Merge branch 'for-linus' into for-next

For updating the HDMI chmap fix.

Conflicts:
	sound/pci/hda/patch_hdmi.c

13 files changed, 150 insertions, 47 deletions

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 445ca481d8d..bf578ba2677 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -175,6 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
 { 0x54524106, 0xffffffff, "TR28026",		NULL,		NULL },
 { 0x54524108, 0xffffffff, "TR28028",		patch_tritech_tr28028,	NULL }, // added by xin jin [07/09/99]
 { 0x54524123, 0xffffffff, "TR28602",		NULL,		NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)]
+{ 0x54584e03, 0xffffffff, "TLV320AIC27",	NULL,		NULL },
 { 0x54584e20, 0xffffffff, "TLC320AD9xC",	NULL,		NULL },
 { 0x56494161, 0xffffffff, "VIA1612A",		NULL,		NULL }, // modified ICE1232 with S/PDIF
 { 0x56494170, 0xffffffff, "VIA1617A",		patch_vt1617a,	NULL }, // modified VT1616 with S/PDIF
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index b524f89a1f1..18d97250158 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -111,6 +111,9 @@ enum {
 /* 0x0009 - 0x0014 -> 12 test regs */
 /* 0x0015 - visibility reg */
 
+/* Cirrus Logic CS4208 */
+#define CS4208_VENDOR_NID	0x24
+
 /*
  * Cirrus Logic CS4210
  *
@@ -223,6 +226,16 @@ static const struct hda_verb cs_coef_init_verbs[] = {
 	{} /* terminator */
 };
 
+static const struct hda_verb cs4208_coef_init_verbs[] = {
+	{0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */
+	{0x24, AC_VERB_SET_PROC_STATE, 0x01},  /* VPW: processing on */
+	{0x24, AC_VERB_SET_COEF_INDEX, 0x0033},
+	{0x24, AC_VERB_SET_PROC_COEF, 0x0001}, /* A1 ICS */
+	{0x24, AC_VERB_SET_COEF_INDEX, 0x0034},
+	{0x24, AC_VERB_SET_PROC_COEF, 0x1C01}, /* A1 Enable, A Thresh = 300mV */
+	{} /* terminator */
+};
+
 /* Errata: CS4207 rev C0/C1/C2 Silicon
  *
  * http://www.cirrus.com/en/pubs/errata/ER880C3.pdf
@@ -295,6 +308,8 @@ static int cs_init(struct hda_codec *codec)
 		/* init_verb sequence for C0/C1/C2 errata*/
 		snd_hda_sequence_write(codec, cs_errata_init_verbs);
 		snd_hda_sequence_write(codec, cs_coef_init_verbs);
+	} else if (spec->vendor_nid == CS4208_VENDOR_NID) {
+		snd_hda_sequence_write(codec, cs4208_coef_init_verbs);
 	}
 
 	snd_hda_gen_init(codec);
@@ -434,6 +449,29 @@ static const struct hda_pintbl mba42_pincfgs[] = {
 	{} /* terminator */
 };
 
+static const struct hda_pintbl mba6_pincfgs[] = {
+	{ 0x10, 0x032120f0 }, /* HP */
+	{ 0x11, 0x500000f0 },
+	{ 0x12, 0x90100010 }, /* Speaker */
+	{ 0x13, 0x500000f0 },
+	{ 0x14, 0x500000f0 },
+	{ 0x15, 0x770000f0 },
+	{ 0x16, 0x770000f0 },
+	{ 0x17, 0x430000f0 },
+	{ 0x18, 0x43ab9030 }, /* Mic */
+	{ 0x19, 0x770000f0 },
+	{ 0x1a, 0x770000f0 },
+	{ 0x1b, 0x770000f0 },
+	{ 0x1c, 0x90a00090 },
+	{ 0x1d, 0x500000f0 },
+	{ 0x1e, 0x500000f0 },
+	{ 0x1f, 0x500000f0 },
+	{ 0x20, 0x500000f0 },
+	{ 0x21, 0x430000f0 },
+	{ 0x22, 0x430000f0 },
+	{} /* terminator */
+};
+
 static void cs420x_fixup_gpio_13(struct hda_codec *codec,
 				 const struct hda_fixup *fix, int action)
 {
@@ -556,22 +594,23 @@ static int patch_cs420x(struct hda_codec *codec)
 
 /*
  * CS4208 support:
- * Its layout is no longer compatible with CS4206/CS4207, and the generic
- * parser seems working fairly well, except for trivial fixups.
+ * Its layout is no longer compatible with CS4206/CS4207
  */
 enum {
+	CS4208_MBA6,
 	CS4208_GPIO0,
 };
 
 static const struct hda_model_fixup cs4208_models[] = {
 	{ .id = CS4208_GPIO0, .name = "gpio0" },
+	{ .id = CS4208_MBA6, .name = "mba6" },
 	{}
 };
 
 static const struct snd_pci_quirk cs4208_fixup_tbl[] = {
 	/* codec SSID */
-	SND_PCI_QUIRK(0x106b, 0x7100, "MacBookPro 6,1", CS4208_GPIO0),
-	SND_PCI_QUIRK(0x106b, 0x7200, "MacBookPro 6,2", CS4208_GPIO0),
+	SND_PCI_QUIRK(0x106b, 0x7100, "MacBookAir 6,1", CS4208_MBA6),
+	SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6),
 	{} /* terminator */
 };
 
@@ -588,18 +627,35 @@ static void cs4208_fixup_gpio0(struct hda_codec *codec,
 }
 
 static const struct hda_fixup cs4208_fixups[] = {
+	[CS4208_MBA6] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = mba6_pincfgs,
+		.chained = true,
+		.chain_id = CS4208_GPIO0,
+	},
 	[CS4208_GPIO0] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cs4208_fixup_gpio0,
 	},
 };
 
+/* correct the 0dB offset of input pins */
+static void cs4208_fix_amp_caps(struct hda_codec *codec, hda_nid_t adc)
+{
+	unsigned int caps;
+
+	caps = query_amp_caps(codec, adc, HDA_INPUT);
+	caps &= ~(AC_AMPCAP_OFFSET);
+	caps |= 0x02;
+	snd_hda_override_amp_caps(codec, adc, HDA_INPUT, caps);
+}
+
 static int patch_cs4208(struct hda_codec *codec)
 {
 	struct cs_spec *spec;
 	int err;
 
-	spec = cs_alloc_spec(codec, 0); /* no specific w/a */
+	spec = cs_alloc_spec(codec, CS4208_VENDOR_NID);
 	if (!spec)
 		return -ENOMEM;
 
@@ -609,6 +665,12 @@ static int patch_cs4208(struct hda_codec *codec)
 			   cs4208_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
+	snd_hda_override_wcaps(codec, 0x18,
+			       get_wcaps(codec, 0x18) | AC_WCAP_STEREO);
+	cs4208_fix_amp_caps(codec, 0x18);
+	cs4208_fix_amp_caps(codec, 0x1b);
+	cs4208_fix_amp_caps(codec, 0x1c);
+
 	err = cs_parse_auto_config(codec);
 	if (err < 0)
 		goto error;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 4edd2d0f9a3..ec68eaea033 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3231,6 +3231,7 @@ enum {
 	CXT_FIXUP_INC_MIC_BOOST,
 	CXT_FIXUP_HEADPHONE_MIC_PIN,
 	CXT_FIXUP_HEADPHONE_MIC,
+	CXT_FIXUP_GPIO1,
 };
 
 static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
@@ -3375,6 +3376,15 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt_fixup_headphone_mic,
 	},
+	[CXT_FIXUP_GPIO1] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{ 0x01, AC_VERB_SET_GPIO_MASK, 0x01 },
+			{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 },
+			{ 0x01, AC_VERB_SET_GPIO_DATA, 0x01 },
+			{ }
+		},
+	},
 };
 
 static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -3384,6 +3394,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = {
 
 static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1),
 	SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 00e04133390..b899eba6f27 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -976,6 +976,14 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
 	}
 
 	/*
+	 * always configure channel mapping, it may have been changed by the
+	 * user in the meantime
+	 */
+	hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca,
+				   channels, per_pin->chmap,
+				   per_pin->chmap_set);
+
+	/*
 	 * sizeof(ai) is used instead of sizeof(*hdmi_ai) or
 	 * sizeof(*dp_ai) to avoid partial match/update problems when
 	 * the user switches between HDMI/DP monitors.
@@ -986,20 +994,10 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
 			    "pin=%d channels=%d ca=0x%02x\n",
 			    pin_nid,
 			    active_channels, ca);
-		hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca,
-					   channels, per_pin->chmap,
-					   per_pin->chmap_set);
 		hdmi_stop_infoframe_trans(codec, pin_nid);
 		hdmi_fill_audio_infoframe(codec, pin_nid,
 					    ai.bytes, sizeof(ai));
 		hdmi_start_infoframe_trans(codec, pin_nid);
-	} else {
-		/* For non-pcm audio switch, setup new channel mapping
-		 * accordingly */
-		if (per_pin->non_pcm != non_pcm)
-			hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca,
-						   channels, per_pin->chmap,
-						   per_pin->chmap_set);
 	}
 
 	per_pin->non_pcm = non_pcm;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bc07d369fac..52c26d3a61d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3439,6 +3439,9 @@ static void alc283_fixup_chromebook(struct hda_codec *codec,
 		/* Set to manual mode */
 		val = alc_read_coef_idx(codec, 0x06);
 		alc_write_coef_idx(codec, 0x06, val & ~0x000c);
+		/* Enable Line1 input control by verb */
+		val = alc_read_coef_idx(codec, 0x1a);
+		alc_write_coef_idx(codec, 0x1a, val | (1 << 4));
 		break;
 	}
 }
@@ -3493,6 +3496,15 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec,
 	}
 }
 
+static void alc290_fixup_mono_speakers(struct hda_codec *codec,
+				       const struct hda_fixup *fix, int action)
+{
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		/* Remove DAC node 0x03, as it seems to be
+		   giving mono output */
+		snd_hda_override_wcaps(codec, 0x03, 0);
+}
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -3519,6 +3531,7 @@ enum {
 	ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT,
 	ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC269_FIXUP_DELL2_MIC_NO_PRESENCE,
+	ALC269_FIXUP_DELL3_MIC_NO_PRESENCE,
 	ALC269_FIXUP_HEADSET_MODE,
 	ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC,
 	ALC269_FIXUP_ASUS_X101_FUNC,
@@ -3531,6 +3544,8 @@ enum {
 	ALC269VB_FIXUP_ORDISSIMO_EVE2,
 	ALC283_FIXUP_CHROME_BOOK,
 	ALC282_FIXUP_ASUS_TX300,
+	ALC283_FIXUP_INT_MIC,
+	ALC290_FIXUP_MONO_SPEAKERS,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -3708,6 +3723,15 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
 	},
+	[ALC269_FIXUP_DELL3_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+	},
 	[ALC269_FIXUP_HEADSET_MODE] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_headset_mode,
@@ -3790,6 +3814,22 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc282_fixup_asus_tx300,
 	},
+	[ALC283_FIXUP_INT_MIC] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{0x20, AC_VERB_SET_COEF_INDEX, 0x1a},
+			{0x20, AC_VERB_SET_PROC_COEF, 0x0011},
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
+	},
+	[ALC290_FIXUP_MONO_SPEAKERS] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc290_fixup_mono_speakers,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -3831,6 +3871,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS),
 	SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -3874,7 +3915,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
-	SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c
index c02405cc007..5810a0603f2 100644
--- a/sound/soc/blackfin/bf6xx-i2s.c
+++ b/sound/soc/blackfin/bf6xx-i2s.c
@@ -88,6 +88,7 @@ static int bfin_i2s_hw_params(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_FORMAT_S8:
 		param.spctl |= 0x70;
 		sport->wdsize = 1;
+		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		param.spctl |= 0xf0;
 		sport->wdsize = 2;
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 8af04343cc1..259d1ac4492 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -349,6 +349,9 @@ static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol,
 	val = ucontrol->value.integer.value[0];
 	val2 = ucontrol->value.integer.value[1];
 
+	if (val >= ARRAY_SIZE(st_table) || val2 >= ARRAY_SIZE(st_table))
+		return -EINVAL;
+
 	err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m);
 	if (err < 0)
 		return err;
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index b8ba0adacfc..80555d7551e 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -1225,13 +1225,18 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
 	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
 	struct device *dev = codec->dev;
 	bool apply_fir, apply_iir;
-	int req, status;
+	unsigned int req;
+	int status;
 
 	dev_dbg(dev, "%s: Enter.\n", __func__);
 
 	mutex_lock(&drvdata->anc_lock);
 
 	req = ucontrol->value.integer.value[0];
+	if (req >= ARRAY_SIZE(enum_anc_state)) {
+		status = -EINVAL;
+		goto cleanup;
+	}
 	if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR &&
 		req != ANC_APPLY_IIR) {
 		dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n",
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 41cdd164297..8dbcacd44e6 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1863,7 +1863,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol,
 	struct max98095_pdata *pdata = max98095->pdata;
 	int channel = max98095_get_eq_channel(kcontrol->id.name);
 	struct max98095_cdata *cdata;
-	int sel = ucontrol->value.integer.value[0];
+	unsigned int sel = ucontrol->value.integer.value[0];
 	struct max98095_eq_cfg *coef_set;
 	int fs, best, best_val, i;
 	int regmask, regsave;
@@ -2016,7 +2016,7 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol,
 	struct max98095_pdata *pdata = max98095->pdata;
 	int channel = max98095_get_bq_channel(codec, kcontrol->id.name);
 	struct max98095_cdata *cdata;
-	int sel = ucontrol->value.integer.value[0];
+	unsigned int sel = ucontrol->value.integer.value[0];
 	struct max98095_biquad_cfg *coef_set;
 	int fs, best, best_val, i;
 	int regmask, regsave;
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index 46c5b4fdfc5..ca1be1d9dcf 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -62,7 +62,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 	struct device_node *ssi_np, *codec_np;
 	struct platform_device *ssi_pdev;
 	struct i2c_client *codec_dev;
-	struct imx_sgtl5000_data *data;
+	struct imx_sgtl5000_data *data = NULL;
 	int int_port, ext_port;
 	int ret;
 
@@ -128,7 +128,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 		goto fail;
 	}
 
-	data->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
+	data->codec_clk = clk_get(&codec_dev->dev, NULL);
 	if (IS_ERR(data->codec_clk)) {
 		ret = PTR_ERR(data->codec_clk);
 		goto fail;
@@ -172,6 +172,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 	return 0;
 
 fail:
+	if (data && !IS_ERR(data->codec_clk))
+		clk_put(data->codec_clk);
 	if (ssi_np)
 		of_node_put(ssi_np);
 	if (codec_np)
@@ -185,6 +187,7 @@ static int imx_sgtl5000_remove(struct platform_device *pdev)
 	struct imx_sgtl5000_data *data = platform_get_drvdata(pdev);
 
 	snd_soc_unregister_card(&data->card);
+	clk_put(data->codec_clk);
 
 	return 0;
 }
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4d0561312f3..1a38be0d0ca 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1380,7 +1380,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
 				return -ENODEV;
 
 			list_add(&cpu_dai->dapm.list, &card->dapm_list);
-			snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai);
 		}
 
 		if (cpu_dai->driver->probe) {
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 63fb5219f0f..6234a51625b 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -299,19 +299,6 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y,
 	usX2Y_clients_stop(usX2Y);
 }
 
-static void usX2Y_error_sequence(struct usX2Ydev *usX2Y,
-				 struct snd_usX2Y_substream *subs, struct urb *urb)
-{
-	snd_printk(KERN_ERR
-"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n"
-"Most probably some urb of usb-frame %i is still missing.\n"
-"Cause could be too long delays in usb-hcd interrupt handling.\n",
-		   usb_get_current_frame_number(usX2Y->dev),
-		   subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out",
-		   usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame);
-	usX2Y_clients_stop(usX2Y);
-}
-
 static void i_usX2Y_urb_complete(struct urb *urb)
 {
 	struct snd_usX2Y_substream *subs = urb->context;
@@ -328,12 +315,9 @@ static void i_usX2Y_urb_complete(struct urb *urb)
 		usX2Y_error_urb_status(usX2Y, subs, urb);
 		return;
 	}
-	if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF)))
-		subs->completed_urb = urb;
-	else {
-		usX2Y_error_sequence(usX2Y, subs, urb);
-		return;
-	}
+
+	subs->completed_urb = urb;
+
 	{
 		struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE],
 			*playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK];
diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c
index f2a1acdc4d8..814d0e887c6 100644
--- a/sound/usb/usx2y/usx2yhwdeppcm.c
+++ b/sound/usb/usx2y/usx2yhwdeppcm.c
@@ -244,13 +244,8 @@ static void i_usX2Y_usbpcm_urb_complete(struct urb *urb)
 		usX2Y_error_urb_status(usX2Y, subs, urb);
 		return;
 	}
-	if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF)))
-		subs->completed_urb = urb;
-	else {
-		usX2Y_error_sequence(usX2Y, subs, urb);
-		return;
-	}
 
+	subs->completed_urb = urb;
 	capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE];
 	capsubs2 = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2];
 	playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK];