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 30468b31cad8..6fd9391b3a6c 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 41b901bde5c7..6e5addeb236b 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 2c41825c836e..eb9fe2e1d291 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 86ee16ca365e..440030818db7 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 2ca6f4f85b41..e3569bdd3b64 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 f16bc8aad6ed..e083122ca55a 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 5e619a84da06..15f0161ce4a2 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 f1de1bac042c..55f0647458c7 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 b05f7be9dc1b..e3e853153d14 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 3e6b9a8539c2..7bbc5f237a5e 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 7a4e10002f56..b48fb43b5448 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 605c99e1e520..f43bb0eaed8b 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 34b24286d279..2692e5ae5f2d 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 949691a876d3..c8e402fc3782 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 7fbfa051f6e1..eda955b15834 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 ea4951cf5526..f79d1655e035 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 b5101efd1c87..f1fd95bb6416 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 ab63d52e36e1..754c496412bd 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_DAC_WORD_LEN_MASK, word_len);
+	snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK,
+		word_len << AD1836_DAC_WORD_LEN_OFFSET);
 
-	snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
-		AD1836_ADC_WORD_LEN_MASK, word_len);
+	snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK,
+		word_len << AD1836_ADC_WORD_OFFSET);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 845596717fdf..9d6a3f8f8aaf 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -25,6 +25,7 @@
 #define AD1836_DAC_SERFMT_PCK256       (0x4 << 5)
 #define AD1836_DAC_SERFMT_PCK128       (0x5 << 5)
 #define AD1836_DAC_WORD_LEN_MASK       0x18
+#define AD1836_DAC_WORD_LEN_OFFSET     3
 
 #define AD1836_DAC_CTRL2               1
 #define AD1836_DACL1_MUTE              0
@@ -51,6 +52,7 @@
 #define AD1836_ADCL2_MUTE 		2
 #define AD1836_ADCR2_MUTE 		3
 #define AD1836_ADC_WORD_LEN_MASK       0x30
+#define AD1836_ADC_WORD_OFFSET         5
 #define AD1836_ADC_SERFMT_MASK	       (7 << 6)
 #define AD1836_ADC_SERFMT_PCK256       (0x4 << 6)
 #define AD1836_ADC_SERFMT_PCK128       (0x5 << 6)
@@ -60,4 +62,8 @@
 
 #define AD1836_NUM_REGS                16
 
+#define AD1836_WORD_LEN_24 0x0
+#define AD1836_WORD_LEN_20 0x1
+#define AD1836_WORD_LEN_16 0x2
+
 #endif
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4be0570e3f1f..65f46047b1cb 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -357,7 +357,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	default:
 		return -EINVAL;
 	}
-	snd_soc_update_bits(codec, PW_MGMT2, MS, data);
+	snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
 	snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
 
 	/* format type */
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index e2a7608d3944..7859bdcc93db 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -161,10 +161,18 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
 		dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
 	}
 
-	/* Configure PLL */
+	/**
+	 * Configure PLL
+	 * fsref = (mclk * PLLM) / 2048
+	 * where PLLM = J.DDDD (DDDD register ranges from 0 to 9999, decimal)
+	 */
 	pval = 1;
-	jval = (fsref == 44100) ? 7 : 8;
-	dval = (fsref == 44100) ? 5264 : 1920;
+	/* compute J portion of multiplier */
+	jval = fsref / (aic26->mclk / 2048);
+	/* compute fractional DDDD component of multiplier */
+	dval = fsref - (jval * (aic26->mclk / 2048));
+	dval = (10000 * dval) / (aic26->mclk / 2048);
+	dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval);
 	qval = 0;
 	reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
 	aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index c3d96fc8c267..789453d44ec5 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1114,12 +1114,19 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power)
 
 		/* Sync reg_cache with the hardware */
 		codec->cache_only = 0;
-		for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++)
+		for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++)
 			snd_soc_write(codec, i, cache[i]);
 		if (aic3x->model == AIC3X_MODEL_3007)
 			aic3x_init_3007(codec);
 		codec->cache_sync = 0;
 	} else {
+		/*
+		 * Do soft reset to this codec instance in order to clear
+		 * possible VDD leakage currents in case the supply regulators
+		 * remain on
+		 */
+		snd_soc_write(codec, AIC3X_RESET, SOFT_RESET);
+		codec->cache_sync = 1;
 		aic3x->power = 0;
 		/* HW writes are needless when bias is off */
 		codec->cache_only = 1;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 2dc964b55e4f..76b4361e9b80 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -175,6 +175,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls =
 SOC_DAPM_ENUM("Input Select", wm8731_insel_enum);
 
 static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("ACTIVE",WM8731_ACTIVE, 0, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0),
 SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1,
 	&wm8731_output_mixer_controls[0],
@@ -204,6 +205,8 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
 static const struct snd_soc_dapm_route wm8731_intercon[] = {
 	{"DAC", NULL, "OSC", wm8731_check_osc},
 	{"ADC", NULL, "OSC", wm8731_check_osc},
+	{"DAC", NULL, "ACTIVE"},
+	{"ADC", NULL, "ACTIVE"},
 
 	/* output mixer */
 	{"Output Mixer", "Line Bypass Switch", "Line Input"},
@@ -315,29 +318,6 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
-			      struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-
-	/* set active */
-	snd_soc_write(codec, WM8731_ACTIVE, 0x0001);
-
-	return 0;
-}
-
-static void wm8731_shutdown(struct snd_pcm_substream *substream,
-			    struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-
-	/* deactivate */
-	if (!codec->active) {
-		udelay(50);
-		snd_soc_write(codec, WM8731_ACTIVE, 0x0);
-	}
-}
-
 static int wm8731_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
@@ -480,7 +460,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, WM8731_PWR, reg | 0x0040);
 		break;
 	case SND_SOC_BIAS_OFF:
-		snd_soc_write(codec, WM8731_ACTIVE, 0x0);
 		snd_soc_write(codec, WM8731_PWR, 0xffff);
 		regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
 				       wm8731->supplies);
@@ -496,9 +475,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 	SNDRV_PCM_FMTBIT_S24_LE)
 
 static struct snd_soc_dai_ops wm8731_dai_ops = {
-	.prepare	= wm8731_pcm_prepare,
 	.hw_params	= wm8731_hw_params,
-	.shutdown	= wm8731_shutdown,
 	.digital_mute	= wm8731_mute,
 	.set_sysclk	= wm8731_set_dai_sysclk,
 	.set_fmt	= wm8731_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 6785688f8806..9a5e67c5a6bd 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -680,20 +680,25 @@ static struct snd_soc_dai_ops wm8804_dai_ops = {
 #define WM8804_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			SNDRV_PCM_FMTBIT_S24_LE)
 
+#define WM8804_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+		      SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
+		      SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+		      SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
 static struct snd_soc_dai_driver wm8804_dai = {
 	.name = "wm8804-spdif",
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000,
+		.rates = WM8804_RATES,
 		.formats = WM8804_FORMATS,
 	},
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 2,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000,
+		.rates = WM8804_RATES,
 		.formats = WM8804_FORMATS,
 	},
 	.ops = &wm8804_dai_ops,
diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c
index a0b1a7278284..e2ab4fac2819 100644
--- a/sound/soc/codecs/wm8915.c
+++ b/sound/soc/codecs/wm8915.c
@@ -1839,7 +1839,7 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai,
 	int old;
 
 	/* Disable SYSCLK while we reconfigure */
-	old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1);
+	old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA;
 	snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
 			    WM8915_SYSCLK_ENA, 0);
 
@@ -2038,6 +2038,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 		break;
 	case WM8915_FLL_MCLK2:
 		reg = 1;
+		break;
 	case WM8915_FLL_DACLRCLK1:
 		reg = 2;
 		break;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index f90ae427242b..5e05eed96c38 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1999,12 +1999,12 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol,
 		return 0;
 
 	/* If the left PGA is enabled hit that VU bit... */
-	if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTL_PGA_ENA)
+	if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA)
 		return snd_soc_write(codec, WM8962_HPOUTL_VOLUME,
 				     reg_cache[WM8962_HPOUTL_VOLUME]);
 
 	/* ...otherwise the right.  The VU is stereo. */
-	if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTR_PGA_ENA)
+	if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA)
 		return snd_soc_write(codec, WM8962_HPOUTR_VOLUME,
 				     reg_cache[WM8962_HPOUTR_VOLUME]);
 
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 3c2ee1bb73cd..6af23d06870f 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -13,7 +13,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 970a95c5360b..c2fc0356c2a4 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1713,6 +1713,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
 		snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
 				    WM8994_FLL1_ENA | WM8994_FLL1_FRAC,
 				    reg);
+
+		msleep(5);
 	}
 
 	wm8994->fll[id].in = freq_in;
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 15dac0f20cd8..6680c0b4d203 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
 	 * should allocate a DMA buffer only for the streams that are valid.
 	 */
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[0].substream) {
 		ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
 			fsl_dma_hardware.buffer_bytes_max,
 			&pcm->streams[0].substream->dma_buffer);
@@ -320,13 +320,13 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
 		}
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[1].substream) {
 		ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
 			fsl_dma_hardware.buffer_bytes_max,
 			&pcm->streams[1].substream->dma_buffer);
 		if (ret) {
-			snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
 			dev_err(card->dev, "can't alloc capture dma buffer\n");
+			snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
 			return ret;
 		}
 	}
@@ -449,7 +449,8 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
 	dma_private->ld_buf_phys = ld_buf_phys;
 	dma_private->dma_buf_phys = substream->dma_buffer.addr;
 
-	ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private);
+	ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "fsldma-audio",
+			  dma_private);
 	if (ret) {
 		dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n",
 			dma_private->irq, ret);
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index d8f130d39dd9..bb699bb55a50 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -11,9 +11,6 @@ menuconfig SND_IMX_SOC
 
 if SND_IMX_SOC
 
-config SND_MXC_SOC_SSI
-	tristate
-
 config SND_MXC_SOC_FIQ
 	tristate
 
@@ -24,7 +21,6 @@ config SND_MXC_SOC_WM1133_EV1
 	tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
 	depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
 	select SND_SOC_WM8350
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
 	help
 	  Enable support for audio on the i.MX31ADS with the WM1133-EV1
@@ -34,7 +30,6 @@ config SND_SOC_MX27VIS_AIC32X4
 	tristate "SoC audio support for Visstrim M10 boards"
 	depends on MACH_IMX27_VISSTRIM_M10
 	select SND_SOC_TVL320AIC32X4
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_MX2
 	help
 	  Say Y if you want to add support for SoC audio on Visstrim SM10
@@ -44,7 +39,6 @@ config SND_SOC_PHYCORE_AC97
 	tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
 	depends on MACH_PCM043 || MACH_PCA100
 	select SND_SOC_WM9712
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
 	help
 	  Say Y if you want to add support for SoC audio on Phytec phyCORE
@@ -57,7 +51,6 @@ config SND_SOC_EUKREA_TLV320
 		|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
 		|| MACH_EUKREA_MBIMXSD51_BASEBOARD
 	select SND_SOC_TLV320AIC23
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
 	help
 	  Enable I2S based access to the TLV320AIC23B codec attached
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index aab7765f401a..4173b3d87f97 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -337,3 +337,5 @@ static void __exit snd_imx_pcm_exit(void)
 	platform_driver_unregister(&imx_pcm_driver);
 }
 module_exit(snd_imx_pcm_exit);
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imx-pcm-audio");
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 5b13feca7537..61fceb09cdb5 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -774,4 +774,4 @@ module_exit(imx_ssi_exit);
 MODULE_AUTHOR("Sascha Hauer, <s.hauer@pengutronix.de>");
 MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface");
 MODULE_LICENSE("GPL");
-
+MODULE_ALIAS("platform:imx-ssi");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 2ce0b2d891d5..fab20a54e863 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -95,14 +95,14 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index ffa09b3b2caa..992a732b5211 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -191,7 +191,7 @@ static inline bool tx_active(struct i2s_dai *i2s)
 	if (!i2s)
 		return false;
 
-	active = readl(i2s->addr + I2SMOD);
+	active = readl(i2s->addr + I2SCON);
 
 	if (is_secondary(i2s))
 		active &= CON_TXSDMA_ACTIVE;
@@ -223,7 +223,7 @@ static inline bool rx_active(struct i2s_dai *i2s)
 	if (!i2s)
 		return false;
 
-	active = readl(i2s->addr + I2SMOD) & CON_RXDMA_ACTIVE;
+	active = readl(i2s->addr + I2SCON) & CON_RXDMA_ACTIVE;
 
 	return active ? true : false;
 }
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 06b7b81a1601..039b9532b270 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -409,9 +409,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 	codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
 
 	switch (control) {
-	case SND_SOC_CUSTOM:
-		break;
-
 	case SND_SOC_I2C:
 #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
 		codec->hw_write = (hw_write_t)i2c_master_send;
@@ -466,6 +463,9 @@ static bool snd_soc_set_cache_val(void *base, unsigned int idx,
 static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx,
 		unsigned int word_size)
 {
+	if (!base)
+		return -1;
+
 	switch (word_size) {
 	case 1: {
 		const u8 *cache = base;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d75043ed7fc0..b194be09e74d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1929,8 +1929,9 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 		 "%s", card->name);
 	snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
 		 "%s", card->long_name ? card->long_name : card->name);
-	snprintf(card->snd_card->driver, sizeof(card->snd_card->driver),
-		 "%s", card->driver_name ? card->driver_name : card->name);
+	if (card->driver_name)
+		strlcpy(card->snd_card->driver, card->driver_name,
+			sizeof(card->snd_card->driver));
 
 	if (card->late_probe) {
 		ret = card->late_probe(card);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 776e6f418306..32ab7fc4579a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -350,9 +350,9 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
 }
 
 /* create new dapm mixer control */
-static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
-	struct snd_soc_dapm_widget *w)
+static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
 {
+	struct snd_soc_dapm_context *dapm = w->dapm;
 	int i, ret = 0;
 	size_t name_len, prefix_len;
 	struct snd_soc_dapm_path *path;
@@ -450,9 +450,9 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
 }
 
 /* create new dapm mux control */
-static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
-	struct snd_soc_dapm_widget *w)
+static int dapm_new_mux(struct snd_soc_dapm_widget *w)
 {
+	struct snd_soc_dapm_context *dapm = w->dapm;
 	struct snd_soc_dapm_path *path = NULL;
 	struct snd_kcontrol *kcontrol;
 	struct snd_card *card = dapm->card->snd_card;
@@ -535,8 +535,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
 }
 
 /* create new dapm volume control */
-static int dapm_new_pga(struct snd_soc_dapm_context *dapm,
-	struct snd_soc_dapm_widget *w)
+static int dapm_new_pga(struct snd_soc_dapm_widget *w)
 {
 	if (w->num_kcontrols)
 		dev_err(w->dapm->dev,
@@ -1826,13 +1825,13 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 		case snd_soc_dapm_mixer:
 		case snd_soc_dapm_mixer_named_ctl:
 			w->power_check = dapm_generic_check_power;
-			dapm_new_mixer(dapm, w);
+			dapm_new_mixer(w);
 			break;
 		case snd_soc_dapm_mux:
 		case snd_soc_dapm_virt_mux:
 		case snd_soc_dapm_value_mux:
 			w->power_check = dapm_generic_check_power;
-			dapm_new_mux(dapm, w);
+			dapm_new_mux(w);
 			break;
 		case snd_soc_dapm_adc:
 		case snd_soc_dapm_aif_out:
@@ -1845,7 +1844,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 		case snd_soc_dapm_pga:
 		case snd_soc_dapm_out_drv:
 			w->power_check = dapm_generic_check_power;
-			dapm_new_pga(dapm, w);
+			dapm_new_pga(w);
 			break;
 		case snd_soc_dapm_input:
 		case snd_soc_dapm_output:
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 6b817e20548c..95f03c10b4f7 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -222,12 +222,18 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream,
 	if (i2sclock % (2 * srate))
 		reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE;
 
+	if (!i2s->clk_refs)
+		clk_enable(i2s->clk_i2s);
+
 	tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg);
 
 	tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR,
 		TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
 		TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
 
+	if (!i2s->clk_refs)
+		clk_disable(i2s->clk_i2s);
+
 	return 0;
 }
 
diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c
index 337a00241a1f..4dd051bdf4fd 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -1124,6 +1124,6 @@ static void __exit at73c213_exit(void)
 }
 module_exit(at73c213_exit);
 
-MODULE_AUTHOR("Hans-Christian Egtvedt <hcegtvedt@atmel.com>");
+MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt@samfundet.no>");
 MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC");
 MODULE_LICENSE("GPL");
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index a91719d5918b..1e3ae3327dd3 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -270,7 +270,6 @@ static int usb6fire_fw_ezusb_upload(
 	data = 0x00; /* resume ezusb cpu */
 	ret = usb6fire_fw_ezusb_write(device, 0xa0, 0xe600, &data, 1);
 	if (ret < 0) {
-		release_firmware(fw);
 		snd_printk(KERN_ERR PREFIX "unable to upload ezusb "
 				"firmware %s: end message.\n", fwname);
 		return ret;
diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c
index b137b25865cc..d144cdb2f159 100644
--- a/sound/usb/6fire/pcm.c
+++ b/sound/usb/6fire/pcm.c
@@ -395,12 +395,12 @@ static int usb6fire_pcm_open(struct snd_pcm_substream *alsa_sub)
 	alsa_rt->hw = pcm_hw;
 
 	if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		if (rt->rate >= 0)
+		if (rt->rate < ARRAY_SIZE(rates))
 			alsa_rt->hw.rates = rates_alsaid[rt->rate];
 		alsa_rt->hw.channels_max = OUT_N_CHANNELS;
 		sub = &rt->playback;
 	} else if (alsa_sub->stream == SNDRV_PCM_STREAM_CAPTURE) {
-		if (rt->rate >= 0)
+		if (rt->rate < ARRAY_SIZE(rates))
 			alsa_rt->hw.rates = rates_alsaid[rt->rate];
 		alsa_rt->hw.channels_max = IN_N_CHANNELS;
 		sub = &rt->capture;