Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus

26 files changed, 452 insertions, 157 deletions

diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c
index f23f331..a57643d 100644
--- a/sound/soc/cirrus/ep93xx-i2s.c
+++ b/sound/soc/cirrus/ep93xx-i2s.c
@@ -408,7 +408,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 	return 0;
 
 fail_put_lrclk:
-	dev_set_drvdata(&pdev->dev, NULL);
 	clk_put(info->lrclk);
 fail_put_sclk:
 	clk_put(info->sclk);
@@ -423,7 +422,6 @@ static int ep93xx_i2s_remove(struct platform_device *pdev)
 	struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev);
 
 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);
 	clk_put(info->lrclk);
 	clk_put(info->sclk);
 	clk_put(info->mclk);
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index 66967ba..b2090b2 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -50,20 +50,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"DMIC AIF", NULL, "DMic"},
 };
 
-static int dmic_probe(struct snd_soc_codec *codec)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
-				  ARRAY_SIZE(dmic_dapm_widgets));
-        snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-	snd_soc_dapm_new_widgets(dapm);
-
-	return 0;
-}
-
 static struct snd_soc_codec_driver soc_dmic = {
-	.probe	= dmic_probe,
+	.dapm_widgets = dmic_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets),
+	.dapm_routes = intercon,
+	.num_dapm_routes = ARRAY_SIZE(intercon),
 };
 
 static int dmic_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 4db7314..c26a8f8 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = {
 
 static struct reg_default init_list[] = {
 	{RT5640_PR_BASE + 0x3d,	0x3600},
-	{RT5640_PR_BASE + 0x1c,	0x0D21},
-	{RT5640_PR_BASE + 0x1b,	0x0000},
 	{RT5640_PR_BASE + 0x12,	0x0aa8},
 	{RT5640_PR_BASE + 0x14,	0x0aaa},
 	{RT5640_PR_BASE + 0x20,	0x6110},
@@ -384,15 +382,11 @@ static const SOC_ENUM_SINGLE_DECL(
 
 static const struct snd_kcontrol_new rt5640_snd_controls[] = {
 	/* Speaker Output Volume */
-	SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL,
-		RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1),
 	SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL,
 		RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1),
 	SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL,
 		RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv),
 	/* Headphone Output Volume */
-	SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL,
-		RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1),
 	SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL,
 		RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1),
 	SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL,
@@ -737,6 +731,22 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = {
 			RT5640_M_BST1_MM_SFT, 1, 1),
 };
 
+static const struct snd_kcontrol_new spk_l_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL,
+		RT5640_L_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new spk_r_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL,
+		RT5640_R_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new hp_l_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL,
+		RT5640_L_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new hp_r_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL,
+		RT5640_R_MUTE_SFT, 1, 1);
+
 /* Stereo ADC source */
 static const char * const rt5640_stereo_adc1_src[] = {
 	"DIG MIX", "ADC"
@@ -868,33 +878,6 @@ static const SOC_ENUM_SINGLE_DECL(
 static const struct snd_kcontrol_new rt5640_sdi_mux =
 	SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum);
 
-static int spk_event(struct snd_soc_dapm_widget *w,
-	struct snd_kcontrol *kcontrol, int event)
-{
-	struct snd_soc_codec *codec = w->codec;
-	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
-
-	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
-		regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1,
-					0x0001, 0x0001);
-		regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c,
-					0xf000, 0xf000);
-		break;
-
-	case SND_SOC_DAPM_PRE_PMD:
-		regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c,
-					0xf000, 0x0000);
-		regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1,
-					0x0001, 0x0000);
-		break;
-
-	default:
-		return 0;
-	}
-	return 0;
-}
-
 static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -943,6 +926,117 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+void hp_amp_power_on(struct snd_soc_codec *codec)
+{
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	/* depop parameters */
+	regmap_update_bits(rt5640->regmap, RT5640_PR_BASE +
+		RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200);
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2,
+		RT5640_DEPOP_MASK, RT5640_DEPOP_MAN);
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1,
+		RT5640_HP_CP_MASK | RT5640_HP_SG_MASK | RT5640_HP_CB_MASK,
+		RT5640_HP_CP_PU | RT5640_HP_SG_DIS | RT5640_HP_CB_PU);
+	regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1,
+			   0x9f00);
+	/* headphone amp power on */
+	regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+		RT5640_PWR_FV1 | RT5640_PWR_FV2, 0);
+	regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+		RT5640_PWR_HA,
+		RT5640_PWR_HA);
+	usleep_range(10000, 15000);
+	regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+		RT5640_PWR_FV1 | RT5640_PWR_FV2 ,
+		RT5640_PWR_FV1 | RT5640_PWR_FV2);
+}
+
+static void rt5640_pmu_depop(struct snd_soc_codec *codec)
+{
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2,
+		RT5640_DEPOP_MASK | RT5640_DIG_DP_MASK,
+		RT5640_DEPOP_AUTO | RT5640_DIG_DP_EN);
+	regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP,
+		RT5640_PM_HP_MASK, RT5640_PM_HP_HV);
+
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3,
+		RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | RT5640_CP_FQ3_MASK,
+		(RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) |
+		(RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) |
+		(RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT));
+
+	regmap_write(rt5640->regmap, RT5640_PR_BASE +
+		RT5640_MAMP_INT_REG2, 0x1c00);
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1,
+		RT5640_HP_CP_MASK | RT5640_HP_SG_MASK,
+		RT5640_HP_CP_PD | RT5640_HP_SG_EN);
+	regmap_update_bits(rt5640->regmap, RT5640_PR_BASE +
+		RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400);
+}
+
+static int rt5640_hp_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		rt5640_pmu_depop(codec);
+		rt5640->hp_mute = 0;
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		rt5640->hp_mute = 1;
+		usleep_range(70000, 75000);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		hp_amp_power_on(codec);
+		break;
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		if (!rt5640->hp_mute)
+			usleep_range(80000, 85000);
+
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2,
 			RT5640_PWR_PLL_BIT, 0, NULL, 0),
@@ -1132,15 +1226,28 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
 		rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)),
 	SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1,
 		RT5640_PWR_MA_BIT, 0, NULL, 0),
-	SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1,
-		SND_SOC_NOPM, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1,
+	SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM,
+		0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0,
+		rt5640_hp_event,
+		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1,
 		RT5640_PWR_HP_L_BIT, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1,
+	SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1,
 		RT5640_PWR_HP_R_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1,
-		SND_SOC_NOPM, 0, spk_event,
-		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		RT5640_PWR_CLS_D_BIT, 0, NULL, 0),
+
+	/* Output Switch */
+	SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0,
+			&spk_l_enable_control),
+	SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0,
+			&spk_r_enable_control),
+	SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0,
+			&hp_l_enable_control),
+	SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0,
+			&hp_r_enable_control),
+	SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event),
 	/* Output Lines */
 	SND_SOC_DAPM_OUTPUT("SPOLP"),
 	SND_SOC_DAPM_OUTPUT("SPOLN"),
@@ -1381,9 +1488,11 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"},
 	{"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"},
 	{"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"},
+	{"HPO MIX L", NULL, "HP L Amp"},
 	{"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"},
 	{"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"},
 	{"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"},
+	{"HPO MIX R", NULL, "HP R Amp"},
 
 	{"LOUT MIX", "DAC L1 Switch", "DAC L1"},
 	{"LOUT MIX", "DAC R1 Switch", "DAC R1"},
@@ -1396,13 +1505,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"Mono MIX", "OUTVOL L Switch", "OUTVOL L"},
 	{"Mono MIX", "BST1 Switch", "BST1"},
 
-	{"HP L Amp", NULL, "HPO MIX L"},
-	{"HP R Amp", NULL, "HPO MIX R"},
+	{"HP Amp", NULL, "HPO MIX L"},
+	{"HP Amp", NULL, "HPO MIX R"},
 
-	{"SPOLP", NULL, "SPOL MIX"},
-	{"SPOLN", NULL, "SPOL MIX"},
-	{"SPORP", NULL, "SPOR MIX"},
-	{"SPORN", NULL, "SPOR MIX"},
+	{"Speaker L Playback", "Switch", "SPOL MIX"},
+	{"Speaker R Playback", "Switch", "SPOR MIX"},
+	{"SPOLP", NULL, "Speaker L Playback"},
+	{"SPOLN", NULL, "Speaker L Playback"},
+	{"SPORP", NULL, "Speaker R Playback"},
+	{"SPORN", NULL, "Speaker R Playback"},
 
 	{"SPOLP", NULL, "Improve SPK Amp Drv"},
 	{"SPOLN", NULL, "Improve SPK Amp Drv"},
@@ -1412,8 +1523,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"HPOL", NULL, "Improve HP Amp Drv"},
 	{"HPOR", NULL, "Improve HP Amp Drv"},
 
-	{"HPOL", NULL, "HP L Amp"},
-	{"HPOR", NULL, "HP R Amp"},
+	{"HP L Playback", "Switch", "HP Amp"},
+	{"HP R Playback", "Switch", "HP Amp"},
+	{"HPOL", NULL, "HP L Playback"},
+	{"HPOR", NULL, "HP R Playback"},
 	{"LOUTL", NULL, "LOUT MIX"},
 	{"LOUTR", NULL, "LOUT MIX"},
 	{"MONOP", NULL, "Mono MIX"},
@@ -1792,17 +1905,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
 				RT5640_PWR_BG | RT5640_PWR_VREF2,
 				RT5640_PWR_VREF1 | RT5640_PWR_MB |
 				RT5640_PWR_BG | RT5640_PWR_VREF2);
-			mdelay(10);
+			usleep_range(10000, 15000);
 			snd_soc_update_bits(codec, RT5640_PWR_ANLG1,
 				RT5640_PWR_FV1 | RT5640_PWR_FV2,
 				RT5640_PWR_FV1 | RT5640_PWR_FV2);
 			regcache_sync(rt5640->regmap);
 			snd_soc_update_bits(codec, RT5640_DUMMY1,
 						0x0301, 0x0301);
-			snd_soc_update_bits(codec, RT5640_DEPOP_M1,
-						0x001d, 0x0019);
-			snd_soc_update_bits(codec, RT5640_DEPOP_M2,
-						0x2000, 0x2000);
 			snd_soc_update_bits(codec, RT5640_MICBIAS,
 						0x0030, 0x0030);
 		}
@@ -1846,8 +1955,6 @@ static int rt5640_probe(struct snd_soc_codec *codec)
 	rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301);
-	snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019);
-	snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000);
 	snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030);
 	snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00);
 
@@ -2069,6 +2176,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c,
 		regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4,
 					RT5640_IN_DF2, RT5640_IN_DF2);
 
+	rt5640->hp_mute = 1;
+
 	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
 			rt5640_dai, ARRAY_SIZE(rt5640_dai));
 	if (ret < 0)
diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h
index c48286d..5e8df25a 100644
--- a/sound/soc/codecs/rt5640.h
+++ b/sound/soc/codecs/rt5640.h
@@ -145,6 +145,8 @@
 
 
 /* Index of Codec Private Register definition */
+#define RT5640_CHPUMP_INT_REG1			0x24
+#define RT5640_MAMP_INT_REG2			0x37
 #define RT5640_3D_SPK				0x63
 #define RT5640_WND_1				0x6c
 #define RT5640_WND_2				0x6d
@@ -153,6 +155,7 @@
 #define RT5640_WND_5				0x70
 #define RT5640_WND_8				0x73
 #define RT5640_DIP_SPK_INF			0x75
+#define RT5640_HP_DCC_INT1			0x77
 #define RT5640_EQ_BW_LOP			0xa0
 #define RT5640_EQ_GN_LOP			0xa1
 #define RT5640_EQ_FC_BP1			0xa2
@@ -1201,6 +1204,14 @@
 #define RT5640_CP_FQ2_SFT			4
 #define RT5640_CP_FQ3_MASK			(0x7)
 #define RT5640_CP_FQ3_SFT			0
+#define RT5640_CP_FQ_1_5_KHZ			0
+#define RT5640_CP_FQ_3_KHZ			1
+#define RT5640_CP_FQ_6_KHZ			2
+#define RT5640_CP_FQ_12_KHZ			3
+#define RT5640_CP_FQ_24_KHZ			4
+#define RT5640_CP_FQ_48_KHZ			5
+#define RT5640_CP_FQ_96_KHZ			6
+#define RT5640_CP_FQ_192_KHZ			7
 
 /* HPOUT charge pump (0x91) */
 #define RT5640_OSW_L_MASK			(0x1 << 11)
@@ -2087,6 +2098,7 @@ struct rt5640_priv {
 	int pll_out;
 
 	int dmic_en;
+	bool hp_mute;
 };
 
 #endif
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index f8d30e5..492644e 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -561,8 +561,9 @@ static int ssm2602_suspend(struct snd_soc_codec *codec)
 
 static int ssm2602_resume(struct snd_soc_codec *codec)
 {
-	snd_soc_cache_sync(codec);
+	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 
+	regcache_sync(ssm2602->regmap);
 	ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 17df4e3..2ed57d4 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -338,18 +338,6 @@ static inline int aic32x4_get_divs(int mclk, int rate)
 	return -EINVAL;
 }
 
-static int aic32x4_add_widgets(struct snd_soc_codec *codec)
-{
-	snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets,
-				  ARRAY_SIZE(aic32x4_dapm_widgets));
-
-	snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes,
-				ARRAY_SIZE(aic32x4_dapm_routes));
-
-	snd_soc_dapm_new_widgets(&codec->dapm);
-	return 0;
-}
-
 static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				  int clk_id, unsigned int freq, int dir)
 {
@@ -683,9 +671,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
 	}
 
 	aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	snd_soc_add_codec_controls(codec, aic32x4_snd_controls,
-			     ARRAY_SIZE(aic32x4_snd_controls));
-	aic32x4_add_widgets(codec);
 
 	/*
 	 * Workaround: for an unknown reason, the ADC needs to be powered up
@@ -714,6 +699,13 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
 	.suspend = aic32x4_suspend,
 	.resume = aic32x4_resume,
 	.set_bias_level = aic32x4_set_bias_level,
+
+	.controls = aic32x4_snd_controls,
+	.num_controls = ARRAY_SIZE(aic32x4_snd_controls),
+	.dapm_widgets = aic32x4_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets),
+	.dapm_routes = aic32x4_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes),
 };
 
 static int aic32x4_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 91dfbfe..4dfa8dc 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1202,7 +1202,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec)
 		break;
 	}
 
-	snd_soc_dapm_new_widgets(dapm);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 36782f0..11d80f3 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3174,7 +3174,7 @@ static ssize_t wm8962_beep_set(struct device *dev,
 	long int time;
 	int ret;
 
-	ret = strict_strtol(buf, 10, &time);
+	ret = kstrtol(buf, 10, &time);
 	if (ret != 0)
 		return ret;
 
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 70eb37a..25c31f1 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -421,13 +421,11 @@ static int dw_i2s_probe(struct platform_device *pdev)
 					 dw_i2s_dai, 1);
 	if (ret != 0) {
 		dev_err(&pdev->dev, "not able to register dai\n");
-		goto err_set_drvdata;
+		goto err_clk_disable;
 	}
 
 	return 0;
 
-err_set_drvdata:
-	dev_set_drvdata(&pdev->dev, NULL);
 err_clk_disable:
 	clk_disable(dev->clk);
 err_clk_put:
@@ -440,7 +438,6 @@ static int dw_i2s_remove(struct platform_device *pdev)
 	struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
 
 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);
 
 	clk_put(dev->clk);
 
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index cd088cc..b7ab71f 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -193,6 +193,17 @@ config SND_SOC_IMX_SGTL5000
 	  Say Y if you want to add support for SoC audio on an i.MX board with
 	  a sgtl5000 codec.
 
+config SND_SOC_IMX_SPDIF
+	tristate "SoC Audio support for i.MX boards with S/PDIF"
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_FSL_SPDIF
+	select SND_SOC_SPDIF
+	select REGMAP_MMIO
+	help
+	  SoC Audio support for i.MX boards with S/PDIF
+	  Say Y if you want to add support for SoC audio on an i.MX board with
+	  a S/DPDIF.
+
 config SND_SOC_IMX_MC13783
 	tristate "SoC Audio support for I.MX boards with mc13783"
 	depends on MFD_MC13783 && ARM
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 4b5970e..8db705b 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -45,6 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
 snd-soc-wm1133-ev1-objs := wm1133-ev1.o
 snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
 snd-soc-imx-wm8962-objs := imx-wm8962.o
+snd-soc-imx-spdif-objs := imx-spdif.o
 snd-soc-imx-mc13783-objs := imx-mc13783.o
 
 obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
@@ -53,4 +54,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
 obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
 obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
 obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
+obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
 obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 42a4382..3920c3e 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -411,8 +411,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-int fsl_spdif_startup(struct snd_pcm_substream *substream,
-			struct snd_soc_dai *cpu_dai)
+static int fsl_spdif_startup(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *cpu_dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
@@ -546,7 +546,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-struct snd_soc_dai_ops fsl_spdif_dai_ops = {
+static struct snd_soc_dai_ops fsl_spdif_dai_ops = {
 	.startup = fsl_spdif_startup,
 	.hw_params = fsl_spdif_hw_params,
 	.trigger = fsl_spdif_trigger,
@@ -555,7 +555,6 @@ struct snd_soc_dai_ops fsl_spdif_dai_ops = {
 
 
 /*
- * ============================================
  * FSL SPDIF IEC958 controller(mixer) functions
  *
  *	Channel status get/put control
@@ -563,7 +562,6 @@ struct snd_soc_dai_ops fsl_spdif_dai_ops = {
  *	Valid bit value get control
  *	DPLL lock status get control
  *	User bit sync mode selection control
- * ============================================
  */
 
 static int fsl_spdif_info(struct snd_kcontrol *kcontrol,
@@ -921,7 +919,7 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
 	return 0;
 }
 
-struct snd_soc_dai_driver fsl_spdif_dai = {
+static struct snd_soc_dai_driver fsl_spdif_dai = {
 	.probe = &fsl_spdif_dai_probe,
 	.playback = {
 		.channels_min = 2,
@@ -942,11 +940,7 @@ static const struct snd_soc_component_driver fsl_spdif_component = {
 	.name		= "fsl-spdif",
 };
 
-/*
- * ================
- * FSL SPDIF REGMAP
- * ================
- */
+/* FSL SPDIF REGMAP */
 
 static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg)
 {
@@ -1077,9 +1071,9 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
 			break;
 	}
 
-	dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate",
+	dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n",
 			spdif_priv->txclk_src[index], rate[index]);
-	dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate",
+	dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate\n",
 			spdif_priv->txclk_div[index], rate[index]);
 
 	return 0;
@@ -1119,10 +1113,8 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	}
 
 	regs = devm_ioremap_resource(&pdev->dev, res);
-	if (IS_ERR(regs)) {
-		dev_err(&pdev->dev, "could not map device resources\n");
+	if (IS_ERR(regs))
 		return PTR_ERR(regs);
-	}
 
 	spdif_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
 			"core", regs, &fsl_spdif_regmap_config);
@@ -1184,7 +1176,7 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 					 &spdif_priv->cpu_dai_drv, 1);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
-		goto error_dev;
+		return ret;
 	}
 
 	ret = imx_pcm_dma_init(pdev);
@@ -1197,8 +1189,6 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 
 error_component:
 	snd_soc_unregister_component(&pdev->dev);
-error_dev:
-	dev_set_drvdata(&pdev->dev, NULL);
 
 	return ret;
 }
@@ -1207,7 +1197,6 @@ static int fsl_spdif_remove(struct platform_device *pdev)
 {
 	imx_pcm_dma_exit(pdev);
 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 5cf626c..c6b7439 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1114,7 +1114,6 @@ error_dai:
 	snd_soc_unregister_component(&pdev->dev);
 
 error_dev:
-	dev_set_drvdata(&pdev->dev, NULL);
 	device_remove_file(&pdev->dev, dev_attr);
 
 error_clk:
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index ab17381..d3bf71a 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -335,7 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev)
 	if (audmux_type == IMX31_AUDMUX)
 		audmux_debugfs_init();
 
-	imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
+	if (of_id)
+		imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
new file mode 100644
index 0000000..816013b
--- /dev/null
+++ b/sound/soc/fsl/imx-spdif.c
@@ -0,0 +1,148 @@
+/*
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+
+struct imx_spdif_data {
+	struct snd_soc_dai_link dai[2];
+	struct snd_soc_card card;
+	struct platform_device *txdev;
+	struct platform_device *rxdev;
+};
+
+static int imx_spdif_audio_probe(struct platform_device *pdev)
+{
+	struct device_node *spdif_np, *np = pdev->dev.of_node;
+	struct imx_spdif_data *data;
+	int ret = 0, num_links = 0;
+
+	spdif_np = of_parse_phandle(np, "spdif-controller", 0);
+	if (!spdif_np) {
+		dev_err(&pdev->dev, "failed to find spdif-controller\n");
+		ret = -EINVAL;
+		goto end;
+	}
+
+	data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		dev_err(&pdev->dev, "failed to allocate memory\n");
+		ret = -ENOMEM;
+		goto end;
+	}
+
+	if (of_property_read_bool(np, "spdif-out")) {
+		data->dai[num_links].name = "S/PDIF TX";
+		data->dai[num_links].stream_name = "S/PDIF PCM Playback";
+		data->dai[num_links].codec_dai_name = "dit-hifi";
+		data->dai[num_links].codec_name = "spdif-dit";
+		data->dai[num_links].cpu_of_node = spdif_np;
+		data->dai[num_links].platform_of_node = spdif_np;
+		num_links++;
+
+		data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0);
+		if (IS_ERR(data->txdev)) {
+			ret = PTR_ERR(data->txdev);
+			dev_err(&pdev->dev, "register dit failed: %d\n", ret);
+			goto end;
+		}
+	}
+
+	if (of_property_read_bool(np, "spdif-in")) {
+		data->dai[num_links].name = "S/PDIF RX";
+		data->dai[num_links].stream_name = "S/PDIF PCM Capture";
+		data->dai[num_links].codec_dai_name = "dir-hifi";
+		data->dai[num_links].codec_name = "spdif-dir";
+		data->dai[num_links].cpu_of_node = spdif_np;
+		data->dai[num_links].platform_of_node = spdif_np;
+		num_links++;
+
+		data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0);
+		if (IS_ERR(data->rxdev)) {
+			ret = PTR_ERR(data->rxdev);
+			dev_err(&pdev->dev, "register dir failed: %d\n", ret);
+			goto error_dit;
+		}
+	}
+
+	if (!num_links) {
+		dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n");
+		goto error_dir;
+	}
+
+	data->card.dev = &pdev->dev;
+	data->card.num_links = num_links;
+	data->card.dai_link = data->dai;
+
+	ret = snd_soc_of_parse_card_name(&data->card, "model");
+	if (ret)
+		goto error_dir;
+
+	ret = snd_soc_register_card(&data->card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret);
+		goto error_dir;
+	}
+
+	platform_set_drvdata(pdev, data);
+
+	goto end;
+
+error_dir:
+	if (data->rxdev)
+		platform_device_unregister(data->rxdev);
+error_dit:
+	if (data->txdev)
+		platform_device_unregister(data->txdev);
+end:
+	if (spdif_np)
+		of_node_put(spdif_np);
+
+	return ret;
+}
+
+static int imx_spdif_audio_remove(struct platform_device *pdev)
+{
+	struct imx_spdif_data *data = platform_get_drvdata(pdev);
+
+	if (data->rxdev)
+		platform_device_unregister(data->rxdev);
+	if (data->txdev)
+		platform_device_unregister(data->txdev);
+
+	snd_soc_unregister_card(&data->card);
+
+	return 0;
+}
+
+static const struct of_device_id imx_spdif_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-spdif", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids);
+
+static struct platform_driver imx_spdif_driver = {
+	.driver = {
+		.name = "imx-spdif",
+		.owner = THIS_MODULE,
+		.of_match_table = imx_spdif_dt_ids,
+	},
+	.probe = imx_spdif_audio_probe,
+	.remove = imx_spdif_audio_remove,
+};
+
+module_platform_driver(imx_spdif_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-spdif");
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 6cf8355..8c49147 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -105,6 +105,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev)
 static struct platform_driver asoc_simple_card = {
 	.driver = {
 		.name	= "asoc-simple-card",
+		.owner = THIS_MODULE,
 	},
 	.probe		= asoc_simple_card_probe,
 	.remove		= asoc_simple_card_remove,
@@ -112,6 +113,7 @@ static struct platform_driver asoc_simple_card = {
 
 module_platform_driver(asoc_simple_card);
 
+MODULE_ALIAS("platform:asoc-simple-card");
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("ASoC Simple Sound Card");
 MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 9e1970c..78ed4a4 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,6 +1,6 @@
 config SND_KIRKWOOD_SOC
-	tristate "SoC Audio for the Marvell Kirkwood chip"
-	depends on ARCH_KIRKWOOD || COMPILE_TEST
+	tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
+	depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Kirkwood I2S interface. You will also need to select the
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index e5f3f7a..7fce340 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -22,6 +22,8 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <linux/platform_data/asoc-kirkwood.h>
+#include <linux/of.h>
+
 #include "kirkwood.h"
 
 #define DRV_NAME	"mvebu-audio"
@@ -453,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai;
 	struct kirkwood_dma_data *priv;
 	struct resource *mem;
+	struct device_node *np = pdev->dev.of_node;
 	int err;
 
 	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
@@ -473,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 		return -ENXIO;
 	}
 
-	if (!data) {
-		dev_err(&pdev->dev, "no platform data ?!\n");
+	if (np) {
+		priv->burst = 128;		/* might be 32 or 128 */
+	} else if (data) {
+		priv->burst = data->burst;
+	} else {
+		dev_err(&pdev->dev, "no DT nor platform data ?!\n");
 		return -EINVAL;
 	}
 
-	priv->burst = data->burst;
-
-	priv->clk = devm_clk_get(&pdev->dev, NULL);
+	priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL);
 	if (IS_ERR(priv->clk)) {
 		dev_err(&pdev->dev, "no clock\n");
 		return PTR_ERR(priv->clk);
@@ -507,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24;
 
 	/* Select the burst size */
-	if (data->burst == 32) {
+	if (priv->burst == 32) {
 		priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32;
 		priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32;
 	} else {
@@ -552,12 +557,21 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev)
 	return 0;
 }
 
+#ifdef CONFIG_OF
+static struct of_device_id mvebu_audio_of_match[] = {
+	{ .compatible = "marvell,mvebu-audio" },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, mvebu_audio_of_match);
+#endif
+
 static struct platform_driver kirkwood_i2s_driver = {
 	.probe  = kirkwood_i2s_dev_probe,
 	.remove = kirkwood_i2s_dev_remove,
 	.driver = {
 		.name = DRV_NAME,
 		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(mvebu_audio_of_match),
 	},
 };
 
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index ce084eb..4bb2737 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -105,11 +105,13 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
 		.stream_name	= "HiFi Playback",
 		.codec_dai_name	= "sgtl5000",
 		.ops		= &mxs_sgtl5000_hifi_ops,
+		.playback_only	= true,
 	}, {
 		.name		= "HiFi Rx",
 		.stream_name	= "HiFi Capture",
 		.codec_dai_name	= "sgtl5000",
 		.ops		= &mxs_sgtl5000_hifi_ops,
+		.capture_only	= true,
 	},
 };
 
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 361e4c0..83433fd 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -781,7 +781,7 @@ static ssize_t prop##_store(struct device *dev,				\
 	unsigned long val;						\
 	int status;							\
 									\
-	status = strict_strtoul(buf, 0, &val);				\
+	status = kstrtoul(buf, 0, &val);				\
 	if (status)							\
 		return status;						\
 									\
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index a0c67f6..9338d11 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -90,6 +90,13 @@ static void dma_enqueue(struct snd_pcm_substream *substream)
 	dma_info.period = prtd->dma_period;
 	dma_info.len = prtd->dma_period*limit;
 
+	if (dma_info.cap == DMA_CYCLIC) {
+		dma_info.buf = pos;
+		prtd->params->ops->prepare(prtd->params->ch, &dma_info);
+		prtd->dma_loaded += limit;
+		return;
+	}
+
 	while (prtd->dma_loaded < limit) {
 		pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
 
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 3039026..b33ca7c 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -235,6 +235,8 @@ struct fsi_stream {
 	struct sh_dmae_slave	slave; /* see fsi_handler_init() */
 	struct work_struct	work;
 	dma_addr_t		dma;
+	int			loop_cnt;
+	int			additional_pos;
 };
 
 struct fsi_clk {
@@ -1289,6 +1291,8 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io)
 	io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
 			 BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);
 
+	io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */
+	io->additional_pos = 0;
 	io->dma = dma_map_single(dai->dev, runtime->dma_area,
 				 snd_pcm_lib_buffer_bytes(io->substream), dir);
 	return 0;
@@ -1305,11 +1309,15 @@ static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io)
 	return 0;
 }
 
-static dma_addr_t fsi_dma_get_area(struct fsi_stream *io)
+static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional)
 {
 	struct snd_pcm_runtime *runtime = io->substream->runtime;
+	int period = io->period_pos + additional;
 
-	return io->dma + samples_to_bytes(runtime, io->buff_sample_pos);
+	if (period >= runtime->periods)
+		period = 0;
+
+	return io->dma + samples_to_bytes(runtime, period * io->period_samples);
 }
 
 static void fsi_dma_complete(void *data)
@@ -1321,7 +1329,7 @@ static void fsi_dma_complete(void *data)
 	enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
 		DMA_TO_DEVICE : DMA_FROM_DEVICE;
 
-	dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io),
+	dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0),
 			samples_to_bytes(runtime, io->period_samples), dir);
 
 	io->buff_sample_pos += io->period_samples;
@@ -1347,7 +1355,7 @@ static void fsi_dma_do_work(struct work_struct *work)
 	struct snd_pcm_runtime *runtime;
 	enum dma_data_direction dir;
 	int is_play = fsi_stream_is_play(fsi, io);
-	int len;
+	int len, i;
 	dma_addr_t buf;
 
 	if (!fsi_stream_is_working(fsi, io))
@@ -1357,26 +1365,33 @@ static void fsi_dma_do_work(struct work_struct *work)
 	runtime	= io->substream->runtime;
 	dir	= is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
 	len	= samples_to_bytes(runtime, io->period_samples);
-	buf	= fsi_dma_get_area(io);
 
-	dma_sync_single_for_device(dai->dev, buf, len, dir);
+	for (i = 0; i < io->loop_cnt; i++) {
+		buf	= fsi_dma_get_area(io, io->additional_pos);
 
-	desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
-					   DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
-	if (!desc) {
-		dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
-		return;
-	}
+		dma_sync_single_for_device(dai->dev, buf, len, dir);
 
-	desc->callback		= fsi_dma_complete;
-	desc->callback_param	= io;
+		desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
+					DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+		if (!desc) {
+			dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
+			return;
+		}
 
-	if (dmaengine_submit(desc) < 0) {
-		dev_err(dai->dev, "tx_submit() fail\n");
-		return;
+		desc->callback		= fsi_dma_complete;
+		desc->callback_param	= io;
+
+		if (dmaengine_submit(desc) < 0) {
+			dev_err(dai->dev, "tx_submit() fail\n");
+			return;
+		}
+
+		dma_async_issue_pending(io->chan);
+
+		io->additional_pos = 1;
 	}
 
-	dma_async_issue_pending(io->chan);
+	io->loop_cnt = 1;
 
 	/*
 	 * FIXME
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 528f870..4d05613 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -203,7 +203,7 @@ static ssize_t pmdown_time_set(struct device *dev,
 	struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
 	int ret;
 
-	ret = strict_strtol(buf, 10, &rtd->pmdown_time);
+	ret = kstrtol(buf, 10, &rtd->pmdown_time);
 	if (ret)
 		return ret;
 
@@ -248,6 +248,7 @@ static ssize_t codec_reg_write_file(struct file *file,
 	char *start = buf;
 	unsigned long reg, value;
 	struct snd_soc_codec *codec = file->private_data;
+	int ret;
 
 	buf_size = min(count, (sizeof(buf)-1));
 	if (copy_from_user(buf, user_buf, buf_size))
@@ -259,8 +260,9 @@ static ssize_t codec_reg_write_file(struct file *file,
 	reg = simple_strtoul(start, &start, 16);
 	while (*start == ' ')
 		start++;
-	if (strict_strtoul(start, 16, &value))
-		return -EINVAL;
+	ret = kstrtoul(start, 16, &value);
+	if (ret)
+		return ret;
 
 	/* Userspace has been fiddling around behind the kernel's back */
 	add_taint(TAINT_USER, LOCKDEP_NOW_UNRELIABLE);
@@ -1243,9 +1245,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
 	}
 	rtd->card = card;
 
-	/* Make sure all DAPM widgets are instantiated */
-	snd_soc_dapm_new_widgets(&codec->dapm);
-
 	/* machine controls, routes and widgets are not prefixed */
 	temp = codec->name_prefix;
 	codec->name_prefix = NULL;
@@ -1741,8 +1740,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 		snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
 					card->num_dapm_routes);
 
-	snd_soc_dapm_new_widgets(&card->dapm);
-
 	for (i = 0; i < card->num_links; i++) {
 		dai_link = &card->dai_link[i];
 		dai_fmt = dai_link->dai_fmt;
@@ -1821,12 +1818,12 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 		}
 	}
 
-	snd_soc_dapm_new_widgets(&card->dapm);
-
 	if (card->fully_routed)
 		list_for_each_entry(codec, &card->codec_dev_list, card_list)
 			snd_soc_dapm_auto_nc_codec_pins(codec);
 
+	snd_soc_dapm_new_widgets(card);
+
 	ret = snd_card_register(card->snd_card);
 	if (ret < 0) {
 		dev_err(card->dev, "ASoC: failed to register soundcard %d\n",
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d84bd0f..c17c14c 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -229,6 +229,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 			template.id = snd_soc_dapm_kcontrol;
 			template.name = kcontrol->id.name;
 
+			data->value = template.on_val;
+
 			data->widget = snd_soc_dapm_new_control(widget->dapm,
 				&template);
 			if (!data->widget) {
@@ -2374,6 +2376,9 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 			wsource->ext = 1;
 	}
 
+	dapm_mark_dirty(wsource, "Route added");
+	dapm_mark_dirty(wsink, "Route added");
+
 	/* connect static paths */
 	if (control == NULL) {
 		list_add(&path->list, &dapm->card->paths);
@@ -2436,9 +2441,6 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 		return 0;
 	}
 
-	dapm_mark_dirty(wsource, "Route added");
-	dapm_mark_dirty(wsink, "Route added");
-
 	return 0;
 err:
 	kfree(path);
@@ -2712,9 +2714,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes);
  *
  * Returns 0 for success.
  */
-int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
+int snd_soc_dapm_new_widgets(struct snd_soc_card *card)
 {
-	struct snd_soc_card *card = dapm->card;
 	struct snd_soc_dapm_widget *w;
 	unsigned int val;
 
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7aa26b5..71358e3 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -183,8 +183,6 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
 		list_add(&(pins[i].list), &jack->pins);
 	}
 
-	snd_soc_dapm_new_widgets(&jack->codec->card->dapm);
-
 	/* Update to reflect the last reported status; canned jack
 	 * implementations are likely to set their state before the
 	 * card has an opportunity to associate pins.
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index fb70fbe..330c9a6 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2020,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 			capture = 1;
 	}
 
+	if (rtd->dai_link->playback_only) {
+		playback = 1;
+		capture = 0;
+	}
+
+	if (rtd->dai_link->capture_only) {
+		playback = 0;
+		capture = 1;
+	}
+
 	/* create the PCM */
 	if (rtd->dai_link->no_pcm) {
 		snprintf(new_name, sizeof(new_name), "(%s)",