44 files changed, 608 insertions, 174 deletions

diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 5d99436dfcae..0cda05c72f50 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -12,9 +12,11 @@ MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_LICENSE("GPL v2");
 
 #define OUI_WEISS               0x001c6a
+#define OUI_LOUD                0x000ff2
 
 #define DICE_CATEGORY_ID        0x04
 #define WEISS_CATEGORY_ID       0x00
+#define LOUD_CATEGORY_ID        0x10
 
 static int dice_interface_check(struct fw_unit *unit)
 {
@@ -57,6 +59,8 @@ static int dice_interface_check(struct fw_unit *unit)
         }
         if (vendor == OUI_WEISS)
                 category = WEISS_CATEGORY_ID;
+        else if (vendor == OUI_LOUD)
+                category = LOUD_CATEGORY_ID;
         else
                 category = DICE_CATEGORY_ID;
         if (device->config_rom[3] != ((vendor << 8) | category) ||
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 8a7fbdcb4072..3b3658297070 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -312,6 +312,10 @@ enum {
         (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\
          AZX_DCAPS_I915_POWERWELL)
 
+#define AZX_DCAPS_INTEL_BROXTON \
+        (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\
+         AZX_DCAPS_I915_POWERWELL)
+
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
         (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\
@@ -351,6 +355,8 @@ enum {
                                         ((pci)->device == 0x0d0c) || \
                                         ((pci)->device == 0x160c))
 
+#define IS_BROXTON(pci) ((pci)->device == 0x5a98)
+
 static char *driver_short_names[] = {
         [AZX_DRIVER_ICH] = "HDA Intel",
         [AZX_DRIVER_PCH] = "HDA Intel PCH",
@@ -502,15 +508,36 @@ static void azx_init_pci(struct azx *chip)
         }
 }
 
+/*
+ * In BXT-P A0, HD-Audio DMA requests is later than expected,
+ * and makes an audio stream sensitive to system latencies when
+ * 24/32 bits are playing.
+ * Adjusting threshold of DMA fifo to force the DMA request
+ * sooner to improve latency tolerance at the expense of power.
+ */
+static void bxt_reduce_dma_latency(struct azx *chip)
+{
+        u32 val;
+
+        val = azx_readl(chip, SKL_EM4L);
+        val &= (0x3 << 20);
+        azx_writel(chip, SKL_EM4L, val);
+}
+
 static void hda_intel_init_chip(struct azx *chip, bool full_reset)
 {
         struct hdac_bus *bus = azx_bus(chip);
+        struct pci_dev *pci = chip->pci;
 
         if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
                 snd_hdac_set_codec_wakeup(bus, true);
         azx_init_chip(chip, full_reset);
         if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
                 snd_hdac_set_codec_wakeup(bus, false);
+
+        /* reduce dma latency to avoid noise */
+        if (IS_BROXTON(pci))
+                bxt_reduce_dma_latency(chip);
 }
 
 /* calculate runtime delay from LPIB */
@@ -927,6 +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)
 {
@@ -1036,6 +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)
 };
 
@@ -2124,6 +2185,9 @@ static const struct pci_device_id azx_ids[] = {
         /* Sunrise Point-LP */
         { PCI_DEVICE(0x8086, 0x9d70),
           .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE },
+        /* Broxton-P(Apollolake) */
+        { PCI_DEVICE(0x8086, 0x5a98),
+          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON },
         /* Haswell */
         { PCI_DEVICE(0x8086, 0x0a0c),
           .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL },
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index f8a12ca477f1..4ef2259f88ca 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -778,7 +778,8 @@ static const struct hda_pintbl alienware_pincfgs[] = {
 };
 
 static const struct snd_pci_quirk ca0132_quirks[] = {
-        SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15", QUIRK_ALIENWARE),
+        SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE),
+        SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE),
         {}
 };
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index c8b8ef5246a6..ef198903c0c3 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -955,6 +955,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
  */
 
 static const struct hda_device_id snd_hda_id_conexant[] = {
+        HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto),
         HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto),
         HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto),
         HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto),
@@ -972,9 +973,9 @@ static const struct hda_device_id snd_hda_id_conexant[] = {
         HDA_CODEC_ENTRY(0x14f150ac, "CX20652", patch_conexant_auto),
         HDA_CODEC_ENTRY(0x14f150b8, "CX20664", patch_conexant_auto),
         HDA_CODEC_ENTRY(0x14f150b9, "CX20665", patch_conexant_auto),
-        HDA_CODEC_ENTRY(0x14f150f1, "CX20721", patch_conexant_auto),
+        HDA_CODEC_ENTRY(0x14f150f1, "CX21722", patch_conexant_auto),
         HDA_CODEC_ENTRY(0x14f150f2, "CX20722", patch_conexant_auto),
-        HDA_CODEC_ENTRY(0x14f150f3, "CX20723", patch_conexant_auto),
+        HDA_CODEC_ENTRY(0x14f150f3, "CX21724", patch_conexant_auto),
         HDA_CODEC_ENTRY(0x14f150f4, "CX20724", patch_conexant_auto),
         HDA_CODEC_ENTRY(0x14f1510f, "CX20751/2", patch_conexant_auto),
         HDA_CODEC_ENTRY(0x14f15110, "CX20751/2", patch_conexant_auto),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 60cd9e700909..4b6fb668c91c 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2352,6 +2352,12 @@ static void intel_pin_eld_notify(void *audio_ptr, int port)
         struct hda_codec *codec = audio_ptr;
         int pin_nid = port + 0x04;
 
+        /* skip notification during system suspend (but not in runtime PM);
+         * the state will be updated at resume
+         */
+        if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0)
+                return;
+
         check_presence_and_report(codec, pin_nid);
 }
 
@@ -2378,7 +2384,8 @@ static int patch_generic_hdmi(struct hda_codec *codec)
          * can cover the codec power request, and so need not set this flag.
          * For previous platforms, there is no such power well feature.
          */
-        if (is_valleyview_plus(codec) || is_skylake(codec))
+        if (is_valleyview_plus(codec) || is_skylake(codec) ||
+                        is_broxton(codec))
                 codec->core.link_power_control = 1;
 
         if (is_haswell_plus(codec) || is_valleyview_plus(codec)) {
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2f7b065f9ac4..fe96428aa403 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -111,6 +111,7 @@ struct alc_spec {
         void (*power_hook)(struct hda_codec *codec);
 #endif
         void (*shutup)(struct hda_codec *codec);
+        void (*reboot_notify)(struct hda_codec *codec);
 
         int init_amp;
         int codec_variant;      /* flag for other variants */
@@ -773,6 +774,25 @@ static inline void alc_shutup(struct hda_codec *codec)
                 snd_hda_shutup_pins(codec);
 }
 
+static void alc_reboot_notify(struct hda_codec *codec)
+{
+        struct alc_spec *spec = codec->spec;
+
+        if (spec && spec->reboot_notify)
+                spec->reboot_notify(codec);
+        else
+                alc_shutup(codec);
+}
+
+/* power down codec to D3 at reboot/shutdown; set as reboot_notify ops */
+static void alc_d3_at_reboot(struct hda_codec *codec)
+{
+        snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
+        snd_hda_codec_write(codec, codec->core.afg, 0,
+                            AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+        msleep(10);
+}
+
 #define alc_free        snd_hda_gen_free
 
 #ifdef CONFIG_PM
@@ -818,7 +838,7 @@ static const struct hda_codec_ops alc_patch_ops = {
         .suspend = alc_suspend,
         .check_power_status = snd_hda_gen_check_power_status,
 #endif
-        .reboot_notify = alc_shutup,
+        .reboot_notify = alc_reboot_notify,
 };
 
 
@@ -1755,10 +1775,12 @@ 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,
         ALC887_FIXUP_BASS_CHMAP,
+        ALC882_FIXUP_DISABLE_AAMIX,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -1842,7 +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)
@@ -1920,6 +1942,8 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
 
 static void alc_fixup_bass_chmap(struct hda_codec *codec,
                                  const struct hda_fixup *fix, int action);
+static void alc_fixup_disable_aamix(struct hda_codec *codec,
+                                    const struct hda_fixup *fix, int action);
 
 static const struct hda_fixup alc882_fixups[] = {
         [ALC882_FIXUP_ABIT_AW9D_MAX] = {
@@ -2130,6 +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,
@@ -2151,6 +2181,10 @@ static const struct hda_fixup alc882_fixups[] = {
                 .type = HDA_FIXUP_FUNC,
                 .v.func = alc_fixup_bass_chmap,
         },
+        [ALC882_FIXUP_DISABLE_AAMIX] = {
+                .type = HDA_FIXUP_FUNC,
+                .v.func = alc_fixup_disable_aamix,
+        },
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -2208,7 +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),
@@ -2218,6 +2252,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
         SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
         SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE),
+        SND_PCI_QUIRK(0x1458, 0xa182, "Gigabyte Z170X-UD3", ALC882_FIXUP_DISABLE_AAMIX),
         SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
         SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD),
         SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD),
@@ -4190,6 +4225,8 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec,
         struct alc_spec *spec = codec->spec;
 
         if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+                spec->shutup = alc_no_shutup; /* reduce click noise */
+                spec->reboot_notify = alc_d3_at_reboot; /* reduce noise */
                 spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
                 codec->power_save_node = 0; /* avoid click noises */
                 snd_hda_apply_pincfgs(codec, pincfgs);
@@ -4570,6 +4607,7 @@ enum {
         ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC,
         ALC293_FIXUP_DELL1_MIC_NO_PRESENCE,
         ALC292_FIXUP_TPT440_DOCK,
+        ALC292_FIXUP_TPT440,
         ALC283_FIXUP_BXBT2807_MIC,
         ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
         ALC282_FIXUP_ASPIRE_V5_PINS,
@@ -4585,8 +4623,11 @@ enum {
         ALC288_FIXUP_DISABLE_AAMIX,
         ALC292_FIXUP_DELL_E7X,
         ALC292_FIXUP_DISABLE_AAMIX,
+        ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK,
         ALC298_FIXUP_DELL1_MIC_NO_PRESENCE,
         ALC275_FIXUP_DELL_XPS,
+        ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE,
+        ALC293_FIXUP_LENOVO_SPK_NOISE,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -5041,6 +5082,12 @@ static const struct hda_fixup alc269_fixups[] = {
                 .chained = true,
                 .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
         },
+        [ALC292_FIXUP_TPT440] = {
+                .type = HDA_FIXUP_FUNC,
+                .v.func = alc_fixup_disable_aamix,
+                .chained = true,
+                .chain_id = ALC292_FIXUP_TPT440_DOCK,
+        },
         [ALC283_FIXUP_BXBT2807_MIC] = {
                 .type = HDA_FIXUP_PINS,
                 .v.pins = (const struct hda_pintbl[]) {
@@ -5140,6 +5187,12 @@ static const struct hda_fixup alc269_fixups[] = {
                 .chained = true,
                 .chain_id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE
         },
+        [ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK] = {
+                .type = HDA_FIXUP_FUNC,
+                .v.func = alc_fixup_disable_aamix,
+                .chained = true,
+                .chain_id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE
+        },
         [ALC292_FIXUP_DELL_E7X] = {
                 .type = HDA_FIXUP_FUNC,
                 .v.func = alc_fixup_dell_xps13,
@@ -5167,6 +5220,23 @@ static const struct hda_fixup alc269_fixups[] = {
                         {}
                 }
         },
+        [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE] = {
+                .type = HDA_FIXUP_VERBS,
+                .v.verbs = (const struct hda_verb[]) {
+                        /* Disable pass-through path for FRONT 14h */
+                        {0x20, AC_VERB_SET_COEF_INDEX, 0x36},
+                        {0x20, AC_VERB_SET_PROC_COEF, 0x1737},
+                        {}
+                },
+                .chained = true,
+                .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
+        },
+        [ALC293_FIXUP_LENOVO_SPK_NOISE] = {
+                .type = HDA_FIXUP_FUNC,
+                .v.func = alc_fixup_disable_aamix,
+                .chained = true,
+                .chain_id = ALC269_FIXUP_THINKPAD_ACPI
+        },
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5180,8 +5250,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
         SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
         SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
+        SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
         SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
         SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
+        SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
         SND_PCI_QUIRK(0x1028, 0x05ca, "Dell Latitude E7240", ALC292_FIXUP_DELL_E7X),
         SND_PCI_QUIRK(0x1028, 0x05cb, "Dell Latitude E7440", ALC292_FIXUP_DELL_E7X),
         SND_PCI_QUIRK(0x1028, 0x05da, "Dell Vostro 5460", ALC290_FIXUP_SUBWOOFER),
@@ -5199,11 +5271,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
-        SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
-        SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
-        SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
-        SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
-        SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC292_FIXUP_DISABLE_AAMIX),
+        SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
+        SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
+        SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
+        SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
+        SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
+        SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE),
         SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -5302,15 +5375,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
         SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
-        SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK),
+        SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440),
         SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x2211, "Thinkpad W541", ALC292_FIXUP_TPT440_DOCK),
         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, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK),
         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, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE),
         SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
         SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP),
         SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -5320,6 +5395,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
         SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK),
+        SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE),
         SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
         SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
         SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
@@ -5400,6 +5476,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
         {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"},
         {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"},
         {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"},
+        {.id = ALC292_FIXUP_TPT440, .name = "tpt440"},
         {}
 };
 
@@ -6386,6 +6463,7 @@ static const struct hda_fixup alc662_fixups[] = {
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2),
         SND_PCI_QUIRK(0x1025, 0x022f, "Acer Aspire One", ALC662_FIXUP_INV_DMIC),
+        SND_PCI_QUIRK(0x1025, 0x0241, "Packard Bell DOTS", ALC662_FIXUP_INV_DMIC),
         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),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 826122d8acee..2c7c5eb8b1e9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3110,6 +3110,29 @@ static void stac92hd71bxx_fixup_hp_hdx(struct hda_codec *codec,
         spec->gpio_led = 0x08;
 }
 
+static bool is_hp_output(struct hda_codec *codec, hda_nid_t pin)
+{
+        unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin);
+
+        /* count line-out, too, as BIOS sets often so */
+        return get_defcfg_connect(pin_cfg) != AC_JACK_PORT_NONE &&
+                (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT ||
+                 get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT);
+}
+
+static void fixup_hp_headphone(struct hda_codec *codec, hda_nid_t pin)
+{
+        unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin);
+
+        /* It was changed in the BIOS to just satisfy MS DTM.
+         * Lets turn it back into slaved HP
+         */
+        pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) |
+                (AC_JACK_HP_OUT << AC_DEFCFG_DEVICE_SHIFT);
+        pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC | AC_DEFCFG_SEQUENCE))) |
+                0x1f;
+        snd_hda_codec_set_pincfg(codec, pin, pin_cfg);
+}
 
 static void stac92hd71bxx_fixup_hp(struct hda_codec *codec,
                                    const struct hda_fixup *fix, int action)
@@ -3119,22 +3142,12 @@ static void stac92hd71bxx_fixup_hp(struct hda_codec *codec,
         if (action != HDA_FIXUP_ACT_PRE_PROBE)
                 return;
 
-        if (hp_blike_system(codec->core.subsystem_id)) {
-                unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f);
-                if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT ||
-                        get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER  ||
-                        get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT) {
-                        /* It was changed in the BIOS to just satisfy MS DTM.
-                         * Lets turn it back into slaved HP
-                         */
-                        pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE))
-                                        | (AC_JACK_HP_OUT <<
-                                                AC_DEFCFG_DEVICE_SHIFT);
-                        pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC
-                                                        | AC_DEFCFG_SEQUENCE)))
-                                                                | 0x1f;
-                        snd_hda_codec_set_pincfg(codec, 0x0f, pin_cfg);
-                }
+        /* when both output A and F are assigned, these are supposedly
+         * dock and built-in headphones; fix both pin configs
+         */
+        if (is_hp_output(codec, 0x0a) && is_hp_output(codec, 0x0f)) {
+                fixup_hp_headphone(codec, 0x0a);
+                fixup_hp_headphone(codec, 0x0f);
         }
 
         if (find_mute_led_cfg(codec, 1))
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 714df906249e..41c31db65039 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -741,10 +741,11 @@ snd_rme96_playback_setrate(struct rme96 *rme96,
         {
                 /* change to/from double-speed: reset the DAC (if available) */
                 snd_rme96_reset_dac(rme96);
+                return 1; /* need to restore volume */
         } else {
                 writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+                return 0;
         }
-        return 0;
 }
 
 static int
@@ -980,6 +981,7 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream,
         struct rme96 *rme96 = snd_pcm_substream_chip(substream);
         struct snd_pcm_runtime *runtime = substream->runtime;
         int err, rate, dummy;
+        bool apply_dac_volume = false;
 
         runtime->dma_area = (void __force *)(rme96->iobase +
                                              RME96_IO_PLAY_BUFFER);
@@ -993,24 +995,26 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream,
         {
                 /* slave clock */
                 if ((int)params_rate(params) != rate) {
-                        spin_unlock_irq(&rme96->lock);
-                        return -EIO;                    
-                }
-        } else if ((err = snd_rme96_playback_setrate(rme96, params_rate(params))) < 0) {
-                spin_unlock_irq(&rme96->lock);
-                return err;
-        }
-        if ((err = snd_rme96_playback_setformat(rme96, params_format(params))) < 0) {
-                spin_unlock_irq(&rme96->lock);
-                return err;
+                        err = -EIO;
+                        goto error;
+                }
+        } else {
+                err = snd_rme96_playback_setrate(rme96, params_rate(params));
+                if (err < 0)
+                        goto error;
+                apply_dac_volume = err > 0; /* need to restore volume later? */
         }
+
+        err = snd_rme96_playback_setformat(rme96, params_format(params));
+        if (err < 0)
+                goto error;
         snd_rme96_setframelog(rme96, params_channels(params), 1);
         if (rme96->capture_periodsize != 0) {
                 if (params_period_size(params) << rme96->playback_frlog !=
                     rme96->capture_periodsize)
                 {
-                        spin_unlock_irq(&rme96->lock);
-                        return -EBUSY;
+                        err = -EBUSY;
+                        goto error;
                 }
         }
         rme96->playback_periodsize =
@@ -1021,9 +1025,16 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream,
                 rme96->wcreg &= ~(RME96_WCR_PRO | RME96_WCR_DOLBY | RME96_WCR_EMP);
                 writel(rme96->wcreg |= rme96->wcreg_spdif_stream, rme96->iobase + RME96_IO_CONTROL_REGISTER);
         }
+
+        err = 0;
+ error:
         spin_unlock_irq(&rme96->lock);
-                
-        return 0;
+        if (apply_dac_volume) {
+                usleep_range(3000, 10000);
+                snd_rme96_apply_dac_volume(rme96);
+        }
+
+        return err;
 }
 
 static int
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 9929efc6b9aa..b3ea24d64c50 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1023,24 +1023,18 @@ void arizona_init_dvfs(struct arizona_priv *priv)
 }
 EXPORT_SYMBOL_GPL(arizona_init_dvfs);
 
-static unsigned int arizona_sysclk_48k_rates[] = {
+static unsigned int arizona_opclk_ref_48k_rates[] = {
         6144000,
         12288000,
         24576000,
         49152000,
-        73728000,
-        98304000,
-        147456000,
 };
 
-static unsigned int arizona_sysclk_44k1_rates[] = {
+static unsigned int arizona_opclk_ref_44k1_rates[] = {
         5644800,
         11289600,
         22579200,
         45158400,
-        67737600,
-        90316800,
-        135475200,
 };
 
 static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk,
@@ -1065,11 +1059,11 @@ static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk,
         }
 
         if (refclk % 8000)
-                rates = arizona_sysclk_44k1_rates;
+                rates = arizona_opclk_ref_44k1_rates;
         else
-                rates = arizona_sysclk_48k_rates;
+                rates = arizona_opclk_ref_48k_rates;
 
-        for (ref = 0; ref < ARRAY_SIZE(arizona_sysclk_48k_rates) &&
+        for (ref = 0; ref < ARRAY_SIZE(arizona_opclk_ref_48k_rates) &&
                      rates[ref] <= refclk; ref++) {
                 div = 1;
                 while (rates[ref] / div >= freq && div < 32) {
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 969e337dc17c..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,
@@ -205,18 +214,18 @@ static const struct snd_kcontrol_new es8328_right_line_controls =
 
 /* Left Mixer */
 static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
-        SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0),
-        SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0),
-        SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0),
-        SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0),
+        SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0),
+        SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0),
+        SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0),
+        SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0),
 };
 
 /* Right Mixer */
 static const struct snd_kcontrol_new es8328_right_mixer_controls[] = {
-        SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0),
-        SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0),
-        SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0),
-        SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0),
+        SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0),
+        SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0),
+        SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0),
+        SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0),
 };
 
 static const char * const es8328_pga_sel[] = {
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/nau8825.c b/sound/soc/codecs/nau8825.c
index 7fc7b4e3f444..c1b87c5800b1 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -1271,6 +1271,36 @@ static int nau8825_i2c_remove(struct i2c_client *client)
         return 0;
 }
 
+#ifdef CONFIG_PM_SLEEP
+static int nau8825_suspend(struct device *dev)
+{
+        struct i2c_client *client = to_i2c_client(dev);
+        struct nau8825 *nau8825 = dev_get_drvdata(dev);
+
+        disable_irq(client->irq);
+        regcache_cache_only(nau8825->regmap, true);
+        regcache_mark_dirty(nau8825->regmap);
+
+        return 0;
+}
+
+static int nau8825_resume(struct device *dev)
+{
+        struct i2c_client *client = to_i2c_client(dev);
+        struct nau8825 *nau8825 = dev_get_drvdata(dev);
+
+        regcache_cache_only(nau8825->regmap, false);
+        regcache_sync(nau8825->regmap);
+        enable_irq(client->irq);
+
+        return 0;
+}
+#endif
+
+static const struct dev_pm_ops nau8825_pm = {
+        SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume)
+};
+
 static const struct i2c_device_id nau8825_i2c_ids[] = {
         { "nau8825", 0 },
         { }
@@ -1297,6 +1327,7 @@ static struct i2c_driver nau8825_driver = {
                 .name = "nau8825",
                 .of_match_table = of_match_ptr(nau8825_of_ids),
                 .acpi_match_table = ACPI_PTR(nau8825_acpi_match),
+                .pm = &nau8825_pm,
         },
         .probe = nau8825_i2c_probe,
         .remove = nau8825_i2c_remove,
diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c
index aca479fa7670..1dc68ab08a17 100644
--- a/sound/soc/codecs/rl6231.c
+++ b/sound/soc/codecs/rl6231.c
@@ -80,8 +80,10 @@ int rl6231_calc_dmic_clk(int rate)
         }
 
         for (i = 0; i < ARRAY_SIZE(div); i++) {
-                /* find divider that gives DMIC frequency below 3MHz */
-                if (3000000 * div[i] >= rate)
+                if ((div[i] % 3) == 0)
+                        continue;
+                /* find divider that gives DMIC frequency below 3.072MHz */
+                if (3072000 * div[i] >= rate)
                         return i;
         }
 
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 28132375e427..ef76940f9dcb 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -245,7 +245,7 @@ struct rt5645_priv {
         struct snd_soc_jack *hp_jack;
         struct snd_soc_jack *mic_jack;
         struct snd_soc_jack *btn_jack;
-        struct delayed_work jack_detect_work;
+        struct delayed_work jack_detect_work, rcclock_work;
         struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)];
         struct rt5645_eq_param_s *eq_param;
 
@@ -565,12 +565,33 @@ static int rt5645_hweq_put(struct snd_kcontrol *kcontrol,
         .put = rt5645_hweq_put \
 }
 
+static int rt5645_spk_put_volsw(struct snd_kcontrol *kcontrol,
+                struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+        struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component);
+        int ret;
+
+        cancel_delayed_work_sync(&rt5645->rcclock_work);
+
+        regmap_update_bits(rt5645->regmap, RT5645_MICBIAS,
+                RT5645_PWR_CLK25M_MASK, RT5645_PWR_CLK25M_PU);
+
+        ret = snd_soc_put_volsw(kcontrol, ucontrol);
+
+        queue_delayed_work(system_power_efficient_wq, &rt5645->rcclock_work,
+                msecs_to_jiffies(200));
+
+        return ret;
+}
+
 static const struct snd_kcontrol_new rt5645_snd_controls[] = {
         /* Speaker Output Volume */
         SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL,
                 RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1),
-        SOC_DOUBLE_TLV("Speaker Playback Volume", RT5645_SPK_VOL,
-                RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv),
+        SOC_DOUBLE_EXT_TLV("Speaker Playback Volume", RT5645_SPK_VOL,
+                RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, snd_soc_get_volsw,
+                rt5645_spk_put_volsw, out_vol_tlv),
 
         /* ClassD modulator Speaker Gain Ratio */
         SOC_SINGLE_TLV("Speaker ClassD Playback Volume", RT5645_SPO_CLSD_RATIO,
@@ -1498,7 +1519,7 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on)
                                 regmap_write(rt5645->regmap, RT5645_PR_BASE +
                                         RT5645_MAMP_INT_REG2, 0xfc00);
                                 snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140);
-                                msleep(40);
+                                msleep(70);
                                 rt5645->hp_on = true;
                         } else {
                                 /* depop parameters */
@@ -3122,6 +3143,15 @@ static void rt5645_jack_detect_work(struct work_struct *work)
                                 SND_JACK_BTN_2 | SND_JACK_BTN_3);
 }
 
+static void rt5645_rcclock_work(struct work_struct *work)
+{
+        struct rt5645_priv *rt5645 =
+                container_of(work, struct rt5645_priv, rcclock_work.work);
+
+        regmap_update_bits(rt5645->regmap, RT5645_MICBIAS,
+                RT5645_PWR_CLK25M_MASK, RT5645_PWR_CLK25M_PD);
+}
+
 static irqreturn_t rt5645_irq(int irq, void *data)
 {
         struct rt5645_priv *rt5645 = data;
@@ -3348,6 +3378,27 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
                         DMI_MATCH(DMI_PRODUCT_NAME, "Reks"),
                 },
         },
+        {
+                .ident = "Google Edgar",
+                .callback = strago_quirk_cb,
+                .matches = {
+                        DMI_MATCH(DMI_PRODUCT_NAME, "Edgar"),
+                },
+        },
+        {
+                .ident = "Google Wizpig",
+                .callback = strago_quirk_cb,
+                .matches = {
+                        DMI_MATCH(DMI_PRODUCT_NAME, "Wizpig"),
+                },
+        },
+        {
+                .ident = "Google Terra",
+                .callback = strago_quirk_cb,
+                .matches = {
+                        DMI_MATCH(DMI_PRODUCT_NAME, "Terra"),
+                },
+        },
         { }
 };
 
@@ -3587,6 +3638,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
         }
 
         INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work);
+        INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work);
 
         if (rt5645->i2c->irq) {
                 ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq,
@@ -3621,6 +3673,7 @@ static int rt5645_i2c_remove(struct i2c_client *i2c)
                 free_irq(i2c->irq, rt5645);
 
         cancel_delayed_work_sync(&rt5645->jack_detect_work);
+        cancel_delayed_work_sync(&rt5645->rcclock_work);
 
         snd_soc_unregister_codec(&i2c->dev);
         regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies);
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index dc2b46236c5c..3f1b0f1df809 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -973,12 +973,12 @@
 #define RT5670_SCLK_SRC_MCLK                    (0x0 << 14)
 #define RT5670_SCLK_SRC_PLL1                    (0x1 << 14)
 #define RT5670_SCLK_SRC_RCCLK                   (0x2 << 14) /* 15MHz */
-#define RT5670_PLL1_SRC_MASK                    (0x3 << 12)
-#define RT5670_PLL1_SRC_SFT                     12
-#define RT5670_PLL1_SRC_MCLK                    (0x0 << 12)
-#define RT5670_PLL1_SRC_BCLK1                   (0x1 << 12)
-#define RT5670_PLL1_SRC_BCLK2                   (0x2 << 12)
-#define RT5670_PLL1_SRC_BCLK3                   (0x3 << 12)
+#define RT5670_PLL1_SRC_MASK                    (0x7 << 11)
+#define RT5670_PLL1_SRC_SFT                     11
+#define RT5670_PLL1_SRC_MCLK                    (0x0 << 11)
+#define RT5670_PLL1_SRC_BCLK1                   (0x1 << 11)
+#define RT5670_PLL1_SRC_BCLK2                   (0x2 << 11)
+#define RT5670_PLL1_SRC_BCLK3                   (0x3 << 11)
 #define RT5670_PLL1_PD_MASK                     (0x1 << 3)
 #define RT5670_PLL1_PD_SFT                      3
 #define RT5670_PLL1_PD_1                        (0x0 << 3)
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index b4cd7e3bf5f8..69d987a9935c 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -1386,90 +1386,90 @@ static const struct snd_kcontrol_new rt5677_dac_r_mix[] = {
 };
 
 static const struct snd_kcontrol_new rt5677_sto1_dac_l_mix[] = {
-        SOC_DAPM_SINGLE("ST L Switch", RT5677_STO1_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("ST L Switch", RT5677_STO1_DAC_MIXER,
                         RT5677_M_ST_DAC1_L_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER,
                         RT5677_M_DAC1_L_STO_L_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_STO1_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_STO1_DAC_MIXER,
                         RT5677_M_DAC2_L_STO_L_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER,
                         RT5677_M_DAC1_R_STO_L_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_sto1_dac_r_mix[] = {
-        SOC_DAPM_SINGLE("ST R Switch", RT5677_STO1_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("ST R Switch", RT5677_STO1_DAC_MIXER,
                         RT5677_M_ST_DAC1_R_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER,
                         RT5677_M_DAC1_R_STO_R_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_STO1_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_STO1_DAC_MIXER,
                         RT5677_M_DAC2_R_STO_R_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER,
                         RT5677_M_DAC1_L_STO_R_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_mono_dac_l_mix[] = {
-        SOC_DAPM_SINGLE("ST L Switch", RT5677_MONO_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("ST L Switch", RT5677_MONO_DAC_MIXER,
                         RT5677_M_ST_DAC2_L_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_MONO_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_MONO_DAC_MIXER,
                         RT5677_M_DAC1_L_MONO_L_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER,
                         RT5677_M_DAC2_L_MONO_L_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER,
                         RT5677_M_DAC2_R_MONO_L_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_mono_dac_r_mix[] = {
-        SOC_DAPM_SINGLE("ST R Switch", RT5677_MONO_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("ST R Switch", RT5677_MONO_DAC_MIXER,
                         RT5677_M_ST_DAC2_R_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_MONO_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_MONO_DAC_MIXER,
                         RT5677_M_DAC1_R_MONO_R_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER,
                         RT5677_M_DAC2_R_MONO_R_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER,
                         RT5677_M_DAC2_L_MONO_R_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_dd1_l_mix[] = {
-        SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5677_DD1_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix L Switch", RT5677_DD1_MIXER,
                         RT5677_M_STO_L_DD1_L_SFT, 1, 1),
-        SOC_DAPM_SINGLE("Mono DAC Mix L Switch", RT5677_DD1_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix L Switch", RT5677_DD1_MIXER,
                         RT5677_M_MONO_L_DD1_L_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC3 L Switch", RT5677_DD1_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC3 L Switch", RT5677_DD1_MIXER,
                         RT5677_M_DAC3_L_DD1_L_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC3 R Switch", RT5677_DD1_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC3 R Switch", RT5677_DD1_MIXER,
                         RT5677_M_DAC3_R_DD1_L_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_dd1_r_mix[] = {
-        SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5677_DD1_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix R Switch", RT5677_DD1_MIXER,
                         RT5677_M_STO_R_DD1_R_SFT, 1, 1),
-        SOC_DAPM_SINGLE("Mono DAC Mix R Switch", RT5677_DD1_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix R Switch", RT5677_DD1_MIXER,
                         RT5677_M_MONO_R_DD1_R_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC3 R Switch", RT5677_DD1_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC3 R Switch", RT5677_DD1_MIXER,
                         RT5677_M_DAC3_R_DD1_R_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC3 L Switch", RT5677_DD1_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC3 L Switch", RT5677_DD1_MIXER,
                         RT5677_M_DAC3_L_DD1_R_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_dd2_l_mix[] = {
-        SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5677_DD2_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix L Switch", RT5677_DD2_MIXER,
                         RT5677_M_STO_L_DD2_L_SFT, 1, 1),
-        SOC_DAPM_SINGLE("Mono DAC Mix L Switch", RT5677_DD2_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix L Switch", RT5677_DD2_MIXER,
                         RT5677_M_MONO_L_DD2_L_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC4 L Switch", RT5677_DD2_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC4 L Switch", RT5677_DD2_MIXER,
                         RT5677_M_DAC4_L_DD2_L_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC4 R Switch", RT5677_DD2_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC4 R Switch", RT5677_DD2_MIXER,
                         RT5677_M_DAC4_R_DD2_L_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5677_dd2_r_mix[] = {
-        SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5677_DD2_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix R Switch", RT5677_DD2_MIXER,
                         RT5677_M_STO_R_DD2_R_SFT, 1, 1),
-        SOC_DAPM_SINGLE("Mono DAC Mix R Switch", RT5677_DD2_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix R Switch", RT5677_DD2_MIXER,
                         RT5677_M_MONO_R_DD2_R_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC4 R Switch", RT5677_DD2_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC4 R Switch", RT5677_DD2_MIXER,
                         RT5677_M_DAC4_R_DD2_R_SFT, 1, 1),
-        SOC_DAPM_SINGLE("DAC4 L Switch", RT5677_DD2_MIXER,
+        SOC_DAPM_SINGLE_AUTODISABLE("DAC4 L Switch", RT5677_DD2_MIXER,
                         RT5677_M_DAC4_L_DD2_R_SFT, 1, 1),
 };
 
@@ -2596,6 +2596,21 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w,
         return 0;
 }
 
+static int rt5677_filter_power_event(struct snd_soc_dapm_widget *w,
+        struct snd_kcontrol *kcontrol, int event)
+{
+        switch (event) {
+        case SND_SOC_DAPM_POST_PMU:
+                msleep(50);
+                break;
+
+        default:
+                return 0;
+        }
+
+        return 0;
+}
+
 static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
         SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT,
                 0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU |
@@ -3072,19 +3087,26 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
 
         /* DAC Mixer */
         SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5677_PWR_DIG2,
-                RT5677_PWR_DAC_S1F_BIT, 0, NULL, 0),
+                RT5677_PWR_DAC_S1F_BIT, 0, rt5677_filter_power_event,
+                SND_SOC_DAPM_POST_PMU),
         SND_SOC_DAPM_SUPPLY("dac mono2 left filter", RT5677_PWR_DIG2,
-                RT5677_PWR_DAC_M2F_L_BIT, 0, NULL, 0),
+                RT5677_PWR_DAC_M2F_L_BIT, 0, rt5677_filter_power_event,
+                SND_SOC_DAPM_POST_PMU),
         SND_SOC_DAPM_SUPPLY("dac mono2 right filter", RT5677_PWR_DIG2,
-                RT5677_PWR_DAC_M2F_R_BIT, 0, NULL, 0),
+                RT5677_PWR_DAC_M2F_R_BIT, 0, rt5677_filter_power_event,
+                SND_SOC_DAPM_POST_PMU),
         SND_SOC_DAPM_SUPPLY("dac mono3 left filter", RT5677_PWR_DIG2,
-                RT5677_PWR_DAC_M3F_L_BIT, 0, NULL, 0),
+                RT5677_PWR_DAC_M3F_L_BIT, 0, rt5677_filter_power_event,
+                SND_SOC_DAPM_POST_PMU),
         SND_SOC_DAPM_SUPPLY("dac mono3 right filter", RT5677_PWR_DIG2,
-                RT5677_PWR_DAC_M3F_R_BIT, 0, NULL, 0),
+                RT5677_PWR_DAC_M3F_R_BIT, 0, rt5677_filter_power_event,
+                SND_SOC_DAPM_POST_PMU),
         SND_SOC_DAPM_SUPPLY("dac mono4 left filter", RT5677_PWR_DIG2,
-                RT5677_PWR_DAC_M4F_L_BIT, 0, NULL, 0),
+                RT5677_PWR_DAC_M4F_L_BIT, 0, rt5677_filter_power_event,
+                SND_SOC_DAPM_POST_PMU),
         SND_SOC_DAPM_SUPPLY("dac mono4 right filter", RT5677_PWR_DIG2,
-                RT5677_PWR_DAC_M4F_R_BIT, 0, NULL, 0),
+                RT5677_PWR_DAC_M4F_R_BIT, 0, rt5677_filter_power_event,
+                SND_SOC_DAPM_POST_PMU),
 
         SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0,
                 rt5677_sto1_dac_l_mix, ARRAY_SIZE(rt5677_sto1_dac_l_mix)),
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/wm8960.c b/sound/soc/codecs/wm8960.c
index 056375339ea3..5380798883b5 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -229,7 +229,7 @@ SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
 SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
         6, 1, 0),
 SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
-        7, 1, 0),
+        7, 1, 1),
 
 SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
                WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 39ebd7bf4f53..a7e79784fc16 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -365,8 +365,8 @@ static const struct reg_default wm8962_reg[] = {
         { 16924, 0x0059 },   /* R16924 - HDBASS_PG_1 */
         { 16925, 0x999A },   /* R16925 - HDBASS_PG_0 */
 
-        { 17048, 0x0083 },   /* R17408 - HPF_C_1 */
-        { 17049, 0x98AD },   /* R17409 - HPF_C_0 */
+        { 17408, 0x0083 },   /* R17408 - HPF_C_1 */
+        { 17409, 0x98AD },   /* R17409 - HPF_C_0 */
 
         { 17920, 0x007F },   /* R17920 - ADCL_RETUNE_C1_1 */
         { 17921, 0xFFFF },   /* R17921 - ADCL_RETUNE_C1_0 */
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 4495a40a9468..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 */
@@ -681,8 +681,8 @@ static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai,
         }
 
         mcasp->tdm_slots = slots;
-        mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = rx_mask;
-        mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = tx_mask;
+        mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask;
+        mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = rx_mask;
         mcasp->slot_width = slot_width;
 
         return davinci_mcasp_set_ch_constraints(mcasp);
@@ -908,6 +908,14 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
                 mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
                 mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
                                FSRMOD(total_slots), FSRMOD(0x1FF));
+                /*
+                 * If McASP is set to be TX/RX synchronous and the playback is
+                 * not running already we need to configure the TX slots in
+                 * order to have correct FSX on the bus
+                 */
+                if (mcasp_is_synchronous(mcasp) && !mcasp->channels)
+                        mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+                                       FSXMOD(total_slots), FSXMOD(0x1FF));
         }
 
         return 0;
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 19c302b0d763..14dfdee05fd5 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -283,6 +283,8 @@ config SND_SOC_IMX_MC13783
 config SND_SOC_FSL_ASOC_CARD
         tristate "Generic ASoC Sound Card with ASRC support"
         depends on OF && I2C
+        # enforce SND_SOC_FSL_ASOC_CARD=m if SND_AC97_CODEC=m:
+        depends on SND_AC97_CODEC || SND_AC97_CODEC=n
         select SND_SOC_IMX_AUDMUX
         select SND_SOC_IMX_PCM_DMA
         select SND_SOC_FSL_ESAI
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index a4435f5e3be9..08b460ba06ef 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -454,7 +454,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
          * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx.
          * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx.
          */
-        regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0);
+        regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC,
+                           sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0);
         regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC,
                            sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0);
 
@@ -504,6 +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/intel/Kconfig b/sound/soc/intel/Kconfig
index 7b778ab85f8b..d430ef5a4f38 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -144,7 +144,7 @@ config SND_SOC_INTEL_SKYLAKE
 
 config SND_SOC_INTEL_SKL_RT286_MACH
         tristate "ASoC Audio driver for SKL with RT286 I2S mode"
-        depends on X86 && ACPI
+        depends on X86 && ACPI && I2C
         select SND_SOC_INTEL_SST
         select SND_SOC_INTEL_SKYLAKE
         select SND_SOC_RT286
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index a7854c8fc523..ffea427aeca8 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -1240,6 +1240,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus)
          */
         ret = snd_soc_tplg_component_load(&platform->component,
                                         &skl_tplg_ops, fw, 0);
+        release_firmware(fw);
         if (ret < 0) {
                 dev_err(bus->dev, "tplg component load failed%d\n", ret);
                 return -EINVAL;
diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c
index a38a3029062c..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;
@@ -280,7 +282,7 @@ static int rk_spdif_probe(struct platform_device *pdev)
         int ret;
 
         match = of_match_node(rk_spdif_match, np);
-        if ((int) match->data == RK_SPDIF_RK3288) {
+        if (match->data == (void *)RK_SPDIF_RK3288) {
                 struct regmap *grf;
 
                 grf = syscon_regmap_lookup_by_phandle(np, "rockchip,grf");
diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h
index 07f86a21046a..3ef12770ae12 100644
--- a/sound/soc/rockchip/rockchip_spdif.h
+++ b/sound/soc/rockchip/rockchip_spdif.h
@@ -28,9 +28,9 @@
 #define SPDIF_CFGR_VDW(x)       (x << SPDIF_CFGR_VDW_SHIFT)
 #define SDPIF_CFGR_VDW_MASK     (0xf << SPDIF_CFGR_VDW_SHIFT)
 
-#define SPDIF_CFGR_VDW_16       SPDIF_CFGR_VDW(0x00)
-#define SPDIF_CFGR_VDW_20       SPDIF_CFGR_VDW(0x01)
-#define SPDIF_CFGR_VDW_24       SPDIF_CFGR_VDW(0x10)
+#define SPDIF_CFGR_VDW_16       SPDIF_CFGR_VDW(0x0)
+#define SPDIF_CFGR_VDW_20       SPDIF_CFGR_VDW(0x1)
+#define SPDIF_CFGR_VDW_24       SPDIF_CFGR_VDW(0x2)
 
 /*
  * DMACR
@@ -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
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 76da7620904c..edcf4cc2e84f 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -235,7 +235,7 @@ static int rsnd_gen2_probe(struct platform_device *pdev,
                 RSND_GEN_S_REG(SCU_SYS_STATUS0, 0x1c8),
                 RSND_GEN_S_REG(SCU_SYS_INT_EN0, 0x1cc),
                 RSND_GEN_S_REG(SCU_SYS_STATUS1, 0x1d0),
-                RSND_GEN_S_REG(SCU_SYS_INT_EN1, 0x1c4),
+                RSND_GEN_S_REG(SCU_SYS_INT_EN1, 0x1d4),
                 RSND_GEN_M_REG(SRC_SWRSR,       0x200,  0x40),
                 RSND_GEN_M_REG(SRC_SRCIR,       0x204,  0x40),
                 RSND_GEN_M_REG(SRC_ADINR,       0x214,  0x40),
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
index 261b50217c48..68b439ed22d7 100644
--- a/sound/soc/sh/rcar/src.c
+++ b/sound/soc/sh/rcar/src.c
@@ -923,6 +923,7 @@ static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod,
                             struct snd_soc_pcm_runtime *rtd)
 {
         struct rsnd_dai *rdai = rsnd_io_to_rdai(io);
+        struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io);
         struct rsnd_src *src = rsnd_mod_to_src(mod);
         int ret;
 
@@ -937,6 +938,12 @@ static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod,
                 return 0;
 
         /*
+         * SRC In doesn't work if DVC was enabled
+         */
+        if (dvc && !rsnd_io_is_play(io))
+                return 0;
+
+        /*
          * enable sync convert
          */
         ret = rsnd_kctrl_new_s(mod, io, rtd,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 24b096066a07..a1305f827a98 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -795,12 +795,12 @@ static void soc_resume_deferred(struct work_struct *work)
 
         dev_dbg(card->dev, "ASoC: resume work completed\n");
 
-        /* userspace can access us now we are back as we were before */
-        snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0);
-
         /* Recheck all endpoints too, their state is affected by suspend */
         dapm_mark_endpoints_dirty(card);
         snd_soc_dapm_sync(&card->dapm);
+
+        /* userspace can access us now we are back as we were before */
+        snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0);
 }
 
 /* powers up audio subsystem after a suspend */
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 016eba10b1ec..7d009428934a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2293,6 +2293,12 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w)
         kfree(w);
 }
 
+void snd_soc_dapm_reset_cache(struct snd_soc_dapm_context *dapm)
+{
+        dapm->path_sink_cache.widget = NULL;
+        dapm->path_source_cache.widget = NULL;
+}
+
 /* free all dapm widgets and resources */
 static void dapm_free_widgets(struct snd_soc_dapm_context *dapm)
 {
@@ -2303,6 +2309,7 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm)
                         continue;
                 snd_soc_dapm_free_widget(w);
         }
+        snd_soc_dapm_reset_cache(dapm);
 }
 
 static struct snd_soc_dapm_widget *dapm_find_widget(
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index ecd38e52285a..2f67ba6d7a8f 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -404,7 +404,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx);
 /**
  * snd_soc_put_volsw_sx - double mixer set callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to set the value of a double mixer control that spans 2 registers.
  *
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 8d7ec80af51b..6963ba20991c 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -531,7 +531,7 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr,
                 /* TLV bytes controls need standard kcontrol info handler,
                  * TLV callback and extended put/get handlers.
                  */
-                k->info = snd_soc_bytes_info;
+                k->info = snd_soc_bytes_info_ext;
                 k->tlv.c = snd_soc_bytes_tlv_callback;
 
                 ext_ops = tplg->bytes_ext_ops;
@@ -1805,6 +1805,7 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm,
                 snd_soc_tplg_widget_remove(w);
                 snd_soc_dapm_free_widget(w);
         }
+        snd_soc_dapm_reset_cache(dapm);
 }
 EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all);
 
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 843f037a317d..5c2bc53f0a9b 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -669,6 +669,7 @@ static int uni_player_startup(struct snd_pcm_substream *substream,
 {
         struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai);
         struct uniperif *player = priv->dai_data.uni;
+        player->substream = substream;
 
         player->clk_adj = 0;
 
@@ -950,6 +951,8 @@ static void uni_player_shutdown(struct snd_pcm_substream *substream,
         if (player->state != UNIPERIF_STATE_STOPPED)
                 /* Stop the player */
                 uni_player_stop(player);
+
+        player->substream = NULL;
 }
 
 static int uni_player_parse_dt_clk_glue(struct platform_device *pdev,
@@ -989,7 +992,7 @@ static int uni_player_parse_dt(struct platform_device *pdev,
         if (!info)
                 return -ENOMEM;
 
-        if (of_property_read_u32(pnode, "version", &player->ver) ||
+        if (of_property_read_u32(pnode, "st,version", &player->ver) ||
             player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
                 dev_err(dev, "Unknown uniperipheral version ");
                 return -EINVAL;
@@ -998,13 +1001,13 @@ static int uni_player_parse_dt(struct platform_device *pdev,
         if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
                 info->underflow_enabled = 1;
 
-        if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) {
+        if (of_property_read_u32(pnode, "st,uniperiph-id", &info->id)) {
                 dev_err(dev, "uniperipheral id not defined");
                 return -EINVAL;
         }
 
         /* Read the device mode property */
-        if (of_property_read_string(pnode, "mode", &mode)) {
+        if (of_property_read_string(pnode, "st,mode", &mode)) {
                 dev_err(dev, "uniperipheral mode not defined");
                 return -EINVAL;
         }
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index f791239a3087..8a0eb2050169 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -316,7 +316,7 @@ static int uni_reader_parse_dt(struct platform_device *pdev,
         if (!info)
                 return -ENOMEM;
 
-        if (of_property_read_u32(node, "version", &reader->ver) ||
+        if (of_property_read_u32(node, "st,version", &reader->ver) ||
             reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
                 dev_err(&pdev->dev, "Unknown uniperipheral version ");
                 return -EINVAL;
@@ -346,7 +346,6 @@ int uni_reader_init(struct platform_device *pdev,
         reader->hw = &uni_reader_pcm_hw;
         reader->dai_ops = &uni_reader_dai_ops;
 
-        dev_err(reader->dev, "%s: enter\n", __func__);
         ret = uni_reader_parse_dt(pdev, reader);
         if (ret < 0) {
                 dev_err(reader->dev, "Failed to parse DeviceTree");
diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index bcbf4da168b6..1bb896d78d09 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -2,6 +2,7 @@
  * Copyright 2014 Emilio López <emilio@elopez.com.ar>
  * Copyright 2014 Jon Smirl <jonsmirl@gmail.com>
  * Copyright 2015 Maxime Ripard <maxime.ripard@free-electrons.com>
+ * Copyright 2015 Adam Sampson <ats@offog.org>
  *
  * Based on the Allwinner SDK driver, released under the GPL.
  *
@@ -404,7 +405,7 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mute =
 static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1);
 
 static const struct snd_kcontrol_new sun4i_codec_widgets[] = {
-        SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL,
+        SOC_SINGLE_TLV("Power Amplifier Volume", SUN4I_CODEC_DAC_ACTL,
                        SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0,
                        sun4i_codec_pa_volume_scale),
 };
@@ -452,12 +453,12 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = {
         SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL,
                             SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0),
 
-        /* Pre-Amplifier */
-        SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL,
+        /* Power Amplifier */
+        SND_SOC_DAPM_MIXER("Power Amplifier", SUN4I_CODEC_ADC_ACTL,
                            SUN4I_CODEC_ADC_ACTL_PA_EN, 0,
                            sun4i_codec_pa_mixer_controls,
                            ARRAY_SIZE(sun4i_codec_pa_mixer_controls)),
-        SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0,
+        SND_SOC_DAPM_SWITCH("Power Amplifier Mute", SND_SOC_NOPM, 0, 0,
                             &sun4i_codec_pa_mute),
 
         SND_SOC_DAPM_OUTPUT("HP Right"),
@@ -480,16 +481,16 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = {
         { "Left Mixer", NULL, "Mixer Enable" },
         { "Left Mixer", "Left DAC Playback Switch", "Left DAC" },
 
-        /* Pre-Amplifier Mixer Routes */
-        { "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" },
-        { "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" },
-        { "Pre-Amplifier", "DAC Playback Switch", "Left DAC" },
-        { "Pre-Amplifier", "DAC Playback Switch", "Right DAC" },
+        /* Power Amplifier Routes */
+        { "Power Amplifier", "Mixer Playback Switch", "Left Mixer" },
+        { "Power Amplifier", "Mixer Playback Switch", "Right Mixer" },
+        { "Power Amplifier", "DAC Playback Switch", "Left DAC" },
+        { "Power Amplifier", "DAC Playback Switch", "Right DAC" },
 
-        /* PA -> HP path */
-        { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" },
-        { "HP Right", NULL, "Pre-Amplifier Mute" },
-        { "HP Left", NULL, "Pre-Amplifier Mute" },
+        /* Headphone Output Routes */
+        { "Power Amplifier Mute", "Switch", "Power Amplifier" },
+        { "HP Right", NULL, "Power Amplifier Mute" },
+        { "HP Left", NULL, "Power Amplifier Mute" },
 };
 
 static struct snd_soc_codec_driver sun4i_codec_codec = {
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 7661616f3636..5b4c58c3e2c5 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -174,6 +174,8 @@ struct snd_usb_midi_in_endpoint {
                 u8 running_status_length;
         } ports[0x10];
         u8 seen_f5;
+        bool in_sysex;
+        u8 last_cin;
         u8 error_resubmit;
         int current_port;
 };
@@ -468,6 +470,39 @@ static void snd_usbmidi_maudio_broken_running_status_input(
 }
 
 /*
+ * QinHeng CH345 is buggy: every second packet inside a SysEx has not CIN 4
+ * but the previously seen CIN, but still with three data bytes.
+ */
+static void ch345_broken_sysex_input(struct snd_usb_midi_in_endpoint *ep,
+                                     uint8_t *buffer, int buffer_length)
+{
+        unsigned int i, cin, length;
+
+        for (i = 0; i + 3 < buffer_length; i += 4) {
+                if (buffer[i] == 0 && i > 0)
+                        break;
+                cin = buffer[i] & 0x0f;
+                if (ep->in_sysex &&
+                    cin == ep->last_cin &&
+                    (buffer[i + 1 + (cin == 0x6)] & 0x80) == 0)
+                        cin = 0x4;
+#if 0
+                if (buffer[i + 1] == 0x90) {
+                        /*
+                         * Either a corrupted running status or a real note-on
+                         * message; impossible to detect reliably.
+                         */
+                }
+#endif
+                length = snd_usbmidi_cin_length[cin];
+                snd_usbmidi_input_data(ep, 0, &buffer[i + 1], length);
+                ep->in_sysex = cin == 0x4;
+                if (!ep->in_sysex)
+                        ep->last_cin = cin;
+        }
+}
+
+/*
  * CME protocol: like the standard protocol, but SysEx commands are sent as a
  * single USB packet preceded by a 0x0F byte.
  */
@@ -660,6 +695,12 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = {
         .output_packet = snd_usbmidi_output_standard_packet,
 };
 
+static struct usb_protocol_ops snd_usbmidi_ch345_broken_sysex_ops = {
+        .input = ch345_broken_sysex_input,
+        .output = snd_usbmidi_standard_output,
+        .output_packet = snd_usbmidi_output_standard_packet,
+};
+
 /*
  * AKAI MPD16 protocol:
  *
@@ -1341,6 +1382,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi,
                  * Various chips declare a packet size larger than 4 bytes, but
                  * do not actually work with larger packets:
                  */
+        case USB_ID(0x0a67, 0x5011): /* Medeli DD305 */
         case USB_ID(0x0a92, 0x1020): /* ESI M4U */
         case USB_ID(0x1430, 0x474b): /* RedOctane GH MIDI INTERFACE */
         case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */
@@ -2378,6 +2420,10 @@ int snd_usbmidi_create(struct snd_card *card,
 
                 err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
                 break;
+        case QUIRK_MIDI_CH345:
+                umidi->usb_protocol_ops = &snd_usbmidi_ch345_broken_sysex_ops;
+                err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+                break;
         default:
                 dev_err(&umidi->dev->dev, "invalid quirk type %d\n",
                         quirk->type);
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index f494dced3c11..4f85757009b3 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1354,6 +1354,8 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
                 }
         }
 
+        snd_usb_mixer_fu_apply_quirk(state->mixer, cval, unitid, kctl);
+
         range = (cval->max - cval->min) / cval->res;
         /*
          * Are there devices with volume range more than 255? I use a bit more
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 6a803eff87f7..ddca6547399b 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -348,13 +348,6 @@ static struct usbmix_name_map bose_companion5_map[] = {
         { 0 }   /* terminator */
 };
 
-/* Dragonfly DAC 1.2, the dB conversion factor is 1 instead of 256 */
-static struct usbmix_dB_map dragonfly_1_2_dB = {0, 5000};
-static struct usbmix_name_map dragonfly_1_2_map[] = {
-        { 7, NULL, .dB = &dragonfly_1_2_dB },
-        { 0 }   /* terminator */
-};
-
 /*
  * Control map entries
  */
@@ -470,11 +463,6 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
                 .id = USB_ID(0x05a7, 0x1020),
                 .map = bose_companion5_map,
         },
-        {
-                /* Dragonfly DAC 1.2 */
-                .id = USB_ID(0x21b4, 0x0081),
-                .map = dragonfly_1_2_map,
-        },
         { 0 } /* terminator */
 };
 
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index fe91184ce832..0ce888dceed0 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -37,6 +37,7 @@
 #include <sound/control.h>
 #include <sound/hwdep.h>
 #include <sound/info.h>
+#include <sound/tlv.h>
 
 #include "usbaudio.h"
 #include "mixer.h"
@@ -1825,3 +1826,39 @@ void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer,
         }
 }
 
+static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer,
+                                         struct snd_kcontrol *kctl)
+{
+        /* Approximation using 10 ranges based on output measurement on hw v1.2.
+         * This seems close to the cubic mapping e.g. alsamixer uses. */
+        static const DECLARE_TLV_DB_RANGE(scale,
+                 0,  1, TLV_DB_MINMAX_ITEM(-5300, -4970),
+                 2,  5, TLV_DB_MINMAX_ITEM(-4710, -4160),
+                 6,  7, TLV_DB_MINMAX_ITEM(-3884, -3710),
+                 8, 14, TLV_DB_MINMAX_ITEM(-3443, -2560),
+                15, 16, TLV_DB_MINMAX_ITEM(-2475, -2324),
+                17, 19, TLV_DB_MINMAX_ITEM(-2228, -2031),
+                20, 26, TLV_DB_MINMAX_ITEM(-1910, -1393),
+                27, 31, TLV_DB_MINMAX_ITEM(-1322, -1032),
+                32, 40, TLV_DB_MINMAX_ITEM(-968, -490),
+                41, 50, TLV_DB_MINMAX_ITEM(-441, 0),
+        );
+
+        usb_audio_info(mixer->chip, "applying DragonFly dB scale quirk\n");
+        kctl->tlv.p = scale;
+        kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+        kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
+}
+
+void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer,
+                                  struct usb_mixer_elem_info *cval, int unitid,
+                                  struct snd_kcontrol *kctl)
+{
+        switch (mixer->chip->usb_id) {
+        case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */
+                if (unitid == 7 && cval->min == 0 && cval->max == 50)
+                        snd_dragonfly_quirk_db_scale(mixer, kctl);
+                break;
+        }
+}
+
diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h
index bdbfab093816..177c329cd4dd 100644
--- a/sound/usb/mixer_quirks.h
+++ b/sound/usb/mixer_quirks.h
@@ -9,5 +9,9 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
 void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer,
                                     int unitid);
 
+void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer,
+                                  struct usb_mixer_elem_info *cval, int unitid,
+                                  struct snd_kcontrol *kctl);
+
 #endif /* SND_USB_MIXER_QUIRKS_H */
 
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 1a1e2e4df35e..c60a776e815d 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2829,6 +2829,17 @@ YAMAHA_DEVICE(0x7010, "UB99"),
         .idProduct = 0x1020,
 },
 
+/* QinHeng devices */
+{
+        USB_DEVICE(0x1a86, 0x752d),
+        .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+                .vendor_name = "QinHeng",
+                .product_name = "CH345",
+                .ifnum = 1,
+                .type = QUIRK_MIDI_CH345
+        }
+},
+
 /* KeithMcMillen Stringport */
 {
         USB_DEVICE(0x1f38, 0x0001),
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 5ca80e7d30cd..b6c0c8e3b450 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -538,6 +538,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
                 [QUIRK_MIDI_CME] = create_any_midi_quirk,
                 [QUIRK_MIDI_AKAI] = create_any_midi_quirk,
                 [QUIRK_MIDI_FTDI] = create_any_midi_quirk,
+                [QUIRK_MIDI_CH345] = create_any_midi_quirk,
                 [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk,
                 [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk,
                 [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk,
@@ -1124,6 +1125,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
         case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */
         case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */
         case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */
+        case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */
                 return true;
         }
         return false;
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 15a12715bd05..b665d85555cb 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -95,6 +95,7 @@ enum quirk_type {
         QUIRK_MIDI_AKAI,
         QUIRK_MIDI_US122L,
         QUIRK_MIDI_FTDI,
+        QUIRK_MIDI_CH345,
         QUIRK_AUDIO_STANDARD_INTERFACE,
         QUIRK_AUDIO_FIXED_ENDPOINT,
         QUIRK_AUDIO_EDIROL_UAXX,