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 445ca481d8d3..bf578ba2677e 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 b524f89a1f13..18d972501585 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 4edd2d0f9a3c..ec68eaea0336 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 00e041333904..b899eba6f279 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 bc07d369fac4..52c26d3a61d4 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 c02405cc007d..5810a0603f2f 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 8af04343cc1a..259d1ac4492f 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 b8ba0adacfce..80555d7551e6 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 41cdd1642970..8dbcacd44e6a 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 46c5b4fdfc52..ca1be1d9dcf0 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 4d0561312f3b..1a38be0d0ca8 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 63fb5219f0f8..6234a51625b1 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 f2a1acdc4d83..814d0e887c62 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];