  otherwise it should be playback (hopefully :)
 		 */
-		if (!mapped_name && !(state->oterm.type >> 16)) {
-			if ((state->oterm.type & 0xff00) == 0x0100)
+		if (!mapped_name && oterm && !(oterm->type >> 16)) {
+			if ((oterm->type & 0xff00) == 0x0100)
 				append_ctl_name(kctl, " Capture");
 			else
 				append_ctl_name(kctl, " Playback");
@@ -1478,7 +1478,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		}
 	}
 
-	snd_usb_mixer_fu_apply_quirk(state->mixer, cval, unitid, kctl);
+	snd_usb_mixer_fu_apply_quirk(mixer, cval, unitid, kctl);
 
 	range = (cval->max - cval->min) / cval->res;
 	/*
@@ -1487,26 +1487,46 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 	 * devices. It will definitively catch all buggy Logitech devices.
 	 */
 	if (range > 384) {
-		usb_audio_warn(state->chip,
+		usb_audio_warn(mixer->chip,
 			       "Warning! Unlikely big volume range (=%u), cval->res is probably wrong.",
 			       range);
-		usb_audio_warn(state->chip,
+		usb_audio_warn(mixer->chip,
 			       "[%d] FU [%s] ch = %d, val = %d/%d/%d",
 			       cval->head.id, kctl->id.name, cval->channels,
 			       cval->min, cval->max, cval->res);
 	}
 
-	usb_audio_dbg(state->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",
+	usb_audio_dbg(mixer->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",
 		      cval->head.id, kctl->id.name, cval->channels,
 		      cval->min, cval->max, cval->res);
 	snd_usb_mixer_add_control(&cval->head, kctl);
 }
 
+static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
+			      unsigned int ctl_mask, int control,
+			      struct usb_audio_term *iterm, int unitid,
+			      int readonly_mask)
+{
+	struct uac_feature_unit_descriptor *desc = raw_desc;
+	int nameid = uac_feature_unit_iFeature(desc);
+
+	__build_feature_ctl(state->mixer, state->map, ctl_mask, control,
+			iterm, &state->oterm, unitid, nameid, readonly_mask);
+}
+
+static void build_feature_ctl_badd(struct usb_mixer_interface *mixer,
+			      unsigned int ctl_mask, int control, int unitid,
+			      const struct usbmix_name_map *badd_map)
+{
+	__build_feature_ctl(mixer, badd_map, ctl_mask, control,
+			NULL, NULL, unitid, 0, 0);
+}
+
 static void get_connector_control_name(struct mixer_build *state,
 				       struct usb_audio_term *term,
 				       bool is_input, char *name, int name_size)
 {
-	int name_len = get_term_name(state, term, name, name_size, 0);
+	int name_len = get_term_name(state->chip, term, name, name_size, 0);
 
 	if (name_len == 0)
 		strlcpy(name, "Unknown", name_size);
@@ -1605,7 +1625,7 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid,
 	}
 
 	kctl->private_free = snd_usb_mixer_elem_free;
-	ret = snd_usb_copy_string_desc(state, hdr->iClockSource,
+	ret = snd_usb_copy_string_desc(state->chip, hdr->iClockSource,
 				       name, sizeof(name));
 	if (ret > 0)
 		snprintf(kctl->id.name, sizeof(kctl->id.name),
@@ -1815,7 +1835,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state,
 	struct snd_kcontrol *kctl;
 	const struct usbmix_name_map *map;
 
-	map = find_map(state, unitid, 0);
+	map = find_map(state->map, unitid, 0);
 	if (check_ignored_ctl(map))
 		return;
 
@@ -1848,7 +1868,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state,
 
 	len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name));
 	if (!len)
-		len = get_term_name(state, iterm, kctl->id.name,
+		len = get_term_name(state->chip, iterm, kctl->id.name,
 				    sizeof(kctl->id.name), 0);
 	if (!len)
 		len = sprintf(kctl->id.name, "Mixer Source %d", in_ch + 1);
@@ -2114,7 +2134,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 
 		if (!(controls[valinfo->control / 8] & (1 << ((valinfo->control % 8) - 1))))
 			continue;
-		map = find_map(state, unitid, valinfo->control);
+		map = find_map(state->map, unitid, valinfo->control);
 		if (check_ignored_ctl(map))
 			continue;
 		cval = kzalloc(sizeof(*cval), GFP_KERNEL);
@@ -2162,7 +2182,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 			nameid = uac_processing_unit_iProcessing(desc, state->mixer->protocol);
 			len = 0;
 			if (nameid)
-				len = snd_usb_copy_string_desc(state, nameid,
+				len = snd_usb_copy_string_desc(state->chip,
+							       nameid,
 							       kctl->id.name,
 							       sizeof(kctl->id.name));
 			if (!len)
@@ -2317,7 +2338,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 	if (desc->bNrInPins == 1) /* only one ? nonsense! */
 		return 0;
 
-	map = find_map(state, unitid, 0);
+	map = find_map(state->map, unitid, 0);
 	if (check_ignored_ctl(map))
 		return 0;
 
@@ -2358,7 +2379,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 		len = check_mapped_selector_name(state, unitid, i, namelist[i],
 						 MAX_ITEM_NAME_LEN);
 		if (! len && check_input_term(state, desc->baSourceID[i], &iterm) >= 0)
-			len = get_term_name(state, &iterm, namelist[i], MAX_ITEM_NAME_LEN, 0);
+			len = get_term_name(state->chip, &iterm, namelist[i],
+					    MAX_ITEM_NAME_LEN, 0);
 		if (! len)
 			sprintf(namelist[i], "Input %u", i);
 	}
@@ -2380,12 +2402,12 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 		/* if iSelector is given, use it */
 		nameid = uac_selector_unit_iSelector(desc);
 		if (nameid)
-			len = snd_usb_copy_string_desc(state, nameid,
+			len = snd_usb_copy_string_desc(state->chip, nameid,
 						       kctl->id.name,
 						       sizeof(kctl->id.name));
 		/* ... or pick up the terminal name at next */
 		if (!len)
-			len = get_term_name(state, &state->oterm,
+			len = get_term_name(state->chip, &state->oterm,
 				    kctl->id.name, sizeof(kctl->id.name), 0);
 		/* ... or use the fixed string "USB" as the last resort */
 		if (!len)
@@ -2503,6 +2525,246 @@ static int snd_usb_mixer_dev_free(struct snd_device *device)
 	return 0;
 }
 
+/* UAC3 predefined channels configuration */
+struct uac3_badd_profile {
+	int subclass;
+	const char *name;
+	int c_chmask;	/* capture channels mask */
+	int p_chmask;	/* playback channels mask */
+	int st_chmask;	/* side tone mixing channel mask */
+};
+
+static struct uac3_badd_profile uac3_badd_profiles[] = {
+	{
+		/*
+		 * BAIF, BAOF or combination of both
+		 * IN: Mono or Stereo cfg, Mono alt possible
+		 * OUT: Mono or Stereo cfg, Mono alt possible
+		 */
+		.subclass = UAC3_FUNCTION_SUBCLASS_GENERIC_IO,
+		.name = "GENERIC IO",
+		.c_chmask = -1,		/* dynamic channels */
+		.p_chmask = -1,		/* dynamic channels */
+	},
+	{
+		/* BAOF; Stereo only cfg, Mono alt possible */
+		.subclass = UAC3_FUNCTION_SUBCLASS_HEADPHONE,
+		.name = "HEADPHONE",
+		.p_chmask = 3,
+	},
+	{
+		/* BAOF; Mono or Stereo cfg, Mono alt possible */
+		.subclass = UAC3_FUNCTION_SUBCLASS_SPEAKER,
+		.name = "SPEAKER",
+		.p_chmask = -1,		/* dynamic channels */
+	},
+	{
+		/* BAIF; Mono or Stereo cfg, Mono alt possible */
+		.subclass = UAC3_FUNCTION_SUBCLASS_MICROPHONE,
+		.name = "MICROPHONE",
+		.c_chmask = -1,		/* dynamic channels */
+	},
+	{
+		/*
+		 * BAIOF topology
+		 * IN: Mono only
+		 * OUT: Mono or Stereo cfg, Mono alt possible
+		 */
+		.subclass = UAC3_FUNCTION_SUBCLASS_HEADSET,
+		.name = "HEADSET",
+		.c_chmask = 1,
+		.p_chmask = -1,		/* dynamic channels */
+		.st_chmask = 1,
+	},
+	{
+		/* BAIOF; IN: Mono only; OUT: Stereo only, Mono alt possible */
+		.subclass = UAC3_FUNCTION_SUBCLASS_HEADSET_ADAPTER,
+		.name = "HEADSET ADAPTER",
+		.c_chmask = 1,
+		.p_chmask = 3,
+		.st_chmask = 1,
+	},
+	{
+		/* BAIF + BAOF; IN: Mono only; OUT: Mono only */
+		.subclass = UAC3_FUNCTION_SUBCLASS_SPEAKERPHONE,
+		.name = "SPEAKERPHONE",
+		.c_chmask = 1,
+		.p_chmask = 1,
+	},
+	{ 0 } /* terminator */
+};
+
+static bool uac3_badd_func_has_valid_channels(struct usb_mixer_interface *mixer,
+					      struct uac3_badd_profile *f,
+					      int c_chmask, int p_chmask)
+{
+	/*
+	 * If both playback/capture channels are dynamic, make sure
+	 * at least one channel is present
+	 */
+	if (f->c_chmask < 0 && f->p_chmask < 0) {
+		if (!c_chmask && !p_chmask) {
+			usb_audio_warn(mixer->chip, "BAAD %s: no channels?",
+				       f->name);
+			return false;
+		}
+		return true;
+	}
+
+	if ((f->c_chmask < 0 && !c_chmask) ||
+	    (f->c_chmask >= 0 && f->c_chmask != c_chmask)) {
+		usb_audio_warn(mixer->chip, "BAAD %s c_chmask mismatch",
+			       f->name);
+		return false;
+	}
+	if ((f->p_chmask < 0 && !p_chmask) ||
+	    (f->p_chmask >= 0 && f->p_chmask != p_chmask)) {
+		usb_audio_warn(mixer->chip, "BAAD %s p_chmask mismatch",
+			       f->name);
+		return false;
+	}
+	return true;
+}
+
+/*
+ * create mixer controls for UAC3 BADD profiles
+ *
+ * UAC3 BADD device doesn't contain CS descriptors thus we will guess everything
+ *
+ * BADD device may contain Mixer Unit, which doesn't have any controls, skip it
+ */
+static int snd_usb_mixer_controls_badd(struct usb_mixer_interface *mixer,
+				       int ctrlif)
+{
+	struct usb_device *dev = mixer->chip->dev;
+	struct usb_interface_assoc_descriptor *assoc;
+	int badd_profile = mixer->chip->badd_profile;
+	struct uac3_badd_profile *f;
+	const struct usbmix_ctl_map *map;
+	int p_chmask = 0, c_chmask = 0, st_chmask = 0;
+	int i;
+
+	assoc = usb_ifnum_to_if(dev, ctrlif)->intf_assoc;
+
+	/* Detect BADD capture/playback channels from AS EP descriptors */
+	for (i = 0; i < assoc->bInterfaceCount; i++) {
+		int intf = assoc->bFirstInterface + i;
+
+		struct usb_interface *iface;
+		struct usb_host_interface *alts;
+		struct usb_interface_descriptor *altsd;
+		unsigned int maxpacksize;
+		char dir_in;
+		int chmask, num;
+
+		if (intf == ctrlif)
+			continue;
+
+		iface = usb_ifnum_to_if(dev, intf);
+		num = iface->num_altsetting;
+
+		if (num < 2)
+			return -EINVAL;
+
+		/*
+		 * The number of Channels in an AudioStreaming interface
+		 * and the audio sample bit resolution (16 bits or 24
+		 * bits) can be derived from the wMaxPacketSize field in
+		 * the Standard AS Audio Data Endpoint descriptor in
+		 * Alternate Setting 1
+		 */
+		alts = &iface->altsetting[1];
+		altsd = get_iface_desc(alts);
+
+		if (altsd->bNumEndpoints < 1)
+			return -EINVAL;
+
+		/* check direction */
+		dir_in = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN);
+		maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+
+		switch (maxpacksize) {
+		default:
+			usb_audio_err(mixer->chip,
+				"incorrect wMaxPacketSize 0x%x for BADD profile\n",
+				maxpacksize);
+			return -EINVAL;
+		case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_16:
+		case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_16:
+		case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_24:
+		case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_24:
+			chmask = 1;
+			break;
+		case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_16:
+		case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_16:
+		case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_24:
+		case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_24:
+			chmask = 3;
+			break;
+		}
+
+		if (dir_in)
+			c_chmask = chmask;
+		else
+			p_chmask = chmask;
+	}
+
+	usb_audio_dbg(mixer->chip,
+		"UAC3 BADD profile 0x%x: detected c_chmask=%d p_chmask=%d\n",
+		badd_profile, c_chmask, p_chmask);
+
+	/* check the mapping table */
+	for (map = uac3_badd_usbmix_ctl_maps; map->id; map++) {
+		if (map->id == badd_profile)
+			break;
+	}
+
+	if (!map->id)
+		return -EINVAL;
+
+	for (f = uac3_badd_profiles; f->name; f++) {
+		if (badd_profile == f->subclass)
+			break;
+	}
+	if (!f->name)
+		return -EINVAL;
+	if (!uac3_badd_func_has_valid_channels(mixer, f, c_chmask, p_chmask))
+		return -EINVAL;
+	st_chmask = f->st_chmask;
+
+	/* Playback */
+	if (p_chmask) {
+		/* Master channel, always writable */
+		build_feature_ctl_badd(mixer, 0, UAC_FU_MUTE,
+				       UAC3_BADD_FU_ID2, map->map);
+		/* Mono/Stereo volume channels, always writable */
+		build_feature_ctl_badd(mixer, p_chmask, UAC_FU_VOLUME,
+				       UAC3_BADD_FU_ID2, map->map);
+	}
+
+	/* Capture */
+	if (c_chmask) {
+		/* Master channel, always writable */
+		build_feature_ctl_badd(mixer, 0, UAC_FU_MUTE,
+				       UAC3_BADD_FU_ID5, map->map);
+		/* Mono/Stereo volume channels, always writable */
+		build_feature_ctl_badd(mixer, c_chmask, UAC_FU_VOLUME,
+				       UAC3_BADD_FU_ID5, map->map);
+	}
+
+	/* Side tone-mixing */
+	if (st_chmask) {
+		/* Master channel, always writable */
+		build_feature_ctl_badd(mixer, 0, UAC_FU_MUTE,
+				       UAC3_BADD_FU_ID7, map->map);
+		/* Mono volume channel, always writable */
+		build_feature_ctl_badd(mixer, 1, UAC_FU_VOLUME,
+				       UAC3_BADD_FU_ID7, map->map);
+	}
+
+	return 0;
+}
+
 /*
  * create mixer controls
  *
@@ -2606,9 +2868,9 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid)
 {
 	struct usb_mixer_elem_list *list;
 
-	for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) {
+	for_each_mixer_elem(list, mixer, unitid) {
 		struct usb_mixer_elem_info *info =
-			(struct usb_mixer_elem_info *)list;
+			mixer_elem_list_to_info(list);
 		/* invalidate cache, so the value is read from the device */
 		info->cached = 0;
 		snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
@@ -2619,7 +2881,7 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid)
 static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer,
 				    struct usb_mixer_elem_list *list)
 {
-	struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list;
+	struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
 	static char *val_types[] = {"BOOLEAN", "INV_BOOLEAN",
 				    "S8", "U8", "S16", "U16"};
 	snd_iprintf(buffer, "    Info: id=%i, control=%i, cmask=0x%x, "
@@ -2645,8 +2907,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry,
 				mixer->ignore_ctl_error);
 		snd_iprintf(buffer, "Card: %s\n", chip->card->longname);
 		for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) {
-			for (list = mixer->id_elems[unitid]; list;
-			     list = list->next_id_elem) {
+			for_each_mixer_elem(list, mixer, unitid) {
 				snd_iprintf(buffer, "  Unit: %i\n", list->id);
 				if (list->kctl)
 					snd_iprintf(buffer,
@@ -2676,19 +2937,19 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer,
 		return;
 	}
 
-	for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem)
+	for_each_mixer_elem(list, mixer, unitid)
 		count++;
 
 	if (count == 0)
 		return;
 
-	for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) {
+	for_each_mixer_elem(list, mixer, unitid) {
 		struct usb_mixer_elem_info *info;
 
 		if (!list->kctl)
 			continue;
 
-		info = (struct usb_mixer_elem_info *)list;
+		info = mixer_elem_list_to_info(list);
 		if (count > 1 && info->control != control)
 			continue;
 
@@ -2809,6 +3070,48 @@ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
 	return 0;
 }
 
+static int keep_iface_ctl_get(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.integer.value[0] = mixer->chip->keep_iface;
+	return 0;
+}
+
+static int keep_iface_ctl_put(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol);
+	bool keep_iface = !!ucontrol->value.integer.value[0];
+
+	if (mixer->chip->keep_iface == keep_iface)
+		return 0;
+	mixer->chip->keep_iface = keep_iface;
+	return 1;
+}
+
+static const struct snd_kcontrol_new keep_iface_ctl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_CARD,
+	.name = "Keep Interface",
+	.info = snd_ctl_boolean_mono_info,
+	.get = keep_iface_ctl_get,
+	.put = keep_iface_ctl_put,
+};
+
+static int create_keep_iface_ctl(struct usb_mixer_interface *mixer)
+{
+	struct snd_kcontrol *kctl = snd_ctl_new1(&keep_iface_ctl, mixer);
+
+	/* need only one control per card */
+	if (snd_ctl_find_id(mixer->chip->card, &kctl->id)) {
+		snd_ctl_free_one(kctl);
+		return 0;
+	}
+
+	return snd_ctl_add(mixer->chip->card, kctl);
+}
+
 int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
 			 int ignore_error)
 {
@@ -2847,8 +3150,16 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
 		break;
 	}
 
-	if ((err = snd_usb_mixer_controls(mixer)) < 0 ||
-	    (err = snd_usb_mixer_status_create(mixer)) < 0)
+	if (mixer->protocol == UAC_VERSION_3 &&
+			chip->badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) {
+		if ((err = snd_usb_mixer_controls_badd(mixer, ctrlif)) < 0)
+			goto _error;
+	} else if ((err = snd_usb_mixer_controls(mixer)) < 0 ||
+			(err = snd_usb_mixer_status_create(mixer)) < 0) {
+		goto _error;
+	}
+	err = create_keep_iface_ctl(mixer);
+	if (err < 0)
 		goto _error;
 
 	snd_usb_mixer_apply_create_quirk(mixer);
@@ -2909,7 +3220,7 @@ int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer)
 
 static int restore_mixer_value(struct usb_mixer_elem_list *list)
 {
-	struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list;
+	struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
 	int c, err, idx;
 
 	if (cval->cmask) {
@@ -2945,8 +3256,7 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume)
 	if (reset_resume) {
 		/* restore cached mixer values */
 		for (id = 0; id < MAX_ID_ELEMS; id++) {
-			for (list = mixer->id_elems[id]; list;
-			     list = list->next_id_elem) {
+			for_each_mixer_elem(list, mixer, id) {
 				if (list->resume) {
 					err = list->resume(list);
 					if (err < 0)
@@ -2956,6 +3266,8 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume)
 		}
 	}
 
+	snd_usb_mixer_resume_quirk(mixer);
+
 	return snd_usb_mixer_activate(mixer);
 }
 #endif
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index ba27f7a..e026534 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -53,6 +53,12 @@ struct usb_mixer_elem_list {
 	usb_mixer_elem_resume_func_t resume;
 };
 
+/* iterate over mixer element list of the given unit id */
+#define for_each_mixer_elem(list, mixer, id)	\
+	for ((list) = (mixer)->id_elems[id]; (list); (list) = (list)->next_id_elem)
+#define mixer_elem_list_to_info(list) \
+	container_of(list, struct usb_mixer_elem_info, head)
+
 struct usb_mixer_elem_info {
 	struct usb_mixer_elem_list head;
 	unsigned int control;	/* CS or ICN (high byte) */
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index eaa03ac..71069e1 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -485,3 +485,68 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
 	{ 0 } /* terminator */
 };
 
+/*
+ * Control map entries for UAC3 BADD profiles
+ */
+
+static struct usbmix_name_map uac3_badd_generic_io_map[] = {
+	{ UAC3_BADD_FU_ID2, "Generic Out Playback" },
+	{ UAC3_BADD_FU_ID5, "Generic In Capture" },
+	{ 0 }					/* terminator */
+};
+static struct usbmix_name_map uac3_badd_headphone_map[] = {
+	{ UAC3_BADD_FU_ID2, "Headphone Playback" },
+	{ 0 }					/* terminator */
+};
+static struct usbmix_name_map uac3_badd_speaker_map[] = {
+	{ UAC3_BADD_FU_ID2, "Speaker Playback" },
+	{ 0 }					/* terminator */
+};
+static struct usbmix_name_map uac3_badd_microphone_map[] = {
+	{ UAC3_BADD_FU_ID5, "Mic Capture" },
+	{ 0 }					/* terminator */
+};
+/* Covers also 'headset adapter' profile */
+static struct usbmix_name_map uac3_badd_headset_map[] = {
+	{ UAC3_BADD_FU_ID2, "Headset Playback" },
+	{ UAC3_BADD_FU_ID5, "Headset Capture" },
+	{ UAC3_BADD_FU_ID7, "Sidetone Mixing" },
+	{ 0 }					/* terminator */
+};
+static struct usbmix_name_map uac3_badd_speakerphone_map[] = {
+	{ UAC3_BADD_FU_ID2, "Speaker Playback" },
+	{ UAC3_BADD_FU_ID5, "Mic Capture" },
+	{ 0 }					/* terminator */
+};
+
+static struct usbmix_ctl_map uac3_badd_usbmix_ctl_maps[] = {
+	{
+		.id = UAC3_FUNCTION_SUBCLASS_GENERIC_IO,
+		.map = uac3_badd_generic_io_map,
+	},
+	{
+		.id = UAC3_FUNCTION_SUBCLASS_HEADPHONE,
+		.map = uac3_badd_headphone_map,
+	},
+	{
+		.id = UAC3_FUNCTION_SUBCLASS_SPEAKER,
+		.map = uac3_badd_speaker_map,
+	},
+	{
+		.id = UAC3_FUNCTION_SUBCLASS_MICROPHONE,
+		.map = uac3_badd_microphone_map,
+	},
+	{
+		.id = UAC3_FUNCTION_SUBCLASS_HEADSET,
+		.map = uac3_badd_headset_map,
+	},
+	{
+		.id = UAC3_FUNCTION_SUBCLASS_HEADSET_ADAPTER,
+		.map = uac3_badd_headset_map,
+	},
+	{
+		.id = UAC3_FUNCTION_SUBCLASS_SPEAKERPHONE,
+		.map = uac3_badd_speakerphone_map,
+	},
+	{ 0 } /* terminator */
+};
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 56537a1..1b94387 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1172,7 +1172,7 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
 	int unitid = 12; /* SamleRate ExtensionUnit ID */
 
 	list_for_each_entry(mixer, &chip->mixer_list, list) {
-		cval = (struct usb_mixer_elem_info *)mixer->id_elems[unitid];
+		cval = mixer_elem_list_to_info(mixer->id_elems[unitid]);
 		if (cval) {
 			snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR,
 						    cval->control << 8,
@@ -1799,6 +1799,26 @@ static int snd_soundblaster_e1_switch_create(struct usb_mixer_interface *mixer)
 					  NULL);
 }
 
+static void dell_dock_init_vol(struct snd_usb_audio *chip, int ch, int id)
+{
+	u16 buf = 0;
+
+	snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR,
+			USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
+			ch, snd_usb_ctrl_intf(chip) | (id << 8),
+			&buf, 2);
+}
+
+static int dell_dock_mixer_init(struct usb_mixer_interface *mixer)
+{
+	/* fix to 0dB playback volumes */
+	dell_dock_init_vol(mixer->chip, 1, 16);
+	dell_dock_init_vol(mixer->chip, 2, 16);
+	dell_dock_init_vol(mixer->chip, 1, 19);
+	dell_dock_init_vol(mixer->chip, 2, 19);
+	return 0;
+}
+
 int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 {
 	int err = 0;
@@ -1884,11 +1904,25 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 	case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */
 		err = snd_soundblaster_e1_switch_create(mixer);
 		break;
+	case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */
+		err = dell_dock_mixer_init(mixer);
+		break;
 	}
 
 	return err;
 }
 
+#ifdef CONFIG_PM
+void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer)
+{
+	switch (mixer->chip->usb_id) {
+	case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */
+		dell_dock_mixer_init(mixer);
+		break;
+	}
+}
+#endif
+
 void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer,
 				    int unitid)
 {
diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h
index b5abd32..52be26d 100644
--- a/sound/usb/mixer_quirks.h
+++ b/sound/usb/mixer_quirks.h
@@ -14,5 +14,9 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer,
 				  struct usb_mixer_elem_info *cval, int unitid,
 				  struct snd_kcontrol *kctl);
 
+#ifdef CONFIG_PM
+void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer);
+#endif
+
 #endif /* SND_USB_MIXER_QUIRKS_H */
 
diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c
index c33e237..4aeb948 100644
--- a/sound/usb/mixer_scarlett.c
+++ b/sound/usb/mixer_scarlett.c
@@ -287,8 +287,7 @@ static int scarlett_ctl_switch_put(struct snd_kcontrol *kctl,
 
 static int scarlett_ctl_resume(struct usb_mixer_elem_list *list)
 {
-	struct usb_mixer_elem_info *elem =
-		container_of(list, struct usb_mixer_elem_info, head);
+	struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list);
 	int i;
 
 	for (i = 0; i < elem->channels; i++)
@@ -447,8 +446,7 @@ static int scarlett_ctl_enum_put(struct snd_kcontrol *kctl,
 
 static int scarlett_ctl_enum_resume(struct usb_mixer_elem_list *list)
 {
-	struct usb_mixer_elem_info *elem =
-		container_of(list, struct usb_mixer_elem_info, head);
+	struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list);
 
 	if (elem->cached)
 		snd_usb_set_cur_mix_value(elem, 0, 0, *elem->cache_val);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 3cbfae6..dc2dfec 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -321,6 +321,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 	struct usb_host_interface *alts;
 	struct usb_interface *iface;
 	unsigned int ep;
+	unsigned int ifnum;
 
 	/* Implicit feedback sync EPs consumers are always playback EPs */
 	if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK)
@@ -330,44 +331,27 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 	case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
 	case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
 		ep = 0x81;
-		iface = usb_ifnum_to_if(dev, 3);
-
-		if (!iface || iface->num_altsetting == 0)
-			return -EINVAL;
-
-		alts = &iface->altsetting[1];
-		goto add_sync_ep;
-		break;
+		ifnum = 3;
+		goto add_sync_ep_from_ifnum;
 	case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
 	case USB_ID(0x0763, 0x2081):
 		ep = 0x81;
-		iface = usb_ifnum_to_if(dev, 2);
-
-		if (!iface || iface->num_altsetting == 0)
-			return -EINVAL;
-
-		alts = &iface->altsetting[1];
-		goto add_sync_ep;
-	case USB_ID(0x2466, 0x8003):
+		ifnum = 2;
+		goto add_sync_ep_from_ifnum;
+	case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */
 		ep = 0x86;
-		iface = usb_ifnum_to_if(dev, 2);
-
-		if (!iface || iface->num_altsetting == 0)
-			return -EINVAL;
-
-		alts = &iface->altsetting[1];
-		goto add_sync_ep;
-	case USB_ID(0x1397, 0x0002):
+		ifnum = 2;
+		goto add_sync_ep_from_ifnum;
+	case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx III */
 		ep = 0x81;
-		iface = usb_ifnum_to_if(dev, 1);
-
-		if (!iface || iface->num_altsetting == 0)
-			return -EINVAL;
-
-		alts = &iface->altsetting[1];
-		goto add_sync_ep;
-
+		ifnum = 2;
+		goto add_sync_ep_from_ifnum;
+	case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */
+		ep = 0x81;
+		ifnum = 1;
+		goto add_sync_ep_from_ifnum;
 	}
+
 	if (attr == USB_ENDPOINT_SYNC_ASYNC &&
 	    altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
 	    altsd->bInterfaceProtocol == 2 &&
@@ -382,6 +366,14 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 	/* No quirk */
 	return 0;
 
+add_sync_ep_from_ifnum:
+	iface = usb_ifnum_to_if(dev, ifnum);
+
+	if (!iface || iface->num_altsetting == 0)
+		return -EINVAL;
+
+	alts = &iface->altsetting[1];
+
 add_sync_ep:
 	subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
 						   alts, ep, !subs->direction,
@@ -507,7 +499,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 	iface = usb_ifnum_to_if(dev, fmt->iface);
 	if (WARN_ON(!iface))
 		return -EINVAL;
-	alts = &iface->altsetting[fmt->altset_idx];
+	alts = usb_altnum_to_altsetting(iface, fmt->altsetting);
 	altsd = get_iface_desc(alts);
 	if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting))
 		return -EINVAL;
@@ -517,21 +509,21 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 
 	/* close the old interface */
 	if (subs->interface >= 0 && subs->interface != fmt->iface) {
-		err = usb_set_interface(subs->dev, subs->interface, 0);
-		if (err < 0) {
-			dev_err(&dev->dev,
-				"%d:%d: return to setting 0 failed (%d)\n",
-				fmt->iface, fmt->altsetting, err);
-			return -EIO;
+		if (!subs->stream->chip->keep_iface) {
+			err = usb_set_interface(subs->dev, subs->interface, 0);
+			if (err < 0) {
+				dev_err(&dev->dev,
+					"%d:%d: return to setting 0 failed (%d)\n",
+					fmt->iface, fmt->altsetting, err);
+				return -EIO;
+			}
 		}
 		subs->interface = -1;
 		subs->altset_idx = 0;
 	}
 
 	/* set interface */
-	if (subs->interface != fmt->iface ||
-	    subs->altset_idx != fmt->altset_idx) {
-
+	if (iface->cur_altsetting != alts) {
 		err = snd_usb_select_mode_quirk(subs, fmt);
 		if (err < 0)
 			return -EIO;
@@ -545,12 +537,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 		}
 		dev_dbg(&dev->dev, "setting usb interface %d:%d\n",
 			fmt->iface, fmt->altsetting);
-		subs->interface = fmt->iface;
-		subs->altset_idx = fmt->altset_idx;
-
 		snd_usb_set_interface_quirk(dev);
 	}
 
+	subs->interface = fmt->iface;
+	subs->altset_idx = fmt->altset_idx;
 	subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip,
 						   alts, fmt->endpoint, subs->direction,
 						   SND_USB_ENDPOINT_TYPE_DATA);
@@ -1264,7 +1255,8 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
 
 	stop_endpoints(subs, true);
 
-	if (subs->interface >= 0 &&
+	if (!as->chip->keep_iface &&
+	    subs->interface >= 0 &&
 	    !snd_usb_lock_shutdown(subs->stream->chip)) {
 		usb_set_interface(subs->dev, subs->interface, 0);
 		subs->interface = -1;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 754e632..0e37e35 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3371,5 +3371,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 		}
 	}
 },
+/* Dell WD15 Dock */
+{
+	USB_DEVICE(0x0bda, 0x4014),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "Dell",
+		.product_name = "WD15 Dock",
+		.profile_name = "Dell-WD15-Dock",
+		.ifnum = QUIRK_NO_INTERFACE
+	}
+},
 
 #undef USB_DEVICE_VENDOR_SPEC
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index acbeb52..f4b6917 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -851,6 +851,36 @@ static int snd_usb_mbox2_boot_quirk(struct usb_device *dev)
 	return 0; /* Successful boot */
 }
 
+static int snd_usb_axefx3_boot_quirk(struct usb_device *dev)
+{
+	int err;
+
+	dev_dbg(&dev->dev, "Waiting for Axe-Fx III to boot up...\n");
+
+	/* If the Axe-Fx III has not fully booted, it will timeout when trying
+	 * to enable the audio streaming interface. A more generous timeout is
+	 * used here to detect when the Axe-Fx III has finished booting as the
+	 * set interface message will be acked once it has
+	 */
+	err = usb_control_msg(dev, usb_sndctrlpipe(dev, 0),
+				USB_REQ_SET_INTERFACE, USB_RECIP_INTERFACE,
+				1, 1, NULL, 0, 120000);
+	if (err < 0) {
+		dev_err(&dev->dev,
+			"failed waiting for Axe-Fx III to boot: %d\n", err);
+		return err;
+	}
+
+	dev_dbg(&dev->dev, "Axe-Fx III is now ready\n");
+
+	err = usb_set_interface(dev, 1, 0);
+	if (err < 0)
+		dev_dbg(&dev->dev,
+			"error stopping Axe-Fx III interface: %d\n", err);
+
+	return 0;
+}
+
 /*
  * Setup quirks
  */
@@ -1026,6 +1056,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
 		return snd_usb_fasttrackpro_boot_quirk(dev);
 	case USB_ID(0x047f, 0xc010): /* Plantronics Gamecom 780 */
 		return snd_usb_gamecon780_boot_quirk(dev);
+	case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */
+		return snd_usb_axefx3_boot_quirk(dev);
 	}
 
 	return 0;
@@ -1327,20 +1359,47 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 
 	/* XMOS based USB DACs */
 	switch (chip->usb_id) {
+	case USB_ID(0x1511, 0x0037): /* AURALiC VEGA */
+	case USB_ID(0x20b1, 0x0002): /* Wyred 4 Sound DAC-2 DSD */
+	case USB_ID(0x20b1, 0x2004): /* Matrix Audio X-SPDIF 2 */
 	case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */
 	case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */
 	case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */
 	case USB_ID(0x22d9, 0x0416): /* OPPO HA-1 */
+	case USB_ID(0x22d9, 0x0436): /* OPPO Sonica */
+	case USB_ID(0x22d9, 0x0461): /* OPPO UDP-205 */
+	case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */
+	case USB_ID(0x25ce, 0x001f): /* Mytek Brooklyn DAC */
+	case USB_ID(0x25ce, 0x0021): /* Mytek Manhattan DAC */
+	case USB_ID(0x25ce, 0x8025): /* Mytek Brooklyn DAC+ */
 	case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */
 		if (fp->altsetting == 2)
 			return SNDRV_PCM_FMTBIT_DSD_U32_BE;
 		break;
 
+	case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */
+	case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */
+	case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */
+	case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */
+	case USB_ID(0x1db5, 0x0003): /* Bryston BDA3 */
 	case USB_ID(0x20b1, 0x000a): /* Gustard DAC-X20U */
+	case USB_ID(0x20b1, 0x2005): /* Denafrips Ares DAC */
 	case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */
 	case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */
+	case USB_ID(0x20b1, 0x3021): /* Eastern El. MiniMax Tube DAC Supreme */
 	case USB_ID(0x20b1, 0x3023): /* Aune X1S 32BIT/384 DSD DAC */
+	case USB_ID(0x20b1, 0x302d): /* Unison Research Unico CD Due */
+	case USB_ID(0x20b1, 0x3036): /* Holo Springs Level 3 R2R DAC */
+	case USB_ID(0x20b1, 0x307b): /* CH Precision C1 DAC */
+	case USB_ID(0x20b1, 0x3086): /* Singxer F-1 converter board */
+	case USB_ID(0x22d9, 0x0426): /* OPPO HA-2 */
+	case USB_ID(0x22e1, 0xca01): /* HDTA Serenade DSD */
+	case USB_ID(0x249c, 0x9326): /* M2Tech Young MkIII */
 	case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */
+	case USB_ID(0x2622, 0x0041): /* Audiolab M-DAC+ */
+	case USB_ID(0x27f7, 0x3002): /* W4S DAC-2v2SE */
+	case USB_ID(0x29a2, 0x0086): /* Mutec MC3+ USB */
+	case USB_ID(0x6b42, 0x0042): /* MSB Technology */
 		if (fp->altsetting == 3)
 			return SNDRV_PCM_FMTBIT_DSD_U32_BE;
 		break;
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 5ed3345..bce3152 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -633,6 +633,391 @@ snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
 	return NULL;
 }
 
+static struct audioformat *
+audio_format_alloc_init(struct snd_usb_audio *chip,
+		       struct usb_host_interface *alts,
+		       int protocol, int iface_no, int altset_idx,
+		       int altno, int num_channels, int clock)
+{
+	struct audioformat *fp;
+
+	fp = kzalloc(sizeof(*fp), GFP_KERNEL);
+	if (!fp)
+		return NULL;
+
+	fp->iface = iface_no;
+	fp->altsetting = altno;
+	fp->altset_idx = altset_idx;
+	fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
+	fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
+	fp->datainterval = snd_usb_parse_datainterval(chip, alts);
+	fp->protocol = protocol;
+	fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+	fp->channels = num_channels;
+	if (snd_usb_get_speed(chip->dev) == USB_SPEED_HIGH)
+		fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
+				* (fp->maxpacksize & 0x7ff);
+	fp->clock = clock;
+	INIT_LIST_HEAD(&fp->list);
+
+	return fp;
+}
+
+static struct audioformat *
+snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip,
+			      struct usb_host_interface *alts,
+			      int protocol, int iface_no, int altset_idx,
+			      int altno, int stream, int bm_quirk)
+{
+	struct usb_device *dev = chip->dev;
+	struct uac_format_type_i_continuous_descriptor *fmt;
+	unsigned int num_channels = 0, chconfig = 0;
+	struct audioformat *fp;
+	int clock = 0;
+	u64 format;
+
+	/* get audio formats */
+	if (protocol == UAC_VERSION_1) {
+		struct uac1_as_header_descriptor *as =
+			snd_usb_find_csint_desc(alts->extra, alts->extralen,
+						NULL, UAC_AS_GENERAL);
+		struct uac_input_terminal_descriptor *iterm;
+
+		if (!as) {
+			dev_err(&dev->dev,
+				"%u:%d : UAC_AS_GENERAL descriptor not found\n",
+				iface_no, altno);
+			return NULL;
+		}
+
+		if (as->bLength < sizeof(*as)) {
+			dev_err(&dev->dev,
+				"%u:%d : invalid UAC_AS_GENERAL desc\n",
+				iface_no, altno);
+			return NULL;
+		}
+
+		format = le16_to_cpu(as->wFormatTag); /* remember the format value */
+
+		iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
+							     as->bTerminalLink);
+		if (iterm) {
+			num_channels = iterm->bNrChannels;
+			chconfig = le16_to_cpu(iterm->wChannelConfig);
+		}
+	} else { /* UAC_VERSION_2 */
+		struct uac2_input_terminal_descriptor *input_term;
+		struct uac2_output_terminal_descriptor *output_term;
+		struct uac2_as_header_descriptor *as =
+			snd_usb_find_csint_desc(alts->extra, alts->extralen,
+						NULL, UAC_AS_GENERAL);
+
+		if (!as) {
+			dev_err(&dev->dev,
+				"%u:%d : UAC_AS_GENERAL descriptor not found\n",
+				iface_no, altno);
+			return NULL;
+		}
+
+		if (as->bLength < sizeof(*as)) {
+			dev_err(&dev->dev,
+				"%u:%d : invalid UAC_AS_GENERAL desc\n",
+				iface_no, altno);
+			return NULL;
+		}
+
+		num_channels = as->bNrChannels;
+		format = le32_to_cpu(as->bmFormats);
+		chconfig = le32_to_cpu(as->bmChannelConfig);
+
+		/*
+		 * lookup the terminal associated to this interface
+		 * to extract the clock
+		 */
+		input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
+								    as->bTerminalLink);
+		if (input_term) {
+			clock = input_term->bCSourceID;
+			if (!chconfig && (num_channels == input_term->bNrChannels))
+				chconfig = le32_to_cpu(input_term->bmChannelConfig);
+			goto found_clock;
+		}
+
+		output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
+								      as->bTerminalLink);
+		if (output_term) {
+			clock = output_term->bCSourceID;
+			goto found_clock;
+		}
+
+		dev_err(&dev->dev,
+			"%u:%d : bogus bTerminalLink %d\n",
+			iface_no, altno, as->bTerminalLink);
+		return NULL;
+	}
+
+found_clock:
+	/* get format type */
+	fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen,
+				      NULL, UAC_FORMAT_TYPE);
+	if (!fmt) {
+		dev_err(&dev->dev,
+			"%u:%d : no UAC_FORMAT_TYPE desc\n",
+			iface_no, altno);
+		return NULL;
+	}
+	if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8))
+			|| ((protocol == UAC_VERSION_2) &&
+					(fmt->bLength < 6))) {
+		dev_err(&dev->dev,
+			"%u:%d : invalid UAC_FORMAT_TYPE desc\n",
+			iface_no, altno);
+		return NULL;
+	}
+
+	/*
+	 * Blue Microphones workaround: The last altsetting is
+	 * identical with the previous one, except for a larger
+	 * packet size, but is actually a mislabeled two-channel
+	 * setting; ignore it.
+	 *
+	 * Part 2: analyze quirk flag and format
+	 */
+	if (bm_quirk && fmt->bNrChannels == 1 && fmt->bSubframeSize == 2)
+		return NULL;
+
+	fp = audio_format_alloc_init(chip, alts, protocol, iface_no,
+				     altset_idx, altno, num_channels, clock);
+	if (!fp)
+		return ERR_PTR(-ENOMEM);
+
+	fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol,
+						       iface_no);
+
+	/* some quirks for attributes here */
+	snd_usb_audioformat_attributes_quirk(chip, fp, stream);
+
+	/* ok, let's parse further... */
+	if (snd_usb_parse_audio_format(chip, fp, format,
+					fmt, stream) < 0) {
+		kfree(fp->rate_table);
+		kfree(fp);
+		return NULL;
+	}
+
+	/* Create chmap */
+	if (fp->channels != num_channels)
+		chconfig = 0;
+
+	fp->chmap = convert_chmap(fp->channels, chconfig, protocol);
+
+	return fp;
+}
+
+static struct audioformat *
+snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip,
+			     struct usb_host_interface *alts,
+			     int iface_no, int altset_idx,
+			     int altno, int stream)
+{
+	struct usb_device *dev = chip->dev;
+	struct uac3_input_terminal_descriptor *input_term;
+	struct uac3_output_terminal_descriptor *output_term;
+	struct uac3_cluster_header_descriptor *cluster;
+	struct uac3_as_header_descriptor *as = NULL;
+	struct uac3_hc_descriptor_header hc_header;
+	struct snd_pcm_chmap_elem *chmap;
+	unsigned char badd_profile;
+	u64 badd_formats = 0;
+	unsigned int num_channels;
+	struct audioformat *fp;
+	u16 cluster_id, wLength;
+	int clock = 0;
+	int err;
+
+	badd_profile = chip->badd_profile;
+
+	if (badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) {
+		unsigned int maxpacksize =
+			le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+
+		switch (maxpacksize) {
+		default:
+			dev_err(&dev->dev,
+				"%u:%d : incorrect wMaxPacketSize for BADD profile\n",
+				iface_no, altno);
+			return NULL;
+		case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_16:
+		case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_16:
+			badd_formats = SNDRV_PCM_FMTBIT_S16_LE;
+			num_channels = 1;
+			break;
+		case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_24:
+		case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_24:
+			badd_formats = SNDRV_PCM_FMTBIT_S24_3LE;
+			num_channels = 1;
+			break;
+		case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_16:
+		case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_16:
+			badd_formats = SNDRV_PCM_FMTBIT_S16_LE;
+			num_channels = 2;
+			break;
+		case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_24:
+		case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_24:
+			badd_formats = SNDRV_PCM_FMTBIT_S24_3LE;
+			num_channels = 2;
+			break;
+		}
+
+		chmap = kzalloc(sizeof(*chmap), GFP_KERNEL);
+		if (!chmap)
+			return ERR_PTR(-ENOMEM);
+
+		if (num_channels == 1) {
+			chmap->map[0] = SNDRV_CHMAP_MONO;
+		} else {
+			chmap->map[0] = SNDRV_CHMAP_FL;
+			chmap->map[1] = SNDRV_CHMAP_FR;
+		}
+
+		chmap->channels = num_channels;
+		clock = UAC3_BADD_CS_ID9;
+		goto found_clock;
+	}
+
+	as = snd_usb_find_csint_desc(alts->extra, alts->extralen,
+				     NULL, UAC_AS_GENERAL);
+	if (!as) {
+		dev_err(&dev->dev,
+			"%u:%d : UAC_AS_GENERAL descriptor not found\n",
+			iface_no, altno);
+		return NULL;
+	}
+
+	if (as->bLength < sizeof(*as)) {
+		dev_err(&dev->dev,
+			"%u:%d : invalid UAC_AS_GENERAL desc\n",
+			iface_no, altno);
+		return NULL;
+	}
+
+	cluster_id = le16_to_cpu(as->wClusterDescrID);
+	if (!cluster_id) {
+		dev_err(&dev->dev,
+			"%u:%d : no cluster descriptor\n",
+			iface_no, altno);
+		return NULL;
+	}
+
+	/*
+	 * Get number of channels and channel map through
+	 * High Capability Cluster Descriptor
+	 *
+	 * First step: get High Capability header and
+	 * read size of Cluster Descriptor
+	 */
+	err = snd_usb_ctl_msg(chip->dev,
+			usb_rcvctrlpipe(chip->dev, 0),
+			UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR,
+			USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+			cluster_id,
+			snd_usb_ctrl_intf(chip),
+			&hc_header, sizeof(hc_header));
+	if (err < 0)
+		return ERR_PTR(err);
+	else if (err != sizeof(hc_header)) {
+		dev_err(&dev->dev,
+			"%u:%d : can't get High Capability descriptor\n",
+			iface_no, altno);
+		return ERR_PTR(-EIO);
+	}
+
+	/*
+	 * Second step: allocate needed amount of memory
+	 * and request Cluster Descriptor
+	 */
+	wLength = le16_to_cpu(hc_header.wLength);
+	cluster = kzalloc(wLength, GFP_KERNEL);
+	if (!cluster)
+		return ERR_PTR(-ENOMEM);
+	err = snd_usb_ctl_msg(chip->dev,
+			usb_rcvctrlpipe(chip->dev, 0),
+			UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR,
+			USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+			cluster_id,
+			snd_usb_ctrl_intf(chip),
+			cluster, wLength);
+	if (err < 0) {
+		kfree(cluster);
+		return ERR_PTR(err);
+	} else if (err != wLength) {
+		dev_err(&dev->dev,
+			"%u:%d : can't get Cluster Descriptor\n",
+			iface_no, altno);
+		kfree(cluster);
+		return ERR_PTR(-EIO);
+	}
+
+	num_channels = cluster->bNrChannels;
+	chmap = convert_chmap_v3(cluster);
+	kfree(cluster);
+
+	/*
+	 * lookup the terminal associated to this interface
+	 * to extract the clock
+	 */
+	input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
+							    as->bTerminalLink);
+	if (input_term) {
+		clock = input_term->bCSourceID;
+		goto found_clock;
+	}
+
+	output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
+							     as->bTerminalLink);
+	if (output_term) {
+		clock = output_term->bCSourceID;
+		goto found_clock;
+	}
+
+	dev_err(&dev->dev, "%u:%d : bogus bTerminalLink %d\n",
+			iface_no, altno, as->bTerminalLink);
+	return NULL;
+
+found_clock:
+	fp = audio_format_alloc_init(chip, alts, UAC_VERSION_3, iface_no,
+				     altset_idx, altno, num_channels, clock);
+	if (!fp)
+		return ERR_PTR(-ENOMEM);
+
+	fp->chmap = chmap;
+
+	if (badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) {
+		fp->attributes = 0; /* No attributes */
+
+		fp->fmt_type = UAC_FORMAT_TYPE_I;
+		fp->formats = badd_formats;
+
+		fp->nr_rates = 0;	/* SNDRV_PCM_RATE_CONTINUOUS */
+		fp->rate_min = UAC3_BADD_SAMPLING_RATE;
+		fp->rate_max = UAC3_BADD_SAMPLING_RATE;
+		fp->rates = SNDRV_PCM_RATE_CONTINUOUS;
+
+	} else {
+		fp->attributes = parse_uac_endpoint_attributes(chip, alts,
+							       UAC_VERSION_3,
+							       iface_no);
+		/* ok, let's parse further... */
+		if (snd_usb_parse_audio_format_v3(chip, fp, as, stream) < 0) {
+			kfree(fp->rate_table);
+			kfree(fp);
+			return NULL;
+		}
+	}
+
+	return fp;
+}
+
 int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 {
 	struct usb_device *dev;
@@ -640,13 +1025,8 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 	struct usb_host_interface *alts;
 	struct usb_interface_descriptor *altsd;
 	int i, altno, err, stream;
-	u64 format = 0;
-	unsigned int num_channels = 0;
 	struct audioformat *fp = NULL;
-	int num, protocol, clock = 0;
-	struct uac_format_type_i_continuous_descriptor *fmt = NULL;
-	struct snd_pcm_chmap_elem *chmap_v3 = NULL;
-	unsigned int chconfig;
+	int num, protocol;
 
 	dev = chip->dev;
 
@@ -695,303 +1075,48 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 		    protocol <= 2)
 			protocol = UAC_VERSION_1;
 
-		chconfig = 0;
-		/* get audio formats */
 		switch (protocol) {
 		default:
 			dev_dbg(&dev->dev, "%u:%d: unknown interface protocol %#02x, assuming v1\n",
 				iface_no, altno, protocol);
 			protocol = UAC_VERSION_1;
 			/* fall through */
-
-		case UAC_VERSION_1: {
-			struct uac1_as_header_descriptor *as =
-				snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
-			struct uac_input_terminal_descriptor *iterm;
-
-			if (!as) {
-				dev_err(&dev->dev,
-					"%u:%d : UAC_AS_GENERAL descriptor not found\n",
-					iface_no, altno);
-				continue;
-			}
-
-			if (as->bLength < sizeof(*as)) {
-				dev_err(&dev->dev,
-					"%u:%d : invalid UAC_AS_GENERAL desc\n",
-					iface_no, altno);
-				continue;
-			}
-
-			format = le16_to_cpu(as->wFormatTag); /* remember the format value */
-
-			iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
-								       as->bTerminalLink);
-			if (iterm) {
-				num_channels = iterm->bNrChannels;
-				chconfig = le16_to_cpu(iterm->wChannelConfig);
-			}
-
-			break;
-		}
-
+		case UAC_VERSION_1:
+			/* fall through */
 		case UAC_VERSION_2: {
-			struct uac2_input_terminal_descriptor *input_term;
-			struct uac2_output_terminal_descriptor *output_term;
-			struct uac2_as_header_descriptor *as =
-				snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
-
-			if (!as) {
-				dev_err(&dev->dev,
-					"%u:%d : UAC_AS_GENERAL descriptor not found\n",
-					iface_no, altno);
-				continue;
-			}
-
-			if (as->bLength < sizeof(*as)) {
-				dev_err(&dev->dev,
-					"%u:%d : invalid UAC_AS_GENERAL desc\n",
-					iface_no, altno);
-				continue;
-			}
-
-			num_channels = as->bNrChannels;
-			format = le32_to_cpu(as->bmFormats);
-			chconfig = le32_to_cpu(as->bmChannelConfig);
-
-			/* lookup the terminal associated to this interface
-			 * to extract the clock */
-			input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
-									    as->bTerminalLink);
-			if (input_term) {
-				clock = input_term->bCSourceID;
-				if (!chconfig && (num_channels == input_term->bNrChannels))
-					chconfig = le32_to_cpu(input_term->bmChannelConfig);
-				break;
-			}
-
-			output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
-									      as->bTerminalLink);
-			if (output_term) {
-				clock = output_term->bCSourceID;
-				break;
-			}
-
-			dev_err(&dev->dev,
-				"%u:%d : bogus bTerminalLink %d\n",
-				iface_no, altno, as->bTerminalLink);
-			continue;
-		}
-
-		case UAC_VERSION_3: {
-			struct uac3_input_terminal_descriptor *input_term;
-			struct uac3_output_terminal_descriptor *output_term;
-			struct uac3_as_header_descriptor *as;
-			struct uac3_cluster_header_descriptor *cluster;
-			struct uac3_hc_descriptor_header hc_header;
-			u16 cluster_id, wLength;
-
-			as = snd_usb_find_csint_desc(alts->extra,
-							alts->extralen,
-							NULL, UAC_AS_GENERAL);
-
-			if (!as) {
-				dev_err(&dev->dev,
-					"%u:%d : UAC_AS_GENERAL descriptor not found\n",
-					iface_no, altno);
-				continue;
-			}
-
-			if (as->bLength < sizeof(*as)) {
-				dev_err(&dev->dev,
-					"%u:%d : invalid UAC_AS_GENERAL desc\n",
-					iface_no, altno);
-				continue;
-			}
-
-			cluster_id = le16_to_cpu(as->wClusterDescrID);
-			if (!cluster_id) {
-				dev_err(&dev->dev,
-					"%u:%d : no cluster descriptor\n",
-					iface_no, altno);
-				continue;
-			}
-
-			/*
-			 * Get number of channels and channel map through
-			 * High Capability Cluster Descriptor
-			 *
-			 * First step: get High Capability header and
-			 * read size of Cluster Descriptor
-			 */
-			err = snd_usb_ctl_msg(chip->dev,
-					usb_rcvctrlpipe(chip->dev, 0),
-					UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR,
-					USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
-					cluster_id,
-					snd_usb_ctrl_intf(chip),
-					&hc_header, sizeof(hc_header));
-			if (err < 0)
-				return err;
-			else if (err != sizeof(hc_header)) {
-				dev_err(&dev->dev,
-					"%u:%d : can't get High Capability descriptor\n",
-					iface_no, altno);
-				return -EIO;
-			}
-
-			/*
-			 * Second step: allocate needed amount of memory
-			 * and request Cluster Descriptor
-			 */
-			wLength = le16_to_cpu(hc_header.wLength);
-			cluster = kzalloc(wLength, GFP_KERNEL);
-			if (!cluster)
-				return -ENOMEM;
-			err = snd_usb_ctl_msg(chip->dev,
-					usb_rcvctrlpipe(chip->dev, 0),
-					UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR,
-					USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
-					cluster_id,
-					snd_usb_ctrl_intf(chip),
-					cluster, wLength);
-			if (err < 0) {
-				kfree(cluster);
-				return err;
-			} else if (err != wLength) {
-				dev_err(&dev->dev,
-					"%u:%d : can't get Cluster Descriptor\n",
-					iface_no, altno);
-				kfree(cluster);
-				return -EIO;
-			}
-
-			num_channels = cluster->bNrChannels;
-			chmap_v3 = convert_chmap_v3(cluster);
-
-			kfree(cluster);
-
-			format = le64_to_cpu(as->bmFormats);
-
-			/* lookup the terminal associated to this interface
-			 * to extract the clock */
-			input_term = snd_usb_find_input_terminal_descriptor(
-							chip->ctrl_intf,
-							as->bTerminalLink);
-
-			if (input_term) {
-				clock = input_term->bCSourceID;
-				break;
-			}
-
-			output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
-									      as->bTerminalLink);
-			if (output_term) {
-				clock = output_term->bCSourceID;
-				break;
-			}
-
-			dev_err(&dev->dev,
-				"%u:%d : bogus bTerminalLink %d\n",
-				iface_no, altno, as->bTerminalLink);
-			continue;
-		}
-		}
-
-		if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) {
-			/* get format type */
-			fmt = snd_usb_find_csint_desc(alts->extra,
-						      alts->extralen,
-						      NULL, UAC_FORMAT_TYPE);
-			if (!fmt) {
-				dev_err(&dev->dev,
-					"%u:%d : no UAC_FORMAT_TYPE desc\n",
-					iface_no, altno);
-				continue;
-			}
-			if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8))
-					|| ((protocol == UAC_VERSION_2) &&
-							(fmt->bLength < 6))) {
-				dev_err(&dev->dev,
-					"%u:%d : invalid UAC_FORMAT_TYPE desc\n",
-					iface_no, altno);
-				continue;
-			}
+			int bm_quirk = 0;
 
 			/*
 			 * Blue Microphones workaround: The last altsetting is
 			 * identical with the previous one, except for a larger
 			 * packet size, but is actually a mislabeled two-channel
 			 * setting; ignore it.
+			 *
+			 * Part 1: prepare quirk flag
 			 */
-			if (fmt->bNrChannels == 1 &&
-			    fmt->bSubframeSize == 2 &&
-			    altno == 2 && num == 3 &&
+			if (altno == 2 && num == 3 &&
 			    fp && fp->altsetting == 1 && fp->channels == 1 &&
 			    fp->formats == SNDRV_PCM_FMTBIT_S16_LE &&
 			    protocol == UAC_VERSION_1 &&
 			    le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) ==
 							fp->maxpacksize * 2)
-				continue;
+				bm_quirk = 1;
+
+			fp = snd_usb_get_audioformat_uac12(chip, alts, protocol,
+							   iface_no, i, altno,
+							   stream, bm_quirk);
+			break;
+		}
+		case UAC_VERSION_3:
+			fp = snd_usb_get_audioformat_uac3(chip, alts,
+						iface_no, i, altno, stream);
+			break;
 		}
 
-		fp = kzalloc(sizeof(*fp), GFP_KERNEL);
 		if (!fp)
-			return -ENOMEM;
-
-		fp->iface = iface_no;
-		fp->altsetting = altno;
-		fp->altset_idx = i;
-		fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
-		fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
-		fp->datainterval = snd_usb_parse_datainterval(chip, alts);
-		fp->protocol = protocol;
-		fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
-		fp->channels = num_channels;
-		if (snd_usb_get_speed(dev) == USB_SPEED_HIGH)
-			fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
-					* (fp->maxpacksize & 0x7ff);
-		fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
-		fp->clock = clock;
-		INIT_LIST_HEAD(&fp->list);
-
-		/* some quirks for attributes here */
-		snd_usb_audioformat_attributes_quirk(chip, fp, stream);
-
-		/* ok, let's parse further... */
-		if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) {
-			if (snd_usb_parse_audio_format(chip, fp, format,
-							fmt, stream) < 0) {
-				kfree(fp->rate_table);
-				kfree(fp);
-				fp = NULL;
-				continue;
-			}
-		} else {
-			struct uac3_as_header_descriptor *as;
-
-			as = snd_usb_find_csint_desc(alts->extra,
-						     alts->extralen,
-						     NULL, UAC_AS_GENERAL);
-
-			if (snd_usb_parse_audio_format_v3(chip, fp, as,
-								stream) < 0) {
-				kfree(fp->rate_table);
-				kfree(fp);
-				fp = NULL;
-				continue;
-			}
-		}
-
-		/* Create chmap */
-		if (fp->channels != num_channels)
-			chconfig = 0;
-
-		if (protocol == UAC_VERSION_3)
-			fp->chmap = chmap_v3;
-		else
-			fp->chmap = convert_chmap(fp->channels, chconfig,
-						  protocol);
+			continue;
+		else if (IS_ERR(fp))
+			return PTR_ERR(fp);
 
 		dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint);
 		err = snd_usb_add_audio_stream(chip, stream, fp);
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 4d5c89a..7b28cbd 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -49,6 +49,8 @@ struct snd_usb_audio {
 	int num_suspended_intf;
 	int sample_rate_read_error;
 
+	int badd_profile;		/* UAC3 BADD profile */
+
 	struct list_head pcm_list;	/* list of pcm streams */
 	struct list_head ep_list;	/* list of audio-related endpoints */
 	int pcm_devs;
@@ -59,6 +61,9 @@ struct snd_usb_audio {
 
 	int setup;			/* from the 'device_setup' module param */
 	bool autoclock;			/* from the 'autoclock' module param */
+	bool keep_iface;		/* keep interface/altset after closing
+					 * or parameter change
+					 */
 
 	struct usb_host_interface *ctrl_intf;	/* the audio control interface */
 };
@@ -109,6 +114,7 @@ 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;
 	const void *data;