Merge branch 'for-linus' into for-next

15 files changed, 69 insertions, 40 deletions

diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c
index 4449d1a99089..2433f7c81472 100644
--- a/sound/hda/ext/hdac_ext_bus.c
+++ b/sound/hda/ext/hdac_ext_bus.c
@@ -19,6 +19,7 @@
 
 #include <linux/module.h>
 #include <linux/slab.h>
+#include <linux/io.h>
 #include <sound/hdaudio_ext.h>
 
 MODULE_DESCRIPTION("HDA extended core");
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 584a0343ab0c..85813de26da8 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -633,6 +633,7 @@ static const struct snd_pci_quirk cs4208_mac_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x5e00, "MacBookPro 11,2", CS4208_MBP11),
 	SND_PCI_QUIRK(0x106b, 0x7100, "MacBookAir 6,1", CS4208_MBA6),
 	SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6),
+	SND_PCI_QUIRK(0x106b, 0x7b00, "MacBookPro 12,1", CS4208_MBP11),
 	{} /* terminator */
 };
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index ca03c40609fc..2f0ec7c45fc7 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -819,6 +819,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD),
+	SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index afec6dc9f91f..16b8dcba5c12 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5306,6 +5306,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 9d947aef2c8b..def5cc8dff02 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4520,7 +4520,11 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 		return err;
 
 	spec = codec->spec;
-	codec->power_save_node = 1;
+	/* enable power_save_node only for new 92HD89xx chips, as it causes
+	 * click noises on old 92HD73xx chips.
+	 */
+	if ((codec->core.vendor_id & 0xfffffff0) != 0x111d7670)
+		codec->power_save_node = 1;
 	spec->linear_tone_beep = 0;
 	spec->gen.mixer_nid = 0x1d;
 	spec->have_spdif_mux = 1;
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 638ca0ba7e6e..5c73061d912a 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -129,6 +129,8 @@ static struct snd_soc_dai_link db1300_i2s_dai = {
 	.cpu_dai_name	= "au1xpsc_i2s.2",
 	.platform_name	= "au1xpsc-pcm.2",
 	.codec_name	= "wm8731.0-001b",
+	.dai_fmt	= SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+			  SND_SOC_DAIFMT_CBM_CFM,
 	.ops		= &db1200_i2s_wm8731_ops,
 };
 
@@ -146,6 +148,8 @@ static struct snd_soc_dai_link db1550_i2s_dai = {
 	.cpu_dai_name	= "au1xpsc_i2s.3",
 	.platform_name	= "au1xpsc-pcm.3",
 	.codec_name	= "wm8731.0-001b",
+	.dai_fmt	= SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+			  SND_SOC_DAIFMT_CBM_CFM,
 	.ops		= &db1200_i2s_wm8731_ops,
 };
 
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 3eb184c4abe5..080cc1ce3963 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -519,11 +519,11 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = {
 		RT5645_L_VOL_SFT + 1, RT5645_R_VOL_SFT + 1, 63, 0, adc_vol_tlv),
 
 	/* ADC Boost Volume Control */
-	SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5645_ADC_BST_VOL1,
+	SOC_DOUBLE_TLV("ADC Boost Capture Volume", RT5645_ADC_BST_VOL1,
 		RT5645_STO1_ADC_L_BST_SFT, RT5645_STO1_ADC_R_BST_SFT, 3, 0,
 		adc_bst_tlv),
-	SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5645_ADC_BST_VOL1,
-		RT5645_STO2_ADC_L_BST_SFT, RT5645_STO2_ADC_R_BST_SFT, 3, 0,
+	SOC_DOUBLE_TLV("Mono ADC Boost Capture Volume", RT5645_ADC_BST_VOL2,
+		RT5645_MONO_ADC_L_BST_SFT, RT5645_MONO_ADC_R_BST_SFT, 3, 0,
 		adc_bst_tlv),
 
 	/* I2S2 function select */
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index f45861c49ef2..61bc8ab77646 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -39,8 +39,8 @@
 #define RT5645_STO1_ADC_DIG_VOL			0x1c
 #define RT5645_MONO_ADC_DIG_VOL			0x1d
 #define RT5645_ADC_BST_VOL1			0x1e
-/* Mixer - D-D */
 #define RT5645_ADC_BST_VOL2			0x20
+/* Mixer - D-D */
 #define RT5645_STO1_ADC_MIXER			0x27
 #define RT5645_MONO_ADC_MIXER			0x28
 #define RT5645_AD_DA_MIXER			0x29
@@ -315,12 +315,14 @@
 #define RT5645_STO1_ADC_R_BST_SFT		12
 #define RT5645_STO1_ADC_COMP_MASK		(0x3 << 10)
 #define RT5645_STO1_ADC_COMP_SFT		10
-#define RT5645_STO2_ADC_L_BST_MASK		(0x3 << 8)
-#define RT5645_STO2_ADC_L_BST_SFT		8
-#define RT5645_STO2_ADC_R_BST_MASK		(0x3 << 6)
-#define RT5645_STO2_ADC_R_BST_SFT		6
-#define RT5645_STO2_ADC_COMP_MASK		(0x3 << 4)
-#define RT5645_STO2_ADC_COMP_SFT		4
+
+/* ADC Boost Volume Control (0x20) */
+#define RT5645_MONO_ADC_L_BST_MASK		(0x3 << 14)
+#define RT5645_MONO_ADC_L_BST_SFT		14
+#define RT5645_MONO_ADC_R_BST_MASK		(0x3 << 12)
+#define RT5645_MONO_ADC_R_BST_SFT		12
+#define RT5645_MONO_ADC_COMP_MASK		(0x3 << 10)
+#define RT5645_MONO_ADC_COMP_SFT		10
 
 /* Stereo2 ADC Mixer Control (0x26) */
 #define RT5645_STO2_ADC_SRC_MASK		(0x1 << 15)
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index bfda25ef0dd4..f540f82b1f27 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1376,8 +1376,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 			sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT);
 
 	snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL,
-			SGTL5000_BIAS_R_MASK,
-			sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT);
+			SGTL5000_BIAS_VOLT_MASK,
+			sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT);
 	/*
 	 * disable DAP
 	 * TODO:
@@ -1549,7 +1549,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
 			else {
 				sgtl5000->micbias_voltage = 0;
 				dev_err(&client->dev,
-					"Unsuitable MicBias resistor\n");
+					"Unsuitable MicBias voltage\n");
 			}
 		} else {
 			sgtl5000->micbias_voltage = 0;
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index e3a0bca28bcf..cc1d3981fa4b 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -549,7 +549,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = {
 /*
  * DAC digital volumes. From -7 to 24 dB in 1 dB steps
  */
-static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 0);
+static DECLARE_TLV_DB_SCALE(dac_tlv, -700, 100, 0);
 
 static const char * const tas2552_din_source_select[] = {
 	"Muted",
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index f1c9fffcd3a6..a564759845f9 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1531,14 +1531,17 @@ static int aic3x_init(struct snd_soc_codec *codec)
 	snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL);
 	snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL);
 
-	/* Line2 to HP Bypass default volume, disconnect from Output Mixer */
-	snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
-	snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
-	snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
-	snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
-	/* Line2 Line Out default volume, disconnect from Output Mixer */
-	snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
-	snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
+	/* On tlv320aic3104, these registers are reserved and must not be written */
+	if (aic3x->model != AIC3X_MODEL_3104) {
+		/* Line2 to HP Bypass default volume, disconnect from Output Mixer */
+		snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
+		snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
+		snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
+		snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
+		/* Line2 Line Out default volume, disconnect from Output Mixer */
+		snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
+		snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
+	}
 
 	switch (aic3x->model) {
 	case AIC3X_MODEL_3X:
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 293e47a6ff59..2fbc6ef8cbdb 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3760,7 +3760,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
 	ret = snd_soc_register_codec(&i2c->dev,
 				     &soc_codec_dev_wm8962, &wm8962_dai, 1);
 	if (ret < 0)
-		goto err_enable;
+		goto err_pm_runtime;
 
 	regcache_cache_only(wm8962->regmap, true);
 
@@ -3769,6 +3769,8 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
 
 	return 0;
 
+err_pm_runtime:
+	pm_runtime_disable(&i2c->dev);
 err_enable:
 	regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
 err:
@@ -3778,6 +3780,7 @@ err:
 static int wm8962_i2c_remove(struct i2c_client *client)
 {
 	snd_soc_unregister_codec(&client->dev);
+	pm_runtime_disable(&client->dev);
 	return 0;
 }
 
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index a3e97b46b64e..ba34252b7bba 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -131,23 +131,32 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		for (i = 0; i < 4; i++)
-			i2s_write_reg(dev->i2s_base, TOR(i), 0);
+			i2s_read_reg(dev->i2s_base, TOR(i));
 	} else {
 		for (i = 0; i < 4; i++)
-			i2s_write_reg(dev->i2s_base, ROR(i), 0);
+			i2s_read_reg(dev->i2s_base, ROR(i));
 	}
 }
 
 static void i2s_start(struct dw_i2s_dev *dev,
 		      struct snd_pcm_substream *substream)
 {
-
+	u32 i, irq;
 	i2s_write_reg(dev->i2s_base, IER, 1);
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		for (i = 0; i < 4; i++) {
+			irq = i2s_read_reg(dev->i2s_base, IMR(i));
+			i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30);
+		}
 		i2s_write_reg(dev->i2s_base, ITER, 1);
-	else
+	} else {
+		for (i = 0; i < 4; i++) {
+			irq = i2s_read_reg(dev->i2s_base, IMR(i));
+			i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03);
+		}
 		i2s_write_reg(dev->i2s_base, IRER, 1);
+	}
 
 	i2s_write_reg(dev->i2s_base, CER, 1);
 }
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 48b2d24dd1f0..b95132e2f9dc 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -95,7 +95,8 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		/* data on rising edge of bclk, frame low 1clk before data */
-		strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0;
+		strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSI |
+			SSI_STCR_TEFS;
 		scr |= SSI_SCR_NET;
 		if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) {
 			scr &= ~SSI_I2S_MODE_MASK;
@@ -104,33 +105,31 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
 		/* data on rising edge of bclk, frame high with data */
-		strcr |= SSI_STCR_TXBIT0;
+		strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
 		/* data on rising edge of bclk, frame high with data */
-		strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0;
+		strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSL;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		/* data on rising edge of bclk, frame high 1clk before data */
-		strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
+		strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSL |
+			SSI_STCR_TEFS;
 		break;
 	}
 
 	/* DAI clock inversion */
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_IB_IF:
-		strcr |= SSI_STCR_TFSI;
-		strcr &= ~SSI_STCR_TSCKP;
+		strcr ^= SSI_STCR_TSCKP | SSI_STCR_TFSI;
 		break;
 	case SND_SOC_DAIFMT_IB_NF:
-		strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI);
+		strcr ^= SSI_STCR_TSCKP;
 		break;
 	case SND_SOC_DAIFMT_NB_IF:
-		strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP;
+		strcr ^= SSI_STCR_TFSI;
 		break;
 	case SND_SOC_DAIFMT_NB_NF:
-		strcr &= ~SSI_STCR_TFSI;
-		strcr |= SSI_STCR_TSCKP;
 		break;
 	}
 
diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c
index 82e350e9501c..ac75816ada7c 100644
--- a/sound/synth/emux/emux_oss.c
+++ b/sound/synth/emux/emux_oss.c
@@ -69,7 +69,8 @@ snd_emux_init_seq_oss(struct snd_emux *emu)
 	struct snd_seq_oss_reg *arg;
 	struct snd_seq_device *dev;
 
-	if (snd_seq_device_new(emu->card, 0, SNDRV_SEQ_DEV_ID_OSS,
+	/* using device#1 here for avoiding conflicts with OPL3 */
+	if (snd_seq_device_new(emu->card, 1, SNDRV_SEQ_DEV_ID_OSS,
 			       sizeof(struct snd_seq_oss_reg), &dev) < 0)
 		return;