Merge branch 'for-linus' into for-next

36 files changed, 328 insertions, 191 deletions

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index fadd3eb..9106d8e 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -74,6 +74,18 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream);
 static DEFINE_RWLOCK(snd_pcm_link_rwlock);
 static DECLARE_RWSEM(snd_pcm_link_rwsem);
 
+/* Writer in rwsem may block readers even during its waiting in queue,
+ * and this may lead to a deadlock when the code path takes read sem
+ * twice (e.g. one in snd_pcm_action_nonatomic() and another in
+ * snd_pcm_stream_lock()).  As a (suboptimal) workaround, let writer to
+ * spin until it gets the lock.
+ */
+static inline void down_write_nonblock(struct rw_semaphore *lock)
+{
+	while (!down_write_trylock(lock))
+		cond_resched();
+}
+
 /**
  * snd_pcm_stream_lock - Lock the PCM stream
  * @substream: PCM substream
@@ -1813,7 +1825,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
 		res = -ENOMEM;
 		goto _nolock;
 	}
-	down_write(&snd_pcm_link_rwsem);
+	down_write_nonblock(&snd_pcm_link_rwsem);
 	write_lock_irq(&snd_pcm_link_rwlock);
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN ||
 	    substream->runtime->status->state != substream1->runtime->status->state ||
@@ -1860,7 +1872,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream)
 	struct snd_pcm_substream *s;
 	int res = 0;
 
-	down_write(&snd_pcm_link_rwsem);
+	down_write_nonblock(&snd_pcm_link_rwsem);
 	write_lock_irq(&snd_pcm_link_rwlock);
 	if (!snd_pcm_stream_linked(substream)) {
 		res = -EALREADY;
diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c
index 8010766..c850345 100644
--- a/sound/core/seq/seq_memory.c
+++ b/sound/core/seq/seq_memory.c
@@ -383,15 +383,20 @@ int snd_seq_pool_init(struct snd_seq_pool *pool)
 
 	if (snd_BUG_ON(!pool))
 		return -EINVAL;
-	if (pool->ptr)			/* should be atomic? */
-		return 0;
 
-	pool->ptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size);
-	if (!pool->ptr)
+	cellptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size);
+	if (!cellptr)
 		return -ENOMEM;
 
 	/* add new cells to the free cell list */
 	spin_lock_irqsave(&pool->lock, flags);
+	if (pool->ptr) {
+		spin_unlock_irqrestore(&pool->lock, flags);
+		vfree(cellptr);
+		return 0;
+	}
+
+	pool->ptr = cellptr;
 	pool->free = NULL;
 
 	for (cell = 0; cell < pool->size; cell++) {
diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c
index 921fb2b..fe686ee 100644
--- a/sound/core/seq/seq_ports.c
+++ b/sound/core/seq/seq_ports.c
@@ -535,19 +535,22 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client,
 					bool is_src, bool ack)
 {
 	struct snd_seq_port_subs_info *grp;
+	struct list_head *list;
+	bool empty;
 
 	grp = is_src ? &port->c_src : &port->c_dest;
+	list = is_src ? &subs->src_list : &subs->dest_list;
 	down_write(&grp->list_mutex);
 	write_lock_irq(&grp->list_lock);
-	if (is_src)
-		list_del(&subs->src_list);
-	else
-		list_del(&subs->dest_list);
+	empty = list_empty(list);
+	if (!empty)
+		list_del_init(list);
 	grp->exclusive = 0;
 	write_unlock_irq(&grp->list_lock);
 	up_write(&grp->list_mutex);
 
-	unsubscribe_port(client, port, grp, &subs->info, ack);
+	if (!empty)
+		unsubscribe_port(client, port, grp, &subs->info, ack);
 }
 
 /* connect two ports */
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index b5a17cb..8c48623 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -426,18 +426,22 @@ EXPORT_SYMBOL_GPL(snd_hdac_bus_stop_chip);
  * @bus: HD-audio core bus
  * @status: INTSTS register value
  * @ask: callback to be called for woken streams
+ *
+ * Returns the bits of handled streams, or zero if no stream is handled.
  */
-void snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
+int snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
 				    void (*ack)(struct hdac_bus *,
 						struct hdac_stream *))
 {
 	struct hdac_stream *azx_dev;
 	u8 sd_status;
+	int handled = 0;
 
 	list_for_each_entry(azx_dev, &bus->stream_list, list) {
 		if (status & azx_dev->sd_int_sta_mask) {
 			sd_status = snd_hdac_stream_readb(azx_dev, SD_STS);
 			snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK);
+			handled |= 1 << azx_dev->index;
 			if (!azx_dev->substream || !azx_dev->running ||
 			    !(sd_status & SD_INT_COMPLETE))
 				continue;
@@ -445,6 +449,7 @@ void snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status,
 				ack(bus, azx_dev);
 		}
 	}
+	return handled;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_bus_handle_stream_irq);
 
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 37cf9ce..27de801 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -930,6 +930,8 @@ irqreturn_t azx_interrupt(int irq, void *dev_id)
 	struct azx *chip = dev_id;
 	struct hdac_bus *bus = azx_bus(chip);
 	u32 status;
+	bool active, handled = false;
+	int repeat = 0; /* count for avoiding endless loop */
 
 #ifdef CONFIG_PM
 	if (azx_has_pm_runtime(chip))
@@ -939,33 +941,36 @@ irqreturn_t azx_interrupt(int irq, void *dev_id)
 
 	spin_lock(&bus->reg_lock);
 
-	if (chip->disabled) {
-		spin_unlock(&bus->reg_lock);
-		return IRQ_NONE;
-	}
-
-	status = azx_readl(chip, INTSTS);
-	if (status == 0 || status == 0xffffffff) {
-		spin_unlock(&bus->reg_lock);
-		return IRQ_NONE;
-	}
+	if (chip->disabled)
+		goto unlock;
 
-	snd_hdac_bus_handle_stream_irq(bus, status, stream_update);
+	do {
+		status = azx_readl(chip, INTSTS);
+		if (status == 0 || status == 0xffffffff)
+			break;
 
-	/* clear rirb int */
-	status = azx_readb(chip, RIRBSTS);
-	if (status & RIRB_INT_MASK) {
-		if (status & RIRB_INT_RESPONSE) {
-			if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND)
-				udelay(80);
-			snd_hdac_bus_update_rirb(bus);
+		handled = true;
+		active = false;
+		if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update))
+			active = true;
+
+		/* clear rirb int */
+		status = azx_readb(chip, RIRBSTS);
+		if (status & RIRB_INT_MASK) {
+			active = true;
+			if (status & RIRB_INT_RESPONSE) {
+				if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND)
+					udelay(80);
+				snd_hdac_bus_update_rirb(bus);
+			}
+			azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
 		}
-		azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
-	}
+	} while (active && ++repeat < 10);
 
+ unlock:
 	spin_unlock(&bus->reg_lock);
 
-	return IRQ_HANDLED;
+	return IRQ_RETVAL(handled);
 }
 EXPORT_SYMBOL_GPL(azx_interrupt);
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4179710..2624cfe 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -363,7 +363,10 @@ enum {
 					((pci)->device == 0x0d0c) || \
 					((pci)->device == 0x160c))
 
-#define IS_BROXTON(pci)	((pci)->device == 0x5a98)
+#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170)
+#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70)
+#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
+#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci))
 
 static char *driver_short_names[] = {
 	[AZX_DRIVER_ICH] = "HDA Intel",
@@ -540,13 +543,13 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset)
 
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
 		snd_hdac_set_codec_wakeup(bus, true);
-	if (IS_BROXTON(pci)) {
+	if (IS_SKL_PLUS(pci)) {
 		pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val);
 		val = val & ~INTEL_HDA_CGCTL_MISCBDCGE;
 		pci_write_config_dword(pci, INTEL_HDA_CGCTL, val);
 	}
 	azx_init_chip(chip, full_reset);
-	if (IS_BROXTON(pci)) {
+	if (IS_SKL_PLUS(pci)) {
 		pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val);
 		val = val | INTEL_HDA_CGCTL_MISCBDCGE;
 		pci_write_config_dword(pci, INTEL_HDA_CGCTL, val);
@@ -555,7 +558,7 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset)
 		snd_hdac_set_codec_wakeup(bus, false);
 
 	/* reduce dma latency to avoid noise */
-	if (IS_BROXTON(pci))
+	if (IS_BXT(pci))
 		bxt_reduce_dma_latency(chip);
 }
 
@@ -977,11 +980,6 @@ static int azx_resume(struct device *dev)
 /* put codec down to D3 at hibernation for Intel SKL+;
  * otherwise BIOS may still access the codec and screw up the driver
  */
-#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170)
-#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70)
-#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
-#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci))
-
 static int azx_freeze_noirq(struct device *dev)
 {
 	struct pci_dev *pci = to_pci_dev(dev);
@@ -2168,10 +2166,10 @@ static void azx_remove(struct pci_dev *pci)
 	struct hda_intel *hda;
 
 	if (card) {
-		/* flush the pending probing work */
+		/* cancel the pending probing work */
 		chip = card->private_data;
 		hda = container_of(chip, struct hda_intel, chip);
-		flush_work(&hda->probe_work);
+		cancel_work_sync(&hda->probe_work);
 
 		snd_card_free(card);
 	}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index efd4980..1f357cd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3801,6 +3801,10 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
 
 static void alc_headset_mode_default(struct hda_codec *codec)
 {
+	static struct coef_fw coef0225[] = {
+		UPDATE_COEF(0x45, 0x3f<<10, 0x34<<10),
+		{}
+	};
 	static struct coef_fw coef0255[] = {
 		WRITE_COEF(0x45, 0xc089),
 		WRITE_COEF(0x45, 0xc489),
@@ -3842,6 +3846,9 @@ static void alc_headset_mode_default(struct hda_codec *codec)
 	};
 
 	switch (codec->core.vendor_id) {
+	case 0x10ec0225:
+		alc_process_coef_fw(codec, coef0225);
+		break;
 	case 0x10ec0255:
 	case 0x10ec0256:
 		alc_process_coef_fw(codec, coef0255);
@@ -4749,6 +4756,9 @@ enum {
 	ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE,
 	ALC293_FIXUP_LENOVO_SPK_NOISE,
 	ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
+	ALC255_FIXUP_DELL_SPK_NOISE,
+	ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
+	ALC280_FIXUP_HP_HEADSET_MIC,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -5368,6 +5378,29 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc233_fixup_lenovo_line2_mic_hotkey,
 	},
+	[ALC255_FIXUP_DELL_SPK_NOISE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_disable_aamix,
+		.chained = true,
+		.chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
+	},
+	[ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			/* Disable pass-through path for FRONT 14h */
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x36 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x57d7 },
+			{}
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
+	},
+	[ALC280_FIXUP_HP_HEADSET_MIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_disable_aamix,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5410,6 +5443,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
 	SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
 	SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
+	SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_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),
@@ -5470,6 +5504,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+	SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -5638,10 +5673,10 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{0x21, 0x03211020}
 
 static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
-	SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
+	SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
 		ALC225_STANDARD_PINS,
 		{0x14, 0x901701a0}),
-	SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
+	SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
 		ALC225_STANDARD_PINS,
 		{0x14, 0x901701b0}),
 	SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE,
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c
index 3191e0a..d1fb035 100644
--- a/sound/soc/amd/acp-pcm-dma.c
+++ b/sound/soc/amd/acp-pcm-dma.c
@@ -635,6 +635,7 @@ static int acp_dma_open(struct snd_pcm_substream *substream)
 					    SNDRV_PCM_HW_PARAM_PERIODS);
 	if (ret < 0) {
 		dev_err(prtd->platform->dev, "set integer constraint failed\n");
+		kfree(adata);
 		return ret;
 	}
 
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 33143fe..9178531 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1929,6 +1929,25 @@ static struct {
 	{ 1000000, 13500000, 0,  1 },
 };
 
+static const unsigned int pseudo_fref_max[ARIZONA_FLL_MAX_FRATIO] = {
+	13500000,
+	 6144000,
+	 6144000,
+	 3072000,
+	 3072000,
+	 2822400,
+	 2822400,
+	 1536000,
+	 1536000,
+	 1536000,
+	 1536000,
+	 1536000,
+	 1536000,
+	 1536000,
+	 1536000,
+	  768000,
+};
+
 static struct {
 	unsigned int min;
 	unsigned int max;
@@ -2042,16 +2061,32 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
 	/* Adjust FRATIO/refdiv to avoid integer mode if possible */
 	refdiv = cfg->refdiv;
 
+	arizona_fll_dbg(fll, "pseudo: initial ratio=%u fref=%u refdiv=%u\n",
+			init_ratio, Fref, refdiv);
+
 	while (div <= ARIZONA_FLL_MAX_REFDIV) {
 		for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO;
 		     ratio++) {
 			if ((ARIZONA_FLL_VCO_CORNER / 2) /
-			    (fll->vco_mult * ratio) < Fref)
+			    (fll->vco_mult * ratio) < Fref) {
+				arizona_fll_dbg(fll, "pseudo: hit VCO corner\n");
 				break;
+			}
+
+			if (Fref > pseudo_fref_max[ratio - 1]) {
+				arizona_fll_dbg(fll,
+					"pseudo: exceeded max fref(%u) for ratio=%u\n",
+					pseudo_fref_max[ratio - 1],
+					ratio);
+				break;
+			}
 
 			if (target % (ratio * Fref)) {
 				cfg->refdiv = refdiv;
 				cfg->fratio = ratio - 1;
+				arizona_fll_dbg(fll,
+					"pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
+					Fref, refdiv, div, ratio);
 				return ratio;
 			}
 		}
@@ -2060,6 +2095,9 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
 			if (target % (ratio * Fref)) {
 				cfg->refdiv = refdiv;
 				cfg->fratio = ratio - 1;
+				arizona_fll_dbg(fll,
+					"pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
+					Fref, refdiv, div, ratio);
 				return ratio;
 			}
 		}
@@ -2068,6 +2106,9 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
 		Fref /= 2;
 		refdiv++;
 		init_ratio = arizona_find_fratio(Fref, NULL);
+		arizona_fll_dbg(fll,
+				"pseudo: change fref=%u refdiv=%d(%d) ratio=%u\n",
+				Fref, refdiv, div, init_ratio);
 	}
 
 	arizona_fll_warn(fll, "Falling back to integer mode operation\n");
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index bc08f0c..1bd3164 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -266,6 +266,8 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
 		} else {
 			*mic = false;
 			regmap_write(rt286->regmap, RT286_SET_MIC1, 0x20);
+			regmap_update_bits(rt286->regmap,
+				RT286_CBJ_CTRL1, 0x0400, 0x0000);
 		}
 	} else {
 		regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf);
@@ -470,24 +472,6 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static int rt286_vref_event(struct snd_soc_dapm_widget *w,
-			     struct snd_kcontrol *kcontrol, int event)
-{
-	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
-
-	switch (event) {
-	case SND_SOC_DAPM_PRE_PMU:
-		snd_soc_update_bits(codec,
-			RT286_CBJ_CTRL1, 0x0400, 0x0000);
-		mdelay(50);
-		break;
-	default:
-		return 0;
-	}
-
-	return 0;
-}
-
 static int rt286_ldo2_event(struct snd_soc_dapm_widget *w,
 			     struct snd_kcontrol *kcontrol, int event)
 {
@@ -536,7 +520,7 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY_S("HV", 1, RT286_POWER_CTRL1,
 		12, 1, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("VREF", RT286_POWER_CTRL1,
-		0, 1, rt286_vref_event, SND_SOC_DAPM_PRE_PMU),
+		0, 1, NULL, 0),
 	SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT286_POWER_CTRL2,
 		2, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY_S("LDO2", 2, RT286_POWER_CTRL1,
@@ -911,8 +895,6 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_ON:
 		mdelay(10);
 		snd_soc_update_bits(codec,
-			RT286_CBJ_CTRL1, 0x0400, 0x0400);
-		snd_soc_update_bits(codec,
 			RT286_DC_GAIN, 0x200, 0x0);
 
 		break;
@@ -920,8 +902,6 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_STANDBY:
 		snd_soc_write(codec,
 			RT286_SET_AUDIO_POWER, AC_PWRST_D3);
-		snd_soc_update_bits(codec,
-			RT286_CBJ_CTRL1, 0x0400, 0x0000);
 		break;
 
 	default:
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index c61d38b..93e8c90 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -776,7 +776,7 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = {
 
 	/* IN1/IN2 Control */
 	SOC_SINGLE_TLV("IN1 Boost", RT5645_IN1_CTRL1,
-		RT5645_BST_SFT1, 8, 0, bst_tlv),
+		RT5645_BST_SFT1, 12, 0, bst_tlv),
 	SOC_SINGLE_TLV("IN2 Boost", RT5645_IN2_CTRL,
 		RT5645_BST_SFT2, 8, 0, bst_tlv),
 
diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index 820d8fa..fb8ea05 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -3985,7 +3985,6 @@ static int rt5659_i2c_probe(struct i2c_client *i2c,
 	if (rt5659 == NULL)
 		return -ENOMEM;
 
-	rt5659->i2c = i2c;
 	i2c_set_clientdata(i2c, rt5659);
 
 	if (pdata)
@@ -4157,24 +4156,17 @@ static int rt5659_i2c_probe(struct i2c_client *i2c,
 
 	INIT_DELAYED_WORK(&rt5659->jack_detect_work, rt5659_jack_detect_work);
 
-	if (rt5659->i2c->irq) {
-		ret = request_threaded_irq(rt5659->i2c->irq, NULL, rt5659_irq,
-			IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
+	if (i2c->irq) {
+		ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL,
+			rt5659_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
 			| IRQF_ONESHOT, "rt5659", rt5659);
 		if (ret)
 			dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret);
 
 	}
 
-	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5659,
+	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5659,
 			rt5659_dai, ARRAY_SIZE(rt5659_dai));
-
-	if (ret) {
-		if (rt5659->i2c->irq)
-			free_irq(rt5659->i2c->irq, rt5659);
-	}
-
-	return 0;
 }
 
 static int rt5659_i2c_remove(struct i2c_client *i2c)
@@ -4191,24 +4183,29 @@ void rt5659_i2c_shutdown(struct i2c_client *client)
 	regmap_write(rt5659->regmap, RT5659_RESET, 0);
 }
 
+#ifdef CONFIG_OF
 static const struct of_device_id rt5659_of_match[] = {
 	{ .compatible = "realtek,rt5658", },
 	{ .compatible = "realtek,rt5659", },
-	{},
+	{ },
 };
+MODULE_DEVICE_TABLE(of, rt5659_of_match);
+#endif
 
+#ifdef CONFIG_ACPI
 static struct acpi_device_id rt5659_acpi_match[] = {
-		{ "10EC5658", 0},
-		{ "10EC5659", 0},
-		{ },
+	{ "10EC5658", 0, },
+	{ "10EC5659", 0, },
+	{ },
 };
 MODULE_DEVICE_TABLE(acpi, rt5659_acpi_match);
+#endif
 
 struct i2c_driver rt5659_i2c_driver = {
 	.driver = {
 		.name = "rt5659",
 		.owner = THIS_MODULE,
-		.of_match_table = rt5659_of_match,
+		.of_match_table = of_match_ptr(rt5659_of_match),
 		.acpi_match_table = ACPI_PTR(rt5659_acpi_match),
 	},
 	.probe = rt5659_i2c_probe,
diff --git a/sound/soc/codecs/rt5659.h b/sound/soc/codecs/rt5659.h
index 8f07ee9..d31c9e5 100644
--- a/sound/soc/codecs/rt5659.h
+++ b/sound/soc/codecs/rt5659.h
@@ -1792,7 +1792,6 @@ struct rt5659_priv {
 	struct snd_soc_codec *codec;
 	struct rt5659_platform_data pdata;
 	struct regmap *regmap;
-	struct i2c_client *i2c;
 	struct gpio_desc *gpiod_ldo1_en;
 	struct gpio_desc *gpiod_reset;
 	struct snd_soc_jack *hs_jack;
diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c
index 21ca3a5..d374c18 100644
--- a/sound/soc/codecs/sigmadsp-i2c.c
+++ b/sound/soc/codecs/sigmadsp-i2c.c
@@ -31,7 +31,10 @@ static int sigmadsp_write_i2c(void *control_data,
 
 	kfree(buf);
 
-	return ret;
+	if (ret < 0)
+		return ret;
+
+	return 0;
 }
 
 static int sigmadsp_read_i2c(void *control_data,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 6088d30..97c0f1e 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -2382,6 +2382,7 @@ error:
 
 static int wm5110_remove(struct platform_device *pdev)
 {
+	snd_soc_unregister_platform(&pdev->dev);
 	snd_soc_unregister_codec(&pdev->dev);
 	pm_runtime_disable(&pdev->dev);
 
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index ff23772..d7f444f 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -240,13 +240,13 @@ SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
 SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
 	7, 1, 1),
 
-SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
 	       WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
-SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
 	       WM8960_INBMIX1, 1, 7, 0, lineinboost_tlv),
-SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
 	       WM8960_INBMIX2, 4, 7, 0, lineinboost_tlv),
-SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
 	       WM8960_INBMIX2, 1, 7, 0, lineinboost_tlv),
 SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT1 Volume",
 		WM8960_RINPATH, 4, 3, 0, micboost_tlv),
@@ -643,29 +643,31 @@ static int wm8960_configure_clocking(struct snd_soc_codec *codec)
 		return -EINVAL;
 	}
 
-	/* check if the sysclk frequency is available. */
-	for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
-		if (sysclk_divs[i] == -1)
-			continue;
-		sysclk = freq_out / sysclk_divs[i];
-		for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) {
-			if (sysclk == dac_divs[j] * lrclk) {
+	if (wm8960->clk_id != WM8960_SYSCLK_PLL) {
+		/* check if the sysclk frequency is available. */
+		for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
+			if (sysclk_divs[i] == -1)
+				continue;
+			sysclk = freq_out / sysclk_divs[i];
+			for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) {
+				if (sysclk != dac_divs[j] * lrclk)
+					continue;
 				for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k)
 					if (sysclk == bclk * bclk_divs[k] / 10)
 						break;
 				if (k != ARRAY_SIZE(bclk_divs))
 					break;
 			}
+			if (j != ARRAY_SIZE(dac_divs))
+				break;
 		}
-		if (j != ARRAY_SIZE(dac_divs))
-			break;
-	}
 
-	if (i != ARRAY_SIZE(sysclk_divs)) {
-		goto configure_clock;
-	} else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) {
-		dev_err(codec->dev, "failed to configure clock\n");
-		return -EINVAL;
+		if (i != ARRAY_SIZE(sysclk_divs)) {
+			goto configure_clock;
+		} else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) {
+			dev_err(codec->dev, "failed to configure clock\n");
+			return -EINVAL;
+		}
 	}
 	/* get a available pll out frequency and set pll */
 	for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index ce664c2..bff258d 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -645,6 +645,8 @@ static int dw_i2s_probe(struct platform_device *pdev)
 
 	dev->dev = &pdev->dev;
 
+	dev->i2s_reg_comp1 = I2S_COMP_PARAM_1;
+	dev->i2s_reg_comp2 = I2S_COMP_PARAM_2;
 	if (pdata) {
 		dev->capability = pdata->cap;
 		clk_id = NULL;
@@ -652,9 +654,6 @@ static int dw_i2s_probe(struct platform_device *pdev)
 		if (dev->quirks & DW_I2S_QUIRK_COMP_REG_OFFSET) {
 			dev->i2s_reg_comp1 = pdata->i2s_reg_comp1;
 			dev->i2s_reg_comp2 = pdata->i2s_reg_comp2;
-		} else {
-			dev->i2s_reg_comp1 = I2S_COMP_PARAM_1;
-			dev->i2s_reg_comp2 = I2S_COMP_PARAM_2;
 		}
 		ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata);
 	} else {
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 40dfd8a..ed8de10 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -112,20 +112,6 @@ struct fsl_ssi_rxtx_reg_val {
 	struct fsl_ssi_reg_val tx;
 };
 
-static const struct reg_default fsl_ssi_reg_defaults[] = {
-	{CCSR_SSI_SCR,     0x00000000},
-	{CCSR_SSI_SIER,    0x00003003},
-	{CCSR_SSI_STCR,    0x00000200},
-	{CCSR_SSI_SRCR,    0x00000200},
-	{CCSR_SSI_STCCR,   0x00040000},
-	{CCSR_SSI_SRCCR,   0x00040000},
-	{CCSR_SSI_SACNT,   0x00000000},
-	{CCSR_SSI_STMSK,   0x00000000},
-	{CCSR_SSI_SRMSK,   0x00000000},
-	{CCSR_SSI_SACCEN,  0x00000000},
-	{CCSR_SSI_SACCDIS, 0x00000000},
-};
-
 static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
@@ -190,8 +176,7 @@ static const struct regmap_config fsl_ssi_regconfig = {
 	.val_bits = 32,
 	.reg_stride = 4,
 	.val_format_endian = REGMAP_ENDIAN_NATIVE,
-	.reg_defaults = fsl_ssi_reg_defaults,
-	.num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults),
+	.num_reg_defaults_raw = CCSR_SSI_SACCDIS / sizeof(uint32_t) + 1,
 	.readable_reg = fsl_ssi_readable_reg,
 	.volatile_reg = fsl_ssi_volatile_reg,
 	.precious_reg = fsl_ssi_precious_reg,
@@ -201,6 +186,7 @@ static const struct regmap_config fsl_ssi_regconfig = {
 
 struct fsl_ssi_soc_data {
 	bool imx;
+	bool imx21regs; /* imx21-class SSI - no SACC{ST,EN,DIS} regs */
 	bool offline_config;
 	u32 sisr_write_mask;
 };
@@ -303,6 +289,7 @@ static struct fsl_ssi_soc_data fsl_ssi_mpc8610 = {
 
 static struct fsl_ssi_soc_data fsl_ssi_imx21 = {
 	.imx = true,
+	.imx21regs = true,
 	.offline_config = true,
 	.sisr_write_mask = 0,
 };
@@ -586,8 +573,12 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private)
 	 */
 	regmap_write(regs, CCSR_SSI_SACNT,
 			CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV);
-	regmap_write(regs, CCSR_SSI_SACCDIS, 0xff);
-	regmap_write(regs, CCSR_SSI_SACCEN, 0x300);
+
+	/* no SACC{ST,EN,DIS} regs on imx21-class SSI */
+	if (!ssi_private->soc->imx21regs) {
+		regmap_write(regs, CCSR_SSI_SACCDIS, 0xff);
+		regmap_write(regs, CCSR_SSI_SACCEN, 0x300);
+	}
 
 	/*
 	 * Enable SSI, Transmit and Receive. AC97 has to communicate with the
@@ -1397,6 +1388,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	struct resource *res;
 	void __iomem *iomem;
 	char name[64];
+	struct regmap_config regconfig = fsl_ssi_regconfig;
 
 	of_id = of_match_device(fsl_ssi_ids, &pdev->dev);
 	if (!of_id || !of_id->data)
@@ -1444,15 +1436,25 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		return PTR_ERR(iomem);
 	ssi_private->ssi_phys = res->start;
 
+	if (ssi_private->soc->imx21regs) {
+		/*
+		 * According to datasheet imx21-class SSI
+		 * don't have SACC{ST,EN,DIS} regs.
+		 */
+		regconfig.max_register = CCSR_SSI_SRMSK;
+		regconfig.num_reg_defaults_raw =
+			CCSR_SSI_SRMSK / sizeof(uint32_t) + 1;
+	}
+
 	ret = of_property_match_string(np, "clock-names", "ipg");
 	if (ret < 0) {
 		ssi_private->has_ipg_clk_name = false;
 		ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem,
-			&fsl_ssi_regconfig);
+			&regconfig);
 	} else {
 		ssi_private->has_ipg_clk_name = true;
 		ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev,
-			"ipg", iomem, &fsl_ssi_regconfig);
+			"ipg", iomem, &regconfig);
 	}
 	if (IS_ERR(ssi_private->regs)) {
 		dev_err(&pdev->dev, "Failed to init register map\n");
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index a407e83..fb896b2 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -72,8 +72,6 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
 		goto end;
 	}
 
-	platform_set_drvdata(pdev, data);
-
 end:
 	of_node_put(spdif_np);
 
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 1ded881..2389ab4 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -99,7 +99,7 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream,
 		if (ret && ret != -ENOTSUPP)
 			goto err;
 	}
-
+	return 0;
 err:
 	return ret;
 }
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 803f95e..7d7c872 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -30,11 +30,15 @@ config SND_SST_IPC_ACPI
 config SND_SOC_INTEL_SST
 	tristate
 	select SND_SOC_INTEL_SST_ACPI if ACPI
+	select SND_SOC_INTEL_SST_MATCH if ACPI
 	depends on (X86 || COMPILE_TEST)
 
 config SND_SOC_INTEL_SST_ACPI
 	tristate
 
+config SND_SOC_INTEL_SST_MATCH
+	tristate
+
 config SND_SOC_INTEL_HASWELL
 	tristate
 
@@ -57,7 +61,7 @@ config SND_SOC_INTEL_HASWELL_MACH
 config SND_SOC_INTEL_BYT_RT5640_MACH
 	tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec"
 	depends on X86_INTEL_LPSS && I2C
-	depends on DW_DMAC_CORE=y && (SND_SOC_INTEL_BYTCR_RT5640_MACH = n)
+	depends on DW_DMAC_CORE=y && (SND_SST_IPC_ACPI = n)
 	select SND_SOC_INTEL_SST
 	select SND_SOC_INTEL_BAYTRAIL
 	select SND_SOC_RT5640
@@ -69,7 +73,7 @@ config SND_SOC_INTEL_BYT_RT5640_MACH
 config SND_SOC_INTEL_BYT_MAX98090_MACH
 	tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec"
 	depends on X86_INTEL_LPSS && I2C
-	depends on DW_DMAC_CORE=y
+	depends on DW_DMAC_CORE=y && (SND_SST_IPC_ACPI = n)
 	select SND_SOC_INTEL_SST
 	select SND_SOC_INTEL_BAYTRAIL
 	select SND_SOC_MAX98090
@@ -97,6 +101,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH
 	select SND_SOC_RT5640
 	select SND_SST_MFLD_PLATFORM
 	select SND_SST_IPC_ACPI
+	select SND_SOC_INTEL_SST_MATCH if ACPI
 	help
           This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR
           platforms with RT5640 audio codec.
@@ -109,6 +114,7 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH
 	select SND_SOC_RT5651
 	select SND_SST_MFLD_PLATFORM
 	select SND_SST_IPC_ACPI
+	select SND_SOC_INTEL_SST_MATCH if ACPI
 	help
           This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR
           platforms with RT5651 audio codec.
@@ -121,6 +127,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
         select SND_SOC_RT5670
         select SND_SST_MFLD_PLATFORM
         select SND_SST_IPC_ACPI
+	select SND_SOC_INTEL_SST_MATCH if ACPI
         help
           This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
           platforms with RT5672 audio codec.
@@ -133,6 +140,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH
 	select SND_SOC_RT5645
 	select SND_SST_MFLD_PLATFORM
 	select SND_SST_IPC_ACPI
+	select SND_SOC_INTEL_SST_MATCH if ACPI
 	help
 	  This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
 	  platforms with RT5645/5650 audio codec.
@@ -145,6 +153,7 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH
 	select SND_SOC_TS3A227E
 	select SND_SST_MFLD_PLATFORM
 	select SND_SST_IPC_ACPI
+	select SND_SOC_INTEL_SST_MATCH if ACPI
 	help
       This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
       platforms with MAX98090 audio codec it also can support TI jack chip as aux device.
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 55c33dc..52ed434 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -528,6 +528,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = {
 	.ops = &sst_compr_dai_ops,
 	.playback = {
 		.stream_name = "Compress Playback",
+		.channels_min = 1,
 	},
 },
 /* BE CPU  Dais */
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 7396ddb..2cbcbe4 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -212,7 +212,10 @@ static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *channels = hw_param_interval(params,
 						SNDRV_PCM_HW_PARAM_CHANNELS);
-	channels->min = channels->max = 4;
+	if (params_channels(params) == 2)
+		channels->min = channels->max = 2;
+	else
+		channels->min = channels->max = 4;
 
 	return 0;
 }
diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile
index 668fdee..fbbb25c 100644
--- a/sound/soc/intel/common/Makefile
+++ b/sound/soc/intel/common/Makefile
@@ -1,13 +1,10 @@
 snd-soc-sst-dsp-objs := sst-dsp.o
-ifneq ($(CONFIG_SND_SST_IPC_ACPI),)
-snd-soc-sst-acpi-objs := sst-match-acpi.o
-else
-snd-soc-sst-acpi-objs := sst-acpi.o sst-match-acpi.o
-endif
-
+snd-soc-sst-acpi-objs := sst-acpi.o
+snd-soc-sst-match-objs := sst-match-acpi.o
 snd-soc-sst-ipc-objs := sst-ipc.o
 
 snd-soc-sst-dsp-$(CONFIG_DW_DMAC_CORE) += sst-firmware.o
 
 obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o
 obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o
+obj-$(CONFIG_SND_SOC_INTEL_SST_MATCH) += snd-soc-sst-match.o
diff --git a/sound/soc/intel/common/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c
index 7a85c57..2c5eda1 100644
--- a/sound/soc/intel/common/sst-acpi.c
+++ b/sound/soc/intel/common/sst-acpi.c
@@ -215,6 +215,7 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = {
 	.dma_size = SST_LPT_DSP_DMA_SIZE,
 };
 
+#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI)
 static struct sst_acpi_mach baytrail_machines[] = {
 	{ "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL },
 	{ "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL },
@@ -231,11 +232,14 @@ static struct sst_acpi_desc sst_acpi_baytrail_desc = {
 	.sst_id = SST_DEV_ID_BYT,
 	.resindex_dma_base = -1,
 };
+#endif
 
 static const struct acpi_device_id sst_acpi_match[] = {
 	{ "INT33C8", (unsigned long)&sst_acpi_haswell_desc },
 	{ "INT3438", (unsigned long)&sst_acpi_broadwell_desc },
+#if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI)
 	{ "80860F28", (unsigned long)&sst_acpi_baytrail_desc },
+#endif
 	{ }
 };
 MODULE_DEVICE_TABLE(acpi, sst_acpi_match);
diff --git a/sound/soc/intel/common/sst-match-acpi.c b/sound/soc/intel/common/sst-match-acpi.c
index dd077e1..3b4539d 100644
--- a/sound/soc/intel/common/sst-match-acpi.c
+++ b/sound/soc/intel/common/sst-match-acpi.c
@@ -41,3 +41,6 @@ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines)
 	return NULL;
 }
 EXPORT_SYMBOL_GPL(sst_acpi_find_machine);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Intel Common ACPI Match module");
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index de6dac4..4629372 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -688,14 +688,14 @@ int skl_unbind_modules(struct skl_sst *ctx,
 	/* get src queue index */
 	src_index = skl_get_queue_index(src_mcfg->m_out_pin, dst_id, out_max);
 	if (src_index < 0)
-		return -EINVAL;
+		return 0;
 
 	msg.src_queue = src_index;
 
 	/* get dst queue index */
 	dst_index  = skl_get_queue_index(dst_mcfg->m_in_pin, src_id, in_max);
 	if (dst_index < 0)
-		return -EINVAL;
+		return 0;
 
 	msg.dst_queue = dst_index;
 
@@ -747,7 +747,7 @@ int skl_bind_modules(struct skl_sst *ctx,
 
 	skl_dump_bind_info(ctx, src_mcfg, dst_mcfg);
 
-	if (src_mcfg->m_state < SKL_MODULE_INIT_DONE &&
+	if (src_mcfg->m_state < SKL_MODULE_INIT_DONE ||
 		dst_mcfg->m_state < SKL_MODULE_INIT_DONE)
 		return 0;
 
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index f355325..b6e6b61 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -863,6 +863,7 @@ static int skl_get_delay_from_lpib(struct hdac_ext_bus *ebus,
 		else
 			delay += hstream->bufsize;
 	}
+	delay = (hstream->bufsize == delay) ? 0 : delay;
 
 	if (delay >= hstream->period_bytes) {
 		dev_info(bus->dev,
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 4624556..a294fee 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -54,12 +54,9 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w)
 
 /*
  * Each pipelines needs memory to be allocated. Check if we have free memory
- * from available pool. Then only add this to pool
- * This is freed when pipe is deleted
- * Note: DSP does actual memory management we only keep track for complete
- * pool
+ * from available pool.
  */
-static bool skl_tplg_alloc_pipe_mem(struct skl *skl,
+static bool skl_is_pipe_mem_avail(struct skl *skl,
 				struct skl_module_cfg *mconfig)
 {
 	struct skl_sst *ctx = skl->skl_sst;
@@ -74,10 +71,20 @@ static bool skl_tplg_alloc_pipe_mem(struct skl *skl,
 				"exceeds ppl memory available %d mem %d\n",
 				skl->resource.max_mem, skl->resource.mem);
 		return false;
+	} else {
+		return true;
 	}
+}
 
+/*
+ * Add the mem to the mem pool. This is freed when pipe is deleted.
+ * Note: DSP does actual memory management we only keep track for complete
+ * pool
+ */
+static void skl_tplg_alloc_pipe_mem(struct skl *skl,
+				struct skl_module_cfg *mconfig)
+{
 	skl->resource.mem += mconfig->pipe->memory_pages;
-	return true;
 }
 
 /*
@@ -85,10 +92,10 @@ static bool skl_tplg_alloc_pipe_mem(struct skl *skl,
  * quantified in MCPS (Million Clocks Per Second) required for module/pipe
  *
  * Each pipelines needs mcps to be allocated. Check if we have mcps for this
- * pipe. This adds the mcps to driver counter
- * This is removed on pipeline delete
+ * pipe.
  */
-static bool skl_tplg_alloc_pipe_mcps(struct skl *skl,
+
+static bool skl_is_pipe_mcps_avail(struct skl *skl,
 				struct skl_module_cfg *mconfig)
 {
 	struct skl_sst *ctx = skl->skl_sst;
@@ -98,13 +105,18 @@ static bool skl_tplg_alloc_pipe_mcps(struct skl *skl,
 			"%s: module_id %d instance %d\n", __func__,
 			mconfig->id.module_id, mconfig->id.instance_id);
 		dev_err(ctx->dev,
-			"exceeds ppl memory available %d > mem %d\n",
+			"exceeds ppl mcps available %d > mem %d\n",
 			skl->resource.max_mcps, skl->resource.mcps);
 		return false;
+	} else {
+		return true;
 	}
+}
 
+static void skl_tplg_alloc_pipe_mcps(struct skl *skl,
+				struct skl_module_cfg *mconfig)
+{
 	skl->resource.mcps += mconfig->mcps;
-	return true;
 }
 
 /*
@@ -411,7 +423,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
 		mconfig = w->priv;
 
 		/* check resource available */
-		if (!skl_tplg_alloc_pipe_mcps(skl, mconfig))
+		if (!skl_is_pipe_mcps_avail(skl, mconfig))
 			return -ENOMEM;
 
 		if (mconfig->is_loadable && ctx->dsp->fw_ops.load_mod) {
@@ -435,6 +447,7 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe)
 		ret = skl_tplg_set_module_params(w, ctx);
 		if (ret < 0)
 			return ret;
+		skl_tplg_alloc_pipe_mcps(skl, mconfig);
 	}
 
 	return 0;
@@ -477,10 +490,10 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 	struct skl_sst *ctx = skl->skl_sst;
 
 	/* check resource available */
-	if (!skl_tplg_alloc_pipe_mcps(skl, mconfig))
+	if (!skl_is_pipe_mcps_avail(skl, mconfig))
 		return -EBUSY;
 
-	if (!skl_tplg_alloc_pipe_mem(skl, mconfig))
+	if (!skl_is_pipe_mem_avail(skl, mconfig))
 		return -ENOMEM;
 
 	/*
@@ -526,11 +539,15 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 		src_module = dst_module;
 	}
 
+	skl_tplg_alloc_pipe_mem(skl, mconfig);
+	skl_tplg_alloc_pipe_mcps(skl, mconfig);
+
 	return 0;
 }
 
 static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w,
 				struct skl *skl,
+				struct snd_soc_dapm_widget *src_w,
 				struct skl_module_cfg *src_mconfig)
 {
 	struct snd_soc_dapm_path *p;
@@ -547,6 +564,10 @@ static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w,
 		dev_dbg(ctx->dev, "%s: sink widget=%s\n", __func__, p->sink->name);
 
 		next_sink = p->sink;
+
+		if (!is_skl_dsp_widget_type(p->sink))
+			return skl_tplg_bind_sinks(p->sink, skl, src_w, src_mconfig);
+
 		/*
 		 * here we will check widgets in sink pipelines, so that
 		 * can be any widgets type and we are only interested if
@@ -576,7 +597,7 @@ static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w,
 	}
 
 	if (!sink)
-		return skl_tplg_bind_sinks(next_sink, skl, src_mconfig);
+		return skl_tplg_bind_sinks(next_sink, skl, src_w, src_mconfig);
 
 	return 0;
 }
@@ -605,7 +626,7 @@ static int skl_tplg_pga_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w,
 	 * if sink is not started, start sink pipe first, then start
 	 * this pipe
 	 */
-	ret = skl_tplg_bind_sinks(w, skl, src_mconfig);
+	ret = skl_tplg_bind_sinks(w, skl, w, src_mconfig);
 	if (ret)
 		return ret;
 
@@ -773,10 +794,7 @@ static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
 			continue;
 		}
 
-		ret = skl_unbind_modules(ctx, src_module, dst_module);
-		if (ret < 0)
-			return ret;
-
+		skl_unbind_modules(ctx, src_module, dst_module);
 		src_module = dst_module;
 	}
 
@@ -814,9 +832,6 @@ static int skl_tplg_pga_dapm_post_pmd_event(struct snd_soc_dapm_widget *w,
 			 * This is a connecter and if path is found that means
 			 * unbind between source and sink has not happened yet
 			 */
-			ret = skl_stop_pipe(ctx, sink_mconfig->pipe);
-			if (ret < 0)
-				return ret;
 			ret = skl_unbind_modules(ctx, src_mconfig,
 							sink_mconfig);
 		}
@@ -842,6 +857,12 @@ static int skl_tplg_vmixer_event(struct snd_soc_dapm_widget *w,
 	case SND_SOC_DAPM_PRE_PMU:
 		return skl_tplg_mixer_dapm_pre_pmu_event(w, skl);
 
+	case SND_SOC_DAPM_POST_PMU:
+		return skl_tplg_mixer_dapm_post_pmu_event(w, skl);
+
+	case SND_SOC_DAPM_PRE_PMD:
+		return skl_tplg_mixer_dapm_pre_pmd_event(w, skl);
+
 	case SND_SOC_DAPM_POST_PMD:
 		return skl_tplg_mixer_dapm_post_pmd_event(w, skl);
 	}
@@ -916,6 +937,13 @@ static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol,
 		skl_get_module_params(skl->skl_sst, (u32 *)bc->params,
 				      bc->max, bc->param_id, mconfig);
 
+	/* decrement size for TLV header */
+	size -= 2 * sizeof(u32);
+
+	/* check size as we don't want to send kernel data */
+	if (size > bc->max)
+		size = bc->max;
+
 	if (bc->params) {
 		if (copy_to_user(data, &bc->param_id, sizeof(u32)))
 			return -EFAULT;
@@ -1510,6 +1538,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus)
 					&skl_tplg_ops, fw, 0);
 	if (ret < 0) {
 		dev_err(bus->dev, "tplg component load failed%d\n", ret);
+		release_firmware(fw);
 		return -EINVAL;
 	}
 
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 443a15d..092705e 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -614,8 +614,6 @@ static int skl_probe(struct pci_dev *pci,
 		goto out_unregister;
 
 	/*configure PM */
-	pm_runtime_set_autosuspend_delay(bus->dev, SKL_SUSPEND_DELAY);
-	pm_runtime_use_autosuspend(bus->dev);
 	pm_runtime_put_noidle(bus->dev);
 	pm_runtime_allow(bus->dev);
 
diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig
index 15c04e2..9769676 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/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index c866ade..a6c7b8d 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -381,9 +381,19 @@ static int mxs_saif_startup(struct snd_pcm_substream *substream,
 	__raw_writel(BM_SAIF_CTRL_CLKGATE,
 		saif->base + SAIF_CTRL + MXS_CLR_ADDR);
 
+	clk_prepare(saif->clk);
+
 	return 0;
 }
 
+static void mxs_saif_shutdown(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *cpu_dai)
+{
+	struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+	clk_unprepare(saif->clk);
+}
+
 /*
  * Should only be called when port is inactive.
  * although can be called multiple times by upper layers.
@@ -424,8 +434,6 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
-	/* prepare clk in hw_param, enable in trigger */
-	clk_prepare(saif->clk);
 	if (saif != master_saif) {
 		/*
 		* Set an initial clock rate for the saif internal logic to work
@@ -611,6 +619,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd,
 
 static const struct snd_soc_dai_ops mxs_saif_dai_ops = {
 	.startup = mxs_saif_startup,
+	.shutdown = mxs_saif_shutdown,
 	.trigger = mxs_saif_trigger,
 	.prepare = mxs_saif_prepare,
 	.hw_params = mxs_saif_hw_params,
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index 79688aa..4aeb8e1 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -440,18 +440,18 @@ static irqreturn_t lpass_platform_lpaif_irq(int irq, void *data)
 }
 
 static int lpass_platform_alloc_buffer(struct snd_pcm_substream *substream,
-		struct snd_soc_pcm_runtime *soc_runtime)
+		struct snd_soc_pcm_runtime *rt)
 {
 	struct snd_dma_buffer *buf = &substream->dma_buffer;
 	size_t size = lpass_platform_pcm_hardware.buffer_bytes_max;
 
 	buf->dev.type = SNDRV_DMA_TYPE_DEV;
-	buf->dev.dev = soc_runtime->dev;
+	buf->dev.dev = rt->platform->dev;
 	buf->private_data = NULL;
-	buf->area = dma_alloc_coherent(soc_runtime->dev, size, &buf->addr,
+	buf->area = dma_alloc_coherent(rt->platform->dev, size, &buf->addr,
 			GFP_KERNEL);
 	if (!buf->area) {
-		dev_err(soc_runtime->dev, "%s: Could not allocate DMA buffer\n",
+		dev_err(rt->platform->dev, "%s: Could not allocate DMA buffer\n",
 				__func__);
 		return -ENOMEM;
 	}
@@ -461,12 +461,12 @@ static int lpass_platform_alloc_buffer(struct snd_pcm_substream *substream,
 }
 
 static void lpass_platform_free_buffer(struct snd_pcm_substream *substream,
-		struct snd_soc_pcm_runtime *soc_runtime)
+		struct snd_soc_pcm_runtime *rt)
 {
 	struct snd_dma_buffer *buf = &substream->dma_buffer;
 
 	if (buf->area) {
-		dma_free_coherent(soc_runtime->dev, buf->bytes, buf->area,
+		dma_free_coherent(rt->dev, buf->bytes, buf->area,
 				buf->addr);
 	}
 	buf->area = NULL;
@@ -499,9 +499,6 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime)
 
 	snd_soc_pcm_set_drvdata(soc_runtime, data);
 
-	soc_runtime->dev->coherent_dma_mask = DMA_BIT_MASK(32);
-	soc_runtime->dev->dma_mask = &soc_runtime->dev->coherent_dma_mask;
-
 	ret = lpass_platform_alloc_buffer(substream, soc_runtime);
 	if (ret)
 		return ret;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 5a2812f..0d37079 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -310,7 +310,7 @@ struct dapm_kcontrol_data {
 };
 
 static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
-	struct snd_kcontrol *kcontrol)
+	struct snd_kcontrol *kcontrol, const char *ctrl_name)
 {
 	struct dapm_kcontrol_data *data;
 	struct soc_mixer_control *mc;
@@ -333,7 +333,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 		if (mc->autodisable) {
 			struct snd_soc_dapm_widget template;
 
-			name = kasprintf(GFP_KERNEL, "%s %s", kcontrol->id.name,
+			name = kasprintf(GFP_KERNEL, "%s %s", ctrl_name,
 					 "Autodisable");
 			if (!name) {
 				ret = -ENOMEM;
@@ -371,7 +371,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 		if (e->autodisable) {
 			struct snd_soc_dapm_widget template;
 
-			name = kasprintf(GFP_KERNEL, "%s %s", kcontrol->id.name,
+			name = kasprintf(GFP_KERNEL, "%s %s", ctrl_name,
 					 "Autodisable");
 			if (!name) {
 				ret = -ENOMEM;
@@ -871,7 +871,7 @@ static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w,
 
 		kcontrol->private_free = dapm_kcontrol_free;
 
-		ret = dapm_kcontrol_data_alloc(w, kcontrol);
+		ret = dapm_kcontrol_data_alloc(w, kcontrol, name);
 		if (ret) {
 			snd_ctl_free_one(kcontrol);
 			goto exit_free;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e898b42..1af4f23 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1810,7 +1810,8 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
 		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
 		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
 		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) &&
-		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) &&
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND))
 			continue;
 
 		dev_dbg(be->dev, "ASoC: hw_free BE %s\n",
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index b79875e..47de8af 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -2458,7 +2458,6 @@ int __snd_usbmidi_create(struct snd_card *card,
 	else
 		err = snd_usbmidi_create_endpoints(umidi, endpoints);
 	if (err < 0) {
-		snd_usbmidi_free(umidi);
 		return err;
 	}