Merge tag 'sound-4.4-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This shouldn't be a nightmare before Christmas: just a handful small
  device-specific fixes for various ASoC and HD-audio drivers.  Most of
  them are stable fixes"

* tag 'sound-4.4-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda/realtek - Fix silent headphone output on MacPro 4,1 (v2)
  ASoC: fsl_sai: fix no frame clk in master mode
  ALSA: hda - Set SKL+ hda controller power at freeze() and thaw()
  ASoC: sgtl5000: fix VAG power up timing
  ASoC: rockchip: spdif: Set transmit data level to 16 samples
  ASoC: wm8974: set cache type for regmap
  ASoC: es8328: Fix shifts for mixer switches
  ASoC: davinci-mcasp: Fix XDATA check in mcasp_start_tx
  ASoC: es8328: Fix deemphasis values

10 files changed, 88 insertions, 15 deletions

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index bff5c8b329d1..3b3658297070 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -954,6 +954,36 @@ static int azx_resume(struct device *dev)
 }
 #endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */
 
+#ifdef CONFIG_PM_SLEEP
+/* put codec down to D3 at hibernation for Intel SKL+;
+ * otherwise BIOS may still access the codec and screw up the driver
+ */
+#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170)
+#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70)
+#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
+#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci))
+
+static int azx_freeze_noirq(struct device *dev)
+{
+        struct pci_dev *pci = to_pci_dev(dev);
+
+        if (IS_SKL_PLUS(pci))
+                pci_set_power_state(pci, PCI_D3hot);
+
+        return 0;
+}
+
+static int azx_thaw_noirq(struct device *dev)
+{
+        struct pci_dev *pci = to_pci_dev(dev);
+
+        if (IS_SKL_PLUS(pci))
+                pci_set_power_state(pci, PCI_D0);
+
+        return 0;
+}
+#endif /* CONFIG_PM_SLEEP */
+
 #ifdef CONFIG_PM
 static int azx_runtime_suspend(struct device *dev)
 {
@@ -1063,6 +1093,10 @@ static int azx_runtime_idle(struct device *dev)
 
 static const struct dev_pm_ops azx_pm = {
         SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume)
+#ifdef CONFIG_PM_SLEEP
+        .freeze_noirq = azx_freeze_noirq,
+        .thaw_noirq = azx_thaw_noirq,
+#endif
         SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle)
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6c268dad143f..fe96428aa403 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1775,6 +1775,7 @@ enum {
         ALC889_FIXUP_MBA11_VREF,
         ALC889_FIXUP_MBA21_VREF,
         ALC889_FIXUP_MP11_VREF,
+        ALC889_FIXUP_MP41_VREF,
         ALC882_FIXUP_INV_DMIC,
         ALC882_FIXUP_NO_PRIMARY_HP,
         ALC887_FIXUP_ASUS_BASS,
@@ -1863,7 +1864,7 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec,
                                   const struct hda_fixup *fix, int action)
 {
         struct alc_spec *spec = codec->spec;
-        static hda_nid_t nids[2] = { 0x14, 0x15 };
+        static hda_nid_t nids[3] = { 0x14, 0x15, 0x19 };
         int i;
 
         if (action != HDA_FIXUP_ACT_INIT)
@@ -2153,6 +2154,12 @@ static const struct hda_fixup alc882_fixups[] = {
                 .chained = true,
                 .chain_id = ALC885_FIXUP_MACPRO_GPIO,
         },
+        [ALC889_FIXUP_MP41_VREF] = {
+                .type = HDA_FIXUP_FUNC,
+                .v.func = alc889_fixup_mbp_vref,
+                .chained = true,
+                .chain_id = ALC885_FIXUP_MACPRO_GPIO,
+        },
         [ALC882_FIXUP_INV_DMIC] = {
                 .type = HDA_FIXUP_FUNC,
                 .v.func = alc_fixup_inv_dmic,
@@ -2235,7 +2242,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
         SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF),
         SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF),
         SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF),
-        SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO),
+        SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 4,1/5,1", ALC889_FIXUP_MP41_VREF),
         SND_PCI_QUIRK(0x106b, 0x4300, "iMac 9,1", ALC889_FIXUP_IMAC91_VREF),
         SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF),
         SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF),
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 84f5eb07a91b..afa6c5db9dcc 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -85,7 +85,15 @@ static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0);
 static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
 static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0);
 
-static const int deemph_settings[] = { 0, 32000, 44100, 48000 };
+static const struct {
+        int rate;
+        unsigned int val;
+} deemph_settings[] = {
+        { 0,     ES8328_DACCONTROL6_DEEMPH_OFF },
+        { 32000, ES8328_DACCONTROL6_DEEMPH_32k },
+        { 44100, ES8328_DACCONTROL6_DEEMPH_44_1k },
+        { 48000, ES8328_DACCONTROL6_DEEMPH_48k },
+};
 
 static int es8328_set_deemph(struct snd_soc_codec *codec)
 {
@@ -97,21 +105,22 @@ static int es8328_set_deemph(struct snd_soc_codec *codec)
          * rate.
          */
         if (es8328->deemph) {
-                best = 1;
-                for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) {
-                        if (abs(deemph_settings[i] - es8328->playback_fs) <
-                            abs(deemph_settings[best] - es8328->playback_fs))
+                best = 0;
+                for (i = 1; i < ARRAY_SIZE(deemph_settings); i++) {
+                        if (abs(deemph_settings[i].rate - es8328->playback_fs) <
+                            abs(deemph_settings[best].rate - es8328->playback_fs))
                                 best = i;
                 }
 
-                val = best << 1;
+                val = deemph_settings[best].val;
         } else {
-                val = 0;
+                val = ES8328_DACCONTROL6_DEEMPH_OFF;
         }
 
         dev_dbg(codec->dev, "Set deemphasis %d\n", val);
 
-        return snd_soc_update_bits(codec, ES8328_DACCONTROL6, 0x6, val);
+        return snd_soc_update_bits(codec, ES8328_DACCONTROL6,
+                        ES8328_DACCONTROL6_DEEMPH_MASK, val);
 }
 
 static int es8328_get_deemph(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h
index cb36afe10c0e..156c748c89c7 100644
--- a/sound/soc/codecs/es8328.h
+++ b/sound/soc/codecs/es8328.h
@@ -153,6 +153,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_DACCONTROL6_CLICKFREE (1 << 3)
 #define ES8328_DACCONTROL6_DAC_INVR (1 << 4)
 #define ES8328_DACCONTROL6_DAC_INVL (1 << 5)
+#define ES8328_DACCONTROL6_DEEMPH_MASK (3 << 6)
 #define ES8328_DACCONTROL6_DEEMPH_OFF (0 << 6)
 #define ES8328_DACCONTROL6_DEEMPH_32k (1 << 6)
 #define ES8328_DACCONTROL6_DEEMPH_44_1k (2 << 6)
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index f540f82b1f27..08b40460663c 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -189,6 +189,7 @@ static int power_vag_event(struct snd_soc_dapm_widget *w,
         case SND_SOC_DAPM_POST_PMU:
                 snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
                         SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
+                msleep(400);
                 break;
 
         case SND_SOC_DAPM_PRE_PMD:
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 0a60677397b3..4c29bd2ae75c 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -574,6 +574,7 @@ static const struct regmap_config wm8974_regmap = {
         .max_register = WM8974_MONOMIX,
         .reg_defaults = wm8974_reg_defaults,
         .num_reg_defaults = ARRAY_SIZE(wm8974_reg_defaults),
+        .cache_type = REGCACHE_FLAT,
 };
 
 static int wm8974_probe(struct snd_soc_codec *codec)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index c1c9c2e3525b..2ccb8bccc9d4 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -223,8 +223,8 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp)
 
         /* wait for XDATA to be cleared */
         cnt = 0;
-        while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) &
-                 ~XRDATA) && (cnt < 100000))
+        while ((mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & XRDATA) &&
+               (cnt < 100000))
                 cnt++;
 
         /* Release TX state machine */
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index ffd5f9acc849..08b460ba06ef 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -505,6 +505,24 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
                                            FSL_SAI_CSR_FR, FSL_SAI_CSR_FR);
                         regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
                                            FSL_SAI_CSR_FR, FSL_SAI_CSR_FR);
+
+                        /*
+                         * For sai master mode, after several open/close sai,
+                         * there will be no frame clock, and can't recover
+                         * anymore. Add software reset to fix this issue.
+                         * This is a hardware bug, and will be fix in the
+                         * next sai version.
+                         */
+                        if (!sai->is_slave_mode) {
+                                /* Software Reset for both Tx and Rx */
+                                regmap_write(sai->regmap,
+                                             FSL_SAI_TCSR, FSL_SAI_CSR_SR);
+                                regmap_write(sai->regmap,
+                                             FSL_SAI_RCSR, FSL_SAI_CSR_SR);
+                                /* Clear SR bit to finish the reset */
+                                regmap_write(sai->regmap, FSL_SAI_TCSR, 0);
+                                regmap_write(sai->regmap, FSL_SAI_RCSR, 0);
+                        }
                 }
                 break;
         default:
diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c
index ac72ff5055bb..5a806da89f42 100644
--- a/sound/soc/rockchip/rockchip_spdif.c
+++ b/sound/soc/rockchip/rockchip_spdif.c
@@ -152,8 +152,10 @@ static int rk_spdif_trigger(struct snd_pcm_substream *substream,
         case SNDRV_PCM_TRIGGER_RESUME:
         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
                 ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR,
-                                   SPDIF_DMACR_TDE_ENABLE,
-                                   SPDIF_DMACR_TDE_ENABLE);
+                                   SPDIF_DMACR_TDE_ENABLE |
+                                   SPDIF_DMACR_TDL_MASK,
+                                   SPDIF_DMACR_TDE_ENABLE |
+                                   SPDIF_DMACR_TDL(16));
 
                 if (ret != 0)
                         return ret;
diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h
index 921b4095fb92..3ef12770ae12 100644
--- a/sound/soc/rockchip/rockchip_spdif.h
+++ b/sound/soc/rockchip/rockchip_spdif.h
@@ -42,7 +42,7 @@
 
 #define SPDIF_DMACR_TDL_SHIFT   0
 #define SPDIF_DMACR_TDL(x)      ((x) << SPDIF_DMACR_TDL_SHIFT)
-#define SPDIF_DMACR_TDL_MASK    (0x1f << SDPIF_DMACR_TDL_SHIFT)
+#define SPDIF_DMACR_TDL_MASK    (0x1f << SPDIF_DMACR_TDL_SHIFT)
 
 /*
  * XFER