13 files changed, 67 insertions, 40 deletions

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_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 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;
         }
 
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;