diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst
index 392875a1b94e..72f97d4b01a7 100644
--- a/Documentation/sound/alsa-configuration.rst
+++ b/Documentation/sound/alsa-configuration.rst
@@ -2234,6 +2234,19 @@ use_vmalloc
     buffers.  If mmap is used on such architectures, turn off this
     option, so that the DMA-coherent buffers are allocated and used
     instead.
+delayed_register
+    The option is needed for devices that have multiple streams
+    defined in multiple USB interfaces.  The driver may invoke
+    registrations multiple times (once per interface) and this may
+    lead to the insufficient device enumeration.
+    This option receives an array of strings, and you can pass
+    ID:INTERFACE like ``0123abcd:4`` for performing the delayed
+    registration to the given device.  In this example, when a USB
+    device 0123:abcd is probed, the driver waits the registration
+    until the USB interface 4 gets probed.
+    The driver prints a message like "Found post-registration device
+    assignment: 1234abcd:04" for such a device, so that user can
+    notice the need.
 
 This module supports multiple devices, autoprobe and hotplugging.
 
diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h
index ba4b3e3327ff..cb9900b34b67 100644
--- a/include/linux/usb/audio-v2.h
+++ b/include/linux/usb/audio-v2.h
@@ -156,6 +156,18 @@ struct uac2_feature_unit_descriptor {
 	__u8 bmaControls[0]; /* variable length */
 } __attribute__((packed));
 
+/* 4.7.2.10 Effect Unit Descriptor */
+
+struct uac2_effect_unit_descriptor {
+	__u8 bLength;
+	__u8 bDescriptorType;
+	__u8 bDescriptorSubtype;
+	__u8 bUnitID;
+	__le16 wEffectType;
+	__u8 bSourceID;
+	__u8 bmaControls[]; /* variable length */
+} __attribute__((packed));
+
 /* 4.9.2 Class-Specific AS Interface Descriptor */
 
 struct uac2_as_header_descriptor {
diff --git a/include/sound/core.h b/include/sound/core.h
index ac8b692b69b4..381a010a1bd4 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -266,6 +266,7 @@ void snd_device_disconnect(struct snd_card *card, void *device_data);
 void snd_device_disconnect_all(struct snd_card *card);
 void snd_device_free(struct snd_card *card, void *device_data);
 void snd_device_free_all(struct snd_card *card);
+int snd_device_get_state(struct snd_card *card, void *device_data);
 
 /* isadma.c */
 
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index f657ff08f317..31a4b300e4c9 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -1415,6 +1415,15 @@ static inline u64 pcm_format_to_bits(snd_pcm_format_t pcm_format)
 	return 1ULL << (__force int) pcm_format;
 }
 
+/**
+ * pcm_for_each_format - helper to iterate for each format type
+ * @f: the iterator variable in snd_pcm_format_t type
+ */
+#define pcm_for_each_format(f)						\
+	for ((f) = SNDRV_PCM_FORMAT_FIRST;				\
+	     (__force int)(f) <= (__force int)SNDRV_PCM_FORMAT_LAST;	\
+	     (f) = (__force snd_pcm_format_t)((__force int)(f) + 1))
+
 /* printk helpers */
 #define pcm_err(pcm, fmt, args...) \
 	dev_err((pcm)->card->dev, fmt, ##args)
diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h
index 661450a2095b..36f94735d23d 100644
--- a/include/sound/pcm_params.h
+++ b/include/sound/pcm_params.h
@@ -133,6 +133,13 @@ static inline int snd_mask_test(const struct snd_mask *mask, unsigned int val)
 	return mask->bits[MASK_OFS(val)] & MASK_BIT(val);
 }
 
+/* Most of drivers need only this one */
+static inline int snd_mask_test_format(const struct snd_mask *mask,
+				       snd_pcm_format_t format)
+{
+	return snd_mask_test(mask, (__force unsigned int)format);
+}
+
 static inline int snd_mask_single(const struct snd_mask *mask)
 {
 	int i, c = 0;
diff --git a/sound/core/device.c b/sound/core/device.c
index cdc5af526739..bf0b04a7ee79 100644
--- a/sound/core/device.c
+++ b/sound/core/device.c
@@ -237,3 +237,24 @@ void snd_device_free_all(struct snd_card *card)
 	list_for_each_entry_safe_reverse(dev, next, &card->devices, list)
 		__snd_device_free(dev);
 }
+
+/**
+ * snd_device_get_state - Get the current state of the given device
+ * @card: the card instance
+ * @device_data: the data pointer to release
+ *
+ * Returns the current state of the given device object.  For the valid
+ * device, either @SNDRV_DEV_BUILD, @SNDRV_DEV_REGISTERED or
+ * @SNDRV_DEV_DISCONNECTED is returned.
+ * Or for a non-existing device, -1 is returned as an error.
+ */
+int snd_device_get_state(struct snd_card *card, void *device_data)
+{
+	struct snd_device *dev;
+
+	dev = look_for_dev(card, device_data);
+	if (dev)
+		return dev->state;
+	return -1;
+}
+EXPORT_SYMBOL_GPL(snd_device_get_state);
diff --git a/sound/core/info.c b/sound/core/info.c
index ca87ae4c30ba..8c6bc5241df5 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -604,7 +604,7 @@ int snd_info_card_free(struct snd_card *card)
  */
 int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len)
 {
-	int c = -1;
+	int c;
 
 	if (snd_BUG_ON(!buffer || !buffer->buffer))
 		return 1;
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 13db77771f0f..930def8201f4 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -884,20 +884,17 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream)
 		sformat = snd_pcm_plug_slave_format(format, sformat_mask);
 
 	if ((__force int)sformat < 0 ||
-	    !snd_mask_test(sformat_mask, (__force int)sformat)) {
-		for (sformat = (__force snd_pcm_format_t)0;
-		     (__force int)sformat <= (__force int)SNDRV_PCM_FORMAT_LAST;
-		     sformat = (__force snd_pcm_format_t)((__force int)sformat + 1)) {
-			if (snd_mask_test(sformat_mask, (__force int)sformat) &&
+	    !snd_mask_test_format(sformat_mask, sformat)) {
+		pcm_for_each_format(sformat) {
+			if (snd_mask_test_format(sformat_mask, sformat) &&
 			    snd_pcm_oss_format_to(sformat) >= 0)
-				break;
-		}
-		if ((__force int)sformat > (__force int)SNDRV_PCM_FORMAT_LAST) {
-			pcm_dbg(substream->pcm, "Cannot find a format!!!\n");
-			err = -EINVAL;
-			goto failure;
+				goto format_found;
 		}
+		pcm_dbg(substream->pcm, "Cannot find a format!!!\n");
+		err = -EINVAL;
+		goto failure;
 	}
+ format_found:
 	err = _snd_pcm_hw_param_set(sparams, SNDRV_PCM_HW_PARAM_FORMAT, (__force int)sformat, 0);
 	if (err < 0)
 		goto failure;
@@ -1220,8 +1217,10 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const
 			if (ret < 0)
 				break;
 		}
+		mutex_unlock(&runtime->oss.params_lock);
 		ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true,
 					 frames, in_kernel);
+		mutex_lock(&runtime->oss.params_lock);
 		if (ret != -EPIPE && ret != -ESTRPIPE)
 			break;
 		/* test, if we can't store new data, because the stream */
@@ -1257,8 +1256,10 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p
 		ret = snd_pcm_oss_capture_position_fixup(substream, &delay);
 		if (ret < 0)
 			break;
+		mutex_unlock(&runtime->oss.params_lock);
 		ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true,
 					 frames, in_kernel);
+		mutex_lock(&runtime->oss.params_lock);
 		if (ret == -EPIPE) {
 			if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
 				ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 752d078908e9..fbda4ebf38b3 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -196,82 +196,74 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin)
 	return 0;
 }
 
+static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug,
+					 snd_pcm_sframes_t frames)
+{
+	struct snd_pcm_plugin *plugin, *plugin_next;
+
+	plugin = snd_pcm_plug_first(plug);
+	while (plugin && frames > 0) {
+		plugin_next = plugin->next;
+		if (plugin->dst_frames) {
+			frames = plugin->dst_frames(plugin, frames);
+			if (frames < 0)
+				return frames;
+		}
+		if (frames > plugin->buf_frames)
+			frames = plugin->buf_frames;
+		plugin = plugin_next;
+	}
+	return frames;
+}
+
+static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug,
+					 snd_pcm_sframes_t frames)
+{
+	struct snd_pcm_plugin *plugin, *plugin_prev;
+
+	plugin = snd_pcm_plug_last(plug);
+	while (plugin && frames > 0) {
+		if (frames > plugin->buf_frames)
+			frames = plugin->buf_frames;
+		plugin_prev = plugin->prev;
+		if (plugin->src_frames) {
+			frames = plugin->src_frames(plugin, frames);
+			if (frames < 0)
+				return frames;
+		}
+		plugin = plugin_prev;
+	}
+	return frames;
+}
+
 snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames)
 {
-	struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
-	int stream;
-
 	if (snd_BUG_ON(!plug))
 		return -ENXIO;
-	if (drv_frames == 0)
-		return 0;
-	stream = snd_pcm_plug_stream(plug);
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		plugin = snd_pcm_plug_last(plug);
-		while (plugin && drv_frames > 0) {
-			if (drv_frames > plugin->buf_frames)
-				drv_frames = plugin->buf_frames;
-			plugin_prev = plugin->prev;
-			if (plugin->src_frames)
-				drv_frames = plugin->src_frames(plugin, drv_frames);
-			plugin = plugin_prev;
-		}
-	} else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
-		plugin = snd_pcm_plug_first(plug);
-		while (plugin && drv_frames > 0) {
-			plugin_next = plugin->next;
-			if (plugin->dst_frames)
-				drv_frames = plugin->dst_frames(plugin, drv_frames);
-			if (drv_frames > plugin->buf_frames)
-				drv_frames = plugin->buf_frames;
-			plugin = plugin_next;
-		}
-	} else
+	switch (snd_pcm_plug_stream(plug)) {
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		return calc_src_frames(plug, drv_frames);
+	case SNDRV_PCM_STREAM_CAPTURE:
+		return calc_dst_frames(plug, drv_frames);
+	default:
 		snd_BUG();
-	return drv_frames;
+		return -EINVAL;
+	}
 }
 
 snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames)
 {
-	struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
-	snd_pcm_sframes_t frames;
-	int stream;
-	
 	if (snd_BUG_ON(!plug))
 		return -ENXIO;
-	if (clt_frames == 0)
-		return 0;
-	frames = clt_frames;
-	stream = snd_pcm_plug_stream(plug);
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		plugin = snd_pcm_plug_first(plug);
-		while (plugin && frames > 0) {
-			plugin_next = plugin->next;
-			if (plugin->dst_frames) {
-				frames = plugin->dst_frames(plugin, frames);
-				if (frames < 0)
-					return frames;
-			}
-			if (frames > plugin->buf_frames)
-				frames = plugin->buf_frames;
-			plugin = plugin_next;
-		}
-	} else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
-		plugin = snd_pcm_plug_last(plug);
-		while (plugin) {
-			if (frames > plugin->buf_frames)
-				frames = plugin->buf_frames;
-			plugin_prev = plugin->prev;
-			if (plugin->src_frames) {
-				frames = plugin->src_frames(plugin, frames);
-				if (frames < 0)
-					return frames;
-			}
-			plugin = plugin_prev;
-		}
-	} else
+	switch (snd_pcm_plug_stream(plug)) {
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		return calc_dst_frames(plug, clt_frames);
+	case SNDRV_PCM_STREAM_CAPTURE:
+		return calc_src_frames(plug, clt_frames);
+	default:
 		snd_BUG();
-	return frames;
+		return -EINVAL;
+	}
 }
 
 static int snd_pcm_plug_formats(const struct snd_mask *mask,
diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c
index 7cd09cef6961..d381f4c967c9 100644
--- a/sound/core/oss/rate.c
+++ b/sound/core/oss/rate.c
@@ -47,7 +47,7 @@ struct rate_priv {
 	unsigned int pos;
 	rate_f func;
 	snd_pcm_sframes_t old_src_frames, old_dst_frames;
-	struct rate_channel channels[0];
+	struct rate_channel channels[];
 };
 
 static void rate_init(struct snd_pcm_plugin *plugin)
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index a141a301369f..b6d2331a82f7 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -1019,7 +1019,7 @@ static ssize_t show_pcm_class(struct device *dev,
 		str = "none";
 	else
 		str = strs[pcm->dev_class];
-        return snprintf(buf, PAGE_SIZE, "%s\n", str);
+	return sprintf(buf, "%s\n", str);
 }
 
 static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL);
diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c
index 5749a8a49784..b37c70864b57 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/sound/core/pcm_dmaengine.c
@@ -426,7 +426,7 @@ int snd_dmaengine_pcm_refine_runtime_hwparams(
 		 * default assumption is that it supports 1, 2 and 4 bytes
 		 * widths.
 		 */
-		for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+		pcm_for_each_format(i) {
 			int bits = snd_pcm_format_physical_width(i);
 
 			/*
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index a6a541511534..cf3e8c26e00a 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -42,6 +42,11 @@ struct pcm_format_data {
 /* we do lots of calculations on snd_pcm_format_t; shut up sparse */
 #define INT	__force int
 
+static bool valid_format(snd_pcm_format_t format)
+{
+	return (INT)format >= 0 && (INT)format <= (INT)SNDRV_PCM_FORMAT_LAST;
+}
+
 static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = {
 	[SNDRV_PCM_FORMAT_S8] = {
 		.width = 8, .phys = 8, .le = -1, .signd = 1,
@@ -259,7 +264,7 @@ static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] =
 int snd_pcm_format_signed(snd_pcm_format_t format)
 {
 	int val;
-	if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+	if (!valid_format(format))
 		return -EINVAL;
 	if ((val = pcm_formats[(INT)format].signd) < 0)
 		return -EINVAL;
@@ -307,7 +312,7 @@ EXPORT_SYMBOL(snd_pcm_format_linear);
 int snd_pcm_format_little_endian(snd_pcm_format_t format)
 {
 	int val;
-	if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+	if (!valid_format(format))
 		return -EINVAL;
 	if ((val = pcm_formats[(INT)format].le) < 0)
 		return -EINVAL;
@@ -343,7 +348,7 @@ EXPORT_SYMBOL(snd_pcm_format_big_endian);
 int snd_pcm_format_width(snd_pcm_format_t format)
 {
 	int val;
-	if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+	if (!valid_format(format))
 		return -EINVAL;
 	if ((val = pcm_formats[(INT)format].width) == 0)
 		return -EINVAL;
@@ -361,7 +366,7 @@ EXPORT_SYMBOL(snd_pcm_format_width);
 int snd_pcm_format_physical_width(snd_pcm_format_t format)
 {
 	int val;
-	if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+	if (!valid_format(format))
 		return -EINVAL;
 	if ((val = pcm_formats[(INT)format].phys) == 0)
 		return -EINVAL;
@@ -394,7 +399,7 @@ EXPORT_SYMBOL(snd_pcm_format_size);
  */
 const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format)
 {
-	if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+	if (!valid_format(format))
 		return NULL;
 	if (! pcm_formats[(INT)format].phys)
 		return NULL;
@@ -418,7 +423,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int
 	unsigned char *dst;
 	const unsigned char *pat;
 
-	if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST)
+	if (!valid_format(format))
 		return -EINVAL;
 	if (samples == 0)
 		return 0;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index d5443eeb8b63..8e1060b084f1 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -228,6 +228,9 @@ int snd_pcm_info_user(struct snd_pcm_substream *substream,
 	return err;
 }
 
+/* macro for simplified cast */
+#define PARAM_MASK_BIT(b)	(1U << (__force int)(b))
+
 static bool hw_support_mmap(struct snd_pcm_substream *substream)
 {
 	if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP))
@@ -257,7 +260,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream,
 			return -EINVAL;
 
 		/* This parameter is not requested to change by a caller. */
-		if (!(params->rmask & (1 << k)))
+		if (!(params->rmask & PARAM_MASK_BIT(k)))
 			continue;
 
 		if (trace_hw_mask_param_enabled())
@@ -271,7 +274,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream,
 
 		/* Set corresponding flag so that the caller gets it. */
 		trace_hw_mask_param(substream, k, 0, &old_mask, m);
-		params->cmask |= 1 << k;
+		params->cmask |= PARAM_MASK_BIT(k);
 	}
 
 	return 0;
@@ -293,7 +296,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream,
 			return -EINVAL;
 
 		/* This parameter is not requested to change by a caller. */
-		if (!(params->rmask & (1 << k)))
+		if (!(params->rmask & PARAM_MASK_BIT(k)))
 			continue;
 
 		if (trace_hw_interval_param_enabled())
@@ -307,7 +310,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream,
 
 		/* Set corresponding flag so that the caller gets it. */
 		trace_hw_interval_param(substream, k, 0, &old_interval, i);
-		params->cmask |= 1 << k;
+		params->cmask |= PARAM_MASK_BIT(k);
 	}
 
 	return 0;
@@ -349,7 +352,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream,
 	 * have 0 so that the parameters are never changed anymore.
 	 */
 	for (k = 0; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++)
-		vstamps[k] = (params->rmask & (1 << k)) ? 1 : 0;
+		vstamps[k] = (params->rmask & PARAM_MASK_BIT(k)) ? 1 : 0;
 
 	/* Due to the above design, actual sequence number starts at 2. */
 	stamp = 2;
@@ -417,7 +420,7 @@ retry:
 					hw_param_interval(params, r->var));
 			}
 
-			params->cmask |= (1 << r->var);
+			params->cmask |= PARAM_MASK_BIT(r->var);
 			vstamps[r->var] = stamp;
 			again = true;
 		}
@@ -486,9 +489,9 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream,
 
 	params->info = 0;
 	params->fifo_size = 0;
-	if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_SAMPLE_BITS))
+	if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_SAMPLE_BITS))
 		params->msbits = 0;
-	if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_RATE)) {
+	if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_RATE)) {
 		params->rate_num = 0;
 		params->rate_den = 0;
 	}
@@ -2293,21 +2296,21 @@ static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params,
 static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params,
 				  struct snd_pcm_hw_rule *rule)
 {
-	unsigned int k;
+	snd_pcm_format_t k;
 	const struct snd_interval *i =
 				hw_param_interval_c(params, rule->deps[0]);
 	struct snd_mask m;
 	struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 	snd_mask_any(&m);
-	for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) {
+	pcm_for_each_format(k) {
 		int bits;
-		if (! snd_mask_test(mask, k))
+		if (!snd_mask_test_format(mask, k))
 			continue;
 		bits = snd_pcm_format_physical_width(k);
 		if (bits <= 0)
 			continue; /* ignore invalid formats */
 		if ((unsigned)bits < i->min || (unsigned)bits > i->max)
-			snd_mask_reset(&m, k);
+			snd_mask_reset(&m, (__force unsigned)k);
 	}
 	return snd_mask_refine(mask, &m);
 }
@@ -2316,14 +2319,15 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params,
 				       struct snd_pcm_hw_rule *rule)
 {
 	struct snd_interval t;
-	unsigned int k;
+	snd_pcm_format_t k;
+
 	t.min = UINT_MAX;
 	t.max = 0;
 	t.openmin = 0;
 	t.openmax = 0;
-	for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) {
+	pcm_for_each_format(k) {
 		int bits;
-		if (! snd_mask_test(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k))
+		if (!snd_mask_test_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k))
 			continue;
 		bits = snd_pcm_format_physical_width(k);
 		if (bits <= 0)
@@ -2505,16 +2509,16 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
 	unsigned int mask = 0;
 
         if (hw->info & SNDRV_PCM_INFO_INTERLEAVED)
-		mask |= 1 << SNDRV_PCM_ACCESS_RW_INTERLEAVED;
+		mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_INTERLEAVED);
         if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED)
-		mask |= 1 << SNDRV_PCM_ACCESS_RW_NONINTERLEAVED;
+		mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_NONINTERLEAVED);
 	if (hw_support_mmap(substream)) {
 		if (hw->info & SNDRV_PCM_INFO_INTERLEAVED)
-			mask |= 1 << SNDRV_PCM_ACCESS_MMAP_INTERLEAVED;
+			mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_INTERLEAVED);
 		if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED)
-			mask |= 1 << SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED;
+			mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED);
 		if (hw->info & SNDRV_PCM_INFO_COMPLEX)
-			mask |= 1 << SNDRV_PCM_ACCESS_MMAP_COMPLEX;
+			mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_COMPLEX);
 	}
 	err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_ACCESS, mask);
 	if (err < 0)
@@ -2524,7 +2528,8 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
 	if (err < 0)
 		return err;
 
-	err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, 1 << SNDRV_PCM_SUBFORMAT_STD);
+	err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT,
+					 PARAM_MASK_BIT(SNDRV_PCM_SUBFORMAT_STD));
 	if (err < 0)
 		return err;
 
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index d78a27271d6d..251eaf1152e2 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -118,7 +118,7 @@ struct loopback_cable {
 struct loopback_setup {
 	unsigned int notify: 1;
 	unsigned int rate_shift;
-	unsigned int format;
+	snd_pcm_format_t format;
 	unsigned int rate;
 	unsigned int channels;
 	struct snd_ctl_elem_id active_id;
@@ -1432,7 +1432,7 @@ static int loopback_format_info(struct snd_kcontrol *kcontrol,
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 	uinfo->count = 1;
 	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = SNDRV_PCM_FORMAT_LAST;
+	uinfo->value.integer.max = (__force int)SNDRV_PCM_FORMAT_LAST;
 	uinfo->value.integer.step = 1;
 	return 0;
 }                                  
@@ -1443,7 +1443,7 @@ static int loopback_format_get(struct snd_kcontrol *kcontrol,
 	struct loopback *loopback = snd_kcontrol_chip(kcontrol);
 	
 	ucontrol->value.integer.value[0] =
-		loopback->setup[kcontrol->id.subdevice]
+		(__force int)loopback->setup[kcontrol->id.subdevice]
 			       [kcontrol->id.device].format;
 	return 0;
 }
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 02ac3f4e0c02..b5486de08b97 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -901,10 +901,10 @@ static int snd_card_dummy_new_mixer(struct snd_dummy *dummy)
 static void print_formats(struct snd_dummy *dummy,
 			  struct snd_info_buffer *buffer)
 {
-	int i;
+	snd_pcm_format_t i;
 
-	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
-		if (dummy->pcm_hw.formats & (1ULL << i))
+	pcm_for_each_format(i) {
+		if (dummy->pcm_hw.formats & pcm_format_to_bits(i))
 			snd_iprintf(buffer, " %s", snd_pcm_format_name(i));
 	}
 }
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 976d8cb9a34f..2c8e3392a490 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -509,7 +509,7 @@ MODULE_DEVICE_TABLE(ieee1394, bebob_id_table);
 static struct fw_driver bebob_driver = {
 	.driver = {
 		.owner	= THIS_MODULE,
-		.name	= "snd-bebob",
+		.name	= KBUILD_MODNAME,
 		.bus	= &fw_bus_type,
 	},
 	.probe    = bebob_probe,
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index 1f5fc0e7c024..c84b913a9fe0 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -192,7 +192,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_dg00x_id_table);
 static struct fw_driver dg00x_driver = {
 	.driver = {
 		.owner = THIS_MODULE,
-		.name = "snd-firewire-digi00x",
+		.name = KBUILD_MODNAME,
 		.bus = &fw_bus_type,
 	},
 	.probe    = snd_dg00x_probe,
diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c
index f5a016560eb8..b62a4fd22407 100644
--- a/sound/firewire/fireface/ff.c
+++ b/sound/firewire/fireface/ff.c
@@ -224,7 +224,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_ff_id_table);
 static struct fw_driver ff_driver = {
 	.driver = {
 		.owner	= THIS_MODULE,
-		.name	= "snd-fireface",
+		.name	= KBUILD_MODNAME,
 		.bus	= &fw_bus_type,
 	},
 	.probe    = snd_ff_probe,
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index 134fc9ee26b9..b1cc013a3540 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -362,7 +362,7 @@ MODULE_DEVICE_TABLE(ieee1394, efw_id_table);
 static struct fw_driver efw_driver = {
 	.driver = {
 		.owner = THIS_MODULE,
-		.name = "snd-fireworks",
+		.name = KBUILD_MODNAME,
 		.bus = &fw_bus_type,
 	},
 	.probe    = efw_probe,
diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c
index c29a97f6f638..6f38335fe10b 100644
--- a/sound/firewire/tascam/tascam-hwdep.c
+++ b/sound/firewire/tascam/tascam-hwdep.c
@@ -17,6 +17,7 @@
 
 static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf,
 				   long count, loff_t *offset)
+	__releases(&tscm->lock)
 {
 	struct snd_firewire_event_lock_status event = {
 		.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS,
@@ -36,6 +37,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf,
 
 static long tscm_hwdep_read_queue(struct snd_tscm *tscm, char __user *buf,
 				  long remained, loff_t *offset)
+	__releases(&tscm->lock)
 {
 	char __user *pos = buf;
 	unsigned int type = SNDRV_FIREWIRE_EVENT_TASCAM_CONTROL;
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index addc464503bc..5dac0d9fc58e 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -224,7 +224,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table);
 static struct fw_driver tscm_driver = {
 	.driver = {
 		.owner = THIS_MODULE,
-		.name = "snd-firewire-tascam",
+		.name = KBUILD_MODNAME,
 		.bus = &fw_bus_type,
 	},
 	.probe    = snd_tscm_probe,
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index 9a526aeef8da..e3119f5cb0d5 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -204,7 +204,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name);
  */
 int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size)
 {
-	return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n",
+	return scnprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n",
 			codec->vendor_id, codec->revision_id, codec->type);
 }
 EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias);
diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c
index e377ac93f37f..8e8257c574b0 100644
--- a/sound/isa/sb/emu8000_pcm.c
+++ b/sound/isa/sb/emu8000_pcm.c
@@ -435,7 +435,7 @@ enum {
 #define LOOP_WRITE(rec, offset, _buf, count, mode)		\
 	do {							\
 		struct snd_emu8000 *emu = (rec)->emu;		\
-		unsigned short *buf = (unsigned short *)(_buf); \
+		unsigned short *buf = (__force unsigned short *)(_buf); \
 		snd_emu8000_write_wait(emu, 1);			\
 		EMU8000_SMALW_WRITE(emu, offset);		\
 		while (count > 0) {				\
@@ -492,7 +492,7 @@ static int emu8k_pcm_silence(struct snd_pcm_substream *subs,
 #define LOOP_WRITE(rec, pos, _buf, count, mode)				\
 	do {								\
 		struct snd_emu8000 *emu = rec->emu;			\
-		unsigned short *buf = (unsigned short *)(_buf);		\
+		unsigned short *buf = (__force unsigned short *)(_buf);	\
 		snd_emu8000_write_wait(emu, 1);				\
 		EMU8000_SMALW_WRITE(emu, pos + rec->loop_start[0]);	\
 		if (rec->voices > 1)					\
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 4f524a9dbbca..4462375d2d82 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -1070,7 +1070,7 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream,
 {
 	struct snd_ali *codec = snd_pcm_substream_chip(substream);
 	struct snd_pcm_substream *s;
-	unsigned int what, whati, capture_flag;
+	unsigned int what, whati;
 	struct snd_ali_voice *pvoice, *evoice;
 	unsigned int val;
 	int do_start;
@@ -1088,7 +1088,7 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	what = whati = capture_flag = 0;
+	what = whati = 0;
 	snd_pcm_group_for_each_entry(s, substream) {
 		if ((struct snd_ali *) snd_pcm_substream_chip(s) == codec) {
 			pvoice = s->runtime->private_data;
@@ -1110,8 +1110,6 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream,
 					evoice->running = 0;
 			}
 			snd_pcm_trigger_done(s, substream);
-			if (pvoice->mode)
-				capture_flag = 1;
 		}
 	}
 	spin_lock(&codec->reg_lock);
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index a89a7e603ca8..6ff581733a19 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1789,6 +1789,7 @@ int snd_emu10k1_create(struct snd_card *card,
 	int idx, err;
 	int is_audigy;
 	size_t page_table_size;
+	__le32 *pgtbl;
 	unsigned int silent_page;
 	const struct snd_emu_chip_details *c;
 	static const struct snd_device_ops ops = {
@@ -2009,8 +2010,9 @@ int snd_emu10k1_create(struct snd_card *card,
 	/* Clear silent pages and set up pointers */
 	memset(emu->silent_page.area, 0, emu->silent_page.bytes);
 	silent_page = emu->silent_page.addr << emu->address_mode;
+	pgtbl = (__le32 *)emu->ptb_pages.area;
 	for (idx = 0; idx < (emu->address_mode ? MAXPAGES1 : MAXPAGES0); idx++)
-		((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx);
+		pgtbl[idx] = cpu_to_le32(silent_page | idx);
 
 	/* set up voice indices */
 	for (idx = 0; idx < NUM_G; idx++) {
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index bd48335d09d7..e1d3082a4fe9 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -184,6 +184,7 @@ comment "Set to Y if you want auto-loading the codec driver"
 config SND_HDA_CODEC_CA0132_DSP
 	bool "Support new DSP code for CA0132 codec"
 	depends on SND_HDA_CODEC_CA0132
+	default y
 	select SND_HDA_DSP_LOADER
 	select FW_LOADER
 	help
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 53e7732ef752..a34a2c9f4bcf 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -88,7 +88,7 @@ struct hda_conn_list {
 	struct list_head list;
 	int len;
 	hda_nid_t nid;
-	hda_nid_t conns[0];
+	hda_nid_t conns[];
 };
 
 /* look up the cached results */
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 2609e391ce54..9765652a73d7 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -373,7 +373,7 @@ static int azx_get_sync_time(ktime_t *device,
 	u32 wallclk_ctr, wallclk_cycles;
 	bool direction;
 	u32 dma_select;
-	u32 timeout = 200;
+	u32 timeout;
 	u32 retry_count = 0;
 
 	runtime = substream->runtime;
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 10223e080d59..34fe753a46fb 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2699,7 +2699,7 @@ struct dsp_image_seg {
 	u32 magic;
 	u32 chip_addr;
 	u32 count;
-	u32 data[0];
+	u32 data[];
 };
 
 static const u32 g_magic_value = 0x4c46584d;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 5119a9ae3d8a..bb287a916dae 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -154,7 +154,6 @@ struct hdmi_spec {
 	struct hda_multi_out multiout;
 	struct hda_pcm_stream pcm_playback;
 
-	bool use_jack_detect; /* jack detection enabled */
 	bool use_acomp_notifier; /* use eld_notify callback for hotplug */
 	bool acomp_registered; /* audio component registered in this driver */
 	struct drm_audio_component_audio_ops drm_audio_ops;
@@ -753,7 +752,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
  * Unsolicited events
  */
 
-static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
+static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
 
 static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid,
 				      int dev_id)
@@ -764,8 +763,7 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid,
 	if (pin_idx < 0)
 		return;
 	mutex_lock(&spec->pcm_lock);
-	if (hdmi_present_sense(get_pin(spec, pin_idx), 1))
-		snd_hda_jack_report_sync(codec);
+	hdmi_present_sense(get_pin(spec, pin_idx), 1);
 	mutex_unlock(&spec->pcm_lock);
 }
 
@@ -779,21 +777,9 @@ static void jack_callback(struct hda_codec *codec,
 	check_presence_and_report(codec, jack->nid, jack->dev_id);
 }
 
-static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
+static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res,
+				 struct hda_jack_tbl *jack)
 {
-	int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
-	struct hda_jack_tbl *jack;
-
-	if (codec->dp_mst) {
-		int dev_entry =
-			(res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;
-
-		jack = snd_hda_jack_tbl_get_from_tag(codec, tag, dev_entry);
-	} else {
-		jack = snd_hda_jack_tbl_get_from_tag(codec, tag, 0);
-	}
-	if (!jack)
-		return;
 	jack->jack_dirty = 1;
 
 	codec_dbg(codec,
@@ -853,7 +839,7 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res)
 	}
 
 	if (subtag == 0)
-		hdmi_intrinsic_event(codec, res);
+		hdmi_intrinsic_event(codec, res, jack);
 	else
 		hdmi_non_intrinsic_event(codec, res);
 }
@@ -1480,21 +1466,60 @@ static void hdmi_pcm_reset_pin(struct hdmi_spec *spec,
 	per_pin->channels = 0;
 }
 
+static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec,
+					    struct hdmi_spec_per_pin *per_pin)
+{
+	struct hdmi_spec *spec = codec->spec;
+
+	if (per_pin->pcm_idx >= 0)
+		return spec->pcm_rec[per_pin->pcm_idx].jack;
+	else
+		return NULL;
+}
+
 /* update per_pin ELD from the given new ELD;
  * setup info frame and notification accordingly
+ * also notify ELD kctl and report jack status changes
  */
-static bool update_eld(struct hda_codec *codec,
+static void update_eld(struct hda_codec *codec,
 		       struct hdmi_spec_per_pin *per_pin,
-		       struct hdmi_eld *eld)
+		       struct hdmi_eld *eld,
+		       int repoll)
 {
 	struct hdmi_eld *pin_eld = &per_pin->sink_eld;
 	struct hdmi_spec *spec = codec->spec;
+	struct snd_jack *pcm_jack;
 	bool old_eld_valid = pin_eld->eld_valid;
 	bool eld_changed;
 	int pcm_idx;
 
+	if (eld->eld_valid) {
+		if (eld->eld_size <= 0 ||
+		    snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer,
+				       eld->eld_size) < 0) {
+			eld->eld_valid = false;
+			if (repoll) {
+				schedule_delayed_work(&per_pin->work,
+						      msecs_to_jiffies(300));
+				return;
+			}
+		}
+	}
+
+	if (!eld->eld_valid || eld->eld_size <= 0) {
+		eld->eld_valid = false;
+		eld->eld_size = 0;
+	}
+
 	/* for monitor disconnection, save pcm_idx firstly */
 	pcm_idx = per_pin->pcm_idx;
+
+	/*
+	 * pcm_idx >=0 before update_eld() means it is in monitor
+	 * disconnected event. Jack must be fetched before update_eld().
+	 */
+	pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
+
 	if (spec->dyn_pcm_assign) {
 		if (eld->eld_valid) {
 			hdmi_attach_hda_pcm(spec, per_pin);
@@ -1509,6 +1534,8 @@ static bool update_eld(struct hda_codec *codec,
 	 */
 	if (pcm_idx == -1)
 		pcm_idx = per_pin->pcm_idx;
+	if (!pcm_jack)
+		pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
 
 	if (eld->eld_valid)
 		snd_hdmi_show_eld(codec, &eld->info);
@@ -1547,42 +1574,17 @@ static bool update_eld(struct hda_codec *codec,
 			       SNDRV_CTL_EVENT_MASK_VALUE |
 			       SNDRV_CTL_EVENT_MASK_INFO,
 			       &get_hdmi_pcm(spec, pcm_idx)->eld_ctl->id);
-	return eld_changed;
+
+	if (eld_changed && pcm_jack)
+		snd_jack_report(pcm_jack,
+				(eld->monitor_present && eld->eld_valid) ?
+				SND_JACK_AVOUT : 0);
 }
 
-static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec,
-					    struct hdmi_spec_per_pin *per_pin)
-{
-	struct hdmi_spec *spec = codec->spec;
-	struct snd_jack *jack = NULL;
-	struct hda_jack_tbl *jack_tbl;
-
-	/* if !dyn_pcm_assign, get jack from hda_jack_tbl
-	 * in !dyn_pcm_assign case, spec->pcm_rec[].jack is not
-	 * NULL even after snd_hda_jack_tbl_clear() is called to
-	 * free snd_jack. This may cause access invalid memory
-	 * when calling snd_jack_report
-	 */
-	if (per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) {
-		jack = spec->pcm_rec[per_pin->pcm_idx].jack;
-	} else if (!spec->dyn_pcm_assign) {
-		/*
-		 * jack tbl doesn't support DP MST
-		 * DP MST will use dyn_pcm_assign,
-		 * so DP MST will never come here
-		 */
-		jack_tbl = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid,
-						    per_pin->dev_id);
-		if (jack_tbl)
-			jack = jack_tbl->jack;
-	}
-	return jack;
-}
 /* update ELD and jack state via HD-audio verbs */
-static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
+static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
 					 int repoll)
 {
-	struct hda_jack_tbl *jack;
 	struct hda_codec *codec = per_pin->codec;
 	struct hdmi_spec *spec = codec->spec;
 	struct hdmi_eld *eld = &spec->temp_eld;
@@ -1597,9 +1599,11 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
 	 * the unsolicited response to avoid custom WARs.
 	 */
 	int present;
-	bool ret;
-	bool do_repoll = false;
-	struct snd_jack *pcm_jack = NULL;
+	int ret;
+
+	ret = snd_hda_power_up_pm(codec);
+	if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec)))
+		goto out;
 
 	present = snd_hda_jack_pin_sense(codec, pin_nid, dev_id);
 
@@ -1618,62 +1622,12 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
 		if (spec->ops.pin_get_eld(codec, pin_nid, dev_id,
 					  eld->eld_buffer, &eld->eld_size) < 0)
 			eld->eld_valid = false;
-		else {
-			if (snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer,
-						    eld->eld_size) < 0)
-				eld->eld_valid = false;
-		}
-		if (!eld->eld_valid && repoll)
-			do_repoll = true;
 	}
 
-	if (do_repoll) {
-		schedule_delayed_work(&per_pin->work, msecs_to_jiffies(300));
-	} else {
-		/*
-		 * pcm_idx >=0 before update_eld() means it is in monitor
-		 * disconnected event. Jack must be fetched before
-		 * update_eld().
-		 */
-		pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
-		update_eld(codec, per_pin, eld);
-		if (!pcm_jack)
-			pcm_jack = pin_idx_to_pcm_jack(codec, per_pin);
-	}
-
-	ret = !repoll || !eld->monitor_present || eld->eld_valid;
-
-	jack = snd_hda_jack_tbl_get_mst(codec, pin_nid, per_pin->dev_id);
-	if (jack) {
-		jack->block_report = !ret;
-		jack->pin_sense = (eld->monitor_present && eld->eld_valid) ?
-			AC_PINSENSE_PRESENCE : 0;
-
-		if (spec->dyn_pcm_assign && pcm_jack && !do_repoll) {
-			int state = 0;
-
-			if (jack->pin_sense & AC_PINSENSE_PRESENCE)
-				state = SND_JACK_AVOUT;
-			snd_jack_report(pcm_jack, state);
-		}
-
-		/*
-		 * snd_hda_jack_pin_sense() call at the beginning of this
-		 * function, updates jack->pins_sense and clears
-		 * jack->jack_dirty, therefore snd_hda_jack_report_sync() will
-		 * not override the jack->pin_sense.
-		 *
-		 * snd_hda_jack_report_sync() is superfluous for dyn_pcm_assign
-		 * case. The jack->pin_sense update was already performed, and
-		 * hda_jack->jack is NULL for dyn_pcm_assign.
-		 *
-		 * Don't call snd_hda_jack_report_sync() for
-		 * dyn_pcm_assign.
-		 */
-		ret = ret && !spec->dyn_pcm_assign;
-	}
+	update_eld(codec, per_pin, eld, repoll);
 	mutex_unlock(&per_pin->lock);
-	return ret;
+ out:
+	snd_hda_power_down_pm(codec);
 }
 
 /* update ELD and jack state via audio component */
@@ -1682,64 +1636,25 @@ static void sync_eld_via_acomp(struct hda_codec *codec,
 {
 	struct hdmi_spec *spec = codec->spec;
 	struct hdmi_eld *eld = &spec->temp_eld;
-	struct snd_jack *jack = NULL;
-	bool changed;
-	int size;
 
 	mutex_lock(&per_pin->lock);
 	eld->monitor_present = false;
-	size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid,
+	eld->eld_size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid,
 				      per_pin->dev_id, &eld->monitor_present,
 				      eld->eld_buffer, ELD_MAX_SIZE);
-	if (size > 0) {
-		size = min(size, ELD_MAX_SIZE);
-		if (snd_hdmi_parse_eld(codec, &eld->info,
-				       eld->eld_buffer, size) < 0)
-			size = -EINVAL;
-	}
-
-	if (size > 0) {
-		eld->eld_valid = true;
-		eld->eld_size = size;
-	} else {
-		eld->eld_valid = false;
-		eld->eld_size = 0;
-	}
-
-	/* pcm_idx >=0 before update_eld() means it is in monitor
-	 * disconnected event. Jack must be fetched before update_eld()
-	 */
-	jack = pin_idx_to_pcm_jack(codec, per_pin);
-	changed = update_eld(codec, per_pin, eld);
-	if (jack == NULL)
-		jack = pin_idx_to_pcm_jack(codec, per_pin);
-	if (changed && jack)
-		snd_jack_report(jack,
-				(eld->monitor_present && eld->eld_valid) ?
-				SND_JACK_AVOUT : 0);
+	eld->eld_valid = (eld->eld_size > 0);
+	update_eld(codec, per_pin, eld, 0);
 	mutex_unlock(&per_pin->lock);
 }
 
-static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
+static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 {
 	struct hda_codec *codec = per_pin->codec;
-	int ret;
 
-	/* no temporary power up/down needed for component notifier */
-	if (!codec_has_acomp(codec)) {
-		ret = snd_hda_power_up_pm(codec);
-		if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) {
-			snd_hda_power_down_pm(codec);
-			return false;
-		}
-		ret = hdmi_present_sense_via_verbs(per_pin, repoll);
-		snd_hda_power_down_pm(codec);
-	} else {
+	if (!codec_has_acomp(codec))
+		hdmi_present_sense_via_verbs(per_pin, repoll);
+	else
 		sync_eld_via_acomp(codec, per_pin);
-		ret = false; /* don't call snd_hda_jack_report_sync() */
-	}
-
-	return ret;
 }
 
 static void hdmi_repoll_eld(struct work_struct *work)
@@ -1759,8 +1674,7 @@ static void hdmi_repoll_eld(struct work_struct *work)
 		per_pin->repoll_count = 0;
 
 	mutex_lock(&spec->pcm_lock);
-	if (hdmi_present_sense(per_pin, per_pin->repoll_count))
-		snd_hda_jack_report_sync(per_pin->codec);
+	hdmi_present_sense(per_pin, per_pin->repoll_count);
 	mutex_unlock(&spec->pcm_lock);
 }
 
@@ -2206,15 +2120,23 @@ static void free_hdmi_jack_priv(struct snd_jack *jack)
 	pcm->jack = NULL;
 }
 
-static int add_hdmi_jack_kctl(struct hda_codec *codec,
-			       struct hdmi_spec *spec,
-			       int pcm_idx,
-			       const char *name)
+static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx)
 {
+	char hdmi_str[32] = "HDMI/DP";
+	struct hdmi_spec *spec = codec->spec;
+	struct hdmi_spec_per_pin *per_pin = get_pin(spec, pcm_idx);
 	struct snd_jack *jack;
+	int pcmdev = get_pcm_rec(spec, pcm_idx)->device;
 	int err;
 
-	err = snd_jack_new(codec->card, name, SND_JACK_AVOUT, &jack,
+	if (pcmdev > 0)
+		sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev);
+	if (!spec->dyn_pcm_assign &&
+	    !is_jack_detectable(codec, per_pin->pin_nid))
+		strncat(hdmi_str, " Phantom",
+			sizeof(hdmi_str) - strlen(hdmi_str) - 1);
+
+	err = snd_jack_new(codec->card, hdmi_str, SND_JACK_AVOUT, &jack,
 			   true, false);
 	if (err < 0)
 		return err;
@@ -2225,48 +2147,6 @@ static int add_hdmi_jack_kctl(struct hda_codec *codec,
 	return 0;
 }
 
-static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx)
-{
-	char hdmi_str[32] = "HDMI/DP";
-	struct hdmi_spec *spec = codec->spec;
-	struct hdmi_spec_per_pin *per_pin;
-	struct hda_jack_tbl *jack;
-	int pcmdev = get_pcm_rec(spec, pcm_idx)->device;
-	bool phantom_jack;
-	int ret;
-
-	if (pcmdev > 0)
-		sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev);
-
-	if (spec->dyn_pcm_assign)
-		return add_hdmi_jack_kctl(codec, spec, pcm_idx, hdmi_str);
-
-	/* for !dyn_pcm_assign, we still use hda_jack for compatibility */
-	/* if !dyn_pcm_assign, it must be non-MST mode.
-	 * This means pcms and pins are statically mapped.
-	 * And pcm_idx is pin_idx.
-	 */
-	per_pin = get_pin(spec, pcm_idx);
-	phantom_jack = !is_jack_detectable(codec, per_pin->pin_nid);
-	if (phantom_jack)
-		strncat(hdmi_str, " Phantom",
-			sizeof(hdmi_str) - strlen(hdmi_str) - 1);
-	ret = snd_hda_jack_add_kctl_mst(codec, per_pin->pin_nid,
-					per_pin->dev_id, hdmi_str, phantom_jack,
-					0, NULL);
-	if (ret < 0)
-		return ret;
-	jack = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid,
-					per_pin->dev_id);
-	if (jack == NULL)
-		return 0;
-	/* assign jack->jack to pcm_rec[].jack to
-	 * align with dyn_pcm_assign mode
-	 */
-	spec->pcm_rec[pcm_idx].jack = jack->jack;
-	return 0;
-}
-
 static int generic_hdmi_build_controls(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
@@ -2355,7 +2235,6 @@ static int generic_hdmi_init(struct hda_codec *codec)
 	int pin_idx;
 
 	mutex_lock(&spec->bind_lock);
-	spec->use_jack_detect = !codec->jackpoll_interval;
 	for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
 		struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
 		hda_nid_t pin_nid = per_pin->pin_nid;
@@ -2365,12 +2244,8 @@ static int generic_hdmi_init(struct hda_codec *codec)
 		hdmi_init_pin(codec, pin_nid);
 		if (codec_has_acomp(codec))
 			continue;
-		if (spec->use_jack_detect)
-			snd_hda_jack_detect_enable(codec, pin_nid, dev_id);
-		else
-			snd_hda_jack_detect_enable_callback_mst(codec, pin_nid,
-								dev_id,
-								jack_callback);
+		snd_hda_jack_detect_enable_callback_mst(codec, pin_nid, dev_id,
+							jack_callback);
 	}
 	mutex_unlock(&spec->bind_lock);
 	return 0;
@@ -2532,12 +2407,6 @@ static void reprogram_jack_detect(struct hda_codec *codec, hda_nid_t nid,
 		unsigned int val = use_acomp ? 0 : (AC_USRSP_EN | tbl->tag);
 		snd_hda_codec_write_cache(codec, nid, 0,
 					  AC_VERB_SET_UNSOLICITED_ENABLE, val);
-	} else {
-		/* if no jack entry was defined beforehand, create a new one
-		 * at need (i.e. only when notifier is cleared)
-		 */
-		if (!use_acomp)
-			snd_hda_jack_detect_enable(codec, nid, dev_id);
 	}
 }
 
@@ -2553,13 +2422,11 @@ static void generic_acomp_notifier_set(struct drm_audio_component *acomp,
 	spec->use_acomp_notifier = use_acomp;
 	spec->codec->relaxed_resume = use_acomp;
 	/* reprogram each jack detection logic depending on the notifier */
-	if (spec->use_jack_detect) {
-		for (i = 0; i < spec->num_pins; i++)
-			reprogram_jack_detect(spec->codec,
-					      get_pin(spec, i)->pin_nid,
-					      get_pin(spec, i)->dev_id,
-					      use_acomp);
-	}
+	for (i = 0; i < spec->num_pins; i++)
+		reprogram_jack_detect(spec->codec,
+				      get_pin(spec, i)->pin_nid,
+				      get_pin(spec, i)->dev_id,
+				      use_acomp);
 	mutex_unlock(&spec->bind_lock);
 }
 
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 21ab9cc50c71..65a887b217ee 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -30,7 +30,7 @@
 #if K1212_DEBUG_LEVEL > 0
 #define K1212_DEBUG_PRINTK(fmt,args...)	printk(KERN_DEBUG fmt,##args)
 #else
-#define K1212_DEBUG_PRINTK(fmt,...)
+#define K1212_DEBUG_PRINTK(fmt,...)	do { } while (0)
 #endif
 #if K1212_DEBUG_LEVEL > 1
 #define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...)	printk(KERN_DEBUG fmt,##args)
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index cc06f0a1a7e4..227aece17e39 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -3353,7 +3353,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 				return;
 			}
 		} else {
-			int err = -EINVAL;
+			int err;
+
 			err = hdsp_request_fw_loader(hdsp);
 			if (err < 0) {
 				snd_iprintf(buffer,
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 799789c8eea9..8b03e2dc503f 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -414,6 +414,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
 {
 	unsigned int i, idx, ofs, rest;
 	struct via82xx *chip = snd_pcm_substream_chip(substream);
+	__le32 *pgtbl;
 
 	if (dev->table.area == NULL) {
 		/* the start of each lists must be aligned to 8 bytes,
@@ -435,6 +436,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
 	/* fill the entries */
 	idx = 0;
 	ofs = 0;
+	pgtbl = (__le32 *)dev->table.area;
 	for (i = 0; i < periods; i++) {
 		rest = fragsize;
 		/* fill descriptors for a period.
@@ -451,7 +453,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
 				return -EINVAL;
 			}
 			addr = snd_pcm_sgbuf_get_addr(substream, ofs);
-			((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr);
+			pgtbl[idx << 1] = cpu_to_le32(addr);
 			r = snd_pcm_sgbuf_get_chunk_size(substream, ofs, rest);
 			rest -= r;
 			if (! rest) {
@@ -466,7 +468,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
 				"tbl %d: at %d  size %d (rest %d)\n",
 				idx, ofs, r, rest);
 			*/
-			((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
+			pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag);
 			dev->idx_table[idx].offset = ofs;
 			dev->idx_table[idx].size = r;
 			ofs += r;
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 84e589803e2e..607b7100db1c 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -267,6 +267,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
 {
 	unsigned int i, idx, ofs, rest;
 	struct via82xx_modem *chip = snd_pcm_substream_chip(substream);
+	__le32 *pgtbl;
 
 	if (dev->table.area == NULL) {
 		/* the start of each lists must be aligned to 8 bytes,
@@ -288,6 +289,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
 	/* fill the entries */
 	idx = 0;
 	ofs = 0;
+	pgtbl = (__le32 *)dev->table.area;
 	for (i = 0; i < periods; i++) {
 		rest = fragsize;
 		/* fill descriptors for a period.
@@ -304,7 +306,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
 				return -EINVAL;
 			}
 			addr = snd_pcm_sgbuf_get_addr(substream, ofs);
-			((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr);
+			pgtbl[idx << 1] = cpu_to_le32(addr);
 			r = PAGE_SIZE - (ofs % PAGE_SIZE);
 			if (rest < r)
 				r = rest;
@@ -321,7 +323,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
 				"tbl %d: at %d  size %d (rest %d)\n",
 				idx, ofs, r, rest);
 			*/
-			((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
+			pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag);
 			dev->idx_table[idx].offset = ofs;
 			dev->idx_table[idx].size = r;
 			ofs += r;
diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c
index 093806d735c6..9554a0c506af 100644
--- a/sound/ppc/keywest.c
+++ b/sound/ppc/keywest.c
@@ -40,6 +40,7 @@ static int keywest_probe(struct i2c_client *client,
 static int keywest_attach_adapter(struct i2c_adapter *adapter)
 {
 	struct i2c_board_info info;
+	struct i2c_client *client;
 
 	if (! keywest_ctx)
 		return -EINVAL;
@@ -50,9 +51,11 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter)
 	memset(&info, 0, sizeof(struct i2c_board_info));
 	strlcpy(info.type, "keywest", I2C_NAME_SIZE);
 	info.addr = keywest_ctx->addr;
-	keywest_ctx->client = i2c_new_device(adapter, &info);
-	if (!keywest_ctx->client)
-		return -ENODEV;
+	client = i2c_new_client_device(adapter, &info);
+	if (IS_ERR(client))
+		return PTR_ERR(client);
+	keywest_ctx->client = client;
+
 	/*
 	 * We know the driver is already loaded, so the device should be
 	 * already bound. If not it means binding failed, and then there
diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index 78edd7d2f418..56031026b113 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -13,6 +13,7 @@ snd-usb-audio-objs := 	card.o \
 			mixer_scarlett.o \
 			mixer_scarlett_gen2.o \
 			mixer_us16x08.o \
+			mixer_s1810c.o \
 			pcm.o \
 			power.o \
 			proc.o \
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 827fb0bc8b56..fd6fd1726ea0 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -72,6 +72,7 @@ static int device_setup[SNDRV_CARDS]; /* device parameter for this card */
 static bool ignore_ctl_error;
 static bool autoclock = true;
 static char *quirk_alias[SNDRV_CARDS];
+static char *delayed_register[SNDRV_CARDS];
 
 bool snd_usb_use_vmalloc = true;
 bool snd_usb_skip_validation;
@@ -95,6 +96,8 @@ module_param(autoclock, bool, 0444);
 MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes).");
 module_param_array(quirk_alias, charp, NULL, 0444);
 MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef.");
+module_param_array(delayed_register, charp, NULL, 0444);
+MODULE_PARM_DESC(delayed_register, "Quirk for delayed registration, given by id:iface, e.g. 0123abcd:4.");
 module_param_named(use_vmalloc, snd_usb_use_vmalloc, bool, 0444);
 MODULE_PARM_DESC(use_vmalloc, "Use vmalloc for PCM intermediate buffers (default: yes).");
 module_param_named(skip_validation, snd_usb_skip_validation, bool, 0444);
@@ -525,6 +528,21 @@ static bool get_alias_id(struct usb_device *dev, unsigned int *id)
 	return false;
 }
 
+static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface)
+{
+	int i;
+	unsigned int id, inum;
+
+	for (i = 0; i < ARRAY_SIZE(delayed_register); i++) {
+		if (delayed_register[i] &&
+		    sscanf(delayed_register[i], "%x:%x", &id, &inum) == 2 &&
+		    id == chip->usb_id)
+			return inum != iface;
+	}
+
+	return false;
+}
+
 static const struct usb_device_id usb_audio_ids[]; /* defined below */
 
 /* look for the corresponding quirk */
@@ -662,10 +680,22 @@ static int usb_audio_probe(struct usb_interface *intf,
 			goto __error;
 	}
 
-	/* we are allowed to call snd_card_register() many times */
-	err = snd_card_register(chip->card);
-	if (err < 0)
-		goto __error;
+	if (chip->need_delayed_register) {
+		dev_info(&dev->dev,
+			 "Found post-registration device assignment: %08x:%02x\n",
+			 chip->usb_id, ifnum);
+		chip->need_delayed_register = false; /* clear again */
+	}
+
+	/* we are allowed to call snd_card_register() many times, but first
+	 * check to see if a device needs to skip it or do anything special
+	 */
+	if (!snd_usb_registration_quirk(chip, ifnum) &&
+	    !check_delayed_register_option(chip, ifnum)) {
+		err = snd_card_register(chip->card);
+		if (err < 0)
+			goto __error;
+	}
 
 	if (quirk && quirk->shares_media_device) {
 		/* don't want to fail when snd_media_device_create() fails */
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index a48313dfa967..b118cf97607f 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -151,16 +151,15 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i
 	return ret;
 }
 
-/*
- * Assume the clock is valid if clock source supports only one single sample
- * rate, the terminal is connected directly to it (there is no clock selector)
- * and clock type is internal. This is to deal with some Denon DJ controllers
- * that always reports that clock is invalid.
- */
 static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip,
 					    struct audioformat *fmt,
 					    int source_id)
 {
+	bool ret = false;
+	int count;
+	unsigned char data;
+	struct usb_device *dev = chip->dev;
+
 	if (fmt->protocol == UAC_VERSION_2) {
 		struct uac_clock_source_descriptor *cs_desc =
 			snd_usb_find_clock_source(chip->ctrl_intf, source_id);
@@ -168,13 +167,51 @@ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip,
 		if (!cs_desc)
 			return false;
 
-		return (fmt->nr_rates == 1 &&
-			(fmt->clock & 0xff) == cs_desc->bClockID &&
-			(cs_desc->bmAttributes & 0x3) !=
-				UAC_CLOCK_SOURCE_TYPE_EXT);
+		/*
+		 * Assume the clock is valid if clock source supports only one
+		 * single sample rate, the terminal is connected directly to it
+		 * (there is no clock selector) and clock type is internal.
+		 * This is to deal with some Denon DJ controllers that always
+		 * reports that clock is invalid.
+		 */
+		if (fmt->nr_rates == 1 &&
+		    (fmt->clock & 0xff) == cs_desc->bClockID &&
+		    (cs_desc->bmAttributes & 0x3) !=
+				UAC_CLOCK_SOURCE_TYPE_EXT)
+			return true;
 	}
 
-	return false;
+	/*
+	 * MOTU MicroBook IIc
+	 * Sample rate changes takes more than 2 seconds for this device. Clock
+	 * validity request returns false during that period.
+	 */
+	if (chip->usb_id == USB_ID(0x07fd, 0x0004)) {
+		count = 0;
+
+		while ((!ret) && (count < 50)) {
+			int err;
+
+			msleep(100);
+
+			err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+					      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+					      UAC2_CS_CONTROL_CLOCK_VALID << 8,
+					      snd_usb_ctrl_intf(chip) | (source_id << 8),
+					      &data, sizeof(data));
+			if (err < 0) {
+				dev_warn(&dev->dev,
+					 "%s(): cannot get clock validity for id %d\n",
+					   __func__, source_id);
+				return false;
+			}
+
+			ret = !!data;
+			count++;
+		}
+	}
+
+	return ret;
 }
 
 static bool uac_clock_source_is_valid(struct snd_usb_audio *chip,
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 9f5cb4ed3a0c..50e1874c847c 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -247,6 +247,36 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
 	return 0;
 }
 
+
+/*
+ * Presonus Studio 1810c supports a limited set of sampling
+ * rates per altsetting but reports the full set each time.
+ * If we don't filter out the unsupported rates and attempt
+ * to configure the card, it will hang refusing to do any
+ * further audio I/O until a hard reset is performed.
+ *
+ * The list of supported rates per altsetting (set of available
+ * I/O channels) is described in the owner's manual, section 2.2.
+ */
+static bool s1810c_valid_sample_rate(struct audioformat *fp,
+				     unsigned int rate)
+{
+	switch (fp->altsetting) {
+	case 1:
+		/* All ADAT ports available */
+		return rate <= 48000;
+	case 2:
+		/* Half of ADAT ports available */
+		return (rate == 88200 || rate == 96000);
+	case 3:
+		/* Analog I/O only (no S/PDIF nor ADAT) */
+		return rate >= 176400;
+	default:
+		return false;
+	}
+	return false;
+}
+
 /*
  * Helper function to walk the array of sample rate triplets reported by
  * the device. The problem is that we need to parse whole array first to
@@ -283,6 +313,12 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip,
 		}
 
 		for (rate = min; rate <= max; rate += res) {
+
+			/* Filter out invalid rates on Presonus Studio 1810c */
+			if (chip->usb_id == USB_ID(0x0194f, 0x010c) &&
+			    !s1810c_valid_sample_rate(fp, rate))
+				goto skip_rate;
+
 			if (fp->rate_table)
 				fp->rate_table[nr_rates] = rate;
 			if (!fp->rate_min || rate < fp->rate_min)
@@ -297,6 +333,7 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip,
 				break;
 			}
 
+skip_rate:
 			/* avoid endless loop */
 			if (res == 0)
 				break;
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 392e5fda680c..be5c597ed40c 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -91,7 +91,7 @@ struct usb_ms_endpoint_descriptor {
 	__u8  bDescriptorType;
 	__u8  bDescriptorSubtype;
 	__u8  bNumEmbMIDIJack;
-	__u8  baAssocJackID[0];
+	__u8  baAssocJackID[];
 } __attribute__ ((packed));
 
 struct snd_usb_midi_in_endpoint;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 81b2db0edd5f..721d12130d0c 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -292,6 +292,11 @@ static int uac2_ctl_value_size(int val_type)
  * retrieve a mixer value
  */
 
+static inline int mixer_ctrl_intf(struct usb_mixer_interface *mixer)
+{
+	return get_iface_desc(mixer->hostif)->bInterfaceNumber;
+}
+
 static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request,
 			    int validx, int *value_ret)
 {
@@ -306,7 +311,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request,
 		return -EIO;
 
 	while (timeout-- > 0) {
-		idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+		idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
 		err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request,
 				      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
 				      validx, idx, buf, val_len);
@@ -354,7 +359,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request,
 	if (ret)
 		goto error;
 
-	idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+	idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
 	ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest,
 			      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
 			      validx, idx, buf, size);
@@ -479,7 +484,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval,
 		return -EIO;
 
 	while (timeout-- > 0) {
-		idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+		idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
 		err = snd_usb_ctl_msg(chip->dev,
 				      usb_sndctrlpipe(chip->dev, 0), request,
 				      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
@@ -901,6 +906,12 @@ static int parse_term_effect_unit(struct mixer_build *state,
 				  struct usb_audio_term *term,
 				  void *p1, int id)
 {
+	struct uac2_effect_unit_descriptor *d = p1;
+	int err;
+
+	err = __check_input_term(state, d->bSourceID, term);
+	if (err < 0)
+		return err;
 	term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */
 	term->id = id;
 	return 0;
@@ -1203,7 +1214,7 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval,
 		    get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) {
 			usb_audio_err(cval->head.mixer->chip,
 				      "%d:%d: cannot get min/max values for control %d (id %d)\n",
-				   cval->head.id, snd_usb_ctrl_intf(cval->head.mixer->chip),
+				   cval->head.id, mixer_ctrl_intf(cval->head.mixer),
 							       cval->control, cval->head.id);
 			return -EINVAL;
 		}
@@ -1422,7 +1433,7 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol,
 	if (ret)
 		goto error;
 
-	idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8);
+	idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8);
 	if (cval->head.mixer->protocol == UAC_VERSION_2) {
 		struct uac2_connectors_ctl_blk uac2_conn;
 
@@ -1674,6 +1685,16 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer,
 	/* get min/max values */
 	get_min_max_with_quirks(cval, 0, kctl);
 
+	/* skip a bogus volume range */
+	if (cval->max <= cval->min) {
+		usb_audio_dbg(mixer->chip,
+			      "[%d] FU [%s] skipped due to invalid volume\n",
+			      cval->head.id, kctl->id.name);
+		snd_ctl_free_one(kctl);
+		return;
+	}
+
+
 	if (control == UAC_FU_VOLUME) {
 		check_mapped_dB(map, cval);
 		if (cval->dBmin < cval->dBmax || !cval->initialized) {
@@ -3203,7 +3224,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry,
 	list_for_each_entry(mixer, &chip->mixer_list, list) {
 		snd_iprintf(buffer,
 			"USB Mixer: usb_id=0x%08x, ctrlif=%i, ctlerr=%i\n",
-				chip->usb_id, snd_usb_ctrl_intf(chip),
+				chip->usb_id, mixer_ctrl_intf(mixer),
 				mixer->ignore_ctl_error);
 		snd_iprintf(buffer, "Card: %s\n", chip->card->longname);
 		for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) {
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index c237e24f08d9..02b036b2aefb 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -34,6 +34,7 @@
 #include "mixer_scarlett.h"
 #include "mixer_scarlett_gen2.h"
 #include "mixer_us16x08.h"
+#include "mixer_s1810c.h"
 #include "helper.h"
 
 struct std_mono_table {
@@ -2277,6 +2278,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 	case USB_ID(0x2a39, 0x3fd4): /* RME */
 		err = snd_rme_controls_create(mixer);
 		break;
+
+	case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */
+		err = snd_sc1810_init_mixer(mixer);
+		break;
 	}
 
 	return err;
diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c
new file mode 100644
index 000000000000..6483e47bafd0
--- /dev/null
+++ b/sound/usb/mixer_s1810c.c
@@ -0,0 +1,595 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Presonus Studio 1810c driver for ALSA
+ * Copyright (C) 2019 Nick Kossifidis <mickflemm@gmail.com>
+ *
+ * Based on reverse engineering of the communication protocol
+ * between the windows driver / Univeral Control (UC) program
+ * and the device, through usbmon.
+ *
+ * For now this bypasses the mixer, with all channels split,
+ * so that the software can mix with greater flexibility.
+ * It also adds controls for the 4 buttons on the front of
+ * the device.
+ */
+
+#include <linux/usb.h>
+#include <linux/usb/audio-v2.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+
+#include "usbaudio.h"
+#include "mixer.h"
+#include "mixer_quirks.h"
+#include "helper.h"
+#include "mixer_s1810c.h"
+
+#define SC1810C_CMD_REQ	160
+#define SC1810C_CMD_REQTYPE \
+	(USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT)
+#define SC1810C_CMD_F1		0x50617269
+#define SC1810C_CMD_F2		0x14
+
+/*
+ * DISCLAIMER: These are just guesses based on the
+ * dumps I got.
+ *
+ * It seems like a selects between
+ * device (0), mixer (0x64) and output (0x65)
+ *
+ * For mixer (0x64):
+ *  * b selects an input channel (see below).
+ *  * c selects an output channel pair (see below).
+ *  * d selects left (0) or right (1) of that pair.
+ *  * e 0-> disconnect, 0x01000000-> connect,
+ *	0x0109-> used for stereo-linking channels,
+ *	e is also used for setting volume levels
+ *	in which case b is also set so I guess
+ *	this way it is possible to set the volume
+ *	level from the specified input to the
+ *	specified output.
+ *
+ * IN Channels:
+ * 0  - 7  Mic/Inst/Line (Analog inputs)
+ * 8  - 9  S/PDIF
+ * 10 - 17 ADAT
+ * 18 - 35 DAW (Inputs from the host)
+ *
+ * OUT Channels (pairs):
+ * 0 -> Main out
+ * 1 -> Line1/2
+ * 2 -> Line3/4
+ * 3 -> S/PDIF
+ * 4 -> ADAT?
+ *
+ * For device (0):
+ *  * b and c are not used, at least not on the
+ *    dumps I got.
+ *  * d sets the control id to be modified
+ *    (see below).
+ *  * e sets the setting for that control.
+ *    (so for the switches I was interested
+ *    in it's 0/1)
+ *
+ * For output (0x65):
+ *   * b is the output channel (see above).
+ *   * c is zero.
+ *   * e I guess the same as with mixer except 0x0109
+ *	 which I didn't see in my dumps.
+ *
+ * The two fixed fields have the same values for
+ * mixer and output but a different set for device.
+ */
+struct s1810c_ctl_packet {
+	u32 a;
+	u32 b;
+	u32 fixed1;
+	u32 fixed2;
+	u32 c;
+	u32 d;
+	u32 e;
+};
+
+#define SC1810C_CTL_LINE_SW	0
+#define SC1810C_CTL_MUTE_SW	1
+#define SC1810C_CTL_AB_SW	3
+#define SC1810C_CTL_48V_SW	4
+
+#define SC1810C_SET_STATE_REQ	161
+#define SC1810C_SET_STATE_REQTYPE SC1810C_CMD_REQTYPE
+#define SC1810C_SET_STATE_F1	0x64656D73
+#define SC1810C_SET_STATE_F2	0xF4
+
+#define SC1810C_GET_STATE_REQ	162
+#define SC1810C_GET_STATE_REQTYPE \
+	(USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN)
+#define SC1810C_GET_STATE_F1	SC1810C_SET_STATE_F1
+#define SC1810C_GET_STATE_F2	SC1810C_SET_STATE_F2
+
+#define SC1810C_STATE_F1_IDX	2
+#define SC1810C_STATE_F2_IDX	3
+
+/*
+ * This packet includes mixer volumes and
+ * various other fields, it's an extended
+ * version of ctl_packet, with a and b
+ * being zero and different f1/f2.
+ */
+struct s1810c_state_packet {
+	u32 fields[63];
+};
+
+#define SC1810C_STATE_48V_SW	58
+#define SC1810C_STATE_LINE_SW	59
+#define SC1810C_STATE_MUTE_SW	60
+#define SC1810C_STATE_AB_SW	62
+
+struct s1810_mixer_state {
+	uint16_t seqnum;
+	struct mutex usb_mutex;
+	struct mutex data_mutex;
+};
+
+static int
+snd_s1810c_send_ctl_packet(struct usb_device *dev, u32 a,
+			   u32 b, u32 c, u32 d, u32 e)
+{
+	struct s1810c_ctl_packet pkt = { 0 };
+	int ret = 0;
+
+	pkt.fixed1 = SC1810C_CMD_F1;
+	pkt.fixed2 = SC1810C_CMD_F2;
+
+	pkt.a = a;
+	pkt.b = b;
+	pkt.c = c;
+	pkt.d = d;
+	/*
+	 * Value for settings 0/1 for this
+	 * output channel is always 0 (probably because
+	 * there is no ADAT output on 1810c)
+	 */
+	pkt.e = (c == 4) ? 0 : e;
+
+	ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
+			      SC1810C_CMD_REQ,
+			      SC1810C_CMD_REQTYPE, 0, 0, &pkt, sizeof(pkt));
+	if (ret < 0) {
+		dev_warn(&dev->dev, "could not send ctl packet\n");
+		return ret;
+	}
+	return 0;
+}
+
+/*
+ * When opening Universal Control the program periodicaly
+ * sends and receives state packets for syncinc state between
+ * the device and the host.
+ *
+ * Note that if we send only the request to get data back we'll
+ * get an error, we need to first send an empty state packet and
+ * then ask to receive a filled. Their seqnumbers must also match.
+ */
+static int
+snd_sc1810c_get_status_field(struct usb_device *dev,
+			     u32 *field, int field_idx, uint16_t *seqnum)
+{
+	struct s1810c_state_packet pkt_out = { { 0 } };
+	struct s1810c_state_packet pkt_in = { { 0 } };
+	int ret = 0;
+
+	pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1;
+	pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2;
+	ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+			      SC1810C_SET_STATE_REQ,
+			      SC1810C_SET_STATE_REQTYPE,
+			      (*seqnum), 0, &pkt_out, sizeof(pkt_out));
+	if (ret < 0) {
+		dev_warn(&dev->dev, "could not send state packet (%d)\n", ret);
+		return ret;
+	}
+
+	ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+			      SC1810C_GET_STATE_REQ,
+			      SC1810C_GET_STATE_REQTYPE,
+			      (*seqnum), 0, &pkt_in, sizeof(pkt_in));
+	if (ret < 0) {
+		dev_warn(&dev->dev, "could not get state field %u (%d)\n",
+			 field_idx, ret);
+		return ret;
+	}
+
+	(*field) = pkt_in.fields[field_idx];
+	(*seqnum)++;
+	return 0;
+}
+
+/*
+ * This is what I got when bypassing the mixer with
+ * all channels split. I'm not 100% sure of what's going
+ * on, I could probably clean this up based on my observations
+ * but I prefer to keep the same behavior as the windows driver.
+ */
+static int snd_s1810c_init_mixer_maps(struct snd_usb_audio *chip)
+{
+	u32 a, b, c, e, n, off;
+	struct usb_device *dev = chip->dev;
+
+	/* Set initial volume levels ? */
+	a = 0x64;
+	e = 0xbc;
+	for (n = 0; n < 2; n++) {
+		off = n * 18;
+		for (b = off, c = 0; b < 18 + off; b++) {
+			/* This channel to all outputs ? */
+			for (c = 0; c <= 8; c++) {
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+			}
+			/* This channel to main output (again) */
+			snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e);
+			snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e);
+		}
+		/*
+		 * I noticed on UC that DAW channels have different
+		 * initial volumes, so this makes sense.
+		 */
+		e = 0xb53bf0;
+	}
+
+	/* Connect analog outputs ? */
+	a = 0x65;
+	e = 0x01000000;
+	for (b = 1; b < 3; b++) {
+		snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e);
+		snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e);
+	}
+	snd_s1810c_send_ctl_packet(dev, a, 0, 0, 0, e);
+	snd_s1810c_send_ctl_packet(dev, a, 0, 0, 1, e);
+
+	/* Set initial volume levels for S/PDIF mappings ? */
+	a = 0x64;
+	e = 0xbc;
+	c = 3;
+	for (n = 0; n < 2; n++) {
+		off = n * 18;
+		for (b = off; b < 18 + off; b++) {
+			snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+			snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+		}
+		e = 0xb53bf0;
+	}
+
+	/* Connect S/PDIF output ? */
+	a = 0x65;
+	e = 0x01000000;
+	snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e);
+	snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e);
+
+	/* Connect all outputs (again) ? */
+	a = 0x65;
+	e = 0x01000000;
+	for (b = 0; b < 4; b++) {
+		snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e);
+		snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e);
+	}
+
+	/* Basic routing to get sound out of the device */
+	a = 0x64;
+	e = 0x01000000;
+	for (c = 0; c < 4; c++) {
+		for (b = 0; b < 36; b++) {
+			if ((c == 0 && b == 18) ||	/* DAW1/2 -> Main */
+			    (c == 1 && b == 20) ||	/* DAW3/4 -> Line3/4 */
+			    (c == 2 && b == 22) ||	/* DAW4/5 -> Line5/6 */
+			    (c == 3 && b == 24)) {	/* DAW5/6 -> S/PDIF */
+				/* Left */
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0);
+				b++;
+				/* Right */
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0);
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+			} else {
+				/* Leave the rest disconnected */
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0);
+				snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0);
+			}
+		}
+	}
+
+	/* Set initial volume levels for S/PDIF (again) ? */
+	a = 0x64;
+	e = 0xbc;
+	c = 3;
+	for (n = 0; n < 2; n++) {
+		off = n * 18;
+		for (b = off; b < 18 + off; b++) {
+			snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e);
+			snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e);
+		}
+		e = 0xb53bf0;
+	}
+
+	/* Connect S/PDIF outputs (again) ? */
+	a = 0x65;
+	e = 0x01000000;
+	snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e);
+	snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e);
+
+	/* Again ? */
+	snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e);
+	snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e);
+
+	return 0;
+}
+
+/*
+ * Sync state with the device and retrieve the requested field,
+ * whose index is specified in (kctl->private_value & 0xFF),
+ * from the received fields array.
+ */
+static int
+snd_s1810c_get_switch_state(struct usb_mixer_interface *mixer,
+			    struct snd_kcontrol *kctl, u32 *state)
+{
+	struct snd_usb_audio *chip = mixer->chip;
+	struct s1810_mixer_state *private = mixer->private_data;
+	u32 field = 0;
+	u32 ctl_idx = (u32) (kctl->private_value & 0xFF);
+	int ret = 0;
+
+	mutex_lock(&private->usb_mutex);
+	ret = snd_sc1810c_get_status_field(chip->dev, &field,
+					   ctl_idx, &private->seqnum);
+	if (ret < 0)
+		goto unlock;
+
+	*state = field;
+ unlock:
+	mutex_unlock(&private->usb_mutex);
+	return ret ? ret : 0;
+}
+
+/*
+ * Send a control packet to the device for the control id
+ * specified in (kctl->private_value >> 8) with value
+ * specified in (kctl->private_value >> 16).
+ */
+static int
+snd_s1810c_set_switch_state(struct usb_mixer_interface *mixer,
+			    struct snd_kcontrol *kctl)
+{
+	struct snd_usb_audio *chip = mixer->chip;
+	struct s1810_mixer_state *private = mixer->private_data;
+	u32 pval = (u32) kctl->private_value;
+	u32 ctl_id = (pval >> 8) & 0xFF;
+	u32 ctl_val = (pval >> 16) & 0x1;
+	int ret = 0;
+
+	mutex_lock(&private->usb_mutex);
+	ret = snd_s1810c_send_ctl_packet(chip->dev, 0, 0, 0, ctl_id, ctl_val);
+	mutex_unlock(&private->usb_mutex);
+	return ret;
+}
+
+/* Generic get/set/init functions for switch controls */
+
+static int
+snd_s1810c_switch_get(struct snd_kcontrol *kctl,
+		      struct snd_ctl_elem_value *ctl_elem)
+{
+	struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl);
+	struct usb_mixer_interface *mixer = list->mixer;
+	struct s1810_mixer_state *private = mixer->private_data;
+	u32 pval = (u32) kctl->private_value;
+	u32 ctl_idx = pval & 0xFF;
+	u32 state = 0;
+	int ret = 0;
+
+	mutex_lock(&private->data_mutex);
+	ret = snd_s1810c_get_switch_state(mixer, kctl, &state);
+	if (ret < 0)
+		goto unlock;
+
+	switch (ctl_idx) {
+	case SC1810C_STATE_LINE_SW:
+	case SC1810C_STATE_AB_SW:
+		ctl_elem->value.enumerated.item[0] = (int)state;
+		break;
+	default:
+		ctl_elem->value.integer.value[0] = (long)state;
+	}
+
+ unlock:
+	mutex_unlock(&private->data_mutex);
+	return (ret < 0) ? ret : 0;
+}
+
+static int
+snd_s1810c_switch_set(struct snd_kcontrol *kctl,
+		      struct snd_ctl_elem_value *ctl_elem)
+{
+	struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl);
+	struct usb_mixer_interface *mixer = list->mixer;
+	struct s1810_mixer_state *private = mixer->private_data;
+	u32 pval = (u32) kctl->private_value;
+	u32 ctl_idx = pval & 0xFF;
+	u32 curval = 0;
+	u32 newval = 0;
+	int ret = 0;
+
+	mutex_lock(&private->data_mutex);
+	ret = snd_s1810c_get_switch_state(mixer, kctl, &curval);
+	if (ret < 0)
+		goto unlock;
+
+	switch (ctl_idx) {
+	case SC1810C_STATE_LINE_SW:
+	case SC1810C_STATE_AB_SW:
+		newval = (u32) ctl_elem->value.enumerated.item[0];
+		break;
+	default:
+		newval = (u32) ctl_elem->value.integer.value[0];
+	}
+
+	if (curval == newval)
+		goto unlock;
+
+	kctl->private_value &= ~(0x1 << 16);
+	kctl->private_value |= (unsigned int)(newval & 0x1) << 16;
+	ret = snd_s1810c_set_switch_state(mixer, kctl);
+
+ unlock:
+	mutex_unlock(&private->data_mutex);
+	return (ret < 0) ? 0 : 1;
+}
+
+static int
+snd_s1810c_switch_init(struct usb_mixer_interface *mixer,
+		       const struct snd_kcontrol_new *new_kctl)
+{
+	struct snd_kcontrol *kctl;
+	struct usb_mixer_elem_info *elem;
+
+	elem = kzalloc(sizeof(struct usb_mixer_elem_info), GFP_KERNEL);
+	if (!elem)
+		return -ENOMEM;
+
+	elem->head.mixer = mixer;
+	elem->control = 0;
+	elem->head.id = 0;
+	elem->channels = 1;
+
+	kctl = snd_ctl_new1(new_kctl, elem);
+	if (!kctl) {
+		kfree(elem);
+		return -ENOMEM;
+	}
+	kctl->private_free = snd_usb_mixer_elem_free;
+
+	return snd_usb_mixer_add_control(&elem->head, kctl);
+}
+
+static int
+snd_s1810c_line_sw_info(struct snd_kcontrol *kctl,
+			struct snd_ctl_elem_info *uinfo)
+{
+	static const char *const texts[2] = {
+		"Preamp On (Mic/Inst)",
+		"Preamp Off (Line in)"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
+}
+
+static const struct snd_kcontrol_new snd_s1810c_line_sw = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Line 1/2 Source Type",
+	.info = snd_s1810c_line_sw_info,
+	.get = snd_s1810c_switch_get,
+	.put = snd_s1810c_switch_set,
+	.private_value = (SC1810C_STATE_LINE_SW | SC1810C_CTL_LINE_SW << 8)
+};
+
+static const struct snd_kcontrol_new snd_s1810c_mute_sw = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Mute Main Out Switch",
+	.info = snd_ctl_boolean_mono_info,
+	.get = snd_s1810c_switch_get,
+	.put = snd_s1810c_switch_set,
+	.private_value = (SC1810C_STATE_MUTE_SW | SC1810C_CTL_MUTE_SW << 8)
+};
+
+static const struct snd_kcontrol_new snd_s1810c_48v_sw = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "48V Phantom Power On Mic Inputs Switch",
+	.info = snd_ctl_boolean_mono_info,
+	.get = snd_s1810c_switch_get,
+	.put = snd_s1810c_switch_set,
+	.private_value = (SC1810C_STATE_48V_SW | SC1810C_CTL_48V_SW << 8)
+};
+
+static int
+snd_s1810c_ab_sw_info(struct snd_kcontrol *kctl,
+		      struct snd_ctl_elem_info *uinfo)
+{
+	static const char *const texts[2] = {
+		"1/2",
+		"3/4"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts);
+}
+
+static const struct snd_kcontrol_new snd_s1810c_ab_sw = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Headphone 1 Source Route",
+	.info = snd_s1810c_ab_sw_info,
+	.get = snd_s1810c_switch_get,
+	.put = snd_s1810c_switch_set,
+	.private_value = (SC1810C_STATE_AB_SW | SC1810C_CTL_AB_SW << 8)
+};
+
+static void snd_sc1810_mixer_state_free(struct usb_mixer_interface *mixer)
+{
+	struct s1810_mixer_state *private = mixer->private_data;
+	kfree(private);
+	mixer->private_data = NULL;
+}
+
+/* Entry point, called from mixer_quirks.c */
+int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer)
+{
+	struct s1810_mixer_state *private = NULL;
+	struct snd_usb_audio *chip = mixer->chip;
+	struct usb_device *dev = chip->dev;
+	int ret = 0;
+
+	/* Run this only once */
+	if (!list_empty(&chip->mixer_list))
+		return 0;
+
+	dev_info(&dev->dev,
+		 "Presonus Studio 1810c, device_setup: %u\n", chip->setup);
+	if (chip->setup == 1)
+		dev_info(&dev->dev, "(8out/18in @ 48KHz)\n");
+	else if (chip->setup == 2)
+		dev_info(&dev->dev, "(6out/8in @ 192KHz)\n");
+	else
+		dev_info(&dev->dev, "(8out/14in @ 96KHz)\n");
+
+	ret = snd_s1810c_init_mixer_maps(chip);
+	if (ret < 0)
+		return ret;
+
+	private = kzalloc(sizeof(struct s1810_mixer_state), GFP_KERNEL);
+	if (!private)
+		return -ENOMEM;
+
+	mutex_init(&private->usb_mutex);
+	mutex_init(&private->data_mutex);
+
+	mixer->private_data = private;
+	mixer->private_free = snd_sc1810_mixer_state_free;
+
+	private->seqnum = 1;
+
+	ret = snd_s1810c_switch_init(mixer, &snd_s1810c_line_sw);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_s1810c_switch_init(mixer, &snd_s1810c_mute_sw);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_s1810c_switch_init(mixer, &snd_s1810c_48v_sw);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw);
+	if (ret < 0)
+		return ret;
+	return ret;
+}
diff --git a/sound/usb/mixer_s1810c.h b/sound/usb/mixer_s1810c.h
new file mode 100644
index 000000000000..a79a3743cff3
--- /dev/null
+++ b/sound/usb/mixer_s1810c.h
@@ -0,0 +1,7 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Presonus Studio 1810c driver for ALSA
+ * Copyright (C) 2019 Nick Kossifidis <mickflemm@gmail.com>
+ */
+
+int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index bd258f1ec2dd..a4e4064f9aee 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -357,7 +357,12 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 		ep = 0x81;
 		ifnum = 1;
 		goto add_sync_ep_from_ifnum;
-	case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */
+	case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II/IIc */
+		/* MicroBook IIc */
+		if (altsd->bInterfaceClass == USB_CLASS_AUDIO)
+			return 0;
+
+		/* MicroBook II */
 		ep = 0x84;
 		ifnum = 0;
 		goto add_sync_ep_from_ifnum;
diff --git a/sound/usb/proc.c b/sound/usb/proc.c
index ffbf4bd9208c..4174ad11fca6 100644
--- a/sound/usb/proc.c
+++ b/sound/usb/proc.c
@@ -70,7 +70,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
 		snd_iprintf(buffer, "  Interface %d\n", fp->iface);
 		snd_iprintf(buffer, "    Altset %d\n", fp->altsetting);
 		snd_iprintf(buffer, "    Format:");
-		for (fmt = 0; fmt <= SNDRV_PCM_FORMAT_LAST; ++fmt)
+		pcm_for_each_format(fmt)
 			if (fp->formats & pcm_format_to_bits(fmt))
 				snd_iprintf(buffer, " %s",
 					    snd_pcm_format_name(fmt));
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index d187aa6d50db..1c8719292eee 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3472,7 +3472,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 },
 /* MOTU Microbook II */
 {
-	USB_DEVICE(0x07fd, 0x0004),
+	USB_DEVICE_VENDOR_SPEC(0x07fd, 0x0004),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "MOTU",
 		.product_name = "MicroBookII",
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 7f558f4b4520..86f192a3043d 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1252,6 +1252,38 @@ static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip,
 	return 0; /* keep this altsetting */
 }
 
+static int s1810c_skip_setting_quirk(struct snd_usb_audio *chip,
+					   int iface, int altno)
+{
+	/*
+	 * Altno settings:
+	 *
+	 * Playback (Interface 1):
+	 * 1: 6 Analog + 2 S/PDIF
+	 * 2: 6 Analog + 2 S/PDIF
+	 * 3: 6 Analog
+	 *
+	 * Capture (Interface 2):
+	 * 1: 8 Analog + 2 S/PDIF + 8 ADAT
+	 * 2: 8 Analog + 2 S/PDIF + 4 ADAT
+	 * 3: 8 Analog
+	 */
+
+	/*
+	 * I'll leave 2 as the default one and
+	 * use device_setup to switch to the
+	 * other two.
+	 */
+	if ((chip->setup == 0 || chip->setup > 2) && altno != 2)
+		return 1;
+	else if (chip->setup == 1 && altno != 1)
+		return 1;
+	else if (chip->setup == 2 && altno != 3)
+		return 1;
+
+	return 0;
+}
+
 int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip,
 				  int iface,
 				  int altno)
@@ -1265,6 +1297,10 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip,
 	/* fasttrackpro usb: skip altsets incompatible with device_setup */
 	if (chip->usb_id == USB_ID(0x0763, 0x2012))
 		return fasttrackpro_skip_setting_quirk(chip, iface, altno);
+	/* presonus studio 1810c: skip altsets incompatible with device_setup */
+	if (chip->usb_id == USB_ID(0x0194f, 0x010c))
+		return s1810c_skip_setting_quirk(chip, iface, altno);
+
 
 	return 0;
 }
@@ -1316,7 +1352,15 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
 	case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */
 		return snd_usb_axefx3_boot_quirk(dev);
 	case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */
-		return snd_usb_motu_microbookii_boot_quirk(dev);
+		/*
+		 * For some reason interface 3 with vendor-spec class is
+		 * detected on MicroBook IIc.
+		 */
+		if (get_iface_desc(intf->altsetting)->bInterfaceClass ==
+		    USB_CLASS_VENDOR_SPEC &&
+		    get_iface_desc(intf->altsetting)->bInterfaceNumber < 3)
+			return snd_usb_motu_microbookii_boot_quirk(dev);
+		break;
 	}
 
 	return 0;
@@ -1754,5 +1798,47 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip,
 		else
 			fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC;
 		break;
+	case USB_ID(0x07fd, 0x0004):  /* MOTU MicroBook IIc */
+		/*
+		 * MaxPacketsOnly attribute is erroneously set in endpoint
+		 * descriptors. As a result this card produces noise with
+		 * all sample rates other than 96 KHz.
+		 */
+		fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX;
+		break;
 	}
 }
+
+/*
+ * registration quirk:
+ * the registration is skipped if a device matches with the given ID,
+ * unless the interface reaches to the defined one.  This is for delaying
+ * the registration until the last known interface, so that the card and
+ * devices appear at the same time.
+ */
+
+struct registration_quirk {
+	unsigned int usb_id;	/* composed via USB_ID() */
+	unsigned int interface;	/* the interface to trigger register */
+};
+
+#define REG_QUIRK_ENTRY(vendor, product, iface) \
+	{ .usb_id = USB_ID(vendor, product), .interface = (iface) }
+
+static const struct registration_quirk registration_quirks[] = {
+	REG_QUIRK_ENTRY(0x0951, 0x16d8, 2),	/* Kingston HyperX AMP */
+	{ 0 }					/* terminator */
+};
+
+/* return true if skipping registration */
+bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface)
+{
+	const struct registration_quirk *q;
+
+	for (q = registration_quirks; q->usb_id; q++)
+		if (chip->usb_id == q->usb_id)
+			return iface != q->interface;
+
+	/* Register as normal */
+	return false;
+}
diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h
index df0355843a4c..c76cf24a640a 100644
--- a/sound/usb/quirks.h
+++ b/sound/usb/quirks.h
@@ -51,4 +51,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip,
 					  struct audioformat *fp,
 					  int stream);
 
+bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface);
+
 #endif /* __USBAUDIO_QUIRKS_H */
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index afd5aa574611..15296f2c902c 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -502,6 +502,9 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip,
 		subs = &as->substream[stream];
 		if (subs->ep_num)
 			continue;
+		if (snd_device_get_state(chip->card, as->pcm) !=
+		    SNDRV_DEV_BUILD)
+			chip->need_delayed_register = true;
 		err = snd_pcm_new_stream(as->pcm, stream, 1);
 		if (err < 0)
 			return err;
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 6fe3ab582ec6..1c892c7f14d7 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -34,6 +34,7 @@ struct snd_usb_audio {
 	unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
 	unsigned int tx_length_quirk:1; /* Put length specifier in transfers */
 	unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */
+	unsigned int need_delayed_register:1; /* warn for delayed registration */
 	int num_interfaces;
 	int num_suspended_intf;
 	int sample_rate_read_error;