Merge branch 'fix/hda' into topic/hda

MAINTAINERS

@@ -5648,7 +5648,6 @@ F:	drivers/media/video/*7146*
 F:	include/media/*7146*
 
 SAMSUNG AUDIO (ASoC) DRIVERS
-M:	Jassi Brar <jassisinghbrar@gmail.com>
 M:	Sangbeom Kim <sbkim73@samsung.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Supported

sound/pci/hda/hda_intel.c

@@ -2507,7 +2507,6 @@ 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),

sound/pci/hda/patch_cirrus.c

@@ -1278,7 +1278,7 @@ static const char * const cs420x_models[CS420X_MODELS] = {
 	[CS420X_MBP53] = "mbp53",
 	[CS420X_MBP55] = "mbp55",
 	[CS420X_IMAC27] = "imac27",
-	[CS420X_IMAC27] = "apple",
+	[CS420X_APPLE] = "apple",
 	[CS420X_AUTO] = "auto",
 };
 

sound/pci/hda/patch_sigmatel.c

@@ -4441,7 +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;
 		}
@@ -5055,20 +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)
 {
@@ -5668,8 +5656,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 		} 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 =
@@ -5983,8 +5969,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 		} 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 =

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)

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),

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;
@@ -501,7 +502,8 @@ static int cs4271_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	ret = snd_soc_update_bits(codec, CS4271_MODE2, 0,
+	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;

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),

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),
 };

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);

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

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;

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;
 

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),
 };

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),
 };

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);
 

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),
 };

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),
 };

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;

sound/usb/quirks-table.h

@@ -1632,6 +1632,37 @@ 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) {