Merge git://git.kernel.org/pub/scm/linux/kernel/git/davem/net

Several conflicts, seemingly all over the place.

I used Stephen Rothwell's sample resolutions for many of these, if not
just to double check my own work, so definitely the credit largely
goes to him.

The NFP conflict consisted of a bug fix (moving operations
past the rhashtable operation) while chaning the initial
argument in the function call in the moved code.

The net/dsa/master.c conflict had to do with a bug fix intermixing of
making dsa_master_set_mtu() static with the fixing of the tagging
attribute location.

cls_flower had a conflict because the dup reject fix from Or
overlapped with the addition of port range classifiction.

__set_phy_supported()'s conflict was relatively easy to resolve
because Andrew fixed it in both trees, so it was just a matter
of taking the net-next copy.  Or at least I think it was :-)

Joe Stringer's fix to the handling of netns id 0 in bpf_sk_lookup()
intermixed with changes on how the sdif and caller_net are calculated
in these code paths in net-next.

The remaining BPF conflicts were largely about the addition of the
__bpf_md_ptr stuff in 'net' overlapping with adjustments and additions
to the relevant data structure where the MD pointer macros are used.

Signed-off-by: David S. Miller <davem@davemloft.net>

34 files changed, 498 insertions, 309 deletions

diff --git a/sound/core/control.c b/sound/core/control.c
index 9aa15bfc7936..649d3217590e 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -348,6 +348,40 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count)
 	return 0;
 }
 
+/* add a new kcontrol object; call with card->controls_rwsem locked */
+static int __snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol)
+{
+	struct snd_ctl_elem_id id;
+	unsigned int idx;
+	unsigned int count;
+
+	id = kcontrol->id;
+	if (id.index > UINT_MAX - kcontrol->count)
+		return -EINVAL;
+
+	if (snd_ctl_find_id(card, &id)) {
+		dev_err(card->dev,
+			"control %i:%i:%i:%s:%i is already present\n",
+			id.iface, id.device, id.subdevice, id.name, id.index);
+		return -EBUSY;
+	}
+
+	if (snd_ctl_find_hole(card, kcontrol->count) < 0)
+		return -ENOMEM;
+
+	list_add_tail(&kcontrol->list, &card->controls);
+	card->controls_count += kcontrol->count;
+	kcontrol->id.numid = card->last_numid + 1;
+	card->last_numid += kcontrol->count;
+
+	id = kcontrol->id;
+	count = kcontrol->count;
+	for (idx = 0; idx < count; idx++, id.index++, id.numid++)
+		snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id);
+
+	return 0;
+}
+
 /**
  * snd_ctl_add - add the control instance to the card
  * @card: the card instance
@@ -364,45 +398,18 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count)
  */
 int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol)
 {
-	struct snd_ctl_elem_id id;
-	unsigned int idx;
-	unsigned int count;
 	int err = -EINVAL;
 
 	if (! kcontrol)
 		return err;
 	if (snd_BUG_ON(!card || !kcontrol->info))
 		goto error;
-	id = kcontrol->id;
-	if (id.index > UINT_MAX - kcontrol->count)
-		goto error;
 
 	down_write(&card->controls_rwsem);
-	if (snd_ctl_find_id(card, &id)) {
-		up_write(&card->controls_rwsem);
-		dev_err(card->dev, "control %i:%i:%i:%s:%i is already present\n",
-					id.iface,
-					id.device,
-					id.subdevice,
-					id.name,
-					id.index);
-		err = -EBUSY;
-		goto error;
-	}
-	if (snd_ctl_find_hole(card, kcontrol->count) < 0) {
-		up_write(&card->controls_rwsem);
-		err = -ENOMEM;
-		goto error;
-	}
-	list_add_tail(&kcontrol->list, &card->controls);
-	card->controls_count += kcontrol->count;
-	kcontrol->id.numid = card->last_numid + 1;
-	card->last_numid += kcontrol->count;
-	id = kcontrol->id;
-	count = kcontrol->count;
+	err = __snd_ctl_add(card, kcontrol);
 	up_write(&card->controls_rwsem);
-	for (idx = 0; idx < count; idx++, id.index++, id.numid++)
-		snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id);
+	if (err < 0)
+		goto error;
 	return 0;
 
  error:
@@ -1361,9 +1368,12 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
 		kctl->tlv.c = snd_ctl_elem_user_tlv;
 
 	/* This function manage to free the instance on failure. */
-	err = snd_ctl_add(card, kctl);
-	if (err < 0)
-		return err;
+	down_write(&card->controls_rwsem);
+	err = __snd_ctl_add(card, kctl);
+	if (err < 0) {
+		snd_ctl_free_one(kctl);
+		goto unlock;
+	}
 	offset = snd_ctl_get_ioff(kctl, &info->id);
 	snd_ctl_build_ioff(&info->id, kctl, offset);
 	/*
@@ -1374,10 +1384,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
 	 * which locks the element.
 	 */
 
-	down_write(&card->controls_rwsem);
 	card->user_ctl_count++;
-	up_write(&card->controls_rwsem);
 
+ unlock:
+	up_write(&card->controls_rwsem);
 	return 0;
 }
 
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 66c90f486af9..818dff1de545 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -36,6 +36,7 @@
 #include <sound/timer.h>
 #include <sound/minors.h>
 #include <linux/uio.h>
+#include <linux/delay.h>
 
 #include "pcm_local.h"
 
@@ -91,12 +92,12 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem);
  * 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.
+ * sleep until all the readers are completed without blocking by writer.
  */
-static inline void down_write_nonblock(struct rw_semaphore *lock)
+static inline void down_write_nonfifo(struct rw_semaphore *lock)
 {
 	while (!down_write_trylock(lock))
-		cond_resched();
+		msleep(1);
 }
 
 #define PCM_LOCK_DEFAULT	0
@@ -1967,7 +1968,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
 		res = -ENOMEM;
 		goto _nolock;
 	}
-	down_write_nonblock(&snd_pcm_link_rwsem);
+	down_write_nonfifo(&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 ||
@@ -2014,7 +2015,7 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream)
 	struct snd_pcm_substream *s;
 	int res = 0;
 
-	down_write_nonblock(&snd_pcm_link_rwsem);
+	down_write_nonfifo(&snd_pcm_link_rwsem);
 	write_lock_irq(&snd_pcm_link_rwlock);
 	if (!snd_pcm_stream_linked(substream)) {
 		res = -EALREADY;
@@ -2369,7 +2370,8 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
 
 static void pcm_release_private(struct snd_pcm_substream *substream)
 {
-	snd_pcm_unlink(substream);
+	if (snd_pcm_stream_linked(substream))
+		snd_pcm_unlink(substream);
 }
 
 void snd_pcm_release_substream(struct snd_pcm_substream *substream)
diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 32453f81b95a..3a5008837576 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -1531,7 +1531,6 @@ static int snd_wss_playback_open(struct snd_pcm_substream *substream)
 	if (err < 0) {
 		if (chip->release_dma)
 			chip->release_dma(chip, chip->dma_private_data, chip->dma1);
-		snd_free_pages(runtime->dma_area, runtime->dma_bytes);
 		return err;
 	}
 	chip->playback_substream = substream;
@@ -1572,7 +1571,6 @@ static int snd_wss_capture_open(struct snd_pcm_substream *substream)
 	if (err < 0) {
 		if (chip->release_dma)
 			chip->release_dma(chip, chip->dma_private_data, chip->dma2);
-		snd_free_pages(runtime->dma_area, runtime->dma_bytes);
 		return err;
 	}
 	chip->capture_substream = substream;
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index f4459d1a9d67..27b468f057dd 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -824,7 +824,7 @@ static int snd_ac97_put_spsa(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
 {
 	struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
 	int reg = kcontrol->private_value & 0xff;
-	int shift = (kcontrol->private_value >> 8) & 0xff;
+	int shift = (kcontrol->private_value >> 8) & 0x0f;
 	int mask = (kcontrol->private_value >> 16) & 0xff;
 	// int invert = (kcontrol->private_value >> 24) & 0xff;
 	unsigned short value, old, new;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index d8eb2b5f51ae..76f03abd15ab 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2169,6 +2169,8 @@ static struct snd_pci_quirk power_save_blacklist[] = {
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
 	SND_PCI_QUIRK(0x1849, 0xc892, "Asrock B85M-ITX", 0),
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
+	SND_PCI_QUIRK(0x1849, 0x0397, "Asrock N68C-S UCC", 0),
+	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
 	SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0),
 	/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
 	SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0),
@@ -2496,6 +2498,10 @@ static const struct pci_device_id azx_ids[] = {
 	/* AMD Hudson */
 	{ PCI_DEVICE(0x1022, 0x780d),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
+	/* AMD Stoney */
+	{ PCI_DEVICE(0x1022, 0x157a),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
+			 AZX_DCAPS_PM_RUNTIME },
 	/* AMD Raven */
 	{ PCI_DEVICE(0x1022, 0x15e3),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 970bc44a378b..8d75597028ee 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -388,6 +388,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 	case 0x10ec0285:
 	case 0x10ec0298:
 	case 0x10ec0289:
+	case 0x10ec0300:
 		alc_update_coef_idx(codec, 0x10, 1<<9, 0);
 		break;
 	case 0x10ec0275:
@@ -2830,6 +2831,7 @@ enum {
 	ALC269_TYPE_ALC215,
 	ALC269_TYPE_ALC225,
 	ALC269_TYPE_ALC294,
+	ALC269_TYPE_ALC300,
 	ALC269_TYPE_ALC700,
 };
 
@@ -2864,6 +2866,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	case ALC269_TYPE_ALC215:
 	case ALC269_TYPE_ALC225:
 	case ALC269_TYPE_ALC294:
+	case ALC269_TYPE_ALC300:
 	case ALC269_TYPE_ALC700:
 		ssids = alc269_ssids;
 		break;
@@ -4985,9 +4988,18 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec,
 		{ 0x19, 0x21a11010 }, /* dock mic */
 		{ }
 	};
+	/* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise
+	 * the speaker output becomes too low by some reason on Thinkpads with
+	 * ALC298 codec
+	 */
+	static hda_nid_t preferred_pairs[] = {
+		0x14, 0x03, 0x17, 0x02, 0x21, 0x02,
+		0
+	};
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->gen.preferred_dacs = preferred_pairs;
 		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 		snd_hda_apply_pincfgs(codec, pincfgs);
 	} else if (action == HDA_FIXUP_ACT_INIT) {
@@ -5358,6 +5370,16 @@ static void alc274_fixup_bind_dacs(struct hda_codec *codec,
 	spec->gen.preferred_dacs = preferred_pairs;
 }
 
+/* The DAC of NID 0x3 will introduce click/pop noise on headphones, so invalidate it */
+static void alc285_fixup_invalidate_dacs(struct hda_codec *codec,
+			      const struct hda_fixup *fix, int action)
+{
+	if (action != HDA_FIXUP_ACT_PRE_PROBE)
+		return;
+
+	snd_hda_override_wcaps(codec, 0x03, 0);
+}
+
 /* for hda_fixup_thinkpad_acpi() */
 #include "thinkpad_helper.c"
 
@@ -5495,6 +5517,9 @@ enum {
 	ALC255_FIXUP_DELL_HEADSET_MIC,
 	ALC295_FIXUP_HP_X360,
 	ALC221_FIXUP_HP_HEADSET_MIC,
+	ALC285_FIXUP_LENOVO_HEADPHONE_NOISE,
+	ALC295_FIXUP_HP_AUTO_MUTE,
+	ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -5659,6 +5684,8 @@ static const struct hda_fixup alc269_fixups[] = {
 	[ALC269_FIXUP_HP_MUTE_LED_MIC3] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc269_fixup_hp_mute_led_mic3,
+		.chained = true,
+		.chain_id = ALC295_FIXUP_HP_AUTO_MUTE
 	},
 	[ALC269_FIXUP_HP_GPIO_LED] = {
 		.type = HDA_FIXUP_FUNC,
@@ -6362,6 +6389,23 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MIC
 	},
+	[ALC285_FIXUP_LENOVO_HEADPHONE_NOISE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_invalidate_dacs,
+	},
+	[ALC295_FIXUP_HP_AUTO_MUTE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_auto_mute_via_amp,
+	},
+	[ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -6376,7 +6420,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
 	SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
 	SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
+	SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
+	SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
 	SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
@@ -6532,6 +6580,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
 	SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
@@ -7034,6 +7083,15 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x12, 0x90a60130},
 		{0x19, 0x03a11020},
 		{0x21, 0x0321101f}),
+	SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_HEADPHONE_NOISE,
+		{0x12, 0x90a60130},
+		{0x14, 0x90170110},
+		{0x19, 0x04a11040},
+		{0x21, 0x04211020}),
+	SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
+		{0x12, 0x90a60130},
+		{0x17, 0x90170110},
+		{0x21, 0x02211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE,
 		{0x12, 0x90a60120},
 		{0x14, 0x90170110},
@@ -7295,6 +7353,10 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->gen.mixer_nid = 0; /* ALC2x4 does not have any loopback mixer path */
 		alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); /* UAJ MIC Vref control by verb */
 		break;
+	case 0x10ec0300:
+		spec->codec_variant = ALC269_TYPE_ALC300;
+		spec->gen.mixer_nid = 0; /* no loopback on ALC300 */
+		break;
 	case 0x10ec0700:
 	case 0x10ec0701:
 	case 0x10ec0703:
@@ -8405,6 +8467,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
 	HDA_CODEC_ENTRY(0x10ec0295, "ALC295", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269),
+	HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269),
 	HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861),
 	HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd),
 	HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861),
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 4e9854889a95..e63d6e33df48 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -2187,11 +2187,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
 	 */
 	snd_hdac_codec_read(hdev, hdev->afg, 0,	AC_VERB_SET_POWER_STATE,
 							AC_PWRST_D3);
-	err = snd_hdac_display_power(bus, false);
-	if (err < 0) {
-		dev_err(dev, "Cannot turn on display power on i915\n");
-		return err;
-	}
 
 	hlink = snd_hdac_ext_bus_get_link(bus, dev_name(dev));
 	if (!hlink) {
@@ -2201,7 +2196,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev)
 
 	snd_hdac_ext_bus_link_put(bus, hlink);
 
-	return 0;
+	err = snd_hdac_display_power(bus, false);
+	if (err < 0)
+		dev_err(dev, "Cannot turn off display power on i915\n");
+
+	return err;
 }
 
 static int hdac_hdmi_runtime_resume(struct device *dev)
diff --git a/sound/soc/codecs/pcm186x.h b/sound/soc/codecs/pcm186x.h
index 2c6ba55bf394..bb3f0c42a1cd 100644
--- a/sound/soc/codecs/pcm186x.h
+++ b/sound/soc/codecs/pcm186x.h
@@ -139,7 +139,7 @@ enum pcm186x_type {
 #define PCM186X_MAX_REGISTER		PCM186X_CURR_TRIM_CTRL
 
 /* PCM186X_PAGE */
-#define PCM186X_RESET			0xff
+#define PCM186X_RESET			0xfe
 
 /* PCM186X_ADCX_INPUT_SEL_X */
 #define PCM186X_ADC_INPUT_SEL_POL	BIT(7)
diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c
index 494d9d662be8..771b46e1974b 100644
--- a/sound/soc/codecs/pcm3060.c
+++ b/sound/soc/codecs/pcm3060.c
@@ -198,20 +198,16 @@ static const struct snd_kcontrol_new pcm3060_dapm_controls[] = {
 };
 
 static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = {
-	SND_SOC_DAPM_OUTPUT("OUTL+"),
-	SND_SOC_DAPM_OUTPUT("OUTR+"),
-	SND_SOC_DAPM_OUTPUT("OUTL-"),
-	SND_SOC_DAPM_OUTPUT("OUTR-"),
+	SND_SOC_DAPM_OUTPUT("OUTL"),
+	SND_SOC_DAPM_OUTPUT("OUTR"),
 
 	SND_SOC_DAPM_INPUT("INL"),
 	SND_SOC_DAPM_INPUT("INR"),
 };
 
 static const struct snd_soc_dapm_route pcm3060_dapm_map[] = {
-	{ "OUTL+", NULL, "Playback" },
-	{ "OUTR+", NULL, "Playback" },
-	{ "OUTL-", NULL, "Playback" },
-	{ "OUTR-", NULL, "Playback" },
+	{ "OUTL", NULL, "Playback" },
+	{ "OUTR", NULL, "Playback" },
 
 	{ "Capture", NULL, "INL" },
 	{ "Capture", NULL, "INR" },
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index a53dc174bbf0..66501b8dc46f 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -765,38 +765,41 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem,
 
 static void wm_adsp2_show_fw_status(struct wm_adsp *dsp)
 {
-	u16 scratch[4];
+	unsigned int scratch[4];
+	unsigned int addr = dsp->base + ADSP2_SCRATCH0;
+	unsigned int i;
 	int ret;
 
-	ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2_SCRATCH0,
-				scratch, sizeof(scratch));
-	if (ret) {
-		adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret);
-		return;
+	for (i = 0; i < ARRAY_SIZE(scratch); ++i) {
+		ret = regmap_read(dsp->regmap, addr + i, &scratch[i]);
+		if (ret) {
+			adsp_err(dsp, "Failed to read SCRATCH%u: %d\n", i, ret);
+			return;
+		}
 	}
 
 	adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n",
-		 be16_to_cpu(scratch[0]),
-		 be16_to_cpu(scratch[1]),
-		 be16_to_cpu(scratch[2]),
-		 be16_to_cpu(scratch[3]));
+		 scratch[0], scratch[1], scratch[2], scratch[3]);
 }
 
 static void wm_adsp2v2_show_fw_status(struct wm_adsp *dsp)
 {
-	u32 scratch[2];
+	unsigned int scratch[2];
 	int ret;
 
-	ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1,
-			      scratch, sizeof(scratch));
-
+	ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1,
+			  &scratch[0]);
 	if (ret) {
-		adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret);
+		adsp_err(dsp, "Failed to read SCRATCH0_1: %d\n", ret);
 		return;
 	}
 
-	scratch[0] = be32_to_cpu(scratch[0]);
-	scratch[1] = be32_to_cpu(scratch[1]);
+	ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH2_3,
+			  &scratch[1]);
+	if (ret) {
+		adsp_err(dsp, "Failed to read SCRATCH2_3: %d\n", ret);
+		return;
+	}
 
 	adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n",
 		 scratch[0] & 0xFFFF,
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 0caa1f4eb94d..18e717703685 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -101,22 +101,42 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
 	  codec, then enable this option by saying Y or m. This is a
 	  recommended option
 
-config SND_SOC_INTEL_SKYLAKE_SSP_CLK
-	tristate
-
 config SND_SOC_INTEL_SKYLAKE
 	tristate "SKL/BXT/KBL/GLK/CNL... Platforms"
 	depends on PCI && ACPI
+	select SND_SOC_INTEL_SKYLAKE_COMMON
+	help
+	  If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/
+	  GeminiLake or CannonLake platform with the DSP enabled in the BIOS
+	  then enable this option by saying Y or m.
+
+if  SND_SOC_INTEL_SKYLAKE
+
+config SND_SOC_INTEL_SKYLAKE_SSP_CLK
+	tristate
+
+config SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
+	bool "HDAudio codec support"
+	help
+	  If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/
+	  GeminiLake or CannonLake platform with an HDaudio codec
+	  then enable this option by saying Y
+
+config SND_SOC_INTEL_SKYLAKE_COMMON
+	tristate
 	select SND_HDA_EXT_CORE
 	select SND_HDA_DSP_LOADER
 	select SND_SOC_TOPOLOGY
 	select SND_SOC_INTEL_SST
+	select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
 	select SND_SOC_ACPI_INTEL_MATCH
 	help
 	  If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/
 	  GeminiLake or CannonLake platform with the DSP enabled in the BIOS
 	  then enable this option by saying Y or m.
 
+endif ## SND_SOC_INTEL_SKYLAKE
+
 config SND_SOC_ACPI_INTEL_MATCH
 	tristate
 	select SND_SOC_ACPI if ACPI
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 73ca1350aa31..b177db2a0dbb 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -293,16 +293,6 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH
 	  Say Y if you have such a device.
 	  If unsure select "N".
 
-config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
-	tristate "SKL/KBL/BXT/APL with HDA Codecs"
-	select SND_SOC_HDAC_HDMI
-	select SND_SOC_HDAC_HDA
-	help
-	  This adds support for ASoC machine driver for Intel platforms
-	  SKL/KBL/BXT/APL with iDisp, HDA audio codecs.
-          Say Y or m if you have such a device. This is a recommended option.
-	  If unsure select "N".
-
 config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
 	tristate "GLK with RT5682 and MAX98357A in I2S Mode"
 	depends on MFD_INTEL_LPSS && I2C && ACPI
@@ -319,4 +309,18 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
 
 endif ## SND_SOC_INTEL_SKYLAKE
 
+if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
+
+config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
+	tristate "SKL/KBL/BXT/APL with HDA Codecs"
+	select SND_SOC_HDAC_HDMI
+	# SND_SOC_HDAC_HDA is already selected
+	help
+	  This adds support for ASoC machine driver for Intel platforms
+	  SKL/KBL/BXT/APL with iDisp, HDA audio codecs.
+          Say Y or m if you have such a device. This is a recommended option.
+	  If unsure select "N".
+
+endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC
+
 endif ## SND_SOC_INTEL_MACH
diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index db6976f4ddaa..9d9f6e41d81c 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -19,6 +19,7 @@
  * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
  */
 
+#include <linux/dmi.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
@@ -35,6 +36,8 @@
 #define CHT_PLAT_CLK_3_HZ	19200000
 #define CHT_CODEC_DAI	"HiFi"
 
+#define QUIRK_PMC_PLT_CLK_0				0x01
+
 struct cht_mc_private {
 	struct clk *mclk;
 	struct snd_soc_jack jack;
@@ -385,11 +388,29 @@ static struct snd_soc_card snd_soc_card_cht = {
 	.num_controls = ARRAY_SIZE(cht_mc_controls),
 };
 
+static const struct dmi_system_id cht_max98090_quirk_table[] = {
+	{
+		/* Swanky model Chromebook (Toshiba Chromebook 2) */
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Swanky"),
+		},
+		.driver_data = (void *)QUIRK_PMC_PLT_CLK_0,
+	},
+	{}
+};
+
 static int snd_cht_mc_probe(struct platform_device *pdev)
 {
+	const struct dmi_system_id *dmi_id;
 	struct device *dev = &pdev->dev;
 	int ret_val = 0;
 	struct cht_mc_private *drv;
+	const char *mclk_name;
+	int quirks = 0;
+
+	dmi_id = dmi_first_match(cht_max98090_quirk_table);
+	if (dmi_id)
+		quirks = (unsigned long)dmi_id->driver_data;
 
 	drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL);
 	if (!drv)
@@ -411,11 +432,16 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
 	snd_soc_card_cht.dev = &pdev->dev;
 	snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
 
-	drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
+	if (quirks & QUIRK_PMC_PLT_CLK_0)
+		mclk_name = "pmc_plt_clk_0";
+	else
+		mclk_name = "pmc_plt_clk_3";
+
+	drv->mclk = devm_clk_get(&pdev->dev, mclk_name);
 	if (IS_ERR(drv->mclk)) {
 		dev_err(&pdev->dev,
-			"Failed to get MCLK from pmc_plt_clk_3: %ld\n",
-			PTR_ERR(drv->mclk));
+			"Failed to get MCLK from %s: %ld\n",
+			mclk_name, PTR_ERR(drv->mclk));
 		return PTR_ERR(drv->mclk);
 	}
 
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 29225623b4b4..7487f388e65d 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -37,7 +37,9 @@
 #include "skl.h"
 #include "skl-sst-dsp.h"
 #include "skl-sst-ipc.h"
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
 #include "../../../soc/codecs/hdac_hda.h"
+#endif
 
 /*
  * initialize the PCI registers
@@ -658,6 +660,8 @@ static void skl_clock_device_unregister(struct skl *skl)
 		platform_device_unregister(skl->clk_dev);
 }
 
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
+
 #define IDISP_INTEL_VENDOR_ID	0x80860000
 
 /*
@@ -676,6 +680,8 @@ static void load_codec_module(struct hda_codec *codec)
 #endif
 }
 
+#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */
+
 /*
  * Probe the given codec address
  */
@@ -685,9 +691,11 @@ static int probe_codec(struct hdac_bus *bus, int addr)
 		(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
 	unsigned int res = -1;
 	struct skl *skl = bus_to_skl(bus);
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
 	struct hdac_hda_priv *hda_codec;
-	struct hdac_device *hdev;
 	int err;
+#endif
+	struct hdac_device *hdev;
 
 	mutex_lock(&bus->cmd_mutex);
 	snd_hdac_bus_send_cmd(bus, cmd);
@@ -697,6 +705,7 @@ static int probe_codec(struct hdac_bus *bus, int addr)
 		return -EIO;
 	dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res);
 
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
 	hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec),
 				 GFP_KERNEL);
 	if (!hda_codec)
@@ -715,6 +724,13 @@ static int probe_codec(struct hdac_bus *bus, int addr)
 		load_codec_module(&hda_codec->codec);
 	}
 	return 0;
+#else
+	hdev = devm_kzalloc(&skl->pci->dev, sizeof(*hdev), GFP_KERNEL);
+	if (!hdev)
+		return -ENOMEM;
+
+	return snd_hdac_ext_bus_device_init(bus, addr, hdev);
+#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */
 }
 
 /* Codec initialization */
@@ -815,6 +831,12 @@ static void skl_probe_work(struct work_struct *work)
 		}
 	}
 
+	/*
+	 * we are done probing so decrement link counts
+	 */
+	list_for_each_entry(hlink, &bus->hlink_list, list)
+		snd_hdac_ext_bus_link_put(bus, hlink);
+
 	if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) {
 		err = snd_hdac_display_power(bus, false);
 		if (err < 0) {
@@ -824,12 +846,6 @@ static void skl_probe_work(struct work_struct *work)
 		}
 	}
 
-	/*
-	 * we are done probing so decrement link counts
-	 */
-	list_for_each_entry(hlink, &bus->hlink_list, list)
-		snd_hdac_ext_bus_link_put(bus, hlink);
-
 	/* configure PM */
 	pm_runtime_put_noidle(bus->dev);
 	pm_runtime_allow(bus->dev);
@@ -870,7 +886,7 @@ static int skl_create(struct pci_dev *pci,
 	hbus = skl_to_hbus(skl);
 	bus = skl_to_bus(skl);
 
-#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDA)
+#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC)
 	ext_ops = snd_soc_hdac_hda_get_ops();
 #endif
 	snd_hdac_ext_bus_init(bus, &pci->dev, &bus_core_ops, io_ops, ext_ops);
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index d5ae9eb8c756..fed45b41f9d3 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -36,6 +36,8 @@
 #include "../codecs/twl6040.h"
 
 struct abe_twl6040 {
+	struct snd_soc_card card;
+	struct snd_soc_dai_link dai_links[2];
 	int	jack_detection;	/* board can detect jack events */
 	int	mclk_freq;	/* MCLK frequency speed for twl6040 */
 };
@@ -208,40 +210,10 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
 				ARRAY_SIZE(dmic_audio_map));
 }
 
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
-	{
-		.name = "TWL6040",
-		.stream_name = "TWL6040",
-		.codec_dai_name = "twl6040-legacy",
-		.codec_name = "twl6040-codec",
-		.init = omap_abe_twl6040_init,
-		.ops = &omap_abe_ops,
-	},
-	{
-		.name = "DMIC",
-		.stream_name = "DMIC Capture",
-		.codec_dai_name = "dmic-hifi",
-		.codec_name = "dmic-codec",
-		.init = omap_abe_dmic_init,
-		.ops = &omap_abe_dmic_ops,
-	},
-};
-
-/* Audio machine driver */
-static struct snd_soc_card omap_abe_card = {
-	.owner = THIS_MODULE,
-
-	.dapm_widgets = twl6040_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
-	.dapm_routes = audio_map,
-	.num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
 static int omap_abe_probe(struct platform_device *pdev)
 {
 	struct device_node *node = pdev->dev.of_node;
-	struct snd_soc_card *card = &omap_abe_card;
+	struct snd_soc_card *card;
 	struct device_node *dai_node;
 	struct abe_twl6040 *priv;
 	int num_links = 0;
@@ -252,12 +224,18 @@ static int omap_abe_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 
-	card->dev = &pdev->dev;
-
 	priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
 	if (priv == NULL)
 		return -ENOMEM;
 
+	card = &priv->card;
+	card->dev = &pdev->dev;
+	card->owner = THIS_MODULE;
+	card->dapm_widgets = twl6040_dapm_widgets;
+	card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets);
+	card->dapm_routes = audio_map;
+	card->num_dapm_routes = ARRAY_SIZE(audio_map);
+
 	if (snd_soc_of_parse_card_name(card, "ti,model")) {
 		dev_err(&pdev->dev, "Card name is not provided\n");
 		return -ENODEV;
@@ -274,14 +252,27 @@ static int omap_abe_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "McPDM node is not provided\n");
 		return -EINVAL;
 	}
-	abe_twl6040_dai_links[0].cpu_of_node = dai_node;
-	abe_twl6040_dai_links[0].platform_of_node = dai_node;
+
+	priv->dai_links[0].name = "DMIC";
+	priv->dai_links[0].stream_name = "TWL6040";
+	priv->dai_links[0].cpu_of_node = dai_node;
+	priv->dai_links[0].platform_of_node = dai_node;
+	priv->dai_links[0].codec_dai_name = "twl6040-legacy";
+	priv->dai_links[0].codec_name = "twl6040-codec";
+	priv->dai_links[0].init = omap_abe_twl6040_init;
+	priv->dai_links[0].ops = &omap_abe_ops;
 
 	dai_node = of_parse_phandle(node, "ti,dmic", 0);
 	if (dai_node) {
 		num_links = 2;
-		abe_twl6040_dai_links[1].cpu_of_node = dai_node;
-		abe_twl6040_dai_links[1].platform_of_node = dai_node;
+		priv->dai_links[1].name = "TWL6040";
+		priv->dai_links[1].stream_name = "DMIC Capture";
+		priv->dai_links[1].cpu_of_node = dai_node;
+		priv->dai_links[1].platform_of_node = dai_node;
+		priv->dai_links[1].codec_dai_name = "dmic-hifi";
+		priv->dai_links[1].codec_name = "dmic-codec";
+		priv->dai_links[1].init = omap_abe_dmic_init;
+		priv->dai_links[1].ops = &omap_abe_dmic_ops;
 	} else {
 		num_links = 1;
 	}
@@ -300,7 +291,7 @@ static int omap_abe_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 
-	card->dai_link = abe_twl6040_dai_links;
+	card->dai_link = priv->dai_links;
 	card->num_links = num_links;
 
 	snd_soc_card_set_drvdata(card, priv);
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index fe966272bd0c..cba9645b6487 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -48,6 +48,8 @@ struct omap_dmic {
 	struct device *dev;
 	void __iomem *io_base;
 	struct clk *fclk;
+	struct pm_qos_request pm_qos_req;
+	int latency;
 	int fclk_freq;
 	int out_freq;
 	int clk_div;
@@ -124,6 +126,8 @@ static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream,
 
 	mutex_lock(&dmic->mutex);
 
+	pm_qos_remove_request(&dmic->pm_qos_req);
+
 	if (!dai->active)
 		dmic->active = 0;
 
@@ -228,6 +232,8 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
 	/* packet size is threshold * channels */
 	dma_data = snd_soc_dai_get_dma_data(dai, substream);
 	dma_data->maxburst = dmic->threshold * channels;
+	dmic->latency = (OMAP_DMIC_THRES_MAX - dmic->threshold) * USEC_PER_SEC /
+			params_rate(params);
 
 	return 0;
 }
@@ -238,6 +244,9 @@ static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream,
 	struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
 	u32 ctrl;
 
+	if (pm_qos_request_active(&dmic->pm_qos_req))
+		pm_qos_update_request(&dmic->pm_qos_req, dmic->latency);
+
 	/* Configure uplink threshold */
 	omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold);
 
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index d0ebb6b9bfac..2d6decbfc99e 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -308,9 +308,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 			pkt_size = channels;
 		}
 
-		latency = ((((buffer_size - pkt_size) / channels) * 1000)
-				 / (params->rate_num / params->rate_den));
-
+		latency = (buffer_size - pkt_size) / channels;
+		latency = latency * USEC_PER_SEC /
+			  (params->rate_num / params->rate_den);
 		mcbsp->latency[substream->stream] = latency;
 
 		omap_mcbsp_set_threshold(substream, pkt_size);
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 4c1be36c2207..7d5bdc5a2890 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -54,6 +54,8 @@ struct omap_mcpdm {
 	unsigned long phys_base;
 	void __iomem *io_base;
 	int irq;
+	struct pm_qos_request pm_qos_req;
+	int latency[2];
 
 	struct mutex mutex;
 
@@ -277,6 +279,9 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
 				  struct snd_soc_dai *dai)
 {
 	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+	int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+	int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
 
 	mutex_lock(&mcpdm->mutex);
 
@@ -289,6 +294,14 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
 		}
 	}
 
+	if (mcpdm->latency[stream2])
+		pm_qos_update_request(&mcpdm->pm_qos_req,
+				      mcpdm->latency[stream2]);
+	else if (mcpdm->latency[stream1])
+		pm_qos_remove_request(&mcpdm->pm_qos_req);
+
+	mcpdm->latency[stream1] = 0;
+
 	mutex_unlock(&mcpdm->mutex);
 }
 
@@ -300,7 +313,7 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 	int stream = substream->stream;
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	u32 threshold;
-	int channels;
+	int channels, latency;
 	int link_mask = 0;
 
 	channels = params_channels(params);
@@ -344,14 +357,25 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 
 		dma_data->maxburst =
 				(MCPDM_DN_THRES_MAX - threshold) * channels;
+		latency = threshold;
 	} else {
 		/* If playback is not running assume a stereo stream to come */
 		if (!mcpdm->config[!stream].link_mask)
 			mcpdm->config[!stream].link_mask = (0x3 << 3);
 
 		dma_data->maxburst = threshold * channels;
+		latency = (MCPDM_DN_THRES_MAX - threshold);
 	}
 
+	/*
+	 * The DMA must act to a DMA request within latency time (usec) to avoid
+	 * under/overflow
+	 */
+	mcpdm->latency[stream] = latency * USEC_PER_SEC / params_rate(params);
+
+	if (!mcpdm->latency[stream])
+		mcpdm->latency[stream] = 10;
+
 	/* Check if we need to restart McPDM with this stream */
 	if (mcpdm->config[stream].link_mask &&
 	    mcpdm->config[stream].link_mask != link_mask)
@@ -366,6 +390,20 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream,
 				  struct snd_soc_dai *dai)
 {
 	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+	struct pm_qos_request *pm_qos_req = &mcpdm->pm_qos_req;
+	int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+	int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+	int latency = mcpdm->latency[stream2];
+
+	/* Prevent omap hardware from hitting off between FIFO fills */
+	if (!latency || mcpdm->latency[stream1] < latency)
+		latency = mcpdm->latency[stream1];
+
+	if (pm_qos_request_active(pm_qos_req))
+		pm_qos_update_request(pm_qos_req, latency);
+	else if (latency)
+		pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency);
 
 	if (!omap_mcpdm_active(mcpdm)) {
 		omap_mcpdm_start(mcpdm);
@@ -427,6 +465,9 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai)
 	free_irq(mcpdm->irq, (void *)mcpdm);
 	pm_runtime_disable(mcpdm->dev);
 
+	if (pm_qos_request_active(&mcpdm->pm_qos_req))
+		pm_qos_remove_request(&mcpdm->pm_qos_req);
+
 	return 0;
 }
 
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index eb1b9da05dd4..4715527054e5 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -13,6 +13,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 	struct device_node *cpu = NULL;
 	struct device *dev = card->dev;
 	struct snd_soc_dai_link *link;
+	struct of_phandle_args args;
 	int ret, num_links;
 
 	ret = snd_soc_of_parse_card_name(card, "model");
@@ -47,12 +48,14 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 			goto err;
 		}
 
-		link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
-		if (!link->cpu_of_node) {
+		ret = of_parse_phandle_with_args(cpu, "sound-dai",
+					"#sound-dai-cells", 0, &args);
+		if (ret) {
 			dev_err(card->dev, "error getting cpu phandle\n");
-			ret = -EINVAL;
 			goto err;
 		}
+		link->cpu_of_node = args.np;
+		link->id = args.args[0];
 
 		ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
 		if (ret) {
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 60ff4a2d3577..8f6c8fc073a9 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -1112,204 +1112,204 @@ static int q6afe_of_xlate_dai_name(struct snd_soc_component *component,
 }
 
 static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
-	SND_SOC_DAPM_AIF_OUT("HDMI_RX", "HDMI Playback", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_RX", "Slimbus Playback", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_RX", "Slimbus1 Playback", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_RX", "Slimbus2 Playback", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_RX", "Slimbus3 Playback", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_2_TX", "Slimbus2 Capture", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_3_TX", "Slimbus3 Capture", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_4_TX", "Slimbus4 Capture", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_5_TX", "Slimbus5 Capture", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SLIMBUS_6_TX", "Slimbus6 Capture", 0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback",
+	SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, 0, 0, 0),
+	SND_SOC_DAPM_AIF_IN("QUAT_MI2S_RX", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture",
+	SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_TX", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("TERT_MI2S_RX", "Tertiary MI2S Playback",
+	SND_SOC_DAPM_AIF_IN("TERT_MI2S_RX", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("TERT_MI2S_TX", "Tertiary MI2S Capture",
+	SND_SOC_DAPM_AIF_OUT("TERT_MI2S_TX", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SEC_MI2S_RX", "Secondary MI2S Playback",
+	SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SEC_MI2S_TX", "Secondary MI2S Capture",
+	SND_SOC_DAPM_AIF_OUT("SEC_MI2S_TX", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SEC_MI2S_RX_SD1",
+	SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX_SD1",
 			"Secondary MI2S Playback SD1",
 			0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("PRI_MI2S_RX", "Primary MI2S Playback",
+	SND_SOC_DAPM_AIF_IN("PRI_MI2S_RX", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("PRI_MI2S_TX", "Primary MI2S Capture",
+	SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL,
 						0, 0, 0, 0),
 
-	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_0", "Primary TDM0 Playback",
+	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_1", "Primary TDM1 Playback",
+	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_2", "Primary TDM2 Playback",
+	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_2", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_3", "Primary TDM3 Playback",
+	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_3", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_4", "Primary TDM4 Playback",
+	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_4", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_5", "Primary TDM5 Playback",
+	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_5", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_6", "Primary TDM6 Playback",
+	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_6", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_RX_7", "Primary TDM7 Playback",
+	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_7", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_0", "Primary TDM0 Capture",
+	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_0", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_1", "Primary TDM1 Capture",
+	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_1", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_2", "Primary TDM2 Capture",
+	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_2", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_3", "Primary TDM3 Capture",
+	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_3", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_4", "Primary TDM4 Capture",
+	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_4", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_5", "Primary TDM5 Capture",
+	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_5", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_6", "Primary TDM6 Capture",
+	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_6", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_TX_7", "Primary TDM7 Capture",
+	SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_7", NULL,
 						0, 0, 0, 0),
 
-	SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_0", "Secondary TDM0 Playback",
+	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_0", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_1", "Secondary TDM1 Playback",
+	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_1", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_2", "Secondary TDM2 Playback",
+	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_2", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_3", "Secondary TDM3 Playback",
+	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_3", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_4", "Secondary TDM4 Playback",
+	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_4", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_5", "Secondary TDM5 Playback",
+	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_5", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_6", "Secondary TDM6 Playback",
+	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_6", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("SEC_TDM_RX_7", "Secondary TDM7 Playback",
+	SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_7", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_0", "Secondary TDM0 Capture",
+	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_0", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_1", "Secondary TDM1 Capture",
+	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_1", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_2", "Secondary TDM2 Capture",
+	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_2", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_3", "Secondary TDM3 Capture",
+	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_3", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_4", "Secondary TDM4 Capture",
+	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_4", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_5", "Secondary TDM5 Capture",
+	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_5", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_6", "Secondary TDM6 Capture",
+	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_6", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("SEC_TDM_TX_7", "Secondary TDM7 Capture",
+	SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_7", NULL,
 						0, 0, 0, 0),
 
-	SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_0", "Tertiary TDM0 Playback",
+	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_0", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_1", "Tertiary TDM1 Playback",
+	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_1", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_2", "Tertiary TDM2 Playback",
+	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_2", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_3", "Tertiary TDM3 Playback",
+	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_3", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_4", "Tertiary TDM4 Playback",
+	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_4", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_5", "Tertiary TDM5 Playback",
+	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_5", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_6", "Tertiary TDM6 Playback",
+	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_6", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("TERT_TDM_RX_7", "Tertiary TDM7 Playback",
+	SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_7", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_0", "Tertiary TDM0 Capture",
+	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_0", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_1", "Tertiary TDM1 Capture",
+	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_1", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_2", "Tertiary TDM2 Capture",
+	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_2", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_3", "Tertiary TDM3 Capture",
+	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_3", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_4", "Tertiary TDM4 Capture",
+	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_4", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_5", "Tertiary TDM5 Capture",
+	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_5", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_6", "Tertiary TDM6 Capture",
+	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_6", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("TERT_TDM_TX_7", "Tertiary TDM7 Capture",
+	SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_7", NULL,
 						0, 0, 0, 0),
 
-	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_0", "Quaternary TDM0 Playback",
+	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_0", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_1", "Quaternary TDM1 Playback",
+	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_1", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_2", "Quaternary TDM2 Playback",
+	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_2", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_3", "Quaternary TDM3 Playback",
+	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_3", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_4", "Quaternary TDM4 Playback",
+	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_4", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_5", "Quaternary TDM5 Playback",
+	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_5", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_6", "Quaternary TDM6 Playback",
+	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_6", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_RX_7", "Quaternary TDM7 Playback",
+	SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_7", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_0", "Quaternary TDM0 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_0", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_1", "Quaternary TDM1 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_1", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_2", "Quaternary TDM2 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_2", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_3", "Quaternary TDM3 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_3", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_4", "Quaternary TDM4 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_4", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_5", "Quaternary TDM5 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_5", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_6", "Quaternary TDM6 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_6", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUAT_TDM_TX_7", "Quaternary TDM7 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_7", NULL,
 						0, 0, 0, 0),
 
-	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_0", "Quinary TDM0 Playback",
+	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_0", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_1", "Quinary TDM1 Playback",
+	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_1", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_2", "Quinary TDM2 Playback",
+	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_2", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_3", "Quinary TDM3 Playback",
+	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_3", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_4", "Quinary TDM4 Playback",
+	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_4", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_5", "Quinary TDM5 Playback",
+	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_5", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_6", "Quinary TDM6 Playback",
+	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_6", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_RX_7", "Quinary TDM7 Playback",
+	SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_7", NULL,
 			     0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_0", "Quinary TDM0 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_0", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_1", "Quinary TDM1 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_1", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_2", "Quinary TDM2 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_2", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_3", "Quinary TDM3 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_3", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_4", "Quinary TDM4 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_4", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_5", "Quinary TDM5 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_5", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_6", "Quinary TDM6 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_6", NULL,
 						0, 0, 0, 0),
-	SND_SOC_DAPM_AIF_IN("QUIN_TDM_TX_7", "Quinary TDM7 Capture",
+	SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_7", NULL,
 						0, 0, 0, 0),
 };
 
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 000775b4bba8..829b5e987b2a 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -49,14 +49,14 @@
 #define AFE_PORT_I2S_SD1		0x2
 #define AFE_PORT_I2S_SD2		0x3
 #define AFE_PORT_I2S_SD3		0x4
-#define AFE_PORT_I2S_SD0_MASK		BIT(0x1)
-#define AFE_PORT_I2S_SD1_MASK		BIT(0x2)
-#define AFE_PORT_I2S_SD2_MASK		BIT(0x3)
-#define AFE_PORT_I2S_SD3_MASK		BIT(0x4)
-#define AFE_PORT_I2S_SD0_1_MASK		GENMASK(2, 1)
-#define AFE_PORT_I2S_SD2_3_MASK		GENMASK(4, 3)
-#define AFE_PORT_I2S_SD0_1_2_MASK	GENMASK(3, 1)
-#define AFE_PORT_I2S_SD0_1_2_3_MASK	GENMASK(4, 1)
+#define AFE_PORT_I2S_SD0_MASK		BIT(0x0)
+#define AFE_PORT_I2S_SD1_MASK		BIT(0x1)
+#define AFE_PORT_I2S_SD2_MASK		BIT(0x2)
+#define AFE_PORT_I2S_SD3_MASK		BIT(0x3)
+#define AFE_PORT_I2S_SD0_1_MASK		GENMASK(1, 0)
+#define AFE_PORT_I2S_SD2_3_MASK		GENMASK(3, 2)
+#define AFE_PORT_I2S_SD0_1_2_MASK	GENMASK(2, 0)
+#define AFE_PORT_I2S_SD0_1_2_3_MASK	GENMASK(3, 0)
 #define AFE_PORT_I2S_QUAD01		0x5
 #define AFE_PORT_I2S_QUAD23		0x6
 #define AFE_PORT_I2S_6CHS		0x7
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index a16c71c03058..86115de5c1b2 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -122,7 +122,6 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
 			.rate_max =	48000,				\
 		},							\
 		.name = "MultiMedia"#num,				\
-		.probe = fe_dai_probe,					\
 		.id = MSM_FRONTEND_DAI_MULTIMEDIA##num,			\
 	}
 
@@ -511,38 +510,6 @@ static void q6asm_dai_pcm_free(struct snd_pcm *pcm)
 	}
 }
 
-static const struct snd_soc_dapm_route afe_pcm_routes[] = {
-	{"MM_DL1",  NULL, "MultiMedia1 Playback" },
-	{"MM_DL2",  NULL, "MultiMedia2 Playback" },
-	{"MM_DL3",  NULL, "MultiMedia3 Playback" },
-	{"MM_DL4",  NULL, "MultiMedia4 Playback" },
-	{"MM_DL5",  NULL, "MultiMedia5 Playback" },
-	{"MM_DL6",  NULL, "MultiMedia6 Playback" },
-	{"MM_DL7",  NULL, "MultiMedia7 Playback" },
-	{"MM_DL7",  NULL, "MultiMedia8 Playback" },
-	{"MultiMedia1 Capture", NULL, "MM_UL1"},
-	{"MultiMedia2 Capture", NULL, "MM_UL2"},
-	{"MultiMedia3 Capture", NULL, "MM_UL3"},
-	{"MultiMedia4 Capture", NULL, "MM_UL4"},
-	{"MultiMedia5 Capture", NULL, "MM_UL5"},
-	{"MultiMedia6 Capture", NULL, "MM_UL6"},
-	{"MultiMedia7 Capture", NULL, "MM_UL7"},
-	{"MultiMedia8 Capture", NULL, "MM_UL8"},
-
-};
-
-static int fe_dai_probe(struct snd_soc_dai *dai)
-{
-	struct snd_soc_dapm_context *dapm;
-
-	dapm = snd_soc_component_get_dapm(dai->component);
-	snd_soc_dapm_add_routes(dapm, afe_pcm_routes,
-				ARRAY_SIZE(afe_pcm_routes));
-
-	return 0;
-}
-
-
 static const struct snd_soc_component_driver q6asm_fe_dai_component = {
 	.name		= DRV_NAME,
 	.ops		= &q6asm_dai_ops,
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index c6b51571be94..d61b8404f7da 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -909,6 +909,25 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"MM_UL6", NULL, "MultiMedia6 Mixer"},
 	{"MM_UL7", NULL, "MultiMedia7 Mixer"},
 	{"MM_UL8", NULL, "MultiMedia8 Mixer"},
+
+	{"MM_DL1",  NULL, "MultiMedia1 Playback" },
+	{"MM_DL2",  NULL, "MultiMedia2 Playback" },
+	{"MM_DL3",  NULL, "MultiMedia3 Playback" },
+	{"MM_DL4",  NULL, "MultiMedia4 Playback" },
+	{"MM_DL5",  NULL, "MultiMedia5 Playback" },
+	{"MM_DL6",  NULL, "MultiMedia6 Playback" },
+	{"MM_DL7",  NULL, "MultiMedia7 Playback" },
+	{"MM_DL8",  NULL, "MultiMedia8 Playback" },
+
+	{"MultiMedia1 Capture", NULL, "MM_UL1"},
+	{"MultiMedia2 Capture", NULL, "MM_UL2"},
+	{"MultiMedia3 Capture", NULL, "MM_UL3"},
+	{"MultiMedia4 Capture", NULL, "MM_UL4"},
+	{"MultiMedia5 Capture", NULL, "MM_UL5"},
+	{"MultiMedia6 Capture", NULL, "MM_UL6"},
+	{"MultiMedia7 Capture", NULL, "MM_UL7"},
+	{"MultiMedia8 Capture", NULL, "MM_UL8"},
+
 };
 
 static int routing_hw_params(struct snd_pcm_substream *substream,
diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c
index 9e7b5fa4cf59..4ac78d7a4b2d 100644
--- a/sound/soc/rockchip/rockchip_pcm.c
+++ b/sound/soc/rockchip/rockchip_pcm.c
@@ -33,6 +33,7 @@ static const struct snd_pcm_hardware snd_rockchip_hardware = {
 
 static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = {
 	.pcm_hardware = &snd_rockchip_hardware,
+	.prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
 	.prealloc_buffer_size = 32 * 1024,
 };
 
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index fcb4df23248c..6ec78f3096dd 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -306,7 +306,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod,
 	if (rsnd_ssi_is_multi_slave(mod, io))
 		return 0;
 
-	if (ssi->rate) {
+	if (ssi->usrcnt > 1) {
 		if (ssi->rate != rate) {
 			dev_err(dev, "SSI parent/child should use same rate\n");
 			return -EINVAL;
diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c
index b8e72b52db30..4fb29f0e561e 100644
--- a/sound/soc/soc-acpi.c
+++ b/sound/soc/soc-acpi.c
@@ -10,11 +10,17 @@ struct snd_soc_acpi_mach *
 snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines)
 {
 	struct snd_soc_acpi_mach *mach;
+	struct snd_soc_acpi_mach *mach_alt;
 
 	for (mach = machines; mach->id[0]; mach++) {
 		if (acpi_dev_present(mach->id, NULL, -1)) {
-			if (mach->machine_quirk)
-				mach = mach->machine_quirk(mach);
+			if (mach->machine_quirk) {
+				mach_alt = mach->machine_quirk(mach);
+				if (!mach_alt)
+					continue; /* not full match, ignore */
+				mach = mach_alt;
+			}
+
 			return mach;
 		}
 	}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6ddcf12bc030..b29d0f65611e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2131,6 +2131,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	card->instantiated = 1;
+	dapm_mark_endpoints_dirty(card);
 	snd_soc_dapm_sync(&card->dapm);
 	mutex_unlock(&card->mutex);
 	mutex_unlock(&client_mutex);
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index ea05cc91aa05..211589b0b2ef 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -390,7 +390,7 @@ static int stm32_sai_add_mclk_provider(struct stm32_sai_sub_data *sai)
 	char *mclk_name, *p, *s = (char *)pname;
 	int ret, i = 0;
 
-	mclk = devm_kzalloc(dev, sizeof(mclk), GFP_KERNEL);
+	mclk = devm_kzalloc(dev, sizeof(*mclk), GFP_KERNEL);
 	if (!mclk)
 		return -ENOMEM;
 
diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig
index 66aad0d3f9c7..8134c3c94229 100644
--- a/sound/soc/sunxi/Kconfig
+++ b/sound/soc/sunxi/Kconfig
@@ -31,7 +31,7 @@ config SND_SUN8I_CODEC_ANALOG
 config SND_SUN50I_CODEC_ANALOG
 	tristate "Allwinner sun50i Codec Analog Controls Support"
 	depends on (ARM64 && ARCH_SUNXI) || COMPILE_TEST
-	select SND_SUNXI_ADDA_PR_REGMAP
+	select SND_SUN8I_ADDA_PR_REGMAP
 	help
 	  Say Y or M if you want to add support for the analog controls for
 	  the codec embedded in Allwinner A64 SoC.
diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index 522a72fde78d..92c5de026c43 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -481,7 +481,11 @@ static const struct snd_soc_dapm_route sun8i_codec_dapm_routes[] = {
 	{ "Right Digital DAC Mixer", "AIF1 Slot 0 Digital DAC Playback Switch",
 	  "AIF1 Slot 0 Right"},
 
-	/* ADC routes */
+	/* ADC Routes */
+	{ "AIF1 Slot 0 Right ADC", NULL, "ADC" },
+	{ "AIF1 Slot 0 Left ADC", NULL, "ADC" },
+
+	/* ADC Mixer Routes */
 	{ "Left Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch",
 	  "AIF1 Slot 0 Left ADC" },
 	{ "Right Digital ADC Mixer", "AIF1 Data Digital ADC Capture Switch",
@@ -605,16 +609,10 @@ err_pm_disable:
 
 static int sun8i_codec_remove(struct platform_device *pdev)
 {
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-	struct sun8i_codec *scodec = snd_soc_card_get_drvdata(card);
-
 	pm_runtime_disable(&pdev->dev);
 	if (!pm_runtime_status_suspended(&pdev->dev))
 		sun8i_codec_runtime_suspend(&pdev->dev);
 
-	clk_disable_unprepare(scodec->clk_module);
-	clk_disable_unprepare(scodec->clk_bus);
-
 	return 0;
 }
 
diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c
index e73c962590eb..079063d8038d 100644
--- a/sound/sparc/cs4231.c
+++ b/sound/sparc/cs4231.c
@@ -1146,10 +1146,8 @@ static int snd_cs4231_playback_open(struct snd_pcm_substream *substream)
 	runtime->hw = snd_cs4231_playback;
 
 	err = snd_cs4231_open(chip, CS4231_MODE_PLAY);
-	if (err < 0) {
-		snd_free_pages(runtime->dma_area, runtime->dma_bytes);
+	if (err < 0)
 		return err;
-	}
 	chip->playback_substream = substream;
 	chip->p_periods_sent = 0;
 	snd_pcm_set_sync(substream);
@@ -1167,10 +1165,8 @@ static int snd_cs4231_capture_open(struct snd_pcm_substream *substream)
 	runtime->hw = snd_cs4231_capture;
 
 	err = snd_cs4231_open(chip, CS4231_MODE_RECORD);
-	if (err < 0) {
-		snd_free_pages(runtime->dma_area, runtime->dma_bytes);
+	if (err < 0)
 		return err;
-	}
 	chip->capture_substream = substream;
 	chip->c_periods_sent = 0;
 	snd_pcm_set_sync(substream);
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 2bfe4e80a6b9..a105947eaf55 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -682,9 +682,12 @@ static int usb_audio_probe(struct usb_interface *intf,
 
  __error:
 	if (chip) {
+		/* chip->active is inside the chip->card object,
+		 * decrement before memory is possibly returned.
+		 */
+		atomic_dec(&chip->active);
 		if (!chip->num_interfaces)
 			snd_card_free(chip->card);
-		atomic_dec(&chip->active);
 	}
 	mutex_unlock(&register_mutex);
 	return err;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 849953e5775c..37fc0447c071 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3382,5 +3382,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 		.ifnum = QUIRK_NO_INTERFACE
 	}
 },
+/* Dell WD19 Dock */
+{
+	USB_DEVICE(0x0bda, 0x402e),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "Dell",
+		.product_name = "WD19 Dock",
+		.profile_name = "Dell-WD15-Dock",
+		.ifnum = QUIRK_NO_INTERFACE
+	}
+},
 
 #undef USB_DEVICE_VENDOR_SPEC
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 8a945ece9869..6623cafc94f2 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1373,6 +1373,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 			return SNDRV_PCM_FMTBIT_DSD_U32_BE;
 		break;
 
+	case USB_ID(0x152a, 0x85de): /* SMSL D1 DAC */
 	case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */
 	case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */
 	case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */