Merge tag 'asoc-fix-v4.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v4.3

Quite a few fixes here but they're all very small and driver specific,
none of them really stand out if you aren't using the relevant hardware
but they're all useful if you do happen to have an affected device.

9 files changed, 58 insertions, 38 deletions

diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 58c3164802b8..8c907ebea189 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 268a28bd1df4..5c101af0ac63 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 0e4cfc6ac649..8c964cfb120d 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 1a82b19b2644..8739126a1f6f 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1509,14 +1509,17 @@ 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;
 	}