Merge tag 'v4.4.121' into toradex_vf_4.4-nextColibri-VF_LXDE-Image_2.8b2.97-20180331

This is the 4.4.121 stable release

27 files changed, 366 insertions, 120 deletions

diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 33e72c809e50..494b7b533366 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -465,7 +465,6 @@ static int snd_pcm_hw_param_near(struct snd_pcm_substream *pcm,
 		v = snd_pcm_hw_param_last(pcm, params, var, dir);
 	else
 		v = snd_pcm_hw_param_first(pcm, params, var, dir);
-	snd_BUG_ON(v < 0);
 	return v;
 }
 
@@ -1370,8 +1369,11 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha
 
 	if ((tmp = snd_pcm_oss_make_ready(substream)) < 0)
 		return tmp;
-	mutex_lock(&runtime->oss.params_lock);
 	while (bytes > 0) {
+		if (mutex_lock_interruptible(&runtime->oss.params_lock)) {
+			tmp = -ERESTARTSYS;
+			break;
+		}
 		if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) {
 			tmp = bytes;
 			if (tmp + runtime->oss.buffer_used > runtime->oss.period_bytes)
@@ -1415,14 +1417,18 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha
 			xfer += tmp;
 			if ((substream->f_flags & O_NONBLOCK) != 0 &&
 			    tmp != runtime->oss.period_bytes)
-				break;
+				tmp = -EAGAIN;
 		}
-	}
-	mutex_unlock(&runtime->oss.params_lock);
-	return xfer;
-
  err:
-	mutex_unlock(&runtime->oss.params_lock);
+		mutex_unlock(&runtime->oss.params_lock);
+		if (tmp < 0)
+			break;
+		if (signal_pending(current)) {
+			tmp = -ERESTARTSYS;
+			break;
+		}
+		tmp = 0;
+	}
 	return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp;
 }
 
@@ -1470,8 +1476,11 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use
 
 	if ((tmp = snd_pcm_oss_make_ready(substream)) < 0)
 		return tmp;
-	mutex_lock(&runtime->oss.params_lock);
 	while (bytes > 0) {
+		if (mutex_lock_interruptible(&runtime->oss.params_lock)) {
+			tmp = -ERESTARTSYS;
+			break;
+		}
 		if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) {
 			if (runtime->oss.buffer_used == 0) {
 				tmp = snd_pcm_oss_read2(substream, runtime->oss.buffer, runtime->oss.period_bytes, 1);
@@ -1502,12 +1511,16 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use
 			bytes -= tmp;
 			xfer += tmp;
 		}
-	}
-	mutex_unlock(&runtime->oss.params_lock);
-	return xfer;
-
  err:
-	mutex_unlock(&runtime->oss.params_lock);
+		mutex_unlock(&runtime->oss.params_lock);
+		if (tmp < 0)
+			break;
+		if (signal_pending(current)) {
+			tmp = -ERESTARTSYS;
+			break;
+		}
+		tmp = 0;
+	}
 	return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp;
 }
 
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 727ac44d39f4..a84a1d3d23e5 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -591,18 +591,26 @@ snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *plug, st
 	snd_pcm_sframes_t frames = size;
 
 	plugin = snd_pcm_plug_first(plug);
-	while (plugin && frames > 0) {
+	while (plugin) {
+		if (frames <= 0)
+			return frames;
 		if ((next = plugin->next) != NULL) {
 			snd_pcm_sframes_t frames1 = frames;
-			if (plugin->dst_frames)
+			if (plugin->dst_frames) {
 				frames1 = plugin->dst_frames(plugin, frames);
+				if (frames1 <= 0)
+					return frames1;
+			}
 			if ((err = next->client_channels(next, frames1, &dst_channels)) < 0) {
 				return err;
 			}
 			if (err != frames1) {
 				frames = err;
-				if (plugin->src_frames)
+				if (plugin->src_frames) {
 					frames = plugin->src_frames(plugin, frames1);
+					if (frames <= 0)
+						return frames;
+				}
 			}
 		} else
 			dst_channels = NULL;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index cd20f91326fe..4c145d6bccd4 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -578,7 +578,6 @@ static inline unsigned int muldiv32(unsigned int a, unsigned int b,
 {
 	u_int64_t n = (u_int64_t) a * b;
 	if (c == 0) {
-		snd_BUG_ON(!n);
 		*r = 0;
 		return UINT_MAX;
 	}
@@ -1664,7 +1663,7 @@ int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm,
 		return changed;
 	if (params->rmask) {
 		int err = snd_pcm_hw_refine(pcm, params);
-		if (snd_BUG_ON(err < 0))
+		if (err < 0)
 			return err;
 	}
 	return snd_pcm_hw_param_value(params, var, dir);
@@ -1711,7 +1710,7 @@ int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm,
 		return changed;
 	if (params->rmask) {
 		int err = snd_pcm_hw_refine(pcm, params);
-		if (snd_BUG_ON(err < 0))
+		if (err < 0)
 			return err;
 	}
 	return snd_pcm_hw_param_value(params, var, dir);
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index b450a27588c8..16f8124b1150 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -579,15 +579,14 @@ static int snd_rawmidi_info_user(struct snd_rawmidi_substream *substream,
 	return 0;
 }
 
-int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info)
+static int __snd_rawmidi_info_select(struct snd_card *card,
+				     struct snd_rawmidi_info *info)
 {
 	struct snd_rawmidi *rmidi;
 	struct snd_rawmidi_str *pstr;
 	struct snd_rawmidi_substream *substream;
 
-	mutex_lock(&register_mutex);
 	rmidi = snd_rawmidi_search(card, info->device);
-	mutex_unlock(&register_mutex);
 	if (!rmidi)
 		return -ENXIO;
 	if (info->stream < 0 || info->stream > 1)
@@ -603,6 +602,16 @@ int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info
 	}
 	return -ENXIO;
 }
+
+int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info)
+{
+	int ret;
+
+	mutex_lock(&register_mutex);
+	ret = __snd_rawmidi_info_select(card, info);
+	mutex_unlock(&register_mutex);
+	return ret;
+}
 EXPORT_SYMBOL(snd_rawmidi_info_select);
 
 static int snd_rawmidi_info_select_user(struct snd_card *card,
diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c
index b36de76f24e2..167b943469ab 100644
--- a/sound/core/seq/seq_clientmgr.c
+++ b/sound/core/seq/seq_clientmgr.c
@@ -236,6 +236,7 @@ static struct snd_seq_client *seq_create_client1(int client_index, int poolsize)
 	rwlock_init(&client->ports_lock);
 	mutex_init(&client->ports_mutex);
 	INIT_LIST_HEAD(&client->ports_list_head);
+	mutex_init(&client->ioctl_mutex);
 
 	/* find free slot in the client table */
 	spin_lock_irqsave(&clients_lock, flags);
@@ -1011,7 +1012,7 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf,
 {
 	struct snd_seq_client *client = file->private_data;
 	int written = 0, len;
-	int err = -EINVAL;
+	int err;
 	struct snd_seq_event event;
 
 	if (!(snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_OUTPUT))
@@ -1026,11 +1027,15 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf,
 
 	/* allocate the pool now if the pool is not allocated yet */ 
 	if (client->pool->size > 0 && !snd_seq_write_pool_allocated(client)) {
-		if (snd_seq_pool_init(client->pool) < 0)
+		mutex_lock(&client->ioctl_mutex);
+		err = snd_seq_pool_init(client->pool);
+		mutex_unlock(&client->ioctl_mutex);
+		if (err < 0)
 			return -ENOMEM;
 	}
 
 	/* only process whole events */
+	err = -EINVAL;
 	while (count >= sizeof(struct snd_seq_event)) {
 		/* Read in the event header from the user */
 		len = sizeof(event);
@@ -2220,11 +2225,15 @@ static int snd_seq_do_ioctl(struct snd_seq_client *client, unsigned int cmd,
 static long snd_seq_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 {
 	struct snd_seq_client *client = file->private_data;
+	long ret;
 
 	if (snd_BUG_ON(!client))
 		return -ENXIO;
 		
-	return snd_seq_do_ioctl(client, cmd, (void __user *) arg);
+	mutex_lock(&client->ioctl_mutex);
+	ret = snd_seq_do_ioctl(client, cmd, (void __user *) arg);
+	mutex_unlock(&client->ioctl_mutex);
+	return ret;
 }
 
 #ifdef CONFIG_COMPAT
diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h
index 20f0a725ec7d..91f8f165bfdc 100644
--- a/sound/core/seq/seq_clientmgr.h
+++ b/sound/core/seq/seq_clientmgr.h
@@ -59,6 +59,7 @@ struct snd_seq_client {
 	struct list_head ports_list_head;
 	rwlock_t ports_lock;
 	struct mutex ports_mutex;
+	struct mutex ioctl_mutex;
 	int convert32;		/* convert 32->64bit */
 
 	/* output pool */
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 54f348a4fb78..cbd20cb8ca11 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -39,6 +39,7 @@
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/pcm.h>
+#include <sound/pcm_params.h>
 #include <sound/info.h>
 #include <sound/initval.h>
 
@@ -305,19 +306,6 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd)
 	return 0;
 }
 
-static void params_change_substream(struct loopback_pcm *dpcm,
-				    struct snd_pcm_runtime *runtime)
-{
-	struct snd_pcm_runtime *dst_runtime;
-
-	if (dpcm == NULL || dpcm->substream == NULL)
-		return;
-	dst_runtime = dpcm->substream->runtime;
-	if (dst_runtime == NULL)
-		return;
-	dst_runtime->hw = dpcm->cable->hw;
-}
-
 static void params_change(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
@@ -329,10 +317,6 @@ static void params_change(struct snd_pcm_substream *substream)
 	cable->hw.rate_max = runtime->rate;
 	cable->hw.channels_min = runtime->channels;
 	cable->hw.channels_max = runtime->channels;
-	params_change_substream(cable->streams[SNDRV_PCM_STREAM_PLAYBACK],
-				runtime);
-	params_change_substream(cable->streams[SNDRV_PCM_STREAM_CAPTURE],
-				runtime);
 }
 
 static int loopback_prepare(struct snd_pcm_substream *substream)
@@ -620,26 +604,29 @@ static unsigned int get_cable_index(struct snd_pcm_substream *substream)
 static int rule_format(struct snd_pcm_hw_params *params,
 		       struct snd_pcm_hw_rule *rule)
 {
+	struct loopback_pcm *dpcm = rule->private;
+	struct loopback_cable *cable = dpcm->cable;
+	struct snd_mask m;
 
-	struct snd_pcm_hardware *hw = rule->private;
-	struct snd_mask *maskp = hw_param_mask(params, rule->var);
-
-	maskp->bits[0] &= (u_int32_t)hw->formats;
-	maskp->bits[1] &= (u_int32_t)(hw->formats >> 32);
-	memset(maskp->bits + 2, 0, (SNDRV_MASK_MAX-64) / 8); /* clear rest */
-	if (! maskp->bits[0] && ! maskp->bits[1])
-		return -EINVAL;
-	return 0;
+	snd_mask_none(&m);
+	mutex_lock(&dpcm->loopback->cable_lock);
+	m.bits[0] = (u_int32_t)cable->hw.formats;
+	m.bits[1] = (u_int32_t)(cable->hw.formats >> 32);
+	mutex_unlock(&dpcm->loopback->cable_lock);
+	return snd_mask_refine(hw_param_mask(params, rule->var), &m);
 }
 
 static int rule_rate(struct snd_pcm_hw_params *params,
 		     struct snd_pcm_hw_rule *rule)
 {
-	struct snd_pcm_hardware *hw = rule->private;
+	struct loopback_pcm *dpcm = rule->private;
+	struct loopback_cable *cable = dpcm->cable;
 	struct snd_interval t;
 
-        t.min = hw->rate_min;
-        t.max = hw->rate_max;
+	mutex_lock(&dpcm->loopback->cable_lock);
+	t.min = cable->hw.rate_min;
+	t.max = cable->hw.rate_max;
+	mutex_unlock(&dpcm->loopback->cable_lock);
         t.openmin = t.openmax = 0;
         t.integer = 0;
 	return snd_interval_refine(hw_param_interval(params, rule->var), &t);
@@ -648,22 +635,44 @@ static int rule_rate(struct snd_pcm_hw_params *params,
 static int rule_channels(struct snd_pcm_hw_params *params,
 			 struct snd_pcm_hw_rule *rule)
 {
-	struct snd_pcm_hardware *hw = rule->private;
+	struct loopback_pcm *dpcm = rule->private;
+	struct loopback_cable *cable = dpcm->cable;
 	struct snd_interval t;
 
-        t.min = hw->channels_min;
-        t.max = hw->channels_max;
+	mutex_lock(&dpcm->loopback->cable_lock);
+	t.min = cable->hw.channels_min;
+	t.max = cable->hw.channels_max;
+	mutex_unlock(&dpcm->loopback->cable_lock);
         t.openmin = t.openmax = 0;
         t.integer = 0;
 	return snd_interval_refine(hw_param_interval(params, rule->var), &t);
 }
 
+static void free_cable(struct snd_pcm_substream *substream)
+{
+	struct loopback *loopback = substream->private_data;
+	int dev = get_cable_index(substream);
+	struct loopback_cable *cable;
+
+	cable = loopback->cables[substream->number][dev];
+	if (!cable)
+		return;
+	if (cable->streams[!substream->stream]) {
+		/* other stream is still alive */
+		cable->streams[substream->stream] = NULL;
+	} else {
+		/* free the cable */
+		loopback->cables[substream->number][dev] = NULL;
+		kfree(cable);
+	}
+}
+
 static int loopback_open(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct loopback *loopback = substream->private_data;
 	struct loopback_pcm *dpcm;
-	struct loopback_cable *cable;
+	struct loopback_cable *cable = NULL;
 	int err = 0;
 	int dev = get_cable_index(substream);
 
@@ -682,7 +691,6 @@ static int loopback_open(struct snd_pcm_substream *substream)
 	if (!cable) {
 		cable = kzalloc(sizeof(*cable), GFP_KERNEL);
 		if (!cable) {
-			kfree(dpcm);
 			err = -ENOMEM;
 			goto unlock;
 		}
@@ -700,19 +708,19 @@ static int loopback_open(struct snd_pcm_substream *substream)
 	/* are cached -> they do not reflect the actual state */
 	err = snd_pcm_hw_rule_add(runtime, 0,
 				  SNDRV_PCM_HW_PARAM_FORMAT,
-				  rule_format, &runtime->hw,
+				  rule_format, dpcm,
 				  SNDRV_PCM_HW_PARAM_FORMAT, -1);
 	if (err < 0)
 		goto unlock;
 	err = snd_pcm_hw_rule_add(runtime, 0,
 				  SNDRV_PCM_HW_PARAM_RATE,
-				  rule_rate, &runtime->hw,
+				  rule_rate, dpcm,
 				  SNDRV_PCM_HW_PARAM_RATE, -1);
 	if (err < 0)
 		goto unlock;
 	err = snd_pcm_hw_rule_add(runtime, 0,
 				  SNDRV_PCM_HW_PARAM_CHANNELS,
-				  rule_channels, &runtime->hw,
+				  rule_channels, dpcm,
 				  SNDRV_PCM_HW_PARAM_CHANNELS, -1);
 	if (err < 0)
 		goto unlock;
@@ -724,6 +732,10 @@ static int loopback_open(struct snd_pcm_substream *substream)
 	else
 		runtime->hw = cable->hw;
  unlock:
+	if (err < 0) {
+		free_cable(substream);
+		kfree(dpcm);
+	}
 	mutex_unlock(&loopback->cable_lock);
 	return err;
 }
@@ -732,20 +744,10 @@ static int loopback_close(struct snd_pcm_substream *substream)
 {
 	struct loopback *loopback = substream->private_data;
 	struct loopback_pcm *dpcm = substream->runtime->private_data;
-	struct loopback_cable *cable;
-	int dev = get_cable_index(substream);
 
 	loopback_timer_stop(dpcm);
 	mutex_lock(&loopback->cable_lock);
-	cable = loopback->cables[substream->number][dev];
-	if (cable->streams[!substream->stream]) {
-		/* other stream is still alive */
-		cable->streams[substream->stream] = NULL;
-	} else {
-		/* free the cable */
-		loopback->cables[substream->number][dev] = NULL;
-		kfree(cable);
-	}
+	free_cable(substream);
 	mutex_unlock(&loopback->cable_lock);
 	return 0;
 }
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 8fef1b8d1fd8..bd7bcf428bcf 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -183,7 +183,7 @@ static int hdac_component_master_match(struct device *dev, void *data)
  */
 int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops)
 {
-	if (WARN_ON(!hdac_acomp))
+	if (!hdac_acomp)
 		return -ENODEV;
 
 	hdac_acomp->audio_ops = aops;
@@ -240,7 +240,8 @@ out_master_del:
 out_err:
 	kfree(acomp);
 	bus->audio_component = NULL;
-	dev_err(dev, "failed to add i915 component master (%d)\n", ret);
+	hdac_acomp = NULL;
+	dev_info(dev, "failed to add i915 component master (%d)\n", ret);
 
 	return ret;
 }
@@ -273,6 +274,7 @@ int snd_hdac_i915_exit(struct hdac_bus *bus)
 
 	kfree(acomp);
 	bus->audio_component = NULL;
+	hdac_acomp = NULL;
 
 	return 0;
 }
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e2e08fc73b50..e2212830df0c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -179,7 +179,7 @@ static const struct kernel_param_ops param_ops_xint = {
 };
 #define param_check_xint param_check_int
 
-static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
+static int power_save = -1;
 module_param(power_save, xint, 0644);
 MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
 		 "(in second, 0 = disable).");
@@ -2055,6 +2055,24 @@ out_free:
 	return err;
 }
 
+#ifdef CONFIG_PM
+/* On some boards setting power_save to a non 0 value leads to clicking /
+ * popping sounds when ever we enter/leave powersaving mode. Ideally we would
+ * figure out how to avoid these sounds, but that is not always feasible.
+ * So we keep a list of devices where we disable powersaving as its known
+ * to causes problems on these devices.
+ */
+static struct snd_pci_quirk power_save_blacklist[] = {
+	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
+	SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0),
+	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
+	SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0),
+	/* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */
+	SND_PCI_QUIRK(0x17aa, 0x2227, "Lenovo X1 Carbon 3rd Gen", 0),
+	{}
+};
+#endif /* CONFIG_PM */
+
 /* number of codec slots for each chipset: 0 = default slots (i.e. 4) */
 static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = {
 	[AZX_DRIVER_NVIDIA] = 8,
@@ -2067,6 +2085,7 @@ static int azx_probe_continue(struct azx *chip)
 	struct hdac_bus *bus = azx_bus(chip);
 	struct pci_dev *pci = chip->pci;
 	int dev = chip->dev_index;
+	int val;
 	int err;
 
 	hda->probe_continued = 1;
@@ -2088,9 +2107,11 @@ static int azx_probe_continue(struct azx *chip)
 			 * for other chips, still continue probing as other
 			 * codecs can be on the same link.
 			 */
-			if (CONTROLLER_IN_GPU(pci))
+			if (CONTROLLER_IN_GPU(pci)) {
+				dev_err(chip->card->dev,
+					"HSW/BDW HD-audio HDMI/DP requires binding with gfx driver\n");
 				goto out_free;
-			else
+			} else
 				goto skip_i915;
 		}
 
@@ -2140,7 +2161,22 @@ static int azx_probe_continue(struct azx *chip)
 
 	chip->running = 1;
 	azx_add_card_list(chip);
-	snd_hda_set_power_save(&chip->bus, power_save * 1000);
+
+	val = power_save;
+#ifdef CONFIG_PM
+	if (val == -1) {
+		const struct snd_pci_quirk *q;
+
+		val = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
+		q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist);
+		if (q && val) {
+			dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n",
+				 q->subvendor, q->subdevice);
+			val = 0;
+		}
+	}
+#endif /* CONFIG_PM */
+	snd_hda_set_power_save(&chip->bus, val * 1000);
 	if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo)
 		pm_runtime_put_noidle(&pci->dev);
 
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index c146d0de53d8..29e1ce2263bc 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1482,6 +1482,9 @@ static int dspio_scp(struct hda_codec *codec,
 		} else if (ret_size != reply_data_size) {
 			codec_dbg(codec, "RetLen and HdrLen .NE.\n");
 			return -EINVAL;
+		} else if (!reply) {
+			codec_dbg(codec, "NULL reply\n");
+			return -EINVAL;
 		} else {
 			*reply_len = ret_size*sizeof(unsigned int);
 			memcpy(reply, scp_reply.data, *reply_len);
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 80bbadc83721..d6e079f4ec09 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -408,6 +408,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = {
 	/*SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),*/
 
 	/* codec SSID */
+	SND_PCI_QUIRK(0x106b, 0x0600, "iMac 14,1", CS420X_IMAC27_122),
 	SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81),
 	SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122),
 	SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101),
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index ac5de4365e15..c92b7ba344ef 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -261,6 +261,7 @@ enum {
 	CXT_FIXUP_HP_530,
 	CXT_FIXUP_CAP_MIX_AMP_5047,
 	CXT_FIXUP_MUTE_LED_EAPD,
+	CXT_FIXUP_HP_DOCK,
 	CXT_FIXUP_HP_SPECTRE,
 	CXT_FIXUP_HP_GATE_MIC,
 };
@@ -778,6 +779,14 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt_fixup_mute_led_eapd,
 	},
+	[CXT_FIXUP_HP_DOCK] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x16, 0x21011020 }, /* line-out */
+			{ 0x18, 0x2181103f }, /* line-in */
+			{ }
+		}
+	},
 	[CXT_FIXUP_HP_SPECTRE] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -839,6 +848,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC),
+	SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE),
 	SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC),
 	SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
@@ -872,6 +882,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = {
 	{ .id = CXT_PINCFG_LEMOTE_A1205, .name = "lemote-a1205" },
 	{ .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" },
 	{ .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" },
+	{ .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" },
 	{}
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e5730a7d0480..b302d056e5d3 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3130,6 +3130,19 @@ static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec,
 		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 }
 
+static void alc269_fixup_pincfg_U7x7_headset_mic(struct hda_codec *codec,
+						 const struct hda_fixup *fix,
+						 int action)
+{
+	unsigned int cfg_headphone = snd_hda_codec_get_pincfg(codec, 0x21);
+	unsigned int cfg_headset_mic = snd_hda_codec_get_pincfg(codec, 0x19);
+
+	if (cfg_headphone && cfg_headset_mic == 0x411111f0)
+		snd_hda_codec_set_pincfg(codec, 0x19,
+			(cfg_headphone & ~AC_DEFCFG_DEVICE) |
+			(AC_JACK_MIC_IN << AC_DEFCFG_DEVICE_SHIFT));
+}
+
 static void alc269_fixup_hweq(struct hda_codec *codec,
 			       const struct hda_fixup *fix, int action)
 {
@@ -4782,6 +4795,7 @@ enum {
 	ALC269_FIXUP_LIFEBOOK_EXTMIC,
 	ALC269_FIXUP_LIFEBOOK_HP_PIN,
 	ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT,
+	ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC,
 	ALC269_FIXUP_AMIC,
 	ALC269_FIXUP_DMIC,
 	ALC269VB_FIXUP_AMIC,
@@ -4839,6 +4853,7 @@ enum {
 	ALC286_FIXUP_HP_GPIO_LED,
 	ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY,
 	ALC280_FIXUP_HP_DOCK_PINS,
+	ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED,
 	ALC280_FIXUP_HP_9480M,
 	ALC288_FIXUP_DELL_HEADSET_MODE,
 	ALC288_FIXUP_DELL1_MIC_NO_PRESENCE,
@@ -4971,6 +4986,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc269_fixup_pincfg_no_hp_to_lineout,
 	},
+	[ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc269_fixup_pincfg_U7x7_headset_mic,
+	},
 	[ALC269_FIXUP_AMIC] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -5377,6 +5396,16 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC280_FIXUP_HP_GPIO4
 	},
+	[ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1b, 0x21011020 }, /* line-out */
+			{ 0x18, 0x2181103f }, /* line-in */
+			{ },
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HP_GPIO_MIC1_LED
+	},
 	[ALC280_FIXUP_HP_9480M] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc280_fixup_hp_9480m,
@@ -5589,6 +5618,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
 	SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME),
 	SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER),
+	SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
 	SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -5629,7 +5659,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2256, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
 	SND_PCI_QUIRK(0x103c, 0x2257, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
 	SND_PCI_QUIRK(0x103c, 0x2259, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
-	SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+	SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED),
 	SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
@@ -5675,6 +5705,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x10cf, 0x159f, "Lifebook E780", ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT),
 	SND_PCI_QUIRK(0x10cf, 0x15dc, "Lifebook T731", ALC269_FIXUP_LIFEBOOK_HP_PIN),
 	SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN),
+	SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC),
 	SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC),
 	SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_BXBT2807_MIC),
@@ -5794,6 +5825,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, .name = "headset-mode-no-hp-mic"},
 	{.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"},
 	{.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"},
+	{.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic1-led"},
 	{.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"},
 	{.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"},
 	{.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"},
@@ -5942,6 +5974,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x1b, 0x01011020},
 		{0x21, 0x02211010}),
 	SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+		{0x12, 0x90a60130},
+		{0x14, 0x90170110},
+		{0x1b, 0x01011020},
+		{0x21, 0x0221101f}),
+	SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
 		{0x12, 0x90a60160},
 		{0x14, 0x90170120},
 		{0x21, 0x02211030}),
@@ -5958,6 +5995,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x14, 0x90170110},
 		{0x21, 0x02211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+		{0x12, 0x90a60130},
+		{0x14, 0x90170110},
+		{0x14, 0x01011020},
+		{0x21, 0x0221101f}),
+	SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
 		ALC256_STANDARD_PINS),
 	SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
 		{0x12, 0x90a60130},
@@ -6013,6 +6055,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x12, 0x90a60120},
 		{0x14, 0x90170110},
 		{0x21, 0x0321101f}),
+	SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
+		{0x12, 0xb7a60130},
+		{0x14, 0x90170110},
+		{0x21, 0x04211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1,
 		ALC290_STANDARD_PINS,
 		{0x15, 0x04211040},
diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c
index 712ed6598c48..ebdf9bd5a64c 100644
--- a/sound/soc/codecs/pcm512x-spi.c
+++ b/sound/soc/codecs/pcm512x-spi.c
@@ -70,3 +70,7 @@ static struct spi_driver pcm512x_spi_driver = {
 };
 
 module_spi_driver(pcm512x_spi_driver);
+
+MODULE_DESCRIPTION("ASoC PCM512x codec driver - SPI");
+MODULE_AUTHOR("Mark Brown <broonie@kernel.org>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index a5a4e9f75c57..a06395507225 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -232,7 +232,7 @@ static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_codec *codec)
 	struct twl4030_codec_data *pdata = dev_get_platdata(codec->dev);
 	struct device_node *twl4030_codec_node = NULL;
 
-	twl4030_codec_node = of_find_node_by_name(codec->dev->parent->of_node,
+	twl4030_codec_node = of_get_child_by_name(codec->dev->parent->of_node,
 						  "codec");
 
 	if (!pdata && twl4030_codec_node) {
@@ -241,9 +241,11 @@ static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_codec *codec)
 				     GFP_KERNEL);
 		if (!pdata) {
 			dev_err(codec->dev, "Can not allocate memory\n");
+			of_node_put(twl4030_codec_node);
 			return NULL;
 		}
 		twl4030_setup_pdata_of(pdata, twl4030_codec_node);
+		of_node_put(twl4030_codec_node);
 	}
 
 	return pdata;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 95d2392303eb..7ca67613e0d4 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1408,12 +1408,6 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 				sizeof(fsl_ssi_ac97_dai));
 
 		fsl_ac97_data = ssi_private;
-
-		ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev);
-		if (ret) {
-			dev_err(&pdev->dev, "could not set AC'97 ops\n");
-			return ret;
-		}
 	} else {
 		/* Initialize this copy of the CPU DAI driver structure */
 		memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template,
@@ -1473,6 +1467,14 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 			return ret;
 	}
 
+	if (fsl_ssi_is_ac97(ssi_private)) {
+		ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev);
+		if (ret) {
+			dev_err(&pdev->dev, "could not set AC'97 ops\n");
+			goto error_ac97_ops;
+		}
+	}
+
 	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_ssi_component,
 					      &ssi_private->cpu_dai_drv, 1);
 	if (ret) {
@@ -1556,6 +1558,10 @@ error_sound_card:
 	fsl_ssi_debugfs_remove(&ssi_private->dbg_stats);
 
 error_asoc_register:
+	if (fsl_ssi_is_ac97(ssi_private))
+		snd_soc_set_ac97_ops(NULL);
+
+error_ac97_ops:
 	if (ssi_private->soc->imx)
 		fsl_ssi_imx_clean(pdev, ssi_private);
 
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index ff6fcd9f92f7..0b1b6fcb7500 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -343,13 +343,19 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
 	snprintf(prop, sizeof(prop), "%scpu", prefix);
 	cpu = of_get_child_by_name(node, prop);
 
+	if (!cpu) {
+		ret = -EINVAL;
+		dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop);
+		goto dai_link_of_err;
+	}
+
 	snprintf(prop, sizeof(prop), "%splat", prefix);
 	plat = of_get_child_by_name(node, prop);
 
 	snprintf(prop, sizeof(prop), "%scodec", prefix);
 	codec = of_get_child_by_name(node, prop);
 
-	if (!cpu || !codec) {
+	if (!codec) {
 		ret = -EINVAL;
 		dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop);
 		goto dai_link_of_err;
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index d430ef5a4f38..79c29330c56a 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -24,7 +24,6 @@ config SND_SST_IPC_PCI
 config SND_SST_IPC_ACPI
 	tristate
 	select SND_SST_IPC
-	depends on ACPI
 
 config SND_SOC_INTEL_SST
 	tristate
@@ -91,7 +90,7 @@ config SND_SOC_INTEL_BROADWELL_MACH
 
 config SND_SOC_INTEL_BYTCR_RT5640_MACH
 	tristate "ASoC Audio DSP Support for MID BYT Platform"
-	depends on X86 && I2C
+	depends on X86 && I2C && ACPI
 	select SND_SOC_RT5640
 	select SND_SST_MFLD_PLATFORM
 	select SND_SST_IPC_ACPI
@@ -103,7 +102,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH
 
 config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
         tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec"
-        depends on X86_INTEL_LPSS && I2C
+        depends on X86_INTEL_LPSS && I2C && ACPI
         select SND_SOC_RT5670
         select SND_SST_MFLD_PLATFORM
         select SND_SST_IPC_ACPI
@@ -115,7 +114,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
 
 config SND_SOC_INTEL_CHT_BSW_RT5645_MACH
 	tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec"
-	depends on X86_INTEL_LPSS && I2C
+	depends on X86_INTEL_LPSS && I2C && ACPI
 	select SND_SOC_RT5645
 	select SND_SST_MFLD_PLATFORM
 	select SND_SST_IPC_ACPI
diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig
index 15c04e2eae34..976967675387 100644
--- a/sound/soc/mediatek/Kconfig
+++ b/sound/soc/mediatek/Kconfig
@@ -9,7 +9,7 @@ config SND_SOC_MEDIATEK
 
 config SND_SOC_MT8173_MAX98090
 	tristate "ASoC Audio driver for MT8173 with MAX98090 codec"
-	depends on SND_SOC_MEDIATEK
+	depends on SND_SOC_MEDIATEK && I2C
 	select SND_SOC_MAX98090
 	help
 	  This adds ASoC driver for Mediatek MT8173 boards
@@ -19,7 +19,7 @@ config SND_SOC_MT8173_MAX98090
 
 config SND_SOC_MT8173_RT5650_RT5676
 	tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs"
-	depends on SND_SOC_MEDIATEK
+	depends on SND_SOC_MEDIATEK && I2C
 	select SND_SOC_RT5645
 	select SND_SOC_RT5677
 	help
diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c
index 5a806da89f42..5e2eb4cc5cf1 100644
--- a/sound/soc/rockchip/rockchip_spdif.c
+++ b/sound/soc/rockchip/rockchip_spdif.c
@@ -54,7 +54,7 @@ static const struct of_device_id rk_spdif_match[] = {
 };
 MODULE_DEVICE_TABLE(of, rk_spdif_match);
 
-static int rk_spdif_runtime_suspend(struct device *dev)
+static int __maybe_unused rk_spdif_runtime_suspend(struct device *dev)
 {
 	struct rk_spdif_dev *spdif = dev_get_drvdata(dev);
 
@@ -64,7 +64,7 @@ static int rk_spdif_runtime_suspend(struct device *dev)
 	return 0;
 }
 
-static int rk_spdif_runtime_resume(struct device *dev)
+static int __maybe_unused rk_spdif_runtime_resume(struct device *dev)
 {
 	struct rk_spdif_dev *spdif = dev_get_drvdata(dev);
 	int ret;
@@ -316,26 +316,30 @@ static int rk_spdif_probe(struct platform_device *pdev)
 	spdif->mclk = devm_clk_get(&pdev->dev, "mclk");
 	if (IS_ERR(spdif->mclk)) {
 		dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n");
-		return PTR_ERR(spdif->mclk);
+		ret = PTR_ERR(spdif->mclk);
+		goto err_disable_hclk;
 	}
 
 	ret = clk_prepare_enable(spdif->mclk);
 	if (ret) {
 		dev_err(spdif->dev, "clock enable failed %d\n", ret);
-		return ret;
+		goto err_disable_clocks;
 	}
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	regs = devm_ioremap_resource(&pdev->dev, res);
-	if (IS_ERR(regs))
-		return PTR_ERR(regs);
+	if (IS_ERR(regs)) {
+		ret = PTR_ERR(regs);
+		goto err_disable_clocks;
+	}
 
 	spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs,
 						  &rk_spdif_regmap_config);
 	if (IS_ERR(spdif->regmap)) {
 		dev_err(&pdev->dev,
 			"Failed to initialise managed register map\n");
-		return PTR_ERR(spdif->regmap);
+		ret = PTR_ERR(spdif->regmap);
+		goto err_disable_clocks;
 	}
 
 	spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR;
@@ -367,6 +371,10 @@ static int rk_spdif_probe(struct platform_device *pdev)
 
 err_pm_runtime:
 	pm_runtime_disable(&pdev->dev);
+err_disable_clocks:
+	clk_disable_unprepare(spdif->mclk);
+err_disable_hclk:
+	clk_disable_unprepare(spdif->hclk);
 
 	return ret;
 }
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 085329878525..5976e3992dd1 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -235,6 +235,7 @@ enum rsnd_mod_type {
 	RSND_MOD_MIX,
 	RSND_MOD_CTU,
 	RSND_MOD_SRC,
+	RSND_MOD_SSIP, /* SSI parent */
 	RSND_MOD_SSI,
 	RSND_MOD_MAX,
 };
@@ -365,6 +366,7 @@ struct rsnd_dai_stream {
 };
 #define rsnd_io_to_mod(io, i)	((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL)
 #define rsnd_io_to_mod_ssi(io)	rsnd_io_to_mod((io), RSND_MOD_SSI)
+#define rsnd_io_to_mod_ssip(io) rsnd_io_to_mod((io), RSND_MOD_SSIP)
 #define rsnd_io_to_mod_src(io)	rsnd_io_to_mod((io), RSND_MOD_SRC)
 #define rsnd_io_to_mod_ctu(io)	rsnd_io_to_mod((io), RSND_MOD_CTU)
 #define rsnd_io_to_mod_mix(io)	rsnd_io_to_mod((io), RSND_MOD_MIX)
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index c62a2947ac14..38aae96267c9 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -550,11 +550,16 @@ static int rsnd_ssi_dma_remove(struct rsnd_mod *mod,
 			       struct rsnd_priv *priv)
 {
 	struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+	struct rsnd_mod *pure_ssi_mod = rsnd_io_to_mod_ssi(io);
 	struct device *dev = rsnd_priv_to_dev(priv);
 	int irq = ssi->info->irq;
 
 	rsnd_dma_quit(io, rsnd_mod_to_dma(mod));
 
+	/* Do nothing if non SSI (= SSI parent, multi SSI) mod */
+	if (pure_ssi_mod != mod)
+		return 0;
+
 	/* PIO will request IRQ again */
 	devm_free_irq(dev, irq, mod);
 
diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
index ba9fc099cf67..503aef8fcde2 100644
--- a/sound/soc/ux500/mop500.c
+++ b/sound/soc/ux500/mop500.c
@@ -164,3 +164,7 @@ static struct platform_driver snd_soc_mop500_driver = {
 };
 
 module_platform_driver(snd_soc_mop500_driver);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("ASoC MOP500 board driver");
+MODULE_AUTHOR("Ola Lilja");
diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c
index f12c01dddc8d..d35ba7700f46 100644
--- a/sound/soc/ux500/ux500_pcm.c
+++ b/sound/soc/ux500/ux500_pcm.c
@@ -165,3 +165,8 @@ int ux500_pcm_unregister_platform(struct platform_device *pdev)
 	return 0;
 }
 EXPORT_SYMBOL_GPL(ux500_pcm_unregister_platform);
+
+MODULE_AUTHOR("Ola Lilja");
+MODULE_AUTHOR("Roger Nilsson");
+MODULE_DESCRIPTION("ASoC UX500 driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 0ed9ae030ce1..c5447ff078b3 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -343,17 +343,20 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request,
 			    int validx, int *value_ret)
 {
 	struct snd_usb_audio *chip = cval->head.mixer->chip;
-	unsigned char buf[4 + 3 * sizeof(__u32)]; /* enough space for one range */
+	/* enough space for one range */
+	unsigned char buf[sizeof(__u16) + 3 * sizeof(__u32)];
 	unsigned char *val;
-	int idx = 0, ret, size;
+	int idx = 0, ret, val_size, size;
 	__u8 bRequest;
 
+	val_size = uac2_ctl_value_size(cval->val_type);
+
 	if (request == UAC_GET_CUR) {
 		bRequest = UAC2_CS_CUR;
-		size = uac2_ctl_value_size(cval->val_type);
+		size = val_size;
 	} else {
 		bRequest = UAC2_CS_RANGE;
-		size = sizeof(buf);
+		size = sizeof(__u16) + 3 * val_size;
 	}
 
 	memset(buf, 0, sizeof(buf));
@@ -386,16 +389,17 @@ error:
 		val = buf + sizeof(__u16);
 		break;
 	case UAC_GET_MAX:
-		val = buf + sizeof(__u16) * 2;
+		val = buf + sizeof(__u16) + val_size;
 		break;
 	case UAC_GET_RES:
-		val = buf + sizeof(__u16) * 3;
+		val = buf + sizeof(__u16) + val_size * 2;
 		break;
 	default:
 		return -EINVAL;
 	}
 
-	*value_ret = convert_signed_value(cval, snd_usb_combine_bytes(val, sizeof(__u16)));
+	*value_ret = convert_signed_value(cval,
+					  snd_usb_combine_bytes(val, val_size));
 
 	return 0;
 }
@@ -2101,20 +2105,25 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
 	kctl->private_value = (unsigned long)namelist;
 	kctl->private_free = usb_mixer_selector_elem_free;
 
-	nameid = uac_selector_unit_iSelector(desc);
+	/* check the static mapping table at first */
 	len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name));
-	if (len)
-		;
-	else if (nameid)
-		len = snd_usb_copy_string_desc(state, nameid, kctl->id.name,
-					 sizeof(kctl->id.name));
-	else
-		len = get_term_name(state, &state->oterm,
-				    kctl->id.name, sizeof(kctl->id.name), 0);
-
 	if (!len) {
-		strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name));
+		/* no mapping ? */
+		/* if iSelector is given, use it */
+		nameid = uac_selector_unit_iSelector(desc);
+		if (nameid)
+			len = snd_usb_copy_string_desc(state, 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,
+				    kctl->id.name, sizeof(kctl->id.name), 0);
+		/* ... or use the fixed string "USB" as the last resort */
+		if (!len)
+			strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name));
 
+		/* and add the proper suffix */
 		if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR)
 			append_ctl_name(kctl, " Clock Source");
 		else if ((state->oterm.type & 0xff00) == 0x0100)
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 48afae053c56..8e8db4ddf365 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -348,6 +348,15 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 
 		alts = &iface->altsetting[1];
 		goto add_sync_ep;
+	case USB_ID(0x1397, 0x0002):
+		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;
 	}
 	if (attr == USB_ENDPOINT_SYNC_ASYNC &&
 	    altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 8a59d4782a0f..69bf5cf1e91e 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3277,4 +3277,51 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 	}
 },
 
+{
+	/*
+	 * Bower's & Wilkins PX headphones only support the 48 kHz sample rate
+	 * even though it advertises more. The capture interface doesn't work
+	 * even on windows.
+	 */
+	USB_DEVICE(0x19b5, 0x0021),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_MIXER,
+			},
+			/* Capture */
+			{
+				.ifnum = 1,
+				.type = QUIRK_IGNORE_INTERFACE,
+			},
+			/* Playback */
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S16_LE,
+					.channels = 2,
+					.iface = 2,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.attributes = UAC_EP_CS_ATTR_FILL_MAX |
+						UAC_EP_CS_ATTR_SAMPLE_RATE,
+					.endpoint = 0x03,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC,
+					.rates = SNDRV_PCM_RATE_48000,
+					.rate_min = 48000,
+					.rate_max = 48000,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) {
+						48000
+					}
+				}
+			},
+		}
+	}
+},
+
 #undef USB_DEVICE_VENDOR_SPEC