Merge branch 'master' into for-next

Sync with Linus' tree to be able to apply pending patches that
are based on newer code already present upstream.

41 files changed, 232 insertions, 154 deletions

diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 30468b31cad..6fd9391b3a6 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -599,4 +599,4 @@ module_exit(atmel_abdac_exit);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Driver for Atmel Audio Bitstream DAC (ABDAC)");
-MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>");
+MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt@samfundet.no>");
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 41b901bde5c..6e5addeb236 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -1199,4 +1199,4 @@ module_exit(atmel_ac97c_exit);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Driver for Atmel AC97 controller");
-MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>");
+MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt@samfundet.no>");
diff --git a/sound/core/misc.c b/sound/core/misc.c
index 2c41825c836..eb9fe2e1d29 100644
--- a/sound/core/misc.c
+++ b/sound/core/misc.c
@@ -58,26 +58,6 @@ static const char *sanity_file_name(const char *path)
 	else
 		return path;
 }
-
-/* print file and line with a certain printk prefix */
-static int print_snd_pfx(unsigned int level, const char *path, int line,
-			 const char *format)
-{
-	const char *file = sanity_file_name(path);
-	char tmp[] = "<0>";
-	const char *pfx = level ? KERN_DEBUG : KERN_DEFAULT;
-	int ret = 0;
-
-	if (format[0] == '<' && format[2] == '>') {
-		tmp[1] = format[1];
-		pfx = tmp;
-		ret = 1;
-	}
-	printk("%sALSA %s:%d: ", pfx, file, line);
-	return ret;
-}
-#else
-#define print_snd_pfx(level, path, line, format)	0
 #endif
 
 #if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
@@ -85,15 +65,29 @@ void __snd_printk(unsigned int level, const char *path, int line,
 		  const char *format, ...)
 {
 	va_list args;
-	
+#ifdef CONFIG_SND_VERBOSE_PRINTK
+	struct va_format vaf;
+	char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV";
+#endif
+
 #ifdef CONFIG_SND_DEBUG	
 	if (debug < level)
 		return;
 #endif
+
 	va_start(args, format);
-	if (print_snd_pfx(level, path, line, format))
-		format += 3; /* skip the printk level-prefix */
+#ifdef CONFIG_SND_VERBOSE_PRINTK
+	vaf.fmt = format;
+	vaf.va = &args;
+	if (format[0] == '<' && format[2] == '>') {
+		memcpy(verbose_fmt, format, 3);
+		vaf.fmt = format + 3;
+	} else if (level)
+		memcpy(verbose_fmt, KERN_DEBUG, 3);
+	printk(verbose_fmt, sanity_file_name(path), line, &vaf);
+#else
 	vprintk(format, args);
+#endif
 	va_end(args);
 }
 EXPORT_SYMBOL_GPL(__snd_printk);
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 86ee16ca365..440030818db 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -209,6 +209,7 @@ static void isight_packet(struct fw_iso_context *context, u32 cycle,
 		isight->packet_index = -1;
 		return;
 	}
+	fw_iso_context_queue_flush(isight->context);
 
 	if (++index >= QUEUE_LENGTH)
 		index = 0;
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 2ca6f4f85b4..e3569bdd3b6 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -27,7 +27,6 @@
 #include "hpioctl.h"
 
 #include <linux/pci.h>
-#include <linux/version.h>
 #include <linux/init.h>
 #include <linux/jiffies.h>
 #include <linux/slab.h>
diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c
index f16bc8aad6e..e083122ca55 100644
--- a/sound/pci/cs5535audio/cs5535audio_pcm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pcm.c
@@ -149,7 +149,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
 			&((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i];
 		desc->addr = cpu_to_le32(addr);
 		desc->size = cpu_to_le32(period_bytes);
-		desc->ctlreserved = cpu_to_le32(PRD_EOP);
+		desc->ctlreserved = cpu_to_le16(PRD_EOP);
 		desc_addr += sizeof(struct cs5535audio_dma_desc);
 		addr += period_bytes;
 	}
@@ -157,7 +157,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
 	lastdesc = &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[periods];
 	lastdesc->addr = cpu_to_le32((u32) dma->desc_buf.addr);
 	lastdesc->size = 0;
-	lastdesc->ctlreserved = cpu_to_le32(PRD_JMP);
+	lastdesc->ctlreserved = cpu_to_le16(PRD_JMP);
 	jmpprd_addr = cpu_to_le32(lastdesc->addr +
 				  (sizeof(struct cs5535audio_dma_desc)*periods));
 
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 5e619a84da0..15f0161ce4a 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1440,6 +1440,14 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .ca0102_chip = 1,
 	 .spk71 = 1,
 	 .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */
+	/* EMU0404 PCIe */
+	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40051102,
+	 .driver = "Audigy2", .name = "E-mu 0404 PCIe [MAEM8984]",
+	 .id = "EMU0404",
+	 .emu10k2_chip = 1,
+	 .ca0108_chip = 1,
+	 .spk71 = 1,
+	 .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 PCIe ver_03 */
 	/* Note that all E-mu cards require kernel 2.6 or newer. */
 	{.vendor = 0x1102, .device = 0x0008,
 	 .driver = "Audigy2", .name = "SB Audigy 2 Value [Unknown]",
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index f1de1bac042..55f0647458c 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -50,7 +50,12 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable);
 int snd_hda_attach_beep_device(struct hda_codec *codec, int nid);
 void snd_hda_detach_beep_device(struct hda_codec *codec);
 #else
-#define snd_hda_attach_beep_device(...)		0
-#define snd_hda_detach_beep_device(...)
+static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
+{
+	return 0;
+}
+static inline void snd_hda_detach_beep_device(struct hda_codec *codec)
+{
+}
 #endif
 #endif
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index b05f7be9dc1..e3e853153d1 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -294,7 +294,7 @@ static int hdmi_update_eld(struct hdmi_eld *e,
 		snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl);
 		goto out_fail;
 	} else
-		strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl);
+		strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1);
 
 	for (i = 0; i < e->sad_count; i++) {
 		if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) {
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3e6b9a8539c..7bbc5f237a5 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3074,6 +3074,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
 };
 
 static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO),
 	SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD),
@@ -3102,6 +3103,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
 	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
+	SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
 	{}
 };
 
@@ -4388,6 +4390,8 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = {
 	  .patch = patch_cxt5066 },
 	{ .id = 0x14f15069, .name = "CX20585",
 	  .patch = patch_cxt5066 },
+	{ .id = 0x14f1506c, .name = "CX20588",
+	  .patch = patch_cxt5066 },
 	{ .id = 0x14f1506e, .name = "CX20590",
 	  .patch = patch_cxt5066 },
 	{ .id = 0x14f15097, .name = "CX20631",
@@ -4416,6 +4420,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066");
 MODULE_ALIAS("snd-hda-codec-id:14f15067");
 MODULE_ALIAS("snd-hda-codec-id:14f15068");
 MODULE_ALIAS("snd-hda-codec-id:14f15069");
+MODULE_ALIAS("snd-hda-codec-id:14f1506c");
 MODULE_ALIAS("snd-hda-codec-id:14f1506e");
 MODULE_ALIAS("snd-hda-codec-id:14f15097");
 MODULE_ALIAS("snd-hda-codec-id:14f15098");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7a4e10002f5..b48fb43b544 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1141,6 +1141,13 @@ static void update_speakers(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	int on;
 
+	/* Control HP pins/amps depending on master_mute state;
+	 * in general, HP pins/amps control should be enabled in all cases,
+	 * but currently set only for master_mute, just to be safe
+	 */
+	do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
+		    spec->autocfg.hp_pins, spec->master_mute, true);
+
 	if (!spec->automute)
 		on = 0;
 	else
@@ -2708,17 +2715,30 @@ typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol,
 
 static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol,
-				 getput_call_t func)
+				 getput_call_t func, bool check_adc_switch)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
-	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-	int err;
+	int i, err = 0;
 
 	mutex_lock(&codec->control_mutex);
-	kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx],
-						      3, 0, HDA_INPUT);
-	err = func(kcontrol, ucontrol);
+	if (check_adc_switch && spec->dual_adc_switch) {
+		for (i = 0; i < spec->num_adc_nids; i++) {
+			kcontrol->private_value =
+				HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+						    3, 0, HDA_INPUT);
+			err = func(kcontrol, ucontrol);
+			if (err < 0)
+				goto error;
+		}
+	} else {
+		i = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+		kcontrol->private_value =
+			HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+					    3, 0, HDA_INPUT);
+		err = func(kcontrol, ucontrol);
+	}
+ error:
 	mutex_unlock(&codec->control_mutex);
 	return err;
 }
@@ -2727,14 +2747,14 @@ static int alc_cap_vol_get(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	return alc_cap_getput_caller(kcontrol, ucontrol,
-				     snd_hda_mixer_amp_volume_get);
+				     snd_hda_mixer_amp_volume_get, false);
 }
 
 static int alc_cap_vol_put(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	return alc_cap_getput_caller(kcontrol, ucontrol,
-				     snd_hda_mixer_amp_volume_put);
+				     snd_hda_mixer_amp_volume_put, true);
 }
 
 /* capture mixer elements */
@@ -2744,14 +2764,14 @@ static int alc_cap_sw_get(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
 	return alc_cap_getput_caller(kcontrol, ucontrol,
-				     snd_hda_mixer_amp_switch_get);
+				     snd_hda_mixer_amp_switch_get, false);
 }
 
 static int alc_cap_sw_put(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
 	return alc_cap_getput_caller(kcontrol, ucontrol,
-				     snd_hda_mixer_amp_switch_put);
+				     snd_hda_mixer_amp_switch_put, true);
 }
 
 #define _DEFINE_CAPMIX(num) \
@@ -4876,7 +4896,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
 	SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
 	SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
-	SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST),
 	SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
 	SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
 	SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
@@ -6201,11 +6220,6 @@ static const struct snd_kcontrol_new alc260_input_mixer[] = {
 /* update HP, line and mono out pins according to the master switch */
 static void alc260_hp_master_update(struct hda_codec *codec)
 {
-	struct alc_spec *spec = codec->spec;
-
-	/* change HP pins */
-	do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
-		    spec->autocfg.hp_pins, spec->master_mute, true);
 	update_speakers(codec);
 }
 
@@ -11924,7 +11938,7 @@ static const struct hda_verb alc262_nec_verbs[] = {
  *  0x1b = port replicator headphone out
  */
 
-#define ALC_HP_EVENT	0x37
+#define ALC_HP_EVENT	ALC880_HP_EVENT
 
 static const struct hda_verb alc262_fujitsu_unsol_verbs[] = {
 	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
@@ -12598,6 +12612,7 @@ static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
  */
 enum {
 	PINFIX_FSC_H270,
+	PINFIX_HP_Z200,
 };
 
 static const struct alc_fixup alc262_fixups[] = {
@@ -12610,9 +12625,17 @@ static const struct alc_fixup alc262_fixups[] = {
 			{ }
 		}
 	},
+	[PINFIX_HP_Z200] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x16, 0x99130120 }, /* internal speaker */
+			{ }
+		}
+	},
 };
 
 static const struct snd_pci_quirk alc262_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", PINFIX_HP_Z200),
 	SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270),
 	{}
 };
@@ -12729,6 +12752,8 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = {
 			   ALC262_HP_BPC),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
 			   ALC262_HP_BPC),
+	SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200",
+			   ALC262_AUTO),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
 			   ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
@@ -13314,9 +13339,8 @@ static void alc268_acer_lc_setup(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute_mixer_nid[0] = 0x0f;
 	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x12;
@@ -13860,6 +13884,7 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
 						ALC268_ACER_ASPIRE_ONE),
 	SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
+	SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO),
 	SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
 			"Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
 	/* almost compatible with toshiba but with optional digital outs;
@@ -13870,7 +13895,6 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
 	SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
 	SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
-	SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
 	SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
 	{}
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 605c99e1e52..f43bb0eaed8 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -745,12 +745,23 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
 	struct via_spec *spec = codec->spec;
 	hda_nid_t nid = kcontrol->private_value;
 	unsigned int pinsel = ucontrol->value.enumerated.item[0];
+	unsigned int parm0, parm1;
 	/* Get Independent Mode index of headphone pin widget */
 	spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel
 		? 1 : 0;
-	if (spec->codec_type == VT1718S)
+	if (spec->codec_type == VT1718S) {
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0);
+		/* Set correct mute switch for MW3 */
+		parm0 = spec->hp_independent_mode ?
+			       AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0);
+		parm1 = spec->hp_independent_mode ?
+			       AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1);
+		snd_hda_codec_write(codec, 0x1b, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, parm0);
+		snd_hda_codec_write(codec, 0x1b, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, parm1);
+	}
 	else
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_CONNECT_SEL, pinsel);
@@ -832,10 +843,13 @@ static int via_hp_build(struct hda_codec *codec)
 	knew->subdevice = HDA_SUBDEV_NID_FLAG | nid;
 	knew->private_value = nid;
 
-	knew = via_clone_control(spec, &via_hp_mixer[1]);
-	if (knew == NULL)
-		return -ENOMEM;
-	knew->subdevice = side_mute_channel(spec);
+	nid = side_mute_channel(spec);
+	if (nid) {
+		knew = via_clone_control(spec, &via_hp_mixer[1]);
+		if (knew == NULL)
+			return -ENOMEM;
+		knew->subdevice = nid;
+	}
 
 	return 0;
 }
@@ -4280,9 +4294,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
-
-	/* Setup default input of Front HP to MW9 */
-	{0x28, AC_VERB_SET_CONNECT_SEL, 0x1},
 	/* PW9 PW10 Output enable */
 	{0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
 	{0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
@@ -4291,10 +4302,10 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	/* Enable Boost Volume backdoor */
 	{0x1, 0xf88, 0x8},
 	/* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
@@ -4304,8 +4315,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	/* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */
 	{0x34, AC_VERB_SET_CONNECT_SEL, 0x2},
 	{0x35, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Unmute MW4's index 0 */
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ }
 };
 
@@ -4453,6 +4462,19 @@ static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec,
 			if (err < 0)
 				return err;
 		} else if (i == AUTO_SEQ_FRONT) {
+			/* add control to mixer index 0 */
+			err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+					      "Master Front Playback Volume",
+					      HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
+								  HDA_INPUT));
+			if (err < 0)
+				return err;
+			err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+					      "Master Front Playback Switch",
+					      HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
+								  HDA_INPUT));
+			if (err < 0)
+				return err;
 			/* Front */
 			sprintf(name, "%s Playback Volume", chname[i]);
 			err = via_add_control(
diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
index 34b24286d27..2692e5ae5f2 100644
--- a/sound/pci/lola/lola.c
+++ b/sound/pci/lola/lola.c
@@ -445,7 +445,7 @@ static void lola_reset_setups(struct lola *chip)
 	lola_setup_all_analog_gains(chip, PLAY, false); /* output, update */
 }
 
-static int lola_parse_tree(struct lola *chip)
+static int __devinit lola_parse_tree(struct lola *chip)
 {
 	unsigned int val;
 	int nid, err;
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 949691a876d..c8e402fc378 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -521,6 +521,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #define HDSPM_DMA_AREA_KILOBYTES (HDSPM_DMA_AREA_BYTES/1024)
 
 /* revisions >= 230 indicate AES32 card */
+#define HDSPM_MADI_OLD_REV	207
 #define HDSPM_MADI_REV		210
 #define HDSPM_RAYDAT_REV	211
 #define HDSPM_AIO_REV		212
@@ -895,11 +896,11 @@ struct hdspm {
 	unsigned char max_channels_in;
 	unsigned char max_channels_out;
 
-	char *channel_map_in;
-	char *channel_map_out;
+	signed char *channel_map_in;
+	signed char *channel_map_out;
 
-	char *channel_map_in_ss, *channel_map_in_ds, *channel_map_in_qs;
-	char *channel_map_out_ss, *channel_map_out_ds, *channel_map_out_qs;
+	signed char *channel_map_in_ss, *channel_map_in_ds, *channel_map_in_qs;
+	signed char *channel_map_out_ss, *channel_map_out_ds, *channel_map_out_qs;
 
 	char **port_names_in;
 	char **port_names_out;
@@ -1143,7 +1144,7 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm)
 
 		/* if wordclock has synced freq and wordclock is valid */
 		if ((status2 & HDSPM_wcLock) != 0 &&
-				(status & HDSPM_SelSyncRef0) == 0) {
+				(status2 & HDSPM_SelSyncRef0) == 0) {
 
 			rate_bits = status2 & HDSPM_wcFreqMask;
 
@@ -1639,12 +1640,14 @@ static int snd_hdspm_midi_input_read (struct hdspm_midi *hmidi)
 		}
 	}
 	hmidi->pending = 0;
+	spin_unlock_irqrestore(&hmidi->lock, flags);
 
+	spin_lock_irqsave(&hmidi->hdspm->lock, flags);
 	hmidi->hdspm->control_register |= hmidi->ie;
 	hdspm_write(hmidi->hdspm, HDSPM_controlRegister,
 		    hmidi->hdspm->control_register);
+	spin_unlock_irqrestore(&hmidi->hdspm->lock, flags);
 
-	spin_unlock_irqrestore (&hmidi->lock, flags);
 	return snd_hdspm_midi_output_write (hmidi);
 }
 
@@ -6377,6 +6380,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
 
 	switch (hdspm->firmware_rev) {
 	case HDSPM_MADI_REV:
+	case HDSPM_MADI_OLD_REV:
 		hdspm->io_type = MADI;
 		hdspm->card_name = "RME MADI";
 		hdspm->midiPorts = 3;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 7fbfa051f6e..eda955b1583 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -848,9 +848,10 @@ int atmel_ssc_set_audio(int ssc_id)
 	if (IS_ERR(ssc))
 		pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n",
 			PTR_ERR(ssc));
-	else
+	else {
 		ssc_pdev->dev.parent = &(ssc->pdev->dev);
-	ssc_free(ssc);
+		ssc_free(ssc);
+	}
 
 	ret = platform_device_add(ssc_pdev);
 	if (ret < 0)
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index ea4951cf552..f79d1655e03 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -75,7 +75,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
 		.cpu_dai_name = "bfin-tdm.0",
 		.codec_dai_name = "ad1836-hifi",
 		.platform_name = "bfin-tdm-pcm-audio",
-		.codec_name = "ad1836.0",
+		.codec_name = "spi0.4",
 		.ops = &bf5xx_ad1836_ops,
 	},
 	{
@@ -84,7 +84,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
 		.cpu_dai_name = "bfin-tdm.1",
 		.codec_dai_name = "ad1836-hifi",
 		.platform_name = "bfin-tdm-pcm-audio",
-		.codec_name = "ad1836.0",
+		.codec_name = "spi0.4",
 		.ops = &bf5xx_ad1836_ops,
 	},
 };
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index b5101efd1c8..f1fd95bb641 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -138,11 +138,20 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
 	pr_debug("%s enter\n", __func__);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		diff = sport_curr_offset_tx(sport);
-		frames = bytes_to_frames(substream->runtime, diff);
 	} else {
 		diff = sport_curr_offset_rx(sport);
-		frames = bytes_to_frames(substream->runtime, diff);
 	}
+
+	/*
+	 * TX at least can report one frame beyond the end of the
+	 * buffer if we hit the wraparound case - clamp to within the
+	 * buffer as the ALSA APIs require.
+	 */
+	if (diff == snd_pcm_lib_buffer_bytes(substream))
+		diff = 0;
+
+	frames = bytes_to_frames(substream->runtime, diff);
+
 	return frames;
 }
 
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index ab63d52e36e..754c496412b 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -145,22 +145,22 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
 	/* bit size */
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-		word_len = 3;
+		word_len = AD1836_WORD_LEN_16;
 		break;
 	case SNDRV_PCM_FORMAT_S20_3LE:
-		word_len = 1;
+		word_len = AD1836_WORD_LEN_20;
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
 	case SNDRV_PCM_FORMAT_S32_LE:
-		word_len = 0;
+		word_len = AD1836_WORD_LEN_24;
 		break;
 	}
 
-	snd_soc_update_bits(codec, AD1836_DAC_CTRL1,
-		AD1836_