Merge remote-tracking branch 'asoc/topic/dmic' into asoc-core

49 files changed, 288 insertions, 152 deletions

diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 76e0d5695075..823359ed95e1 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -166,7 +166,9 @@ void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
 	} else {
 		printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n",
 			rtd->params->name, dma_ch, dcsr);
+		snd_pcm_stream_lock(substream);
 		snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+		snd_pcm_stream_unlock(substream);
 	}
 }
 EXPORT_SYMBOL(pxa2xx_pcm_dma_irq);
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 99db892d7299..98969541cbcc 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -743,7 +743,7 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg)
 	mutex_lock(&stream->device->lock);
 	switch (_IOC_NR(cmd)) {
 	case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION):
-		put_user(SNDRV_COMPRESS_VERSION,
+		retval = put_user(SNDRV_COMPRESS_VERSION,
 				(int __user *)arg) ? -EFAULT : 0;
 		break;
 	case _IOC_NR(SNDRV_COMPRESS_GET_CAPS):
diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c
index e3cb46fef2c7..b3f39b5ed742 100644
--- a/sound/core/seq/oss/seq_oss_init.c
+++ b/sound/core/seq/oss/seq_oss_init.c
@@ -31,6 +31,7 @@
 #include <linux/export.h>
 #include <linux/moduleparam.h>
 #include <linux/slab.h>
+#include <linux/workqueue.h>
 
 /*
  * common variables
@@ -60,6 +61,14 @@ static void free_devinfo(void *private);
 #define call_ctl(type,rec) snd_seq_kernel_client_ctl(system_client, type, rec)
 
 
+/* call snd_seq_oss_midi_lookup_ports() asynchronously */
+static void async_call_lookup_ports(struct work_struct *work)
+{
+	snd_seq_oss_midi_lookup_ports(system_client);
+}
+
+static DECLARE_WORK(async_lookup_work, async_call_lookup_ports);
+
 /*
  * create sequencer client for OSS sequencer
  */
@@ -85,9 +94,6 @@ snd_seq_oss_create_client(void)
 	system_client = rc;
 	debug_printk(("new client = %d\n", rc));
 
-	/* look up midi devices */
-	snd_seq_oss_midi_lookup_ports(system_client);
-
 	/* create annoucement receiver port */
 	memset(port, 0, sizeof(*port));
 	strcpy(port->name, "Receiver");
@@ -115,6 +121,9 @@ snd_seq_oss_create_client(void)
 	}
 	rc = 0;
 
+	/* look up midi devices */
+	schedule_work(&async_lookup_work);
+
  __error:
 	kfree(port);
 	return rc;
@@ -160,6 +169,7 @@ receive_announce(struct snd_seq_event *ev, int direct, void *private, int atomic
 int
 snd_seq_oss_delete_client(void)
 {
+	cancel_work_sync(&async_lookup_work);
 	if (system_client >= 0)
 		snd_seq_delete_kernel_client(system_client);
 
diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c
index 677dc84590c7..862d84893ee8 100644
--- a/sound/core/seq/oss/seq_oss_midi.c
+++ b/sound/core/seq/oss/seq_oss_midi.c
@@ -72,7 +72,7 @@ static int send_midi_event(struct seq_oss_devinfo *dp, struct snd_seq_event *ev,
  * look up the existing ports
  * this looks a very exhausting job.
  */
-int __init
+int
 snd_seq_oss_midi_lookup_ports(int client)
 {
 	struct snd_seq_client_info *clinfo;
diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c
index 7e814a5c3677..4bbcc0fcd4eb 100644
--- a/sound/oss/vwsnd.c
+++ b/sound/oss/vwsnd.c
@@ -149,17 +149,19 @@
 #include <linux/interrupt.h>
 #include <linux/mutex.h>
 #include <linux/slab.h>
+#include <linux/delay.h>
 
 #include <asm/visws/cobalt.h>
 
 #include "sound_config.h"
 
+static DEFINE_MUTEX(vwsnd_mutex);
+
 /*****************************************************************************/
 /* debug stuff */
 
 #ifdef VWSND_DEBUG
 
-static DEFINE_MUTEX(vwsnd_mutex);
 static int shut_up = 1;
 
 /*
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 185d54a5cb1a..dc632cdc3870 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -769,7 +769,10 @@ static void snd_card_asihpi_timer_function(unsigned long data)
 						s->number);
 				ds->drained_count++;
 				if (ds->drained_count > 20) {
+					unsigned long flags;
+					snd_pcm_stream_lock_irqsave(s, flags);
 					snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN);
+					snd_pcm_stream_unlock_irqrestore(s, flags);
 					continue;
 				}
 			} else {
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index fe4c61bdb8ba..f6dec3ea371f 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -689,7 +689,9 @@ static void snd_atiixp_xrun_dma(struct atiixp *chip, struct atiixp_dma *dma)
 	if (! dma->substream || ! dma->running)
 		return;
 	snd_printdd("atiixp: XRUN detected (DMA %d)\n", dma->ops->type);
+	snd_pcm_stream_lock(dma->substream);
 	snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN);
+	snd_pcm_stream_unlock(dma->substream);
 }
 
 /*
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index cf29b9a1d65d..289563ecb6dd 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -638,7 +638,9 @@ static void snd_atiixp_xrun_dma(struct atiixp_modem *chip,
 	if (! dma->substream || ! dma->running)
 		return;
 	snd_printdd("atiixp-modem: XRUN detected (DMA %d)\n", dma->ops->type);
+	snd_pcm_stream_lock(dma->substream);
 	snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN);
+	snd_pcm_stream_unlock(dma->substream);
 }
 
 /*
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 7c11d46b84d3..48a9d004d6d9 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -860,7 +860,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
 		}
 	}
 	if (id < 0 && quirk) {
-		for (q = quirk; q->subvendor; q++) {
+		for (q = quirk; q->subvendor || q->subdevice; q++) {
 			unsigned int vendorid =
 				q->subdevice | (q->subvendor << 16);
 			unsigned int mask = 0xffff0000 | q->subdevice_mask;
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 8e77cbbad871..e3c7ba8d7582 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -522,7 +522,7 @@ static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1,
 }
 
 #define nid_has_mute(codec, nid, dir) \
-	check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE)
+	check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE))
 #define nid_has_volume(codec, nid, dir) \
 	check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS)
 
@@ -624,7 +624,7 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid,
 		if (enable)
 			val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
 	}
-	if (caps & AC_AMPCAP_MUTE) {
+	if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) {
 		if (!enable)
 			val |= HDA_AMP_MUTE;
 	}
@@ -648,7 +648,7 @@ static unsigned int get_amp_mask_to_modify(struct hda_codec *codec,
 {
 	unsigned int mask = 0xff;
 
-	if (caps & AC_AMPCAP_MUTE) {
+	if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) {
 		if (is_ctl_associated(codec, nid, dir, idx, NID_PATH_MUTE_CTL))
 			mask &= ~0x80;
 	}
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 540bdef2f904..030ca8652a1c 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2622,6 +2622,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
 { .id = 0x10de0043, .name = "GPU 43 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x10de0044, .name = "GPU 44 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x10de0051, .name = "GPU 51 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x10de0067, .name = "MCP67 HDMI",	.patch = patch_nvhdmi_2ch },
 { .id = 0x10de8001, .name = "MCP73 HDMI",	.patch = patch_nvhdmi_2ch },
 { .id = 0x11069f80, .name = "VX900 HDMI/DP",	.patch = patch_via_hdmi },
@@ -2674,6 +2675,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0042");
 MODULE_ALIAS("snd-hda-codec-id:10de0043");
 MODULE_ALIAS("snd-hda-codec-id:10de0044");
 MODULE_ALIAS("snd-hda-codec-id:10de0051");
+MODULE_ALIAS("snd-hda-codec-id:10de0060");
 MODULE_ALIAS("snd-hda-codec-id:10de0067");
 MODULE_ALIAS("snd-hda-codec-id:10de8001");
 MODULE_ALIAS("snd-hda-codec-id:11069f80");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8bd226149868..f303cd898515 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1031,6 +1031,7 @@ enum {
 	ALC880_FIXUP_GPIO2,
 	ALC880_FIXUP_MEDION_RIM,
 	ALC880_FIXUP_LG,
+	ALC880_FIXUP_LG_LW25,
 	ALC880_FIXUP_W810,
 	ALC880_FIXUP_EAPD_COEF,
 	ALC880_FIXUP_TCL_S700,
@@ -1089,6 +1090,14 @@ static const struct hda_fixup alc880_fixups[] = {
 			{ }
 		}
 	},
+	[ALC880_FIXUP_LG_LW25] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x0181344f }, /* line-in */
+			{ 0x1b, 0x0321403f }, /* headphone */
+			{ }
+		}
+	},
 	[ALC880_FIXUP_W810] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -1341,6 +1350,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG),
 	SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG),
 	SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG),
+	SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_FIXUP_LG_LW25),
 	SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700),
 
 	/* Below is the copied entries from alc880_quirks.c.
@@ -4329,6 +4339,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index e2f83591161b..6d1924c19abf 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -417,9 +417,11 @@ static void stac_update_outputs(struct hda_codec *codec)
 			val &= ~spec->eapd_mask;
 		else
 			val |= spec->eapd_mask;
-		if (spec->gpio_data != val)
+		if (spec->gpio_data != val) {
+			spec->gpio_data = val;
 			stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir,
 				      val);
+		}
 	}
 }
 
@@ -2817,6 +2819,7 @@ static const struct hda_pintbl ecs202_pin_configs[] = {
 
 /* codec SSIDs for Intel Mac sharing the same PCI SSID 8384:7680 */
 static const struct snd_pci_quirk stac922x_intel_mac_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x0000, 0x0100, "Mac Mini", STAC_INTEL_MAC_V3),
 	SND_PCI_QUIRK(0x106b, 0x0800, "Mac", STAC_INTEL_MAC_V1),
 	SND_PCI_QUIRK(0x106b, 0x0600, "Mac", STAC_INTEL_MAC_V2),
 	SND_PCI_QUIRK(0x106b, 0x0700, "Mac", STAC_INTEL_MAC_V2),
@@ -3231,7 +3234,7 @@ static const struct hda_fixup stac927x_fixups[] = {
 			/* configure the analog microphone on some laptops */
 			{ 0x0c, 0x90a79130 },
 			/* correct the front output jack as a hp out */
-			{ 0x0f, 0x0227011f },
+			{ 0x0f, 0x0221101f },
 			/* correct the front input jack as a mic */
 			{ 0x0e, 0x02a79130 },
 			{}
@@ -3612,20 +3615,18 @@ static int stac_parse_auto_config(struct hda_codec *codec)
 static int stac_init(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	unsigned int gpio;
 	int i;
 
 	/* override some hints */
 	stac_store_hints(codec);
 
 	/* set up GPIO */
-	gpio = spec->gpio_data;
 	/* turn on EAPD statically when spec->eapd_switch isn't set.
 	 * otherwise, unsol event will turn it on/off dynamically
 	 */
 	if (!spec->eapd_switch)
-		gpio |= spec->eapd_mask;
-	stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, gpio);
+		spec->gpio_data |= spec->eapd_mask;
+	stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data);
 
 	snd_hda_gen_init(codec);
 
@@ -3915,6 +3916,7 @@ static void stac_setup_gpio(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
 
+	spec->gpio_mask |= spec->eapd_mask;
 	if (spec->gpio_led) {
 		if (!spec->vref_mute_led_nid) {
 			spec->gpio_mask |= spec->gpio_led;
diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c
index 1d38fd0bc4e2..d12826526798 100644
--- a/sound/soc/atmel/atmel-pcm-dma.c
+++ b/sound/soc/atmel/atmel-pcm-dma.c
@@ -81,7 +81,9 @@ static void atmel_pcm_dma_irq(u32 ssc_sr,
 
 		/* stop RX and capture: will be enabled again at restart */
 		ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_disable);
+		snd_pcm_stream_lock(substream);
 		snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+		snd_pcm_stream_unlock(substream);
 
 		/* now drain RHR and read status to remove xrun condition */
 		ssc_readx(prtd->ssc->regs, SSC_RHR);
diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c
index d6f7694fcad4..c8a2de103c5f 100644
--- a/sound/soc/au1x/ac97c.c
+++ b/sound/soc/au1x/ac97c.c
@@ -341,7 +341,7 @@ static struct platform_driver au1xac97c_driver = {
 	.remove		= au1xac97c_drvremove,
 };
 
-module_platform_driver(&au1xac97c_driver);
+module_platform_driver(au1xac97c_driver);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver");
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index efb1daecd0dd..e82eb373a731 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -294,11 +294,12 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev)
 	/* Request PB3 as reset pin */
 	ret = devm_gpio_request_one(&pdev->dev,
 				    CONFIG_SND_BF5XX_RESET_GPIO_NUM,
-				    GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET") {
+				    GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET");
+	if (ret) {
 		dev_err(&pdev->dev,
 			"Failed to request GPIO_%d for reset: %d\n",
 			CONFIG_SND_BF5XX_RESET_GPIO_NUM, ret);
-		goto gpio_err;
+		return ret;
 	}
 #endif
 
diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h
index 15c635e33f4d..0c3e22d90a8d 100644
--- a/sound/soc/blackfin/bf5xx-ac97.h
+++ b/sound/soc/blackfin/bf5xx-ac97.h
@@ -9,7 +9,6 @@
 #ifndef _BF5XX_AC97_H
 #define _BF5XX_AC97_H
 
-extern struct snd_ac97_bus_ops bf5xx_ac97_ops;
 extern struct snd_ac97 *ac97;
 /* Frame format in memory, only support stereo currently */
 struct ac97_frame {
diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c
index ac73c607410a..04491f0e8d1b 100644
--- a/sound/soc/cirrus/ep93xx-ac97.c
+++ b/sound/soc/cirrus/ep93xx-ac97.c
@@ -102,13 +102,13 @@ static struct ep93xx_ac97_info *ep93xx_ac97_info;
 
 static struct ep93xx_dma_data ep93xx_ac97_pcm_out = {
 	.name		= "ac97-pcm-out",
-	.dma_port	= EP93XX_DMA_AAC1,
+	.port		= EP93XX_DMA_AAC1,
 	.direction	= DMA_MEM_TO_DEV,
 };
 
 static struct ep93xx_dma_data ep93xx_ac97_pcm_in = {
 	.name		= "ac97-pcm-in",
-	.dma_port	= EP93XX_DMA_AAC1,
+	.port		= EP93XX_DMA_AAC1,
 	.direction	= DMA_DEV_TO_MEM,
 };
 
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 987f728718c5..be2ba1b6fe4a 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -195,6 +195,8 @@ static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0);
 
 static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0);
 
+static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0);
+
 static const unsigned int limiter_tlv[] = {
 	TLV_DB_RANGE_HEAD(2),
 	0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
@@ -451,7 +453,8 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 	SOC_ENUM("Beep Pitch", beep_pitch_enum),
 	SOC_ENUM("Beep on Time", beep_ontime_enum),
 	SOC_ENUM("Beep off Time", beep_offtime_enum),
-	SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv),
+	SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL,
+			0, 0x07, 0x1f, beep_tlv),
 	SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1),
 	SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum),
 	SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum),
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index 66967ba6f757..b2090b2a5e2d 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -50,20 +50,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"DMIC AIF", NULL, "DMic"},
 };
 
-static int dmic_probe(struct snd_soc_codec *codec)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
-				  ARRAY_SIZE(dmic_dapm_widgets));
-        snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-	snd_soc_dapm_new_widgets(dapm);
-
-	return 0;
-}
-
 static struct snd_soc_codec_driver soc_dmic = {
-	.probe	= dmic_probe,
+	.dapm_widgets = dmic_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets),
+	.dapm_routes = intercon,
+	.num_dapm_routes = ARRAY_SIZE(intercon),
 };
 
 static int dmic_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 3eeada57e87d..566a367c94fa 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute)
 
 static void max98088_sync_cache(struct snd_soc_codec *codec)
 {
-       u16 *reg_cache = codec->reg_cache;
+       u8 *reg_cache = codec->reg_cache;
        int i;
 
        if (!codec->cache_sync)
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d441559dc92c..760e8bfeacaa 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -38,7 +38,7 @@
 static const struct reg_default sgtl5000_reg_defaults[] = {
 	{ SGTL5000_CHIP_CLK_CTRL,		0x0008 },
 	{ SGTL5000_CHIP_I2S_CTRL,		0x0010 },
-	{ SGTL5000_CHIP_SSS_CTRL,		0x0008 },
+	{ SGTL5000_CHIP_SSS_CTRL,		0x0010 },
 	{ SGTL5000_CHIP_DAC_VOL,		0x3c3c },
 	{ SGTL5000_CHIP_PAD_STRENGTH,		0x015f },
 	{ SGTL5000_CHIP_ANA_HP_CTRL,		0x1818 },
@@ -153,6 +153,8 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
 static int power_vag_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
+	const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP;
+
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
 		snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
@@ -160,9 +162,17 @@ static int power_vag_event(struct snd_soc_dapm_widget *w,
 		break;
 
 	case SND_SOC_DAPM_PRE_PMD:
-		snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
-			SGTL5000_VAG_POWERUP, 0);
-		msleep(400);
+		/*
+		 * Don't clear VAG_POWERUP, when both DAC and ADC are
+		 * operational to prevent inadvertently starving the
+		 * other one of them.
+		 */
+		if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) &
+				mask) != mask) {
+			snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_VAG_POWERUP, 0);
+			msleep(400);
+		}
 		break;
 	default:
 		break;
@@ -388,7 +398,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
 	SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0),
 	SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)",
 			SGTL5000_CHIP_ANA_ADC_CTRL,
-			8, 2, 0, capture_6db_attenuate),
+			8, 1, 0, capture_6db_attenuate),
 	SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0),
 
 	SOC_DOUBLE_TLV("Headphone Playback Volume",
@@ -1527,6 +1537,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
 	if (IS_ERR(sgtl5000->mclk)) {
 		ret = PTR_ERR(sgtl5000->mclk);
 		dev_err(&client->dev, "Failed to get mclock: %d\n", ret);
+		/* Defer the probe to see if the clk will be provided later */
+		if (ret == -ENOENT)
+			return -EPROBE_DEFER;
 		return ret;
 	}
 
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
index 4b69229a9818..2f8c88931f69 100644
--- a/sound/soc/codecs/sgtl5000.h
+++ b/sound/soc/codecs/sgtl5000.h
@@ -347,7 +347,7 @@
 #define SGTL5000_PLL_INT_DIV_MASK		0xf800
 #define SGTL5000_PLL_INT_DIV_SHIFT		11
 #define SGTL5000_PLL_INT_DIV_WIDTH		5
-#define SGTL5000_PLL_FRAC_DIV_MASK		0x0700
+#define SGTL5000_PLL_FRAC_DIV_MASK		0x07ff
 #define SGTL5000_PLL_FRAC_DIV_SHIFT		0
 #define SGTL5000_PLL_FRAC_DIV_WIDTH		11
 
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index f5e835662cdc..10adc4145d46 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -410,6 +410,16 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec)
 			rec->command, rec->length);
 		len = rec->length + 8;
 
+		xfer = kzalloc(sizeof(*xfer), GFP_KERNEL);
+		if (!xfer) {
+			dev_err(codec->dev, "Failed to allocate xfer\n");
+			ret = -ENOMEM;
+			goto abort;
+		}
+
+		xfer->codec = codec;
+		list_add_tail(&xfer->list, &xfer_list);
+
 		out = kzalloc(len, GFP_KERNEL);
 		if (!out) {
 			dev_err(codec->dev,
@@ -417,6 +427,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec)
 			ret = -ENOMEM;
 			goto abort1;
 		}
+		xfer->t.rx_buf = out;
 
 		img = kzalloc(len, GFP_KERNEL);
 		if (!img) {
@@ -425,24 +436,13 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec)
 			ret = -ENOMEM;
 			goto abort1;
 		}
+		xfer->t.tx_buf = img;
 
 		byte_swap_64((u64 *)&rec->command, img, len);
 
-		xfer = kzalloc(sizeof(*xfer), GFP_KERNEL);
-		if (!xfer) {
-			dev_err(codec->dev, "Failed to allocate xfer\n");
-			ret = -ENOMEM;
-			goto abort1;
-		}
-
-		xfer->codec = codec;
-		list_add_tail(&xfer->list, &xfer_list);
-
 		spi_message_init(&xfer->m);
 		xfer->m.complete = wm0010_boot_xfer_complete;
 		xfer->m.context = xfer;
-		xfer->t.tx_buf = img;
-		xfer->t.rx_buf = out;
 		xfer->t.len = len;
 		xfer->t.bits_per_word = 8;
 
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 029f31c8e703..d8fc531c0e59 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -921,6 +921,7 @@ static struct snd_soc_dai_driver wm8978_dai = {
 		.formats = WM8978_FORMATS,
 	},
 	.ops = &wm8978_dai_ops,
+	.symmetri