Merge branch 'fix/misc' into topic/misc

47 files changed, 642 insertions, 828 deletions

diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 5dbab38d04af..130cfe677d60 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -52,6 +52,7 @@ struct link_slave {
 	struct link_ctl_info info;
 	int vals[2];		/* current values */
 	unsigned int flags;
+	struct snd_kcontrol *kctl; /* original kcontrol pointer */
 	struct snd_kcontrol slave; /* the copy of original control entry */
 };
 
@@ -252,6 +253,7 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
 		       slave->count * sizeof(*slave->vd), GFP_KERNEL);
 	if (!srec)
 		return -ENOMEM;
+	srec->kctl = slave;
 	srec->slave = *slave;
 	memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd));
 	srec->master = master_link;
@@ -333,10 +335,18 @@ static int master_put(struct snd_kcontrol *kcontrol,
 static void master_free(struct snd_kcontrol *kcontrol)
 {
 	struct link_master *master = snd_kcontrol_chip(kcontrol);
-	struct link_slave *slave;
-
-	list_for_each_entry(slave, &master->slaves, list)
-		slave->master = NULL;
+	struct link_slave *slave, *n;
+
+	/* free all slave links and retore the original slave kctls */
+	list_for_each_entry_safe(slave, n, &master->slaves, list) {
+		struct snd_kcontrol *sctl = slave->kctl;
+		struct list_head olist = sctl->list;
+		memcpy(sctl, &slave->slave, sizeof(*sctl));
+		memcpy(sctl->vd, slave->slave.vd,
+		       sctl->count * sizeof(*sctl->vd));
+		sctl->list = olist; /* keep the current linked-list */
+		kfree(slave);
+	}
 	kfree(master);
 }
 
diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c
index e083122ca55a..dbf94b189e75 100644
--- a/sound/pci/cs5535audio/cs5535audio_pcm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pcm.c
@@ -148,7 +148,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
 		struct cs5535audio_dma_desc *desc =
 			&((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i];
 		desc->addr = cpu_to_le32(addr);
-		desc->size = cpu_to_le32(period_bytes);
+		desc->size = cpu_to_le16(period_bytes);
 		desc->ctlreserved = cpu_to_le16(PRD_EOP);
 		desc_addr += sizeof(struct cs5535audio_dma_desc);
 		addr += period_bytes;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 916a1863af73..4562e9de6a1a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2331,6 +2331,39 @@ int snd_hda_codec_reset(struct hda_codec *codec)
 	return 0;
 }
 
+typedef int (*map_slave_func_t)(void *, struct snd_kcontrol *);
+
+/* apply the function to all matching slave ctls in the mixer list */
+static int map_slaves(struct hda_codec *codec, const char * const *slaves,
+		      map_slave_func_t func, void *data) 
+{
+	struct hda_nid_item *items;
+	const char * const *s;
+	int i, err;
+
+	items = codec->mixers.list;
+	for (i = 0; i < codec->mixers.used; i++) {
+		struct snd_kcontrol *sctl = items[i].kctl;
+		if (!sctl || !sctl->id.name ||
+		    sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER)
+			continue;
+		for (s = slaves; *s; s++) {
+			if (!strcmp(sctl->id.name, *s)) {
+				err = func(data, sctl);
+				if (err)
+					return err;
+				break;
+			}
+		}
+	}
+	return 0;
+}
+
+static int check_slave_present(void *data, struct snd_kcontrol *sctl)
+{
+	return 1;
+}
+
 /**
  * snd_hda_add_vmaster - create a virtual master control and add slaves
  * @codec: HD-audio codec
@@ -2351,12 +2384,10 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 			unsigned int *tlv, const char * const *slaves)
 {
 	struct snd_kcontrol *kctl;
-	const char * const *s;
 	int err;
 
-	for (s = slaves; *s && !snd_hda_find_mixer_ctl(codec, *s); s++)
-		;
-	if (!*s) {
+	err = map_slaves(codec, slaves, check_slave_present, NULL);
+	if (err != 1) {
 		snd_printdd("No slave found for %s\n", name);
 		return 0;
 	}
@@ -2367,23 +2398,10 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 	if (err < 0)
 		return err;
 
-	for (s = slaves; *s; s++) {
-		struct snd_kcontrol *sctl;
-		int i = 0;
-		for (;;) {
-			sctl = _snd_hda_find_mixer_ctl(codec, *s, i);
-			if (!sctl) {
-				if (!i)
-					snd_printdd("Cannot find slave %s, "
-						    "skipped\n", *s);
-				break;
-			}
-			err = snd_ctl_add_slave(kctl, sctl);
-			if (err < 0)
-				return err;
-			i++;
-		}
-	}
+	err = map_slaves(codec, slaves, (map_slave_func_t)snd_ctl_add_slave,
+			 kctl);
+	if (err < 0)
+		return err;
 	return 0;
 }
 EXPORT_SYMBOL_HDA(snd_hda_add_vmaster);
@@ -4028,9 +4046,9 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec,
 
 	/* Search for codec ID */
 	for (q = tbl; q->subvendor; q++) {
-		unsigned long vendorid = (q->subdevice) | (q->subvendor << 16);
-
-		if (vendorid == codec->subsystem_id)
+		unsigned int mask = 0xffff0000 | q->subdevice_mask;
+		unsigned int id = (q->subdevice | (q->subvendor << 16)) & mask;
+		if ((codec->subsystem_id & mask) == id)
 			break;
 	}
 
@@ -4752,6 +4770,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 	memset(sequences_hp, 0, sizeof(sequences_hp));
 	assoc_line_out = 0;
 
+	codec->ignore_misc_bit = true;
 	end_nid = codec->start_nid + codec->num_nodes;
 	for (nid = codec->start_nid; nid < end_nid; nid++) {
 		unsigned int wid_caps = get_wcaps(codec, nid);
@@ -4767,6 +4786,9 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 			continue;
 
 		def_conf = snd_hda_codec_get_pincfg(codec, nid);
+		if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
+		      AC_DEFCFG_MISC_NO_PRESENCE))
+			codec->ignore_misc_bit = false;
 		conn = get_defcfg_connect(def_conf);
 		if (conn == AC_JACK_PORT_NONE)
 			continue;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 755f2b0f9d8e..564471169cae 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -854,6 +854,7 @@ struct hda_codec {
 	unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */
 	unsigned int pins_shutup:1;	/* pins are shut up */
 	unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
+	unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	unsigned int power_on :1;	/* current (global) power-state */
 	unsigned int power_transition :1; /* power-state in transition */
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 1c8ddf547a2d..c1da422e085a 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -297,10 +297,18 @@ static int hdmi_update_eld(struct hdmi_eld *e,
 					buf + ELD_FIXED_BYTES + mnl + 3 * i);
 	}
 
+	/*
+	 * HDMI sink's ELD info cannot always be retrieved for now, e.g.
+	 * in console or for audio devices. Assume the highest speakers
+	 * configuration, to _not_ prohibit multi-channel audio playback.
+	 */
+	if (!e->spk_alloc)
+		e->spk_alloc = 0xffff;
+
+	e->eld_valid = true;
 	return 0;
 
 out_fail:
-	e->eld_ver = 0;
 	return -EINVAL;
 }
 
@@ -323,9 +331,6 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
 	 * ELD is valid, actual eld_size is assigned in hdmi_update_eld()
 	 */
 
-	if (!eld->eld_valid)
-		return -ENOENT;
-
 	size = snd_hdmi_get_eld_size(codec, nid);
 	if (size == 0) {
 		/* wfg: workaround for ASUS P5E-VM HDMI board */
@@ -342,18 +347,28 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
 
 	for (i = 0; i < size; i++) {
 		unsigned int val = hdmi_get_eld_data(codec, nid, i);
+		/*
+		 * Graphics driver might be writing to ELD buffer right now.
+		 * Just abort. The caller will repoll after a while.
+		 */
 		if (!(val & AC_ELDD_ELD_VALID)) {
-			if (!i) {
-				snd_printd(KERN_INFO
-					   "HDMI: invalid ELD data\n");
-				ret = -EINVAL;
-				goto error;
-			}
 			snd_printd(KERN_INFO
 				  "HDMI: invalid ELD data byte %d\n", i);
-			val = 0;
-		} else
-			val &= AC_ELDD_ELD_DATA;
+			ret = -EINVAL;
+			goto error;
+		}
+		val &= AC_ELDD_ELD_DATA;
+		/*
+		 * The first byte cannot be zero. This can happen on some DVI
+		 * connections. Some Intel chips may also need some 250ms delay
+		 * to return non-zero ELD data, even when the graphics driver
+		 * correctly writes ELD content before setting ELD_valid bit.
+		 */
+		if (!val && !i) {
+			snd_printdd(KERN_INFO "HDMI: 0 ELD data\n");
+			ret = -EINVAL;
+			goto error;
+		}
 		buf[i] = val;
 	}
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 096507d2ca9a..7d98240def0b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2508,7 +2508,6 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
-	SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index dcbea0da0fa2..618ddad17236 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -510,13 +510,15 @@ int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
 
 static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
 {
-	return (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT) &&
-		/* disable MISC_NO_PRESENCE check because it may break too
-		 * many devices
-		 */
-		/*(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid) &
-		  AC_DEFCFG_MISC_NO_PRESENCE)) &&*/
-		(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP);
+	if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT))
+		return false;
+	if (!codec->ignore_misc_bit &&
+	    (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
+	     AC_DEFCFG_MISC_NO_PRESENCE))
+		return false;
+	if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
+		return false;
+	return true;
 }
 
 /* flags for hda_nid_item */
@@ -651,6 +653,9 @@ struct hdmi_eld {
 	int	spk_alloc;
 	int	sad_count;
 	struct cea_sad sad[ELD_MAX_SAD];
+	/*
+	 * all fields above eld_buffer will be cleared before updating ELD
+	 */
 	char    eld_buffer[ELD_MAX_SIZE];
 #ifdef CONFIG_PROC_FS
 	struct snd_info_entry *proc_entry;
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 2a2d8645ba09..70a7abda7e22 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -58,6 +58,8 @@ struct cs_spec {
 	unsigned int gpio_mask;
 	unsigned int gpio_dir;
 	unsigned int gpio_data;
+	unsigned int gpio_eapd_hp; /* EAPD GPIO bit for headphones */
+	unsigned int gpio_eapd_speaker; /* EAPD GPIO bit for speakers */
 
 	struct hda_pcm pcm_rec[2];	/* PCM information */
 
@@ -76,6 +78,7 @@ enum {
 	CS420X_MBP53,
 	CS420X_MBP55,
 	CS420X_IMAC27,
+	CS420X_APPLE,
 	CS420X_AUTO,
 	CS420X_MODELS
 };
@@ -237,6 +240,15 @@ static int cs_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
 }
 
+static void cs_update_input_select(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	if (spec->cur_adc)
+		snd_hda_codec_write(codec, spec->cur_adc, 0,
+				    AC_VERB_SET_CONNECT_SEL,
+				    spec->adc_idx[spec->cur_input]);
+}
+
 /*
  * Analog capture
  */
@@ -250,6 +262,7 @@ static int cs_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
 	spec->cur_adc = spec->adc_nid[spec->cur_input];
 	spec->cur_adc_stream_tag = stream_tag;
 	spec->cur_adc_format = format;
+	cs_update_input_select(codec);
 	snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format);
 	return 0;
 }
@@ -689,10 +702,8 @@ static int change_cur_input(struct hda_codec *codec, unsigned int idx,
 					   spec->cur_adc_stream_tag, 0,
 					   spec->cur_adc_format);
 	}
-	snd_hda_codec_write(codec, spec->cur_adc, 0,
-			    AC_VERB_SET_CONNECT_SEL,
-			    spec->adc_idx[idx]);
 	spec->cur_input = idx;
+	cs_update_input_select(codec);
 	return 1;
 }
 
@@ -920,10 +931,9 @@ static void cs_automute(struct hda_codec *codec)
 					spdif_present ? 0 : PIN_OUT);
 		}
 	}
-	if (spec->board_config == CS420X_MBP53 ||
-	    spec->board_config == CS420X_MBP55 ||
-	    spec->board_config == CS420X_IMAC27) {
-		unsigned int gpio = hp_present ? 0x02 : 0x08;
+	if (spec->gpio_eapd_hp) {
+		unsigned int gpio = hp_present ?
+			spec->gpio_eapd_hp : spec->gpio_eapd_speaker;
 		snd_hda_codec_write(codec, 0x01, 0,
 				    AC_VERB_SET_GPIO_DATA, gpio);
 	}
@@ -973,10 +983,7 @@ static void cs_automic(struct hda_codec *codec)
 		} else  {
 			spec->cur_input = spec->last_input;
 		}
-
-		snd_hda_codec_write_cache(codec, spec->cur_adc, 0,
-					AC_VERB_SET_CONNECT_SEL,
-					spec->adc_idx[spec->cur_input]);
+		cs_update_input_select(codec);
 	} else {
 		if (present)
 			change_cur_input(codec, spec->automic_idx, 0);
@@ -1073,9 +1080,7 @@ static void init_input(struct hda_codec *codec)
 			cs_automic(codec);
 		else  {
 			spec->cur_adc = spec->adc_nid[spec->cur_input];
-			snd_hda_codec_write(codec, spec->cur_adc, 0,
-					AC_VERB_SET_CONNECT_SEL,
-					spec->adc_idx[spec->cur_input]);
+			cs_update_input_select(codec);
 		}
 	} else {
 		change_cur_input(codec, spec->cur_input, 1);
@@ -1273,6 +1278,7 @@ static const char * const cs420x_models[CS420X_MODELS] = {
 	[CS420X_MBP53] = "mbp53",
 	[CS420X_MBP55] = "mbp55",
 	[CS420X_IMAC27] = "imac27",
+	[CS420X_APPLE] = "apple",
 	[CS420X_AUTO] = "auto",
 };
 
@@ -1282,7 +1288,13 @@ static const struct snd_pci_quirk cs420x_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55),
 	SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55),
 	SND_PCI_QUIRK(0x10de, 0xcb89, "MacBookPro 7,1", CS420X_MBP55),
-	SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),
+	/* this conflicts with too many other models */
+	/*SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),*/
+	{} /* terminator */
+};
+
+static const struct snd_pci_quirk cs420x_codec_cfg_tbl[] = {
+	SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE),
 	{} /* terminator */
 };
 
@@ -1364,6 +1376,10 @@ static int patch_cs420x(struct hda_codec *codec)
 	spec->board_config =
 		snd_hda_check_board_config(codec, CS420X_MODELS,
 					   cs420x_models, cs420x_cfg_tbl);
+	if (spec->board_config < 0)
+		spec->board_config =
+			snd_hda_check_board_codec_sid_config(codec,
+				CS420X_MODELS, NULL, cs420x_codec_cfg_tbl);
 	if (spec->board_config >= 0)
 		fix_pincfg(codec, spec->board_config, cs_pincfgs);
 
@@ -1371,10 +1387,11 @@ static int patch_cs420x(struct hda_codec *codec)
 	case CS420X_IMAC27:
 	case CS420X_MBP53:
 	case CS420X_MBP55:
-		/* GPIO1 = headphones */
-		/* GPIO3 = speakers */
-		spec->gpio_mask = 0x0a;
-		spec->gpio_dir = 0x0a;
+	case CS420X_APPLE:
+		spec->gpio_eapd_hp = 2; /* GPIO1 = headphones */
+		spec->gpio_eapd_speaker = 8; /* GPIO3 = speakers */
+		spec->gpio_mask = spec->gpio_dir =
+			spec->gpio_eapd_hp | spec->gpio_eapd_speaker;
 		break;
 	}
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 5e706e4d1737..0de21193a2b0 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3062,7 +3062,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
-	SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
 		      CXT5066_LAPTOP),
 	SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 81b7b791b3c3..c505fd5d338c 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -65,7 +65,11 @@ struct hdmi_spec_per_pin {
 	hda_nid_t pin_nid;
 	int num_mux_nids;
 	hda_nid_t mux_nids[HDA_MAX_CONNECTIONS];
+
+	struct hda_codec *codec;
 	struct hdmi_eld sink_eld;
+	struct delayed_work work;
+	int repoll_count;
 };
 
 struct hdmi_spec {
@@ -745,8 +749,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx,
  * Unsolicited events
  */
 
-static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid,
-			       struct hdmi_eld *eld);
+static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
 
 static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 {
@@ -755,7 +758,6 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	int pd = !!(res & AC_UNSOL_RES_PD);
 	int eldv = !!(res & AC_UNSOL_RES_ELDV);
 	int pin_idx;
-	struct hdmi_eld *eld;
 
 	printk(KERN_INFO
 		"HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
@@ -764,17 +766,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	pin_idx = pin_nid_to_pin_index(spec, pin_nid);
 	if (pin_idx < 0)
 		return;
-	eld = &spec->pins[pin_idx].sink_eld;
-
-	hdmi_present_sense(codec, pin_nid, eld);
 
-	/*
-	 * HDMI sink's ELD info cannot always be retrieved for now, e.g.
-	 * in console or for audio devices. Assume the highest speakers
-	 * configuration, to _not_ prohibit multi-channel audio playback.
-	 */
-	if (!eld->spk_alloc)
-		eld->spk_alloc = 0xffff;
+	hdmi_present_sense(&spec->pins[pin_idx], 1);
 }
 
 static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res)
@@ -968,9 +961,11 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx)
 	return 0;
 }
 
-static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid,
-			       struct hdmi_eld *eld)
+static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 {
+	struct hda_codec *codec = per_pin->codec;
+	struct hdmi_eld *eld = &per_pin->sink_eld;
+	hda_nid_t pin_nid = per_pin->pin_nid;
 	/*
 	 * Always execute a GetPinSense verb here, even when called from
 	 * hdmi_intrinsic_event; for some NVIDIA HW, the unsolicited
@@ -980,26 +975,42 @@ static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid,
 	 * the unsolicited response to avoid custom WARs.
 	 */
 	int present = snd_hda_pin_sense(codec, pin_nid);
+	bool eld_valid = false;
 
-	memset(eld, 0, sizeof(*eld));
+	memset(eld, 0, offsetof(struct hdmi_eld, eld_buffer));
 
 	eld->monitor_present	= !!(present & AC_PINSENSE_PRESENCE);
 	if (eld->monitor_present)
-		eld->eld_valid	= !!(present & AC_PINSENSE_ELDV);
-	else
-		eld->eld_valid	= 0;
+		eld_valid	= !!(present & AC_PINSENSE_ELDV);
 
 	printk(KERN_INFO
 		"HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-		codec->addr, pin_nid, eld->monitor_present, eld->eld_valid);
+		codec->addr, pin_nid, eld->monitor_present, eld_valid);
 
-	if (eld->eld_valid)
+	if (eld_valid) {
 		if (!snd_hdmi_get_eld(eld, codec, pin_nid))
 			snd_hdmi_show_eld(eld);
+		else if (repoll) {
+			queue_delayed_work(codec->bus->workq,
+					   &per_pin->work,
+					   msecs_to_jiffies(300));
+		}
+	}
 
 	snd_hda_input_jack_report(codec, pin_nid);
 }
 
+static void hdmi_repoll_eld(struct work_struct *work)
+{
+	struct hdmi_spec_per_pin *per_pin =
+	container_of(to_delayed_work(work), struct hdmi_spec_per_pin, work);
+
+	if (per_pin->repoll_count++ > 6)
+		per_pin->repoll_count = 0;
+
+	hdmi_present_sense(per_pin, per_pin->repoll_count);
+}
+
 static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
 {
 	struct hdmi_spec *spec = codec->spec;
@@ -1228,7 +1239,7 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx)
 	if (err < 0)
 		return err;
 
-	hdmi_present_sense(codec, per_pin->pin_nid, &per_pin->sink_eld);
+	hdmi_present_sense(per_pin, 0);
 	return 0;
 }
 
@@ -1279,6 +1290,8 @@ static int generic_hdmi_init(struct hda_codec *codec)
 				    AC_VERB_SET_UNSOLICITED_ENABLE,
 				    AC_USRSP_EN | pin_nid);
 
+		per_pin->codec = codec;
+		INIT_DELAYED_WORK(&per_pin->work, hdmi_repoll_eld);
 		snd_hda_eld_proc_new(codec, eld, pin_idx);
 	}
 	return 0;
@@ -1293,10 +1306,12 @@ static void generic_hdmi_free(struct hda_codec *codec)
 		struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
 		struct hdmi_eld *eld = &per_pin->sink_eld;
 
+		cancel_delayed_work(&per_pin->work);
 		snd_hda_eld_proc_free(codec, eld);
 	}
 	snd_hda_input_jack_free(codec);
 
+	flush_workqueue(codec->bus->workq);
 	kfree(spec);
 }
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a24e068a021b..cbde019d3d52 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -277,6 +277,12 @@ static bool alc_dyn_adc_pcm_resetup(struct hda_codec *codec, int cur)
 	return false;
 }
 
+static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx)
+{
+	return spec->capsrc_nids ?
+		spec->capsrc_nids[idx] : spec->adc_nids[idx];
+}
+
 /* select the given imux item; either unmute exclusively or select the route */
 static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
 			  unsigned int idx, bool force)
@@ -284,7 +290,7 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
 	struct alc_spec *spec = codec->spec;
 	const struct hda_input_mux *imux;
 	unsigned int mux_idx;
-	int i, type;
+	int i, type, num_conns;
 	hda_nid_t nid;
 
 	mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
@@ -303,20 +309,20 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
 		adc_idx = spec->dyn_adc_idx[idx];
 	}
 
-	nid = spec->capsrc_nids ?
-		spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx];
+	nid = get_capsrc(spec, adc_idx);
 
 	/* no selection? */
-	if (snd_hda_get_conn_list(codec, nid, NULL) <= 1)
+	num_conns = snd_hda_get_conn_list(codec, nid, NULL);
+	if (num_conns <= 1)
 		return 1;
 
 	type = get_wcaps_type(get_wcaps(codec, nid));
 	if (type == AC_WID_AUD_MIX) {
 		/* Matrix-mixer style (e.g. ALC882) */
-		for (i = 0; i < imux->num_items; i++) {
-			unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
-			snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
-						 imux->items[i].index,
+		int active = imux->items[idx].index;
+		for (i = 0; i < num_conns; i++) {
+			unsigned int v = (i == active) ? 0 : HDA_AMP_MUTE;
+			snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, i,
 						 HDA_AMP_MUTE, v);
 		}
 	} else {
@@ -1053,8 +1059,19 @@ static bool alc_rebuild_imux_for_auto_mic(struct hda_codec *codec)
 	spec->imux_pins[2] = spec->dock_mic_pin;
 	for (i = 0; i < 3; i++) {
 		strcpy(imux->items[i].label, texts[i]);
-		if (spec->imux_pins[i])
+		if (spec->imux_pins[i]) {
+			hda_nid_t pin = spec->imux_pins[i];
+			int c;
+			for (c = 0; c < spec->num_adc_nids; c++) {
+				hda_nid_t cap = get_capsrc(spec, c);
+				int idx = get_connection_index(codec, cap, pin);
+				if (idx >= 0) {
+					imux->items[i].index = idx;
+					break;
+				}
+			}
 			imux->num_items = i + 1;
+		}
 	}
 	spec->num_mux_defs = 1;
 	spec->input_mux = imux;
@@ -1451,7 +1468,7 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
 		switch (fix->type) {
 		case ALC_FIXUP_SKU:
 			if (action != ALC_FIXUP_ACT_PRE_PROBE || !fix->v.sku)
-				break;;
+				break;
 			snd_printdd(KERN_INFO "hda_codec: %s: "
 				    "Apply sku override for %s\n",
 				    codec->chip_name, modelname);
@@ -1956,10 +1973,8 @@ static int alc_build_controls(struct hda_codec *codec)
 		if (!kctl)
 			kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
 		for (i = 0; kctl && i < kctl->count; i++) {
-			const hda_nid_t *nids = spec->capsrc_nids;
-			if (!nids)
-				nids = spec->adc_nids;
-			err = snd_hda_add_nid(codec, kctl, i, nids[i]);
+			err = snd_hda_add_nid(codec, kctl, i,
+					      get_capsrc(spec, i));
 			if (err < 0)
 				return err;
 		}
@@ -2746,8 +2761,7 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec)
 		}
 
 		for (c = 0; c < num_adcs; c++) {
-			hda_nid_t cap = spec->capsrc_nids ?
-				spec->capsrc_nids[c] : spec->adc_nids[c];
+			hda_nid_t cap = get_capsrc(spec, c);
 			idx = get_connection_index(codec, cap, pin);
 			if (idx >= 0) {
 				spec->imux_pins[imux->num_items] = pin;
@@ -3693,8 +3707,7 @@ static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin)
 	if (!pin)
 		return 0;
 	for (i = 0; i < spec->num_adc_nids; i++) {
-		hda_nid_t cap = spec->capsrc_nids ?
-			spec->capsrc_nids[i] : spec->adc_nids[i];
+		hda_nid_t cap = get_capsrc(spec, i);
 		int idx;
 
 		idx = get_connection_index(codec, cap, pin);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4e715fefebef..d8d2f9dccd9b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -95,6 +95,7 @@ enum {
 	STAC_92HD83XXX_REF,
 	STAC_92HD83XXX_PWR_REF,
 	STAC_DELL_S14,
+	STAC_DELL_VOSTRO_3500,
 	STAC_92HD83XXX_HP,
 	STAC_92HD83XXX_HP_cNB11_INTQUAD,
 	STAC_HP_DV7_4000,
@@ -226,7 +227,6 @@ struct sigmatel_spec {
 
 	/* power management */
 	unsigned int num_pwrs;
-	const unsigned int *pwr_mapping;
 	const hda_nid_t *pwr_nids;
 	const hda_nid_t *dac_list;
 
@@ -373,18 +373,15 @@ static const unsigned long stac92hd73xx_capvols[] = {
 
 #define STAC92HD83_DAC_COUNT 3
 
-static const hda_nid_t stac92hd83xxx_pwr_nids[4] = {
-	0xa, 0xb, 0xd, 0xe,
+static const hda_nid_t stac92hd83xxx_pwr_nids[7] = {
+	0x0a, 0x0b, 0x0c, 0xd, 0x0e,
+	0x0f, 0x10
 };
 
 static const hda_nid_t stac92hd83xxx_slave_dig_outs[2] = {
 	0x1e, 0,
 };
 
-static const unsigned int stac92hd83xxx_pwr_mapping[4] = {
-	0x03, 0x0c, 0x20, 0x40,
-};
-
 static const hda_nid_t stac92hd83xxx_dmic_nids[] = {
 		0x11, 0x20,
 };
@@ -1644,6 +1641,8 @@ static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
 		      "Alienware M17x", STAC_ALIENWARE_M17X),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a,
 		      "Alienware M17x", STAC_ALIENWARE_M17X),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490,
+		      "Alienware M17x", STAC_ALIENWARE_M17X),
 	{} /* terminator */
 };
 
@@ -1659,6 +1658,12 @@ static const unsigned int dell_s14_pin_configs[10] = {
 	0x40f000f0, 0x40f000f0,
 };
 
+static const unsigned int dell_vostro_3500_pin_configs[10] = {
+	0x02a11020, 0x0221101f, 0x400000f0, 0x90170110,
+	0x400000f1, 0x400000f2, 0x400000f3, 0x90a60160,
+	0x400000f4, 0x400000f5,
+};
+
 static const unsigned int hp_dv7_4000_pin_configs[10] = {
 	0x03a12050, 0x0321201f, 0x40f000f0, 0x90170110,
 	0x40f000f0, 0x40f000f0, 0x90170110, 0xd5a30140,
@@ -1675,6 +1680,7 @@ static const unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs,
 	[STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs,
 	[STAC_DELL_S14] = dell_s14_pin_configs,
+	[STAC_DELL_VOSTRO_3500] = dell_vostro_3500_pin_configs,
 	[STAC_92HD83XXX_HP_cNB11_INTQUAD] = hp_cNB11_intquad_pin_configs,
 	[STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs,
 };
@@ -1684,6 +1690,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_REF] = "ref",
 	[STAC_92HD83XXX_PWR_REF] = "mic-ref",
 	[STAC_DELL_S14] = "dell-s14",
+	[STAC_DELL_VOSTRO_3500] = "dell-vostro-3500",
 	[STAC_92HD83XXX_HP] = "hp",
 	[STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad",
 	[STAC_HP_DV7_4000] = "hp-dv7-4000",
@@ -1697,6 +1704,8 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 		      "DFI LanParty", STAC_92HD83XXX_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba,
 		      "unknown Dell", STAC_DELL_S14),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x1028,
+		      "Dell Vostro 3500", STAC_DELL_VOSTRO_3500),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x3600,
 			  "HP", STAC_92HD83XXX_HP),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1656,
@@ -4432,7 +4441,9 @@ static int stac92xx_init(struct hda_codec *codec)
 		int pinctl, def_conf;
 
 		/* power on when no jack detection is available */
-		if (!spec->hp_detect) {
+		/* or when the VREF is used for controlling LED */
+		if (!spec->hp_detect ||
+		    (spec->gpio_led > 8 && spec->gpio_led == nid)) {
 			stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
@@ -4459,8 +4470,12 @@ static int stac92xx_init(struct hda_codec *codec)
 				stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
-		if (enable_pin_detect(codec, nid, STAC_PWR_EVENT))
+		if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) {
 			stac_issue_unsol_event(codec, nid);
+			continue;
+		}
+		/* none of the above, turn the port OFF */
+		stac_toggle_power_map(codec, nid, 0);
 	}
 
 	/* sync mute LED */
@@ -4716,11 +4731,7 @@ static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid,
 	if (idx >= spec->num_pwrs)
 		return;
 
-	/* several codecs have two power down bits */
-	if (spec->pwr_mapping)
-		idx = spec->pwr_mapping[idx];
-	else
-		idx = 1 << idx;
+	idx = 1 << idx;
 
 	val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0) & 0xff;
 	if (enable)
@@ -5046,20 +5057,6 @@ static int stac92xx_pre_resume(struct hda_codec *codec)
 	return 0;
 }
 
-static int stac92xx_post_suspend(struct hda_codec *codec)
-{
-	struct sigmatel_spec *spec = codec->spec;
-	if (spec->gpio_led > 8) {
-		/* with vref-out pin used for mute led control
-		 * codec AFG is prevented from D3 state, but on
-		 * system suspend it can (and should) be used
-		 */
-		snd_hda_codec_read(codec, codec->afg, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	}
-	return 0;
-}
-
 static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 				unsigned int power_state)
 {
@@ -5618,9 +5615,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 		snd_hda_codec_set_pincfg(codec, 0xf, 0x2181205e);
 	}
 
-	/* reset pin power-down; Windows may leave these bits after reboot */
-	snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7EC, 0);
-	snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7ED, 0);
 	codec->no_trigger_sense = 1;
 	codec->spec = spec;
 
@@ -5630,7 +5624,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs;
 	spec->digbeep_nid = 0x21;
 	spec->pwr_nids = stac92hd83xxx_pwr_nids;
-	spec->pwr_mapping = stac92hd83xxx_pwr_mapping;
 	spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
 	spec->multiout.dac_nids = spec->dac_nids;
 	spec->init = stac92hd83xxx_core_init;
@@ -5647,9 +5640,6 @@ again:
 		stac92xx_set_config_regs(codec,
 				stac92hd83xxx_brd_tbl[spec->board_config]);
 
-	if (spec->board_config != STAC_92HD83XXX_PWR_REF)
-		spec->num_pwrs = 0;
-
 	codec->patch_ops = stac92xx_patch_ops;
 
 	if (find_mute_led_gpio(codec, 0))
@@ -5666,8 +5656,6 @@ again:
 		} else {
 			codec->patch_ops.set_power_state =
 					stac92xx_set_power_state;
-			codec->patch_ops.post_suspend =
-					stac92xx_post_suspend;
 		}
 		codec->patch_ops.pre_resume = stac92xx_pre_resume;
 		codec->patch_ops.check_power_status =
@@ -5858,8 +5846,6 @@ again:
 		    (codec->revision_id & 0xf) == 1)
 			spec->stream_delay = 40; /* 40 milliseconds */
 
-		/* no output amps */
-		spec->num_pwrs = 0;
 		/* disable VSW */
 		spec->init = stac92hd71bxx_core_init;
 		unmute_init++;
@@ -5874,8 +5860,6 @@ again:
 		if ((codec->revision_id & 0xf) == 1)
 			spec->stream_delay = 40; /* 40 milliseconds */
 
-		/* no output amps */
-		spec->num_pwrs = 0;
 		/* fallthru */
 	default:
 		spec->init = stac92hd71bxx_core_init;
@@ -5985,8 +5969,6 @@ again:
 		} else {
 			codec->patch_ops.set_power_state =
 					stac92xx_set_power_state;
-			codec->patch_ops.post_suspend =
-					stac92xx_post_suspend;
 		}
 		codec->patch_ops.pre_resume = stac92xx_pre_resume;
 		codec->patch_ops.check_power_status =
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 431c0d417eeb..b5137629f8e9 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -208,6 +208,7 @@ struct via_spec {
 	/* work to check hp jack state */
 	struct hda_codec *codec;
 	struct delayed_work vt1708_hp_work;
+	int hp_work_active;
 	int vt1708_jack_detect;
 	int vt1708_hp_present;
 
@@ -305,27 +306,35 @@ enum {
 static void analog_low_current_mode(struct hda_codec *codec);
 static bool is_aa_path_mute(struct hda_codec *codec);
 
-static void vt1708_start_hp_work(struct via_spec *spec)
+#define hp_detect_with_aa(codec) \
+	(snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1 && \
+	 !is_aa_path_mute(codec))
+
+static void vt1708_stop_hp_work(struct via_spec *spec)
 {
 	if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0)
 		return;
-	snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
-			    !spec->vt1708_jack_detect);
-	if (!delayed_work_pending(&spec->vt1708_hp_work))
-		schedule_delayed_work(&spec->vt1708_hp_work,
-				      msecs_to_jiffies(100));
+	if (spec->hp_work_active) {
+		snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, 1);
+		cancel_delayed_work_sync(&spec->vt1708_hp_work);
+		spec->hp_work_active = 0;
+	}
 }
 
-static void vt1708_stop_hp_work(struct via_spec *spec)
+static void vt1708_update_hp_work(struct via_spec *spec)
 {
 	if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0)
 		return;
-	if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1
-	    && !is_aa_path_mute(spec->codec))
-		return;
-	snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
-			    !spec->vt1708_jack_detect);
-	cancel_delayed_work_sync(&spec->vt1708_hp_work);
+	if (spec->vt1708_jack_detect &&
+	    (spec->active_streams || hp_detect_with_aa(spec->codec))) {
+		if (!spec->hp_work_active) {
+			snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, 0);
+			schedule_delayed_work(&spec->vt1708_hp_work,
+					      msecs_to_jiffies(100));
+			spec->hp_work_active = 1;
+		}
+	} else if (!hp_detect_with_aa(spec->codec))
+		vt1708_stop_hp_work(spec);
 }
 
 static void set_widgets_power_state(struct hda_codec *codec)
@@ -343,12 +352,7 @@ static int analog_input_switch_put(struct snd_kcontrol *kcontrol,
 
 	set_widgets_power_state(codec);
 	analog_low_current_mode(snd_kcontrol_chip(kcontrol));
-	if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) {
-		if (is_aa_path_mute(codec))
-			vt1708_start_hp_work(codec->spec);
-		else
-			vt1708_stop_hp_work(codec->spec);
-	}
+	vt1708_update_hp_work(codec->spec);
 	return change;
 }
 
@@ -1154,7 +1158,7 @@ static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo,
 	spec->cur_dac_stream_tag = stream_tag;
 	spec->cur_dac_format = format;
 	mutex_unlock(&spec->config_mutex);
-	vt1708_start_hp_work(spec);
+	vt1708_update_hp_work(spec);
 	return 0;
 }
 
@@ -1174,7 +1178,7 @@ static int via_playback_hp_pcm_prepare(struct hda_pcm_stream *hinfo,
 	spec->cur_hp_stream_tag = stream_tag;
 	spec->cur_hp_format = format;
 	mutex_unlock(&spec->config_mutex);
-	vt1708_start_hp_work(spec);
+	vt1708_update_hp_work(spec);
 	return 0;
 }
 
@@ -1188,7 +1192,7 @@ static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
 	spec->active_streams &= ~STREAM_MULTI_OUT;
 	mutex_unlock(&spec->config_mutex);
-	vt1708_stop_hp_work(spec);
+	vt1708_update_hp_work(spec);
 	return 0;
 }
 
@@ -1203,7 +1207,7 @@ static int via_playback_hp_pcm_cleanup(struct hda_pcm_stream *hinfo,
 		snd_hda_codec_setup_stream(codec, spec->hp_dac_nid, 0, 0, 0);
 	spec->active_streams &= ~STREAM_INDEP_HP;
 	mutex_unlock(&spec->config_mutex);
-	vt1708_stop_hp_work(spec);
+	vt1708_update_hp_work(spec);
 	return 0;
 }
 
@@ -1645,7 +1649,8 @@ static void via_hp_automute(struct hda_codec *codec)
 	int nums;
 	struct via_spec *spec = codec->spec;
 
-	if (!spec->hp_independent_mode && spec->autocfg.hp_pins[0])
+	if (!spec->hp_independent_mode && spec->autocfg.hp_pins[0] &&
+	    (spec->codec_type != VT1708 || spec->vt1708_jack_detect))
 		present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
 
 	if (spec->smart51_enabled)
@@ -2612,8 +2617,6 @@ static int vt1708_jack_detect_get(struct snd_kcontrol *kcontrol,
 
 	if (spec->codec_type != VT1708)
 		return 0;
-	spec->vt1708_jack_detect =
-		!((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1);
 	ucontrol->value.integer.value[0] = spec->vt1708_jack_detect;
 	return 0;
 }
@@ -2623,18 +2626,22 @@ static int vt1708_jack_detect_put(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct via_spec *spec = codec->spec;
-	int change;
+	int val;
 
 	if (spec->codec_type != VT1708)
 		return 0;
-	spec->vt1708_jack_detect = ucontrol->value.integer.value[0];
-	change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8))
-		== !spec->vt1708_jack_detect;
-	if (spec->vt1708_jack_detect) {
+	val = !!ucontrol->value.integer.value[0];
+	if (spec->vt1708_jack_detect == val)
+		return 0;
+	spec->vt1708_jack_detect = val;
+	if (spec->vt1708_jack_detect &&
+	    snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") != 1) {
 		mute_aa_path(codec, 1);
 		notify_aa_path_ctls(codec);
 	}
-	return change;
+	via_hp_automute(codec);
+	vt1708_update_hp_work(spec);
+	return 1;
 }
 
 static const struct snd_kcontrol_new vt1708_jack_detect_ctl = {
@@ -2771,6 +2778,7 @@ static int via_init(struct hda_codec *codec)
 	via_auto_init_unsol_event(codec);
 
 	via_hp_automute(codec);
+	vt1708_update_hp_work(spec);
 
 	return 0;
 }
@@ -2787,7 +2795,9 @@ static void vt1708_update_hp_jack_state(struct work_struct *work)
 		spec->vt1708_hp_present ^= 1;
 		via_hp_automute(spec->codec);
 	}
-	vt1708_start_hp_work(spec);
+	if (spec->vt1708_jack_detect)
+		schedule_delayed_work(&spec->vt1708_hp_work,
+				      msecs_to_jiffies(100));
 }
 
 static int get_mux_nids(struct hda_codec *codec)
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 29e312597f20..11718b49b2e2 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1077,6 +1077,13 @@ static snd_pcm_uframes_t snd_intel8x0_pcm_pointer(struct snd_pcm_substream *subs
 		}
 		if (civ != igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV))
 			continue;
+
+		/* IO read operation is very expensive inside virtual machine
+		 * as it is emulated. The probability that subsequent PICB read
+		 * will return different result is high enough to loop till
+		 * timeout here.
+		 * Same CIV is strict enough condition to be sure that PICB
+		 * is valid inside VM on emulated card. */
 		if (chip->inside_vm)
 			break;
 		if (ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb))
@@ -2930,6 +2937,45 @@ static unsigned int sis_codec_bits[3] = {
 	ICH_PCR, ICH_SCR, ICH_SIS_TCR
 };
 
+static int __devinit snd_intel8x0_inside_vm(struct pci_dev *pci)
+{
+	int result  = inside_vm;
+	char *msg   = NULL;
+
+	/* check module parameter first (override detection) */
+	if (result >= 0) {
+		msg = result ? "enable (forced) VM" : "disable (forced) VM";
+		goto fini;
+	}
+
+	/* detect KVM and Parallels virtual environments */
+	result = kvm_para_available();
+#ifdef X86_FEATURE_HYPERVISOR
+	result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR);
+#endif
+	if (!result)
+		goto fini;
+
+	/* check for known (emulated) devices */
+	if (pci->subsystem_vendor == 0x1af4 &&
+	    pci->subsystem_device == 0x1100) {
+		/* KVM emulated sound, PCI SSID: 1af4:1100 */
+		msg = "enable KVM";
+	} else if (pci->subsystem_vendor == 0x1ab8) {
+		/* Parallels VM emulated sound, PCI SSID: 1ab8:xxxx */
+		msg = "enable Parallels VM";
+	} else {
+		msg = "disable (unknown or VT-d) VM";
+		result = 0;
+	}
+
+fini:
+	if (msg != NULL)
+		printk(KERN_INFO "intel8x0: %s optimization\n", msg);
+
+	return result;
+}
+
 static int __devinit snd_intel8x0_create(struct snd_card *card,
 					 struct pci_dev *pci,
 					 unsigned long device_type,
@@ -2997,9 +3043,7 @@ static int __devinit snd_intel8x0_create(struct snd_card *card,
 	if (xbox)
 		chip->xbox = 1;
 
-	chip->inside_vm = inside_vm;
-	if (inside_vm)
-		printk(KERN_INFO "intel8x0: enable KVM optimization\n");
+	chip->inside_vm = snd_intel8x0_inside_vm(pci);
 
 	if (pci->vendor == PCI_VENDOR_ID_INTEL &&
 	    pci->device == PCI_DEVICE_ID_INTEL_440MX)
@@ -3243,14 +3287,6 @@ static int __devinit snd_intel8x0_probe(struct pci_dev *pci,
 			buggy_irq = 0;
 	}
 
-	if (inside_vm < 0) {
-		/* detect KVM and Parallels virtual environments */
-		inside_vm = kvm_para_available();
-#if defined(__i386__) || defined(__x86_64__)
-		inside_vm = inside_vm || boot_cpu_has(X86_FEATURE_HYPERVISOR);
-#endif
-	}
-
 	if ((err = snd_intel8x0_create(card, pci, pci_id->driver_data,
 				       &chip)) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c
index 5c8717e29eeb..8c3e7fcefd99 100644
--- a/sound/pci/lx6464es/lx_core.c
+++ b/sound/pci/lx6464es/lx_core.c
@@ -78,10 +78,15 @@ unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port)
 	return ioread32(address);
 }
 
-void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len)
+static void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data,
+			       u32 len)
 {
-	void __iomem *address = lx_dsp_register(chip, port);
-	memcpy_fromio(data, address, len*sizeof(u32));
+	u32 __iomem *address = lx_dsp_register(chip, port);
+	int i;
+
+	/* we cannot use memcpy_fromio */
+	for (i = 0; i != len; ++i)
+		data[i] = ioread32(address + i);
 }
 
 
@@ -91,11 +96,15 @@ void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data)
 	iowrite32(data, address);
 }
 
-void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data,
-			 u32 len)
+static void lx_dsp_reg_writebuf(struct lx6464es *chip, int port,
+				const u32 *data, u32 len)
 {
-	void __iomem *address = lx_dsp_register(chip, port);
-	memcpy_toio(address, data, len*sizeof(u32));
+	u32 __iomem *address = lx_dsp_register(chip, port);
+	int i;
+
+	/* we cannot use memcpy_to */
+	for (i = 0; i != len; ++i)
+		iowrite32(data[i], address + i);
 }
 
 
diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h
index 1dd562980b6c..4d7ff797a646 100644
--- a/sound/pci/lx6464es/lx_core.h
+++ b/sound/pci/lx6464es/lx_core.h
@@ -72,10 +72,7 @@ enum {
 };
 
 unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port);
-void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len);
 void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data);
-void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data,
-			 u32 len);
 
 /* plx register access */
 enum {
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index e760adad9523..19ee2203cbb5 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6518,7 +6518,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
 			hdspm->io_type = AES32;
 			hdspm->card_name = "RME AES32";
 			hdspm->midiPorts = 2;
-		} else if ((hdspm->firmware_rev == 0xd5) ||
+		} else if ((hdspm->firmware_rev == 0xd2) ||
 			((hdspm->firmware_rev >= 0xc8)  &&
 				(hdspm->firmware_rev <= 0xcf))) {
 			hdspm->io_type = MADI;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index a391e622a192..28dfafb56dd1 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -41,6 +41,7 @@ MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}");
 static int index = SNDRV_DEFAULT_IDX1;	/* Index 0-MAX */
 static char *id = SNDRV_DEFAULT_STR1;	/* ID for this card */
 static int enable = 1;
+static int codecs = 1;
 
 module_param(index, int, 0444);
 MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator.");
@@ -48,6 +49,8 @@ module_param(id, charp, 0444);
 MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator.");
 module_param(enable, bool, 0444);
 MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator.");
+module_param(codecs, int, 0444);
+MODULE_PARM_DESC(codecs, "Set bit to indicate that codec number is expected to be present (default 1)");
 
 static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = {
 	{ PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) },
@@ -140,6 +143,9 @@ struct sis7019 {
 	dma_addr_t silence_dma_addr;
 };
 
+/* These values are also used by the module param 'codecs' to indicate
+ * which codecs should be present.
+ */
 #define SIS_PRIMARY_CODEC_PRESENT	0x0001
 #define SIS_SECONDARY_CODEC_PRESENT	0x0002
 #define SIS_TERTIARY_CODEC_PRESENT	0x0004
@@ -1078,6 +1084,7 @@ static int sis_chip_init(struct sis7019 *sis)
 {
 	unsigned long io = sis->ioport;
 	void __iomem *ioaddr = sis->ioaddr;
+	unsigned long timeout;
 	u16 status;
 	int count;
 	int i;
@@ -1104,21 +1111,45 @@ static int sis_chip_init(struct sis7019 *sis)
 	while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count)
 		udelay(1);
 
+	/* Command complete, we can let go of the semaphore now.
+	 */
+	outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA);
+	if (!count)
+		return -EIO;
+
 	/* Now that we've finished the reset, find out what's attached.
+	 * There are some codec/board combinations that take an extremely
+	 * long time to come up. 350+ ms has been observed in the field,
+	 * so we'll give them up to 500ms.
 	 */
-	status = inl(io + SIS_AC97_STATUS);
-	if (status & SIS_AC97_STATUS_CODEC_READY)
-		sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT;
-	if (status & SIS_AC97_STATUS_CODEC2_READY)
-		sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT;
-	if (status & SIS_AC97_STATUS_CODEC3_READY)
-		sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT;
-
-	/* All done, let go of the semaphore, and check for errors
+	sis->codecs_present = 0;
+	timeout = msecs_to_jiffies(500) + jiffies;
+	while (time_before_eq(jiffies, timeout)) {
+		status = inl(io + SIS_AC97_STATUS);
+		if (status & SIS_AC97_STATUS_CODEC_READY)
+			sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT;
+		if (status & SIS_AC97_STATUS_CODEC2_READY)
+			sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT;
+		if (status & SIS_AC97_STATUS_CODEC3_READY)
+			sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT;
+
+		if (sis->codecs_present == codecs)
+			break;
+
+		msleep(1);
+	}
+
+	/* All done, check for errors.
 	 */
-	outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA);
-	if (!sis->codecs_present || !count)
+	if (!sis->codecs_present) {
+		printk(KERN_ERR "sis7019: could not find any codecs\n");
 		return -EIO;
+	}
+
+	if (sis->codecs_present != codecs) {
+		printk(KERN_WARNING "sis7019: missing codecs, found %0x, expected %0x\n",
+		       sis->codecs_present, codecs);
+	}
 
 	/* Let the hardware know that the audio driver is alive,
 	 * and enable PCM slots on the AC-link for L/R playback (3 & 4) and
@@ -1390,6 +1421,17 @@ static int __devinit snd_sis7019_probe(struct pci_dev *pci,
 	if (!enable)
 		goto error_out;
 
+	/* The user can specify which codecs should be present so that we
+	 * can wait for them to show up if they are slow to recover from
+	 * the AC97 cold reset. We default to a single codec, the primary.
+	 *
+	 * We assume that SIS_PRIMARY_*_PRESENT matches bits 0-2.
+	 */
+	codecs &= SIS_PRIMARY_CODEC_PRESENT | SIS_SECONDARY_CODEC_PRESENT |
+		  SIS_TERTIARY_CODEC_PRESENT;
+	if (!codecs)
+		codecs = SIS_PRIMARY_CODEC_PRESENT;
+
 	rc = snd_card_create(index, id, THIS_MODULE, sizeof(*sis), &card);
 	if (rc < 0)
 		goto error_out;
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index bee3c94f58b0..d1fcc816ce97 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -1,6 +1,6 @@
 config SND_ATMEL_SOC
 	tristate "SoC Audio for the Atmel System-on-Chip"
-	depends on ARCH_AT91 || AVR32
+	depends on ARCH_AT91
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the ATMEL SSC interface. You will also need
@@ -24,25 +24,6 @@ config SND_AT91_SOC_SAM9G20_WM8731
 	  Say Y if you want to add support for SoC audio on WM8731-based
 	  AT91sam9g20 evaluation board.
 
-config SND_AT32_SOC_PLAYPAQ
-        tristate "SoC Audio support for PlayPaq with WM8510"
-        depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS
-        select SND_ATMEL_SOC_SSC
-        select SND_SOC_WM8510
-        help
-          Say Y or M here if you want to add support for SoC audio
-          on the LRS PlayPaq.
-
-config SND_AT32_SOC_PLAYPAQ_SLAVE
-        bool "Run CODEC on PlayPaq in slave mode"
-        depends on SND_AT32_SOC_PLAYPAQ
-        default n
-        help
-          Say Y if you want to run with the AT32 SSC generating the BCLK
-          and FRAME signals on the PlayPaq.  Unless you want to play
-          with the AT32 as the SSC master, you probably want to say N here,
-          as this will give you better sound quality.
-
 config SND_AT91_SOC_AFEB9260
 	tristate "SoC Audio support for AFEB9260 board"
 	depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index e7ea56bd5f82..a5c0bf19da78 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -8,9 +8,5 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
 # AT91 Machine Support
 snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
 
-# AT32 Machine Support
-snd-soc-playpaq-objs := playpaq_wm8510.o
-
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
-obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
 obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
deleted file mode 100644
index 73ae99ad4578..000000000000
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ /dev/null
@@ -1,473 +0,0 @@
-/* sound/soc/at32/playpaq_wm8510.c
- * ASoC machine driver for PlayPaq using WM8510 codec
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
- * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c
- *
- * NOTE: If you don't have the AT32 enhanced portmux configured (which
- * isn't currently in the mainline or Atmel patched kernel), you will
- * need to set the MCLK pin (PA30) to peripheral A in your board initialization
- * code.  Something like:
- *	at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0);
- *
- */
-
-/* #define DEBUG */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/kernel.h>
-#include <linux/errno.h>
-#include <linux/clk.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <mach/at32ap700x.h>
-#include <mach/portmux.h>
-
-#include "../codecs/wm8510.h"
-#include "atmel-pcm.h"
-#include "atmel_ssc_dai.h"
-
-
-/*-------------------------------------------------------------------------*\
- * constants
-\*-------------------------------------------------------------------------*/
-#define MCLK_PIN		GPIO_PIN_PA(30)
-#define MCLK_PERIPH		GPIO_PERIPH_A
-
-
-/*-------------------------------------------------------------------------*\
- * data types
-\*-------------------------------------------------------------------------*/
-/* SSC clocking data */
-struct ssc_clock_data {
-	/* CMR div */
-	unsigned int cmr_div;
-
-	/* Frame period (as needed by xCMR.PERIOD) */
-	unsigned int period;
-
-	/* The SSC clock rate these settings where calculated for */
-	unsigned long ssc_rate;
-};
-
-
-/*-------------------------------------------------------------------------*\
- * module data
-\*-------------------------------------------------------------------------*/
-static struct clk *_gclk0;
-static struct clk *_pll0;
-
-#define CODEC_CLK (_gclk0)
-
-
-/*-------------------------------------------------------------------------*\
- * Sound SOC operations
-\*-------------------------------------------------------------------------*/
-#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock(
-	struct snd_pcm_hw_params *params,
-	struct snd_soc_dai *cpu_dai)
-{
-	struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai);
-	struct ssc_device *ssc = ssc_p->ssc;
-	struct ssc_clock_data cd;
-	unsigned int rate, width_bits, channels;
-	unsigned int bitrate, ssc_div;
-	unsigned actual_rate;
-
-
-	/*
-	 * Figure out required bitrate
-	 */
-	rate = params_rate(params);
-	channels = params_channels(params);
-	width_bits = snd_pcm_format_physical_width(params_format(params));
-	bitrate = rate * width_bits * channels;
-
-
-	/*
-	 * Figure out required SSC divider and period for required bitrate
-	 */
-	cd.ssc_rate = clk_get_rate(ssc->clk);
-	ssc_div = cd.ssc_rate / bitrate;
-	cd.cmr_div = ssc_div / 2;
-	if (ssc_div & 1) {
-		/* round cmr_div up */
-		cd.cmr_div++;
-	}
-	cd.period = width_bits - 1;
-
-
-	/*
-	 * Find actual rate, compare to requested rate
-	 */
-	actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
-	pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n",
-		 rate, actual_rate);
-
-
-	return cd;
-}
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
-
-static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream,
-				    struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai);
-	struct ssc_device *ssc = ssc_p->ssc;
-	unsigned int pll_out = 0, bclk = 0, mclk_div = 0;
-	int ret;
-
-
-	/* Due to difficulties with getting the correct clocks from the AT32's
-	 * PLL0, we're going to let the CODEC be in charge of all the clocks
-	 */
-#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-	const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
-#else
-	struct ssc_clock_data cd;
-	const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBS_CFS);
-#endif
-
-	if (ssc == NULL) {
-		pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n");
-		return -EINVAL;
-	}
-
-
-	/*
-	 * Figure out PLL and BCLK dividers for WM8510
-	 */
-	switch (params_rate(params)) {
-	case 48000:
-		pll_out = 24576000;
-		mclk_div = WM8510_MCLKDIV_2;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 44100:
-		pll_out = 22579200;
-		mclk_div = WM8510_MCLKDIV_2;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 22050:
-		pll_out = 22579200;
-		mclk_div = WM8510_MCLKDIV_4;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 16000:
-		pll_out = 24576000;
-		mclk_div = WM8510_MCLKDIV_6;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 11025:
-		pll_out = 22579200;
-		mclk_div = WM8510_MCLKDIV_8;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 8000:
-		pll_out = 24576000;
-		mclk_div = WM8510_MCLKDIV_12;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	default:
-		pr_warning("playpaq_wm8510: Unsupported sample rate %d\n",
-			   params_rate(params));
-		return -EINVAL;
-	}
-
-
-	/*
-	 * set CPU and CODEC DAI configuration
-	 */
-	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: "
-			   "Failed to set CODEC DAI format (%d)\n",
-			   ret);
-		return ret;
-	}
-	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: "
-			   "Failed to set CPU DAI format (%d)\n",
-			   ret);
-		return ret;
-	}
-
-
-	/*
-	 * Set CPU clock configuration
-	 */
-#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-	cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai);
-	pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n",
-		 cd.cmr_div, cd.period);
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n",
-			   ret);
-		return ret;
-	}
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD,
-					  cd.period);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: "
-			   "Failed to set CPU transmit period (%d)\n",
-			   ret);
-		return ret;
-	}
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
-	/*
-	 * Set CODEC clock configuration
-	 */
-	pr_debug("playpaq_wm8510: "
-		 "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n",
-		 clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div);
-
-
-#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk);
-	if (ret < 0) {
-		pr_warning
-		    ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n",
-		     ret);
-		return ret;
-	}
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
-	ret = snd_soc_dai_set_pll(codec_dai, 0, 0,
-					 clk_get_rate(CODEC_CLK), pll_out);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n",
-			   ret);
-		return ret;
-	}
-
-
-	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n",
-			   ret);
-		return ret;
-	}
-
-
-	return 0;
-}
-
-
-
-static struct snd_soc_ops playpaq_wm8510_ops = {
-	.hw_params = playpaq_wm8510_hw_params,
-};
-
-
-
-static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = {
-	SND_SOC_DAPM_MIC("Int Mic", NULL),
-	SND_SOC_DAPM_SPK("Ext Spk", NULL),
-};
-
-
-
-static const struct snd_soc_dapm_route intercon[] = {
-	/* speaker connected to SPKOUT */
-	{"Ext Spk", NULL, "SPKOUTP"},
-	{"Ext Spk", NULL, "SPKOUTN"},
-
-	{"Mic Bias", NULL, "Int Mic"},
-	{"MICN", NULL, "Mic Bias"},
-	{"MICP", NULL, "Mic Bias"},
-};
-
-
-
-static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int i;
-
-	/*
-	 * Add DAPM widgets
-	 */
-	for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++)
-		snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]);
-
-
-
-	/*
-	 * Setup audio path interconnects
-	 */
-	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-
-
-
-	/* always connected pins */
-	snd_soc_dapm_enable_pin(dapm, "Int Mic");
-	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
-
-
-
-	/* Make CSB show PLL rate */
-	snd_soc_dai_set_clkdiv(rtd->codec_dai, WM8510_OPCLKDIV,
-				       WM8510_OPCLKDIV_1 | 4);
-
-	return 0;
-}
-
-
-
-static struct snd_soc_dai_link playpaq_wm8510_dai = {
-	.name = "WM8510",
-	.stream_name = "WM8510 PCM",
-	.cpu_dai_name= "atmel-ssc-dai.0",
-	.platform_name = "atmel-pcm-audio",
-	.codec_name = "wm8510-codec.0-0x1a",
-	.codec_dai_name = "wm8510-hifi",
-	.init = playpaq_wm8510_init,
-	.ops = &playpaq_wm8510_ops,
-};
-
-
-
-static struct snd_soc_card snd_soc_playpaq = {
-	.name = "LRS_PlayPaq_WM8510",
-	.dai_link = &playpaq_wm8510_dai,
-	.num_links = 1,
-};
-
-static struct platform_device *playpaq_snd_device;
-
-
-static int __init playpaq_asoc_init(void)
-{
-	int ret = 0;
-
-	/*
-	 * Configure MCLK for WM8510
-	 */
-	_gclk0 = clk_get(NULL, "gclk0");
-	if (IS_ERR(_gclk0)) {
-		_gclk0 = NULL;
-		ret = PTR_ERR(_gclk0);
-		goto err_gclk0;
-	}
-	_pll0 = clk_get(NULL, "pll0");
-	if (IS_ERR(_pll0)) {
-		_pll0 = NULL;
-		ret = PTR_ERR(_pll0);
-		goto err_pll0;
-	}
-	ret = clk_set_parent(_gclk0, _pll0);
-	if (ret) {
-		pr_warning("snd-soc-playpaq: "
-			   "Failed to set PLL0 as parent for DAC clock\n");
-		goto err_set_clk;
-	}
-	clk_set_rate(CODEC_CLK, 12000000);
-	clk_enable(CODEC_CLK);
-
-#if defined CONFIG_AT32_ENHANCED_PORTMUX
-	at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0);
-#endif
-
-
-	/*
-	 * Create and register platform device
-	 */
-	playpaq_snd_device = platform_device_alloc("soc-audio", 0);
-	if (playpaq_snd_device == NULL) {
-		ret = -ENOMEM;
-		goto err_device_alloc;
-	}
-
-	platform_set_drvdata(playpaq_snd_device, &snd_soc_playpaq);
-
-	ret = platform_device_add(playpaq_snd_device);
-	if (ret) {
-		pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n",
-			   ret);
-		goto err_device_add;
-	}
-
-	return 0;
-
-
-err_device_add:
-	if (playpaq_snd_device != NULL) {
-		platform_device_put(playpaq_snd_device);
-		playpaq_snd_device = NULL;
-	}
-err_device_alloc:
-err_set_clk:
-	if (_pll0 != NULL) {
-		clk_put(_pll0);
-		_pll0 = NULL;
-	}
-err_pll0:
-	if (_gclk0 != NULL) {
-		clk_put(_gclk0);
-		_gclk0 = NULL;
-	}
-	return ret;
-}
-
-
-static void __exit playpaq_asoc_exit(void)
-{
-	if (_gclk0 != NULL) {
-		clk_put(_gclk0);
-		_gclk0 = NULL;
-	}
-	if (_pll0 != NULL) {
-		clk_put(_pll0);
-		_pll0 = NULL;
-	}
-
-#if defined CONFIG_AT32_ENHANCED_PORTMUX
-	at32_free_pin(MCLK_PIN);
-#endif
-
-	platform_device_unregister(playpaq_snd_device);
-	playpaq_snd_device = NULL;
-}
-
-module_init(playpaq_asoc_init);
-module_exit(playpaq_asoc_exit);
-
-MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
-MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 444747f0db26..dd7be0dbbc58 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -34,7 +34,7 @@
 
 #define AD1836_ADC_CTRL2               13
 #define AD1836_ADC_WORD_LEN_MASK       0x30
-#define AD1836_ADC_WORD_OFFSET         5
+#define AD1836_ADC_WORD_OFFSET         4
 #define AD1836_ADC_SERFMT_MASK         (7 << 6)
 #define AD1836_ADC_SERFMT_PCK256       (0x4 << 6)
 #define AD1836_ADC_SERFMT_PCK128       (0x5 << 6)
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 1ccf8dd47576..45c63028b40d 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -245,7 +245,7 @@ static const char *adau1373_bass_hpf_cutoff_text[] = {
 };
 
 static const unsigned int adau1373_bass_tlv[] = {
-	TLV_DB_RANGE_HEAD(4),
+	TLV_DB_RANGE_HEAD(3),
 	0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1),
 	3, 4, TLV_DB_SCALE_ITEM(950, 250, 0),
 	5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0),
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index f1f237ecec2a..73f46eb459f1 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -601,7 +601,6 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
 static int cs4270_soc_resume(struct snd_soc_codec *codec)
 {
 	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
-	struct i2c_client *i2c_client = to_i2c_client(codec->dev);
 	int reg;
 
 	regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
@@ -612,14 +611,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec)
 	ndelay(500);
 
 	/* first restore the entire register cache ... */
-	for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
-		u8 val = snd_soc_read(codec, reg);
-
-		if (i2c_smbus_write_byte_data(i2c_client, reg, val)) {
-			dev_err(codec->dev, "i2c write failed\n");
-			return -EIO;
-		}
-	}
+	snd_soc_cache_sync(codec);
 
 	/* ... then disable the power-down bits */
 	reg = snd_soc_read(codec, CS4270_PWRCTL);
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 23d1bd5dadda..69fde1506fe1 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -434,7 +434,8 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
 {
 	int ret;
 	/* Set power-down bit */
-	ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN);
+	ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN,
+				  CS4271_MODE2_PDN);
 	if (ret < 0)
 		return ret;
 	return 0;
@@ -501,8 +502,9 @@ static int cs4271_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	ret = snd_soc_update_bits(codec, CS4271_MODE2, 0,
-		CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
+	ret = snd_soc_update_bits(codec, CS4271_MODE2,
+				  CS4271_MODE2_PDN | CS4271_MODE2_CPEN,
+				  CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
 	if (ret < 0)
 		return ret;
 	ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0);
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 8c3c8205d19e..1ee66361f61b 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -555,7 +555,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec)
 
 static struct snd_soc_codec_driver soc_codec_device_cs42l51 = {
 	.probe =	cs42l51_probe,
-	.reg_cache_size = CS42L51_NUMREGS,
+	.reg_cache_size = CS42L51_NUMREGS + 1,
 	.reg_word_size = sizeof(u8),
 };
 
diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c
index 9e7e964a5fa3..dcf6f2a1600a 100644
--- a/sound/soc/codecs/max9877.c
+++ b/sound/soc/codecs/max9877.c
@@ -106,13 +106,13 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol,
 	unsigned int mask = mc->max;
 	unsigned int val = (ucontrol->value.integer.value[0] & mask);
 	unsigned int val2 = (ucontrol->value.integer.value[1] & mask);
-	unsigned int change = 1;
+	unsigned int change = 0;
 
-	if (((max9877_regs[reg] >> shift) & mask) == val)
-		change = 0;
+	if (((max9877_regs[reg] >> shift) & mask) != val)
+		change = 1;
 
-	if (((max9877_regs[reg2] >> shift) & mask) == val2)
-		change = 0;
+	if (((max9877_regs[reg2] >> shift) & mask) != val2)
+		change = 1;
 
 	if (change) {
 		max9877_regs[reg] &= ~(mask << shift);
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 27a078cbb6eb..4646e808b90a 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -177,7 +177,7 @@ static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -95625, 375, 0);
 static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
 /* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */
 static unsigned int mic_bst_tlv[] = {
-	TLV_DB_RANGE_HEAD(6),
+	TLV_DB_RANGE_HEAD(7),
 	0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
 	1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
 	2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0),
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d15695d1c273..bbcf921166f7 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -365,7 +365,7 @@ static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0);
 
 /* tlv for mic gain, 0db 20db 30db 40db */
 static const unsigned int mic_gain_tlv[] = {
-	TLV_DB_RANGE_HEAD(4),
+	TLV_DB_RANGE_HEAD(2),
 	0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
 	1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0),
 };
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index bb82408ab8e1..d2f37152f940 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -76,6 +76,8 @@ struct sta32x_priv {
 
 	unsigned int mclk;
 	unsigned int format;
+
+	u32 coef_shadow[STA32X_COEF_COUNT];
 };
 
 static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1);
@@ -227,6 +229,7 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
 	int numcoef = kcontrol->private_value >> 16;
 	int index = kcontrol->private_value & 0xffff;
 	unsigned int cfud;
@@ -239,6 +242,11 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
 	snd_soc_write(codec, STA32X_CFUD, cfud);
 
 	snd_soc_write(codec, STA32X_CFADDR2, index);
+	for (i = 0; i < numcoef && (index + i < STA32X_COEF_COUNT); i++)
+		sta32x->coef_shadow[index + i] =
+			  (ucontrol->value.bytes.data[3 * i] << 16)
+			| (ucontrol->value.bytes.data[3 * i + 1] << 8)
+			| (ucontrol->value.bytes.data[3 * i + 2]);
 	for (i = 0; i < 3 * numcoef; i++)
 		snd_soc_write(codec, STA32X_B1CF1 + i,
 			      ucontrol->value.bytes.data[i]);
@@ -252,6 +260,48 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
+int sta32x_sync_coef_shadow(struct snd_soc_codec *codec)
+{
+	struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+	unsigned int cfud;
+	int i;
+
+	/* preserve reserved bits in STA32X_CFUD */
+	cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0;
+
+	for (i = 0; i < STA32X_COEF_COUNT; i++) {
+		snd_soc_write(codec, STA32X_CFADDR2, i);
+		snd_soc_write(codec, STA32X_B1CF1,
+			      (sta32x->coef_shadow[i] >> 16) & 0xff);
+		snd_soc_write(codec, STA32X_B1CF2,
+			      (sta32x->coef_shadow[i] >> 8) & 0xff);
+		snd_soc_write(codec, STA32X_B1CF3,
+			      (sta32x->coef_shadow[i]) & 0xff);
+		/* chip documentation does not say if the bits are
+		 * self-clearing, so do it explicitly */
+		snd_soc_write(codec, STA32X_CFUD, cfud);
+		snd_soc_write(codec, STA32X_CFUD, cfud | 0x01);
+	}
+	return 0;
+}
+
+int sta32x_cache_sync(struct snd_soc_codec *codec)
+{
+	unsigned int mute;
+	int rc;
+
+	if (!codec->cache_sync)
+		return 0;
+
+	/* mute during register sync */
+	mute = snd_soc_read(codec, STA32X_MMUTE);
+	snd_soc_write(codec, STA32X_MMUTE, mute | STA32X_MMUTE_MMUTE);
+	sta32x_sync_coef_shadow(codec);
+	rc = snd_soc_cache_sync(codec);
+	snd_soc_write(codec, STA32X_MMUTE, mute);
+	return rc;
+}
+
 #define SINGLE_COEF(xname, index) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = sta32x_coefficient_info, \
@@ -661,7 +711,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec,
 				return ret;
 			}
 
-			snd_soc_cache_sync(codec);
+			sta32x_cache_sync(codec);
 		}
 
 		/* Power up to mute */
@@ -790,6 +840,17 @@ static int sta32x_probe(struct snd_soc_codec *codec)
 			    STA32X_CxCFG_OM_MASK,
 			    2 << STA32X_CxCFG_OM_SHIFT);
 
+	/* initialize coefficient shadow RAM with reset values */
+	for (i = 4; i <= 49; i += 5)
+		sta32x->coef_shadow[i] = 0x400000;
+	for (i = 50; i <= 54; i++)
+		sta32x->coef_shadow[i] = 0x7fffff;
+	sta32x->coef_shadow[55] = 0x5a9df7;
+	sta32x->coef_shadow[56] = 0x7fffff;
+	sta32x->coef_shadow[59] = 0x7fffff;
+	sta32x->coef_shadow[60] = 0x400000;
+	sta32x->coef_shadow[61] = 0x400000;
+
 	sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	/* Bias level configuration will have done an extra enable */
 	regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h
index b97ee5a75667..d8e32a6262ee 100644
--- a/sound/soc/codecs/sta32x.h
+++ b/sound/soc/codecs/sta32x.h
@@ -19,6 +19,7 @@
 /* STA326 register addresses */
 
 #define STA32X_REGISTER_COUNT	0x2d
+#define STA32X_COEF_COUNT 62
 
 #define STA32X_CONFA	0x00
 #define STA32X_CONFB    0x01
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7e5ec03f6f8d..a7c9ae17fc7e 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -453,6 +453,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, WM8731_PWR, 0xffff);
 		regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
 				       wm8731->supplies);
+		codec->cache_sync = 1;
 		break;
 	}
 	codec->dapm.bias_level = level;
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index a9504710bb69..3a629d0d690e 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -190,6 +190,9 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
 	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
 	u16 ioctl;
 
+	if (wm8753->dai_func == ucontrol->value.integer.value[0])
+		return 0;
+
 	if (codec->active)
 		return -EBUSY;
 
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 91d3c6dbeba3..53edd9a8c758 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1973,7 +1973,7 @@ static int wm8962_reset(struct snd_soc_codec *codec)
 static const DECLARE_TLV_DB_SCALE(inpga_tlv, -2325, 75, 0);
 static const DECLARE_TLV_DB_SCALE(mixin_tlv, -1500, 300, 0);
 static const unsigned int mixinpga_tlv[] = {
-	TLV_DB_RANGE_HEAD(7),
+	TLV_DB_RANGE_HEAD(5),
 	0, 1, TLV_DB_SCALE_ITEM(0, 600, 0),
 	2, 2, TLV_DB_SCALE_ITEM(1300, 1300, 0),
 	3, 4, TLV_DB_SCALE_ITEM(1800, 200, 0),
@@ -1988,7 +1988,7 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
 static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
 static const DECLARE_TLV_DB_SCALE(hp_tlv, -700, 100, 0);
 static const unsigned int classd_tlv[] = {
-	TLV_DB_RANGE_HEAD(7),
+	TLV_DB_RANGE_HEAD(2),
 	0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
 	7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
 };
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index eec8e1435116..d1a142f48b09 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -512,7 +512,7 @@ static const DECLARE_TLV_DB_SCALE(drc_comp_threash, -4500, 75, 0);
 static const DECLARE_TLV_DB_SCALE(drc_comp_amp, -2250, 75, 0);
 static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0);
 static const unsigned int drc_max_tlv[] = {
-	TLV_DB_RANGE_HEAD(4),
+	TLV_DB_RANGE_HEAD(2),
 	0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0),
 	3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0),
 };
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 6b73efd26991..6c2988549003 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -56,7 +56,7 @@ static int wm8994_retune_mobile_base[] = {
 static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg)
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
-	struct wm8994 *control = wm8994->control_data;
+	struct wm8994 *control = codec->control_data;
 
 	switch (reg) {
 	case WM8994_GPIO_1:
@@ -2357,6 +2357,11 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
 	bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT;
 
 	lrclk = bclk_rate / params_rate(params);
+	if (!lrclk) {
+		dev_err(dai->dev, "Unable to generate LRCLK from %dHz BCLK\n",
+			bclk_rate);
+		return -EINVAL;
+	}
 	dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n",
 		lrclk, bclk_rate / lrclk);
 
@@ -3030,19 +3035,34 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
 {
 	struct wm8994_priv *wm8994 = data;
 	struct snd_soc_codec *codec = wm8994->codec;
-	int reg;
+	int reg, count;
 
-	reg = snd_soc_read(codec, WM8958_MIC_DETECT_3);
-	if (reg < 0) {
-		dev_err(codec->dev, "Failed to read mic detect status: %d\n",
-			reg);
-		return IRQ_NONE;
-	}
+	/* We may occasionally read a detection without an impedence
+	 * range being provided - if that happens loop again.
+	 */
+	count = 10;
+	do {
+		reg = snd_soc_read(codec, WM8958_MIC_DETECT_3);
+		if (reg < 0) {
+			dev_err(codec->dev,
+				"Failed to read mic detect status: %d\n",
+				reg);
+			return IRQ_NONE;
+		}
 
-	if (!(reg & WM8958_MICD_VALID)) {
-		dev_dbg(codec->dev, "Mic detect data not valid\n");
-		goto out;
-	}
+		if (!(reg & WM8958_MICD_VALID)) {
+			dev_dbg(codec->dev, "Mic detect data not valid\n");
+			goto out;
+		}
+
+		if (!(reg & WM8958_MICD_STS) || (reg & WM8958_MICD_LVL_MASK))
+			break;
+
+		msleep(1);
+	} while (count--);
+
+	if (count == 0)
+		dev_warn(codec->dev, "No impedence range reported for jack\n");
 
 #ifndef CONFIG_SND_SOC_WM8994_MODULE
 	trace_snd_soc_jack_irq(dev_name(codec->dev));
@@ -3163,6 +3183,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		switch (wm8994->revision) {
 		case 0:
 		case 1:
+		case 2:
+		case 3:
 			wm8994->hubs.dcs_codes_l = -9;
 			wm8994->hubs.dcs_codes_r = -5;
 			break;
@@ -3180,9 +3202,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 
 	wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR,
 			   wm8994_fifo_error, "FIFO error", codec);
-	wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_WARN,
+	wm8994_request_irq(codec->control_data, WM8994_IRQ_TEMP_WARN,
 			   wm8994_temp_warn, "Thermal warning", codec);
-	wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_SHUT,
+	wm8994_request_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT,
 			   wm8994_temp_shut, "Thermal shutdown", codec);
 
 	ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE,
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 3cd35a02c28c..4a398c3bfe84 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -807,7 +807,6 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
 			mdelay(100);
 
 			/* Normal bias enable & soft start off */
-			reg |= WM9081_BIAS_ENA;
 			reg &= ~WM9081_VMID_RAMP;
 			snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
 
@@ -818,7 +817,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
 		}
 
 		/* VMID 2*240k */
-		reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
+		reg = snd_soc_read(codec, WM9081_VMID_CONTROL);
 		reg &= ~WM9081_VMID_SEL_MASK;
 		reg |= 0x04;
 		snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
@@ -830,14 +829,15 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_OFF:
-		/* Startup bias source */
+		/* Startup bias source and disable bias */
 		reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
 		reg |= WM9081_BIAS_SRC;
+		reg &= ~WM9081_BIAS_ENA;
 		snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg);
 
-		/* Disable VMID and biases with soft ramping */
+		/* Disable VMID with soft ramping */
 		reg = snd_soc_read(codec, WM9081_VMID_CONTROL);
-		reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA);
+		reg &= ~WM9081_VMID_SEL_MASK;
 		reg |= WM9081_VMID_RAMP;
 		snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
 
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 2b5252c9e377..f94c06057c64 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -177,19 +177,19 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec)
 }
 
 static const unsigned int in_tlv[] = {
-	TLV_DB_RANGE_HEAD(6),
+	TLV_DB_RANGE_HEAD(3),
 	0, 0, TLV_DB_SCALE_ITEM(-600, 0, 0),
 	1, 3, TLV_DB_SCALE_ITEM(-350, 350, 0),
 	4, 6, TLV_DB_SCALE_ITEM(600, 600, 0),
 };
 static const unsigned int mix_tlv[] = {
-	TLV_DB_RANGE_HEAD(4),
+	TLV_DB_RANGE_HEAD(2),
 	0, 2, TLV_DB_SCALE_ITEM(-1200, 300, 0),
 	3, 3, TLV_DB_SCALE_ITEM(0, 0, 0),
 };
 static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
 static const unsigned int spkboost_tlv[] = {
-	TLV_DB_RANGE_HEAD(7),
+	TLV_DB_RANGE_HEAD(2),
 	0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
 	7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
 };
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 84f33d4ea2cd..48e61e912400 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -40,7 +40,7 @@ static const DECLARE_TLV_DB_SCALE(outmix_tlv, -2100, 300, 0);
 static const DECLARE_TLV_DB_SCALE(spkmixout_tlv, -1800, 600, 1);
 static const DECLARE_TLV_DB_SCALE(outpga_tlv, -5700, 100, 0);
 static const unsigned int spkboost_tlv[] = {
-	TLV_DB_RANGE_HEAD(7),
+	TLV_DB_RANGE_HEAD(2),
 	0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
 	7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
 };
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 0268cf989736..83c4bd5b2dd7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -694,6 +694,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 
 	/* Initialize the the device_attribute structure */
 	dev_attr = &ssi_private->dev_attr;
+	sysfs_attr_init(&dev_attr->attr);
 	dev_attr->attr.name = "statistics";
 	dev_attr->attr.mode = S_IRUGO;
 	dev_attr->show = fsl_sysfs_ssi_show;
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 31af405bda84..ae49f1c78c6d 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -392,7 +392,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 	}
 
 	if (strcasecmp(sprop, "i2s-slave") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
 
@@ -409,31 +410,38 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 		}
 		machine_data->clk_frequency = be32_to_cpup(iprop);
 	} else if (strcasecmp(sprop, "i2s-master") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
 	} else if (strcasecmp(sprop, "lj-slave") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
 	} else if (strcasecmp(sprop, "lj-master") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
 	} else if (strcasecmp(sprop, "rj-slave") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
 	} else if (strcasecmp(sprop, "rj-master") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
 	} else if (strcasecmp(sprop, "ac97-slave") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
 	} else if (strcasecmp(sprop, "ac97-master") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
 	} else {
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index f75e43997d5b..ad9ac42522e2 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -9,6 +9,7 @@
 
 #include "../codecs/wm8994.h"
 #include <sound/pcm_params.h>
+#include <linux/module.h>
 
  /*
   * Default CFG switch settings to use this driver:
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 85bf541a771d..4b8e35410eb1 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -191,7 +191,7 @@ static int speyside_late_probe(struct snd_soc_card *card)
 	snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic");
 	snd_soc_dapm_ignore_suspend(&card->dapm, "Main AMIC");
 	snd_soc_dapm_ignore_suspend(&card->dapm, "Main DMIC");
-	snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Main Speaker");
 	snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Output");
 	snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Input");
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a5d3685a5d38..a25fa63ce9a2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -709,6 +709,12 @@ int snd_soc_resume(struct device *dev)
 	struct snd_soc_card *card = dev_get_drvdata(dev);
 	int i, ac97_control = 0;
 
+	/* If the initialization of this soc device failed, there is no codec
+	 * associated with it. Just bail out in this case.
+	 */
+	if (list_empty(&card->codec_dev_list))
+		return 0;
+
 	/* AC97 devices might have other drivers hanging off them so
 	 * need to resume immediately.  Other drivers don't have that
 	 * problem and may take a substantial amount of time to resume
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 60f65ace7474..ab23869c01bb 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -765,10 +765,61 @@ static void usb_mixer_elem_free(struct snd_kcontrol *kctl)
  * interface to ALSA control for feature/mixer units
  */
 
+/* volume control quirks */
+static void volume_control_quirks(struct usb_mixer_elem_info *cval,
+				  struct snd_kcontrol *kctl)
+{
+	switch (cval->mixer->chip->usb_id) {
+	case USB_ID(0x0471, 0x0101):
+	case USB_ID(0x0471, 0x0104):
+	case USB_ID(0x0471, 0x0105):
+	case USB_ID(0x0672, 0x1041):
+	/* quirk for UDA1321/N101.
+	 * note that detection between firmware 2.1.1.7 (N101)
+	 * and later 2.1.1.21 is not very clear from datasheets.
+	 * I hope that the min value is -15360 for newer firmware --jk
+	 */
+		if (!strcmp(kctl->id.name, "PCM Playback Volume") &&
+		    cval->min == -15616) {
+			snd_printk(KERN_INFO
+				 "set volume quirk for UDA1321/N101 chip\n");
+			cval->max = -256;
+		}
+		break;
+
+	case USB_ID(0x046d, 0x09a4):
+		if (!strcmp(kctl->id.name, "Mic Capture Volume")) {
+			snd_printk(KERN_INFO
+				"set volume quirk for QuickCam E3500\n");
+			cval->min = 6080;
+			cval->max = 8768;
+			cval->res = 192;
+		}
+		break;
+
+	case USB_ID(0x046d, 0x0808):
+	case USB_ID(0x046d, 0x0809):
+	case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */
+	case USB_ID(0x046d, 0x0991):
+	/* Most audio usb devices lie about volume resolution.
+	 * Most Logitech webcams have res = 384.
+	 * Proboly there is some logitech magic behind this number --fishor
+	 */
+		if (!strcmp(kctl->id.name, "Mic Capture Volume")) {
+			snd_printk(KERN_INFO
+				"set resolution quirk: cval->res = 384\n");
+			cval->res = 384;
+		}
+		break;
+
+	}
+}
+
 /*
  * retrieve the minimum and maximum values for the specified control
  */
-static int get_min_max(struct usb_mixer_elem_info *cval, int default_min)
+static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval,
+				   int default_min, struct snd_kcontrol *kctl)
 {
 	/* for failsafe */
 	cval->min = default_min;
@@ -844,6 +895,9 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min)
 		cval->initialized = 1;
 	}
 
+	if (kctl)
+		volume_control_quirks(cval, kctl);
+
 	/* USB descriptions contain the dB scale in 1/256 dB unit
 	 * while ALSA TLV contains in 1/100 dB unit
 	 */
@@ -864,6 +918,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min)
 	return 0;
 }
 
+#define get_min_max(cval, def)	get_min_max_with_quirks(cval, def, NULL)
 
 /* get a feature/mixer unit info */
 static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
@@ -882,7 +937,7 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_
 		uinfo->value.integer.max = 1;
 	} else {
 		if (!cval->initialized) {
-			get_min_max(cval, 0);
+			get_min_max_with_quirks(cval, 0, kcontrol);
 			if (cval->initialized && cval->dBmin >= cval->dBmax) {
 				kcontrol->vd[0].access &= 
 					~(SNDRV_CTL_ELEM_ACCESS_TLV_READ |
@@ -1045,9 +1100,6 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		cval->ch_readonly = readonly_mask;
 	}
 
-	/* get min/max values */
-	get_min_max(cval, 0);
-
 	/* if all channels in the mask are marked read-only, make the control
 	 * read-only. set_cur_mix_value() will check the mask again and won't
 	 * issue write commands to read-only channels. */
@@ -1069,6 +1121,9 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		len = snd_usb_copy_string_desc(state, nameid,
 				kctl->id.name, sizeof(kctl->id.name));
 
+	/* get min/max values */
+	get_min_max_with_quirks(cval, 0, kctl);
+
 	switch (control) {
 	case UAC_FU_MUTE:
 	case UAC_FU_VOLUME:
@@ -1118,51 +1173,6 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 		break;
 	}
 
-	/* volume control quirks */
-	switch (state->chip->usb_id) {
-	case USB_ID(0x0471, 0x0101):
-	case USB_ID(0x0471, 0x0104):
-	case USB_ID(0x0471, 0x0105):
-	case USB_ID(0x0672, 0x1041):
-	/* quirk for UDA1321/N101.
-	 * note that detection between firmware 2.1.1.7 (N101)
-	 * and later 2.1.1.21 is not very clear from datasheets.
-	 * I hope that the min value is -15360 for newer firmware --jk
-	 */
-		if (!strcmp(kctl->id.name, "PCM Playback Volume") &&
-		    cval->min == -15616) {
-			snd_printk(KERN_INFO
-				 "set volume quirk for UDA1321/N101 chip\n");
-			cval->max = -256;
-		}
-		break;
-
-	case USB_ID(0x046d, 0x09a4):
-		if (!strcmp(kctl->id.name, "Mic Capture Volume")) {
-			snd_printk(KERN_INFO
-				"set volume quirk for QuickCam E3500\n");
-			cval->min = 6080;
-			cval->max = 8768;
-			cval->res = 192;
-		}
-		break;
-
-	case USB_ID(0x046d, 0x0808):
-	case USB_ID(0x046d, 0x0809):
-	case USB_ID(0x046d, 0x0991):
-	/* Most audio usb devices lie about volume resolution.
-	 * Most Logitech webcams have res = 384.
-	 * Proboly there is some logitech magic behind this number --fishor
-	 */
-		if (!strcmp(kctl->id.name, "Mic Capture Volume")) {
-			snd_printk(KERN_INFO
-				"set resolution quirk: cval->res = 384\n");
-			cval->res = 384;
-		}
-		break;
-
-	}
-
 	range = (cval->max - cval->min) / cval->res;
 	/* Are there devices with volume range more than 255? I use a bit more
 	 * to be sure. 384 is a resolution magic number found on Logitech
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index b61945f3af9e..32d2a21f2e3b 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1633,6 +1633,37 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* Roland GAIA SH-01 */
+	USB_DEVICE(0x0582, 0x0111),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "Roland",
+		.product_name = "GAIA",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_MIDI_FIXED_ENDPOINT,
+				.data = &(const struct snd_usb_midi_endpoint_info) {
+				.out_cables = 0x0003,
+				.in_cables  = 0x0003
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
+{
 	USB_DEVICE(0x0582, 0x0113),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
 		/* .vendor_name = "BOSS", */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 2e5bc7344026..a3ddac0deffd 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -137,12 +137,12 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 		return -ENOMEM;
 	}
 	if (fp->nr_rates > 0) {
-		rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL);
+		rate_table = kmemdup(fp->rate_table,
+				     sizeof(int) * fp->nr_rates, GFP_KERNEL);
 		if (!rate_table) {
 			kfree(fp);
 			return -ENOMEM;
 		}
-		memcpy(rate_table, fp->rate_table, sizeof(int) * fp->nr_rates);
 		fp->rate_table = rate_table;
 	}
 
@@ -224,10 +224,9 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
 	if (altsd->bNumEndpoints != 1)
 		return -ENXIO;
 
-	fp = kmalloc(sizeof(*fp), GFP_KERNEL);
+	fp = kmemdup(&ua_format, sizeof(*fp), GFP_KERNEL);
 	if (!fp)
 		return -ENOMEM;
-	memcpy(fp, &ua_format, sizeof(*fp));
 
 	fp->iface = altsd->bInterfaceNumber;
 	fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;