Merge tag 'sound-fix-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Here are a few fixes that have landed after the previous pull request.
  All are driver specific fixes including:

   - error/int value fixes in OXFW,
   - Intel Skylake HD-audio HDMI codec support,
   - Additional HD-audio Realtek codecs and AD1986A codec fixes/quirks,
   - a few more DSD support and a quirk for Arcam rPAC in usb-audio,
   - a typo fix for Scarlett 6i6,
   - fixes for new ASIHPI firmware,
   - ASoC Exynos7 cleanups,
   - Intel ACPI support, and
   - a fix for PCM512 register cache sync"

* tag 'sound-fix-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (24 commits)
  ALSA: usb-audio: extend KEF X300A FU 10 tweak to Arcam rPAC
  ALSA: hda/realtek - New codec support for ALC298
  ALSA: asihpi: update to HPI version 4.14
  ALSA: asihpi: increase tuner pad cache size
  ALSA: asihpi: relax firmware version check
  ALSA: usb-audio: Fix Scarlett 6i6 initialization typo
  ALSA: hda - Add quirk for Packard Bell EasyNote MX65
  ALSA: usb-audio: add native DSD support for Matrix Audio DACs
  ALSA: hda/realtek - New codec support for ALC256
  ALSA: hda/realtek - Add new Dell desktop for ALC3234 headset mode
  ASoC: Intel: fix possible acpi enumeration panic
  ALSA: hda/hdmi - apply Haswell fix-ups to Skylake display codec
  ASoC: Intel: fix return value check in sst_acpi_probe()
  ALSA: hda - Make add_stereo_mix_input flag tristate
  ALSA: hda - Create capture source ctls when stereo mix input is added
  ALSA: hda - Fix typos in snd_hda_get_int_hint() kerneldoc comments
  ALSA: hda - add codec ID for Skylake display audio codec
  ALSA: oxfw: some signedness bugs
  ALSA: oxfw: fix detect_loud_models() return value
  ASoC: rt5677: add REGMAP_I2C and REGMAP_IRQ dependency
  ...

24 files changed, 139 insertions, 51 deletions

diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index 9bc556b15a9..67ade0775a5 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -19,7 +19,7 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params,
 		.min = UINT_MAX, .max = 0, .integer = 1
 	};
 	struct snd_oxfw_stream_formation formation;
-	unsigned int i, err;
+	int i, err;
 
 	for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
 		if (formats[i] == NULL)
@@ -47,7 +47,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params,
 	const struct snd_interval *r =
 		hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
 	struct snd_oxfw_stream_formation formation;
-	unsigned int i, j, err;
+	int i, j, err;
 	unsigned int count, list[SND_OXFW_STREAM_FORMAT_ENTRIES] = {0};
 
 	count = 0;
@@ -80,7 +80,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params,
 static void limit_channels_and_rates(struct snd_pcm_hardware *hw, u8 **formats)
 {
 	struct snd_oxfw_stream_formation formation;
-	unsigned int i, err;
+	int i, err;
 
 	hw->channels_min = UINT_MAX;
 	hw->channels_max = 0;
diff --git a/sound/firewire/oxfw/oxfw-proc.c b/sound/firewire/oxfw/oxfw-proc.c
index 604808e5526..8ba4f9f262b 100644
--- a/sound/firewire/oxfw/oxfw-proc.c
+++ b/sound/firewire/oxfw/oxfw-proc.c
@@ -15,7 +15,7 @@ static void proc_read_formation(struct snd_info_entry *entry,
 	struct snd_oxfw_stream_formation formation, curr;
 	u8 *format;
 	char flag;
-	unsigned int i, err;
+	int i, err;
 
 	/* Show input. */
 	err = snd_oxfw_stream_get_current_formation(oxfw,
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index b77cf80f167..bda845afb47 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -61,7 +61,8 @@ static int set_stream_format(struct snd_oxfw *oxfw, struct amdtp_stream *s,
 	u8 **formats;
 	struct snd_oxfw_stream_formation formation;
 	enum avc_general_plug_dir dir;
-	unsigned int i, err, len;
+	unsigned int len;
+	int i, err;
 
 	if (s == &oxfw->tx_stream) {
 		formats = oxfw->tx_stream_formats;
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index cf1d0b55e82..60e5cad0531 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -43,7 +43,7 @@ static bool detect_loud_models(struct fw_unit *unit)
 	err = fw_csr_string(unit->directory, CSR_MODEL,
 			    model, sizeof(model));
 	if (err < 0)
-		return err;
+		return false;
 
 	for (i = 0; i < ARRAY_SIZE(models); i++) {
 		if (strcmp(models[i], model) == 0)
diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index 48380ce2c81..aeea679b228 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -1367,9 +1367,9 @@ struct hpi_control_cache_single {
 struct hpi_control_cache_pad {
 	struct hpi_control_cache_info i;
 	u32 field_valid_flags;
-	u8 c_channel[8];
-	u8 c_artist[40];
-	u8 c_title[40];
+	u8 c_channel[40];
+	u8 c_artist[100];
+	u8 c_title[100];
 	u8 c_comment[200];
 	u32 pTY;
 	u32 pI;
diff --git a/sound/pci/asihpi/hpi_version.h b/sound/pci/asihpi/hpi_version.h
index e9146e53bd5..6623ab11003 100644
--- a/sound/pci/asihpi/hpi_version.h
+++ b/sound/pci/asihpi/hpi_version.h
@@ -11,13 +11,13 @@ Production releases have even minor version.
 /* Use single digits for versions less that 10 to avoid octal. */
 /* *** HPI_VER is the only edit required to update version *** */
 /** HPI version */
-#define HPI_VER HPI_VERSION_CONSTRUCTOR(4, 10, 1)
+#define HPI_VER HPI_VERSION_CONSTRUCTOR(4, 14, 3)
 
 /** HPI version string in dotted decimal format */
-#define HPI_VER_STRING "4.10.01"
+#define HPI_VER_STRING "4.14.03"
 
 /** Library version as documented in hpi-api-versions.txt */
-#define HPI_LIB_VER  HPI_VERSION_CONSTRUCTOR(10, 2, 0)
+#define HPI_LIB_VER  HPI_VERSION_CONSTRUCTOR(10, 4, 0)
 
 /** Construct hpi version number from major, minor, release numbers */
 #define HPI_VERSION_CONSTRUCTOR(maj, min, r) ((maj << 16) + (min << 8) + r)
diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c
index ac916377001..3603c24f34d 100644
--- a/sound/pci/asihpi/hpidspcd.c
+++ b/sound/pci/asihpi/hpidspcd.c
@@ -1,8 +1,9 @@
-/***********************************************************************/
-/**
+/***********************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Functions for reading DSP code using hotplug firmware loader
+
+    Copyright (C) 1997-2014  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -17,11 +18,7 @@
     along with this program; if not, write to the Free Software
     Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 
-\file
-Functions for reading DSP code using
-hotplug firmware loader from individual dsp code files
-*/
-/***********************************************************************/
+***********************************************************************/
 #define SOURCEFILE_NAME "hpidspcd.c"
 #include "hpidspcd.h"
 #include "hpidebug.h"
@@ -68,17 +65,18 @@ short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code,
 		goto error2;
 	}
 
-	if ((header.version >> 9) != (HPI_VER >> 9)) {
-		/* Consider even and subsequent odd minor versions to be compatible */
-		dev_err(&dev->dev, "Incompatible firmware version DSP image %X != Driver %X\n",
+	if (HPI_VER_MAJOR(header.version) != HPI_VER_MAJOR(HPI_VER)) {
+		/* Major version change probably means Host-DSP protocol change */
+		dev_err(&dev->dev,
+			"Incompatible firmware version DSP image %X != Driver %X\n",
 			header.version, HPI_VER);
 		goto error2;
 	}
 
 	if (header.version != HPI_VER) {
-		dev_info(&dev->dev,
-			 "Firmware: release version mismatch  DSP image %X != Driver %X\n",
-			 header.version, HPI_VER);
+		dev_warn(&dev->dev,
+			"Firmware version mismatch: DSP image %X != Driver %X\n",
+			header.version, HPI_VER);
 	}
 
 	HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name);
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 63b69f750d8..b680b4ec633 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -3218,12 +3218,13 @@ static int create_input_ctls(struct hda_codec *codec)
 	}
 
 	/* add stereo mix when explicitly enabled via hint */
-	if (mixer && spec->add_stereo_mix_input &&
-	    snd_hda_get_bool_hint(codec, "add_stereo_mix_input") > 0) {
+	if (mixer && spec->add_stereo_mix_input == HDA_HINT_STEREO_MIX_ENABLE) {
 		err = parse_capture_source(codec, mixer, CFG_IDX_MIX, num_adcs,
 					   "Stereo Mix", 0);
 		if (err < 0)
 			return err;
+		else
+			spec->suppress_auto_mic = 1;
 	}
 
 	return 0;
@@ -4542,9 +4543,8 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
 
 	/* add stereo mix if available and not enabled yet */
 	if (!spec->auto_mic && spec->mixer_nid &&
-	    spec->add_stereo_mix_input &&
-	    spec->input_mux.num_items > 1 &&
-	    snd_hda_get_bool_hint(codec, "add_stereo_mix_input") < 0) {
+	    spec->add_stereo_mix_input == HDA_HINT_STEREO_MIX_AUTO &&
+	    spec->input_mux.num_items > 1) {
 		err = parse_capture_source(codec, spec->mixer_nid,
 					   CFG_IDX_MIX, spec->num_all_adcs,
 					   "Stereo Mix", 0);
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 61dd5153f51..3d852660443 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -222,7 +222,7 @@ struct hda_gen_spec {
 	unsigned int vmaster_mute_enum:1; /* add vmaster mute mode enum */
 	unsigned int indep_hp:1; /* independent HP supported */
 	unsigned int prefer_hp_amp:1; /* enable HP amp for speaker if any */
-	unsigned int add_stereo_mix_input:1; /* add aamix as a capture src */
+	unsigned int add_stereo_mix_input:2; /* add aamix as a capture src */
 	unsigned int add_jack_modes:1; /* add i/o jack mode enum ctls */
 	unsigned int power_down_unused:1; /* power down unused widgets */
 	unsigned int dac_min_mute:1; /* minimal = mute for DACs */
@@ -291,6 +291,13 @@ struct hda_gen_spec {
 				    struct hda_jack_callback *cb);
 };
 
+/* values for add_stereo_mix_input flag */
+enum {
+	HDA_HINT_STEREO_MIX_DISABLE,	/* No stereo mix input */
+	HDA_HINT_STEREO_MIX_ENABLE,	/* Add stereo mix input */
+	HDA_HINT_STEREO_MIX_AUTO,	/* Add only if auto-mic is disabled */
+};
+
 int snd_hda_gen_spec_init(struct hda_gen_spec *spec);
 
 int snd_hda_gen_init(struct hda_codec *codec);
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index bef721592c3..ccc962a1699 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -468,7 +468,7 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
 EXPORT_SYMBOL_GPL(snd_hda_get_bool_hint);
 
 /**
- * snd_hda_get_bool_hint - Get a boolean hint value
+ * snd_hda_get_int_hint - Get an integer hint value
  * @codec: the HDA codec
  * @key: the hint key string
  * @valp: pointer to store a value
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index c81b715d6c9..a9d78e27513 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -195,7 +195,8 @@ static int ad198x_parse_auto_config(struct hda_codec *codec, bool indep_hp)
 	codec->no_sticky_stream = 1;
 
 	spec->gen.indep_hp = indep_hp;
-	spec->gen.add_stereo_mix_input = 1;
+	if (!spec->gen.add_stereo_mix_input)
+		spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO;
 
 	err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0);
 	if (err < 0)
@@ -256,6 +257,18 @@ static void ad1986a_fixup_eapd(struct hda_codec *codec,
 	}
 }
 
+/* enable stereo-mix input for avoiding regression on KDE (bko#88251) */
+static void ad1986a_fixup_eapd_mix_in(struct hda_codec *codec,
+				      const struct hda_fixup *fix, int action)
+{
+	struct ad198x_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		ad1986a_fixup_eapd(codec, fix, action);
+		spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_ENABLE;
+	}
+}
+
 enum {
 	AD1986A_FIXUP_INV_JACK_DETECT,
 	AD1986A_FIXUP_ULTRA,
@@ -264,6 +277,8 @@ enum {
 	AD1986A_FIXUP_LAPTOP,
 	AD1986A_FIXUP_LAPTOP_IMIC,
 	AD1986A_FIXUP_EAPD,
+	AD1986A_FIXUP_EAPD_MIX_IN,
+	AD1986A_FIXUP_EASYNOTE,
 };
 
 static const struct hda_fixup ad1986a_fixups[] = {
@@ -328,6 +343,30 @@ static const struct hda_fixup ad1986a_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = ad1986a_fixup_eapd,
 	},
+	[AD1986A_FIXUP_EAPD_MIX_IN] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = ad1986a_fixup_eapd_mix_in,
+	},
+	[AD1986A_FIXUP_EASYNOTE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x0421402f }, /* headphone */
+			{ 0x1b, 0x90170110 }, /* speaker */
+			{ 0x1c, 0x411111f0 }, /* N/A */
+			{ 0x1d, 0x90a70130 }, /* int mic */
+			{ 0x1e, 0x411111f0 }, /* N/A */
+			{ 0x1f, 0x04a19040 }, /* mic */
+			{ 0x20, 0x411111f0 }, /* N/A */
+			{ 0x21, 0x411111f0 }, /* N/A */
+			{ 0x22, 0x411111f0 }, /* N/A */
+			{ 0x23, 0x411111f0 }, /* N/A */
+			{ 0x24, 0x411111f0 }, /* N/A */
+			{ 0x25, 0x411111f0 }, /* N/A */
+			{}
+		},
+		.chained = true,
+		.chain_id = AD1986A_FIXUP_EAPD_MIX_IN,
+	},
 };
 
 static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
@@ -341,6 +380,7 @@ static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP),
 	SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG),
 	SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA),
+	SND_PCI_QUIRK(0x1631, 0xc022, "PackardBell EasyNote MX65", AD1986A_FIXUP_EASYNOTE),
 	SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT),
 	SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_FIXUP_3STACK),
 	SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_FIXUP_3STACK),
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index e9ebc7bd752..fd3ed18670e 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -855,14 +855,14 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	case 0x14f15045:
 		codec->single_adc_amp = 1;
 		spec->gen.mixer_nid = 0x17;
-		spec->gen.add_stereo_mix_input = 1;
+		spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO;
 		snd_hda_pick_fixup(codec, cxt5045_fixup_models,
 				   cxt5045_fixups, cxt_fixups);
 		break;
 	case 0x14f15047:
 		codec->pin_amp_workaround = 1;
 		spec->gen.mixer_nid = 0x19;
-		spec->gen.add_stereo_mix_input = 1;
+		spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO;
 		snd_hda_pick_fixup(codec, cxt5047_fixup_models,
 				   cxt5047_fixups, cxt_fixups);
 		break;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 9dc9cf8c90e..5f13d2d1807 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -47,7 +47,9 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info");
 
 #define is_haswell(codec)  ((codec)->vendor_id == 0x80862807)
 #define is_broadwell(codec)    ((codec)->vendor_id == 0x80862808)
-#define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec))
+#define is_skylake(codec) ((codec)->vendor_id == 0x80862809)
+#define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \
+					|| is_skylake(codec))
 
 #define is_valleyview(codec) ((codec)->vendor_id == 0x80862882)
 #define is_cherryview(codec) ((codec)->vendor_id == 0x80862883)
@@ -3365,6 +3367,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
 { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi },
 { .id = 0x80862807, .name = "Haswell HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862808, .name = "Broadwell HDMI",	.patch = patch_generic_hdmi },
+{ .id = 0x80862809, .name = "Skylake HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862880, .name = "CedarTrail HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862882, .name = "Valleyview2 HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862883, .name = "Braswell HDMI",	.patch = patch_generic_hdmi },
@@ -3425,6 +3428,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862805");
 MODULE_ALIAS("snd-hda-codec-id:80862806");
 MODULE_ALIAS("snd-hda-codec-id:80862807");
 MODULE_ALIAS("snd-hda-codec-id:80862808");
+MODULE_ALIAS("snd-hda-codec-id:80862809");
 MODULE_ALIAS("snd-hda-codec-id:80862880");
 MODULE_ALIAS("snd-hda-codec-id:80862882");
 MODULE_ALIAS("snd-hda-codec-id:80862883");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a722067c491..65f1f4e18ea 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -321,10 +321,12 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 		break;
 	case 0x10ec0233:
 	case 0x10ec0255:
+	case 0x10ec0256:
 	case 0x10ec0282:
 	case 0x10ec0283:
 	case 0x10ec0286:
 	case 0x10ec0288:
+	case 0x10ec0298:
 		alc_update_coef_idx(codec, 0x10, 1<<9, 0);
 		break;
 	case 0x10ec0285:
@@ -2659,7 +2661,9 @@ enum {
 	ALC269_TYPE_ALC284,
 	ALC269_TYPE_ALC285,
 	ALC269_TYPE_ALC286,
+	ALC269_TYPE_ALC298,
 	ALC269_TYPE_ALC255,
+	ALC269_TYPE_ALC256,
 };
 
 /*
@@ -2686,7 +2690,9 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	case ALC269_TYPE_ALC282:
 	case ALC269_TYPE_ALC283:
 	case ALC269_TYPE_ALC286:
+	case ALC269_TYPE_ALC298:
 	case ALC269_TYPE_ALC255:
+	case ALC269_TYPE_ALC256:
 		ssids = alc269_ssids;
 		break;
 	default:
@@ -4829,6 +4835,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK),
 	SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
+	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, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -5417,9 +5424,15 @@ static int patch_alc269(struct hda_codec *codec)
 		spec->codec_variant = ALC269_TYPE_ALC286;
 		spec->shutup = alc286_shutup;
 		break;
+	case 0x10ec0298:
+		spec->codec_variant = ALC269_TYPE_ALC298;
+		break;
 	case 0x10ec0255:
 		spec->codec_variant = ALC269_TYPE_ALC255;
 		break;
+	case 0x10ec0256:
+		spec->codec_variant = ALC269_TYPE_ALC256;
+		break;
 	}
 
 	if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) {
@@ -6341,6 +6354,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 },
 	{ .id = 0x10ec0235, .name = "ALC233", .patch = patch_alc269 },
 	{ .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 },
+	{ .id = 0x10ec0256, .name = "ALC256", .patch = patch_alc269 },
 	{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
 	{ .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
 	{ .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
@@ -6360,6 +6374,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 },
 	{ .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 },
 	{ .id = 0x10ec0293, .name = "ALC293", .patch = patch_alc269 },
+	{ .id = 0x10ec0298, .name = "ALC298", .patch = patch_alc269 },
 	{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
 	  .patch = patch_alc861 },
 	{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 6c206b6c8d6..3de6d3d779c 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -137,7 +137,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec)
 	spec->gen.indep_hp = 1;
 	spec->gen.keep_eapd_on = 1;
 	spec->gen.pcm_playback_hook = via_playback_pcm_hook;
-	spec->gen.add_stereo_mix_input = 1;
+	spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO;
 	return spec;
 }
 
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index b1cc2a4a7fc..99ff35e2a25 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -267,7 +267,7 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream,
 	if (!ssc_p->dir_mask) {
 		if (ssc_p->initialized) {
 			/* Shutdown the SSC clock. */
-			pr_debug("atmel_ssc_dau: Stopping clock\n");
+			pr_debug("atmel_ssc_dai: Stopping clock\n");
 			clk_disable(ssc_p->ssc->clk);
 
 			free_irq(ssc_p->ssc->irq, ssc_p);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 883c5778b30..8349f982a58 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -520,6 +520,8 @@ config SND_SOC_RT5670
 
 config SND_SOC_RT5677
 	tristate
+	select REGMAP_I2C
+	select REGMAP_IRQ
 
 config SND_SOC_RT5677_SPI
 	tristate
diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c
index 4d62230bd37..d0547fa275f 100644
--- a/sound/soc/codecs/pcm512x-i2c.c
+++ b/sound/soc/codecs/pcm512x-i2c.c
@@ -24,8 +24,13 @@ static int pcm512x_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
 	struct regmap *regmap;
+	struct regmap_config config = pcm512x_regmap;
 
-	regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap);
+	/* msb needs to be set to enable auto-increment of addresses */
+	config.read_flag_mask = 0x80;
+	config.write_flag_mask = 0x80;
+
+	regmap = devm_regmap_init_i2c(i2c, &config);
 	if (IS_ERR(regmap))
 		return PTR_ERR(regmap);
 
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index a7789a8726e..27141e2df87 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -2209,6 +2209,10 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec)
 	int gpio_state, jack_type = 0;
 	unsigned int val;
 
+	if (!gpio_is_valid(rt5645->pdata.hp_det_gpio)) {
+		dev_err(codec->dev, "invalid gpio\n");
+		return -EINVAL;
+	}
 	gpio_state = gpio_get_value(rt5645->pdata.hp_det_gpio);
 
 	dev_dbg(codec->dev, "gpio = %d(%d)\n", rt5645->pdata.hp_det_gpio,
diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c
index 31124aa4434..3abc29e8a92 100644
--- a/sound/soc/intel/sst/sst_acpi.c
+++ b/sound/soc/intel/sst/sst_acpi.c
@@ -43,7 +43,7 @@
 #include "sst.h"
 
 struct sst_machines {
-	char codec_id[32];
+	char *codec_id;
 	char board[32];
 	char machine[32];
 	void (*machine_quirk)(void);
@@ -277,16 +277,16 @@ int sst_acpi_probe(struct platform_device *pdev)
 	dev_dbg(dev, "ACPI device id: %x\n", dev_id);
 
 	plat_dev = platform_device_register_data(dev, mach->pdata->platform, -1, NULL, 0);
-	if (plat_dev == NULL) {
+	if (IS_ERR(plat_dev)) {
 		dev_err(dev, "Failed to create machine device: %s\n", mach->pdata->platform);
-		return -ENODEV;
+		return PTR_ERR(plat_dev);
 	}
 
 	/* Create platform device for sst machine driver */
 	mdev = platform_device_register_data(dev, mach->machine, -1, NULL, 0);
-	if (mdev == NULL) {
+	if (IS_ERR(mdev)) {
 		dev_err(dev, "Failed to create machine device: %s\n", mach->machine);
-		return -ENODEV;
+		return PTR_ERR(mdev);
 	}
 
 	ret = sst_alloc_drv_context(&ctx, dev, dev_id);
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index b1a7c5bce4a..b5a80c528d8 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -1261,6 +1261,8 @@ static int samsung_i2s_probe(struct platform_device *pdev)
 			ret = -ENOMEM;
 			goto err;
 		}
+
+		sec_dai->variant_regs = pri_dai->variant_regs;
 		sec_dai->dma_playback.dma_addr = regs_base + I2STXDS;
 		sec_dai->dma_playback.ch_name = "tx-sec";
 
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 1994d41348f..b703cb3cda1 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -333,8 +333,11 @@ static struct usbmix_name_map gamecom780_map[] = {
 	{}
 };
 
-static const struct usbmix_name_map kef_x300a_map[] = {
-	{ 10, NULL }, /* firmware locks up (?) when we try to access this FU */
+/* some (all?) SCMS USB3318 devices are affected by a firmware lock up
+ * when anything attempts to access FU 10 (control)
+ */
+static const struct usbmix_name_map scms_usb3318_map[] = {
+	{ 10, NULL },
 	{ 0 }
 };
 
@@ -434,8 +437,14 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.map = ebox44_map,
 	},
 	{
+		/* KEF X300A */
 		.id = USB_ID(0x27ac, 0x1000),
-		.map = kef_x300a_map,
+		.map = scms_usb3318_map,
+	},
+	{
+		/* Arcam rPAC */
+		.id = USB_ID(0x25c4, 0x0003),
+		.map = scms_usb3318_map,
 	},
 	{ 0 } /* terminator */
 };
diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c
index 9109652b88b..7438e7c4a84 100644
--- a/sound/usb/mixer_scarlett.c
+++ b/sound/usb/mixer_scarlett.c
@@ -655,7 +655,7 @@ static struct scarlett_device_info s6i6_info = {
 		.names = NULL
 	},
 
-	.num_controls = 0,
+	.num_controls = 9,
 	.controls = {
 		{ .num = 0, .type = SCARLETT_OUTPUTS, .name = "Monitor" },
 		{ .num = 1, .type = SCARLETT_OUTPUTS, .name = "Headphone" },
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 4dbfb3d18ee..a7398412310 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1245,8 +1245,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
 
 	/* XMOS based USB DACs */
 	switch (chip->usb_id) {
-	/* iFi Audio micro/nano iDSD */
-	case USB_ID(0x20b1, 0x3008):
+	case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */
+	case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */
+	case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */
 		if (fp->altsetting == 2)
 			return SNDRV_PCM_FMTBIT_DSD_U32_BE;
 		break;