22 files changed, 145 insertions, 221 deletions

diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index 58804c7acfcf..fd2188c3df2b 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -170,7 +170,7 @@ static void tas_set_volume(struct tas *tas)
         /* analysing the volume and mixer tables shows
          * that they are similar enough when we shift
          * the mixer table down by 4 bits. The error
-         * is minuscule, in just one item the error
+         * is miniscule, in just one item the error
          * is 1, at a value of 0x07f17b (mixer table
          * value is 0x07f17a) */
         tmp = tas_gaintable[left];
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index 33f0ba5559a7..62e959120c44 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -44,10 +44,10 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_adb = {
         .channels_min = 1,
         .channels_max = 2,
         .buffer_bytes_max = 0x10000,
-        .period_bytes_min = 0x1,
+        .period_bytes_min = 0x20,
         .period_bytes_max = 0x1000,
         .periods_min = 2,
-        .periods_max = 32,
+        .periods_max = 1024,
 };
 
 #ifndef CHIP_AU8820
@@ -140,6 +140,9 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream)
                                         SNDRV_PCM_HW_PARAM_PERIOD_BYTES)) < 0)
                 return err;
 
+        snd_pcm_hw_constraint_step(runtime, 0,
+                                        SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 64);
+
         if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
 #ifndef CHIP_AU8820
                 if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_A3D) {
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 430f41db6044..759ade12e758 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -937,6 +937,7 @@ void snd_hda_shutup_pins(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_HDA(snd_hda_shutup_pins);
 
+#ifdef SND_HDA_NEEDS_RESUME
 /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
 static void restore_shutup_pins(struct hda_codec *codec)
 {
@@ -953,6 +954,7 @@ static void restore_shutup_pins(struct hda_codec *codec)
         }
         codec->pins_shutup = 0;
 }
+#endif
 
 static void init_hda_cache(struct hda_cache_rec *cache,
                            unsigned int record_size);
@@ -1329,6 +1331,7 @@ static void purify_inactive_streams(struct hda_codec *codec)
         }
 }
 
+#ifdef SND_HDA_NEEDS_RESUME
 /* clean up all streams; called from suspend */
 static void hda_cleanup_all_streams(struct hda_codec *codec)
 {
@@ -1340,6 +1343,7 @@ static void hda_cleanup_all_streams(struct hda_codec *codec)
                         really_cleanup_stream(codec, p);
         }
 }
+#endif
 
 /*
  * amp access functions
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 52928d9a72da..c82979a8cd09 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1704,11 +1704,11 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
                                    codec->chip_name, fix->type);
                         break;
                 }
-                if (!fix[id].chained)
+                if (!fix->chained)
                         break;
                 if (++depth > 10)
                         break;
-                id = fix[id].chain_id;
+                id = fix->chain_id;
         }
 }
 
@@ -5645,6 +5645,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
 static struct snd_pci_quirk beep_white_list[] = {
         SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
         SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1),
+        SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1),
         SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
         {}
 };
@@ -9863,6 +9864,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
         SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
         SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
+        SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
 
         SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
         SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
@@ -10699,7 +10701,6 @@ enum {
         PINFIX_LENOVO_Y530,
         PINFIX_PB_M5210,
         PINFIX_ACER_ASPIRE_7736,
-        PINFIX_GIGABYTE_880GM,
 };
 
 static const struct alc_fixup alc882_fixups[] = {
@@ -10731,13 +10732,6 @@ static const struct alc_fixup alc882_fixups[] = {
                 .type = ALC_FIXUP_SKU,
                 .v.sku = ALC_FIXUP_SKU_IGNORE,
         },
-        [PINFIX_GIGABYTE_880GM] = {
-                .type = ALC_FIXUP_PINS,
-                .v.pins = (const struct alc_pincfg[]) {
-                        { 0x14, 0x1114410 }, /* set as speaker */
-                        { }
-                }
-        },
 };
 
 static struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -10745,7 +10739,6 @@ static struct snd_pci_quirk alc882_fixup_tbl[] = {
         SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530),
         SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
         SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736),
-        SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", PINFIX_GIGABYTE_880GM),
         {}
 };
 
@@ -14868,6 +14861,23 @@ static void alc269_fixup_hweq(struct hda_codec *codec,
         alc_write_coef_idx(codec, 0x1e, coef | 0x80);
 }
 
+static void alc271_fixup_dmic(struct hda_codec *codec,
+                              const struct alc_fixup *fix, int action)
+{
+        static struct hda_verb verbs[] = {
+                {0x20, AC_VERB_SET_COEF_INDEX, 0x0d},
+                {0x20, AC_VERB_SET_PROC_COEF, 0x4000},
+                {}
+        };
+        unsigned int cfg;
+
+        if (strcmp(codec->chip_name, "ALC271X"))
+                return;
+        cfg = snd_hda_codec_get_pincfg(codec, 0x12);
+        if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED)
+                snd_hda_sequence_write(codec, verbs);
+}
+
 enum {
         ALC269_FIXUP_SONY_VAIO,
         ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -14876,6 +14886,7 @@ enum {
         ALC269_FIXUP_ASUS_G73JW,
         ALC269_FIXUP_LENOVO_EAPD,
         ALC275_FIXUP_SONY_HWEQ,
+        ALC271_FIXUP_DMIC,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -14929,7 +14940,11 @@ static const struct alc_fixup alc269_fixups[] = {
                 .v.func = alc269_fixup_hweq,
                 .chained = true,
                 .chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2
-        }
+        },
+        [ALC271_FIXUP_DMIC] = {
+                .type = ALC_FIXUP_FUNC,
+                .v.func = alc271_fixup_dmic,
+        },
 };
 
 static struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -14938,6 +14953,7 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
         SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
         SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
         SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
+        SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
         SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
         SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
         SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
@@ -18782,6 +18798,8 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
                       ALC662_3ST_6ch_DIG),
         SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
         SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
+        SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
+                      ALC662_3ST_6ch_DIG),
         SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13),
         SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
         SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA),
@@ -19455,7 +19473,7 @@ enum {
         ALC662_FIXUP_IDEAPAD,
         ALC272_FIXUP_MARIO,
         ALC662_FIXUP_CZC_P10T,
-        ALC662_FIXUP_GIGABYTE,
+        ALC662_FIXUP_SKU_IGNORE,
 };
 
 static const struct alc_fixup alc662_fixups[] = {
@@ -19484,20 +19502,17 @@ static const struct alc_fixup alc662_fixups[] = {
                         {}
                 }
         },
-        [ALC662_FIXUP_GIGABYTE] = {
-                .type = ALC_FIXUP_PINS,
-                .v.pins = (const struct alc_pincfg[]) {
-                        { 0x14, 0x1114410 }, /* set as speaker */
-                        { }
-                }
+        [ALC662_FIXUP_SKU_IGNORE] = {
+                .type = ALC_FIXUP_SKU,
+                .v.sku = ALC_FIXUP_SKU_IGNORE,
         },
 };
 
 static struct snd_pci_quirk alc662_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
+        SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
         SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
         SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
-        SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", ALC662_FIXUP_GIGABYTE),
         SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
         SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
         SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 1371b57c11e8..0997031c48d2 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1292,14 +1292,18 @@ static void notify_aa_path_ctls(struct hda_codec *codec)
 {
         int i;
         struct snd_ctl_elem_id id;
-        const char *labels[] = {"Mic", "Front Mic", "Line"};
+        const char *labels[] = {"Mic", "Front Mic", "Line", "Rear Mic"};
+        struct snd_kcontrol *ctl;
 
         memset(&id, 0, sizeof(id));
         id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
         for (i = 0; i < ARRAY_SIZE(labels); i++) {
                 sprintf(id.name, "%s Playback Volume", labels[i]);
-                snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE,
-                               &id);
+                ctl = snd_hda_find_mixer_ctl(codec, id.name);
+                if (ctl)
+                        snd_ctl_notify(codec->bus->card,
+                                        SNDRV_CTL_EVENT_MASK_VALUE,
+                                        &ctl->id);
         }
 }
 
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index f7cd346fd727..f5ccdbf7ebc6 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -308,8 +308,6 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec)
         snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes,
                 ARRAY_SIZE(jz4740_codec_dapm_routes));
 
-        snd_soc_dapm_new_widgets(codec);
-
         jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
         return 0;
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index a54d2a5b28f6..4d9fb279e146 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -927,7 +927,7 @@ static struct platform_driver sn95031_codec_driver = {
                 .owner          = THIS_MODULE,
         },
         .probe          = sn95031_device_probe,
-        .remove         = sn95031_device_remove,
+        .remove         = __devexit_p(sn95031_device_remove),
 };
 
 static int __init sn95031_init(void)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ae1cadfae84c..f52b623bb692 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -247,8 +247,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re
         case WM8903_REVISION_NUMBER:
         case WM8903_INTERRUPT_STATUS_1:
         case WM8903_WRITE_SEQUENCER_4:
-        case WM8903_POWER_MANAGEMENT_3:
-        case WM8903_POWER_MANAGEMENT_2:
         case WM8903_DC_SERVO_READBACK_1:
         case WM8903_DC_SERVO_READBACK_2:
         case WM8903_DC_SERVO_READBACK_3:
@@ -875,34 +873,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0,
 SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
                    right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
 
-SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
-                   4, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
+SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2,
+                   1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2,
                    0, 0, NULL, 0),
 
-SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
+SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 1, 0,
                    NULL, 0),
-SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
+SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 0, 0,
                    NULL, 0),
 
 SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0),
 SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA", 1, WM8903_ANALOGUE_HP_0, 4, 0, NULL, 0),
 SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0),
 SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA", 1, WM8903_ANALOGUE_HP_0, 0, 0, NULL, 0),
 
 SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0,
                    NULL, 0),
 SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0,
                    NULL, 0),
-SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+                   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
                    NULL, 0),
 SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0,
                    NULL, 0),
 SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0,
                    NULL, 0),
-SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+                   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
                    NULL, 0),
 
 SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0),
@@ -1037,10 +1041,14 @@ static const struct snd_soc_dapm_route intercon[] = {
         { "Left Speaker PGA", NULL, "Left Speaker Mixer" },
         { "Right Speaker PGA", NULL, "Right Speaker Mixer" },
 
-        { "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" },
-        { "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" },
-        { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" },
-        { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" },
+        { "HPL_ENA", NULL, "Left Headphone Output PGA" },
+        { "HPR_ENA", NULL, "Right Headphone Output PGA" },
+        { "HPL_ENA_DLY", NULL, "HPL_ENA" },
+        { "HPR_ENA_DLY", NULL, "HPR_ENA" },
+        { "LINEOUTL_ENA", NULL, "Left Line Output PGA" },
+        { "LINEOUTR_ENA", NULL, "Right Line Output PGA" },
+        { "LINEOUTL_ENA_DLY", NULL, "LINEOUTL_ENA" },
+        { "LINEOUTR_ENA_DLY", NULL, "LINEOUTR_ENA" },
 
         { "HPL_DCS", NULL, "DCS Master" },
         { "HPR_DCS", NULL, "DCS Master" },
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3290333b2bb9..84e1bd1d2822 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3261,20 +3261,36 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
         wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
         /* Latch volume updates (right only; we always do left then right). */
+        snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME,
+                            WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU);
         snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME,
                             WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU);
+        snd_soc_update_bits(codec, WM8994_AIF1_DAC2_LEFT_VOLUME,
+                            WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU);
         snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME,
                             WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU);
+        snd_soc_update_bits(codec, WM8994_AIF2_DAC_LEFT_VOLUME,
+                            WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU);
         snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME,
                             WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU);
+        snd_soc_update_bits(codec, WM8994_AIF1_ADC1_LEFT_VOLUME,
+                            WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU);
         snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME,
                             WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU);
+        snd_soc_update_bits(codec, WM8994_AIF1_ADC2_LEFT_VOLUME,
+                            WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU);
         snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME,
                             WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU);
+        snd_soc_update_bits(codec, WM8994_AIF2_ADC_LEFT_VOLUME,
+                            WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU);
         snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME,
                             WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU);
+        snd_soc_update_bits(codec, WM8994_DAC1_LEFT_VOLUME,
+                            WM8994_DAC1_VU, WM8994_DAC1_VU);
         snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME,
                             WM8994_DAC1_VU, WM8994_DAC1_VU);
+        snd_soc_update_bits(codec, WM8994_DAC2_LEFT_VOLUME,
+                            WM8994_DAC2_VU, WM8994_DAC2_VU);
         snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME,
                             WM8994_DAC2_VU, WM8994_DAC2_VU);
 
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 7b6b3c18e299..4005e9af5d61 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -740,12 +740,12 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 
         { "SPKL", "Input Switch", "MIXINL" },
         { "SPKL", "IN1LP Switch", "IN1LP" },
-        { "SPKL", "Output Switch", "Left Output Mixer" },
+        { "SPKL", "Output Switch", "Left Output PGA" },
         { "SPKL", NULL, "TOCLK" },
 
         { "SPKR", "Input Switch", "MIXINR" },
         { "SPKR", "IN1RP Switch", "IN1RP" },
-        { "SPKR", "Output Switch", "Right Output Mixer" },
+        { "SPKR", "Output Switch", "Right Output PGA" },
         { "SPKR", NULL, "TOCLK" },
 
         { "SPKL Boost", "Direct Voice Switch", "Direct Voice" },
@@ -767,8 +767,8 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
         { "SPKOUTRP", NULL, "SPKR Driver" },
         { "SPKOUTRN", NULL, "SPKR Driver" },
 
-        { "Left Headphone Mux", "Mixer", "Left Output Mixer" },
-        { "Right Headphone Mux", "Mixer", "Right Output Mixer" },
+        { "Left Headphone Mux", "Mixer", "Left Output PGA" },
+        { "Right Headphone Mux", "Mixer", "Right Output PGA" },
 
         { "Headphone PGA", NULL, "Left Headphone Mux" },
         { "Headphone PGA", NULL, "Right Headphone Mux" },
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index a5af834c8ef5..4ddc6d3b6678 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -434,17 +434,21 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
                 mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
                 mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
 
-                mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26));
+                mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
+                                ACLKX | AHCLKX | AFSX);
                 break;
         case SND_SOC_DAIFMT_CBM_CFS:
                 /* codec is clock master and frame slave */
-                mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+                mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
                 mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
 
-                mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+                mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
                 mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
 
-                mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 26));
+                mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG,
+                                ACLKX | ACLKR);
+                mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
+                                AFSX | AFSR);
                 break;
         case SND_SOC_DAIFMT_CBM_CFM:
                 /* codec is clock and frame master */
@@ -454,7 +458,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
                 mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
                 mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
 
-                mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26));
+                mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG,
+                                ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR);
                 break;
 
         default:
@@ -644,7 +649,7 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
                 mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask);
                 mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD);
 
-                if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32))
+                if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
                         mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG,
                                         FSXMOD(dev->tdm_slots), FSXMOD(0x1FF));
                 else
@@ -660,7 +665,7 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
                                 AHCLKRE);
                 mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask);
 
-                if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32))
+                if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
                         mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG,
                                         FSRMOD(dev->tdm_slots), FSRMOD(0x1FF));
                 else
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index b2e9198a983a..d567c322a2fb 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -116,18 +116,20 @@ struct snd_soc_dai_driver sst_platform_dai[] = {
 static inline void sst_set_stream_status(struct sst_runtime_stream *stream,
                                         int state)
 {
-        spin_lock(&stream->status_lock);
+        unsigned long flags;
+        spin_lock_irqsave(&stream->status_lock, flags);
         stream->stream_status = state;
-        spin_unlock(&stream->status_lock);
+        spin_unlock_irqrestore(&stream->status_lock, flags);
 }
 
 static inline int sst_get_stream_status(struct sst_runtime_stream *stream)
 {
         int state;
+        unsigned long flags;
 
-        spin_lock(&stream->status_lock);
+        spin_lock_irqsave(&stream->status_lock, flags);
         state = stream->stream_status;
-        spin_unlock(&stream->status_lock);
+        spin_unlock_irqrestore(&stream->status_lock, flags);
         return state;
 }
 
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index b5922984eac6..99054cf1f68f 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -65,14 +65,6 @@ config SND_OMAP_SOC_OVERO
           Say Y if you want to add support for SoC audio on the
           Gumstix Overo or CompuLab CM-T35
 
-config SND_OMAP_SOC_OMAP2EVM
-        tristate "SoC Audio support for OMAP2EVM board"
-        depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP2EVM
-        select SND_OMAP_SOC_MCBSP
-        select SND_SOC_TWL4030
-        help
-          Say Y if you want to add support for SoC audio on the omap2evm board.
-
 config SND_OMAP_SOC_OMAP3EVM
         tristate "SoC Audio support for OMAP3EVM board"
         depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index ba9fc650db28..6c2c87eed5bb 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -13,7 +13,6 @@ snd-soc-rx51-objs := rx51.o
 snd-soc-ams-delta-objs := ams-delta.o
 snd-soc-osk5912-objs := osk5912.o
 snd-soc-overo-objs := overo.o
-snd-soc-omap2evm-objs := omap2evm.o
 snd-soc-omap3evm-objs := omap3evm.o
 snd-soc-am3517evm-objs := am3517evm.o
 snd-soc-sdp3430-objs := sdp3430.o
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
deleted file mode 100644
index 29b60d6796e7..000000000000
--- a/sound/soc/omap/omap2evm.c
+++ /dev/null
@@ -1,139 +0,0 @@
-/*
- * omap2evm.c  --  SoC audio machine driver for omap2evm board
- *
- * Author: Arun KS <arunks@mistralsolutions.com>
- *
- * Based on sound/soc/omap/overo.c by Steve Sakoman
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License
- * version 2 as published by the Free Software Foundation.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-
-#include <linux/clk.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <mach/hardware.h>
-#include <mach/gpio.h>
-#include <plat/mcbsp.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-static int omap2evm_hw_params(struct snd_pcm_substream *substream,
-        struct snd_pcm_hw_params *params)
-{
-        struct snd_soc_pcm_runtime *rtd = substream->private_data;
-        struct snd_soc_dai *codec_dai = rtd->codec_dai;
-        struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-        int ret;
-
-        /* Set codec DAI configuration */
-        ret = snd_soc_dai_set_fmt(codec_dai,
-                                  SND_SOC_DAIFMT_I2S |
-                                  SND_SOC_DAIFMT_NB_NF |
-                                  SND_SOC_DAIFMT_CBM_CFM);
-        if (ret < 0) {
-                printk(KERN_ERR "can't set codec DAI configuration\n");
-                return ret;
-        }
-
-        /* Set cpu DAI configuration */
-        ret = snd_soc_dai_set_fmt(cpu_dai,
-                                  SND_SOC_DAIFMT_I2S |
-                                  SND_SOC_DAIFMT_NB_NF |
-                                  SND_SOC_DAIFMT_CBM_CFM);
-        if (ret < 0) {
-                printk(KERN_ERR "can't set cpu DAI configuration\n");
-                return ret;
-        }
-
-        /* Set the codec system clock for DAC and ADC */
-        ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
-                                            SND_SOC_CLOCK_IN);
-        if (ret < 0) {
-                printk(KERN_ERR "can't set codec system clock\n");
-                return ret;
-        }
-
-        return 0;
-}
-
-static struct snd_soc_ops omap2evm_ops = {
-        .hw_params = omap2evm_hw_params,
-};
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link omap2evm_dai = {
-        .name = "TWL4030",
-        .stream_name = "TWL4030",
-        .cpu_dai_name = "omap-mcbsp-dai.1",
-        .codec_dai_name = "twl4030-hifi",
-        .platform_name = "omap-pcm-audio",
-        .codec_name = "twl4030-codec",
-        .ops = &omap2evm_ops,
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_omap2evm = {
-        .name = "omap2evm",
-        .dai_link = &omap2evm_dai,
-        .num_links = 1,
-};
-
-static struct platform_device *omap2evm_snd_device;
-
-static int __init omap2evm_soc_init(void)
-{
-        int ret;
-
-        if (!machine_is_omap2evm())
-                return -ENODEV;
-        printk(KERN_INFO "omap2evm SoC init\n");
-
-        omap2evm_snd_device = platform_device_alloc("soc-audio", -1);
-        if (!omap2evm_snd_device) {
-                printk(KERN_ERR "Platform device allocation failed\n");
-                return -ENOMEM;
-        }
-
-        platform_set_drvdata(omap2evm_snd_device, &snd_soc_omap2evm);
-
-        ret = platform_device_add(omap2evm_snd_device);
-        if (ret)
-                goto err1;
-
-        return 0;
-
-err1:
-        printk(KERN_ERR "Unable to add platform device\n");
-        platform_device_put(omap2evm_snd_device);
-
-        return ret;
-}
-module_init(omap2evm_soc_init);
-
-static void __exit omap2evm_soc_exit(void)
-{
-        platform_device_unregister(omap2evm_snd_device);
-}
-module_exit(omap2evm_soc_exit);
-
-MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
-MODULE_DESCRIPTION("ALSA SoC omap2evm");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c
index f6b3a3ce5919..0e80daee8b6f 100644
--- a/sound/soc/samsung/goni_wm8994.c
+++ b/sound/soc/samsung/goni_wm8994.c
@@ -236,18 +236,18 @@ static struct snd_soc_dai_link goni_dai[] = {
         .name = "WM8994",
         .stream_name = "WM8994 HiFi",
         .cpu_dai_name = "samsung-i2s.0",
-        .codec_dai_name = "wm8994-hifi",
+        .codec_dai_name = "wm8994-aif1",
         .platform_name = "samsung-audio",
-        .codec_name = "wm8994-codec.0-0x1a",
+        .codec_name = "wm8994-codec.0-001a",
         .init = goni_wm8994_init,
         .ops = &goni_hifi_ops,
 }, {
         .name = "WM8994 Voice",
         .stream_name = "Voice",
         .cpu_dai_name = "goni-voice-dai",
-        .codec_dai_name = "wm8994-voice",
+        .codec_dai_name = "wm8994-aif2",
         .platform_name = "samsung-audio",
-        .codec_name = "wm8994-codec.0-0x1a",
+        .codec_name = "wm8994-codec.0-001a",
         .ops = &goni_voice_ops,
 },
 };
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index 38aac7d57a59..9c7e8b48aed6 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -350,8 +350,8 @@ static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai,
         ctl = readl(regs + S3C_PCM_CTL);
 
         switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-        case SND_SOC_DAIFMT_NB_NF:
-                /* Nothing to do, NB_NF by default */
+        case SND_SOC_DAIFMT_IB_NF:
+                /* Nothing to do, IB_NF by default */
                 break;
         default:
                 dev_err(pcm->dev, "Unsupported clock inversion!\n");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 0c9997e2d8c0..23c0e83d4c19 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1200,10 +1200,11 @@ static int fsi_probe(struct platform_device *pdev)
         master->fsib.master     = master;
 
         pm_runtime_enable(&pdev->dev);
-        pm_runtime_resume(&pdev->dev);
         dev_set_drvdata(&pdev->dev, master);
 
+        pm_runtime_get_sync(&pdev->dev);
         fsi_soft_all_reset(master);
+        pm_runtime_put_sync(&pdev->dev);
 
         ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
                           id_entry->name, master);
@@ -1218,8 +1219,17 @@ static int fsi_probe(struct platform_device *pdev)
                 goto exit_free_irq;
         }
 
-        return snd_soc_register_dais(&pdev->dev, fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai));
+        ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai,
+                                    ARRAY_SIZE(fsi_soc_dai));
+        if (ret < 0) {
+                dev_err(&pdev->dev, "cannot snd dai register\n");
+                goto exit_snd_soc;
+        }
+
+        return ret;
 
+exit_snd_soc:
+        snd_soc_unregister_platform(&pdev->dev);
 exit_free_irq:
         free_irq(irq, master);
 exit_iounmap:
@@ -1238,12 +1248,11 @@ static int fsi_remove(struct platform_device *pdev)
 
         master = dev_get_drvdata(&pdev->dev);
 
-        snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
-        snd_soc_unregister_platform(&pdev->dev);
-
+        free_irq(master->irq, master);
         pm_runtime_disable(&pdev->dev);
 
-        free_irq(master->irq, master);
+        snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
+        snd_soc_unregister_platform(&pdev->dev);
 
         iounmap(master->base);
         kfree(master);
@@ -1321,3 +1330,4 @@ module_exit(fsi_mobile_exit);
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip FSI audio driver");
 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
+MODULE_ALIAS("platform:fsi-pcm-audio");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b76b74db0968..d8562ce4de7a 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -629,6 +629,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
                         runtime->hw.rates |= codec_dai_drv->capture.rates;
         }
 
+        ret = -EINVAL;
         snd_pcm_limit_hw_rates(runtime);
         if (!runtime->hw.rates) {
                 printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
@@ -640,7 +641,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
                         codec_dai->name, cpu_dai->name);
                 goto config_err;
         }
-        if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
+        if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
+            runtime->hw.channels_min > runtime->hw.channels_max) {
                 printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
                                 codec_dai->name, cpu_dai->name);
                 goto config_err;
@@ -2060,6 +2062,7 @@ const struct dev_pm_ops snd_soc_pm_ops = {
         .resume = snd_soc_resume,
         .poweroff = snd_soc_poweroff,
 };
+EXPORT_SYMBOL_GPL(snd_soc_pm_ops);
 
 /* ASoC platform driver */
 static struct platform_driver soc_driver = {
diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c
index 8585957477eb..556a57133925 100644
--- a/sound/soc/tegra/harmony.c
+++ b/sound/soc/tegra/harmony.c
@@ -370,6 +370,7 @@ static struct platform_driver tegra_snd_harmony_driver = {
         .driver = {
                 .name = DRV_NAME,
                 .owner = THIS_MODULE,
+                .pm = &snd_soc_pm_ops,
         },
         .probe = tegra_snd_harmony_probe,
         .remove = __devexit_p(tegra_snd_harmony_remove),
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 5b792d2c8061..f079b5e2ab28 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -176,9 +176,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
                         if (!rate)
                                 continue;
                         /* C-Media CM6501 mislabels its 96 kHz altsetting */
+                        /* Terratec Aureon 7.1 USB C-Media 6206, too */
                         if (rate == 48000 && nr_rates == 1 &&
                             (chip->usb_id == USB_ID(0x0d8c, 0x0201) ||
-                             chip->usb_id == USB_ID(0x0d8c, 0x0102)) &&
+                             chip->usb_id == USB_ID(0x0d8c, 0x0102) ||
+                             chip->usb_id == USB_ID(0x0ccd, 0x00b1)) &&
                             fp->altsetting == 5 && fp->maxpacksize == 392)
                                 rate = 96000;
                         /* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index ec07e62e53f3..1b94ec3a3368 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -533,6 +533,7 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
 
         case USB_ID(0x0d8c, 0x0102):
                 /* C-Media CM6206 / CM106-Like Sound Device */
+        case USB_ID(0x0ccd, 0x00b1): /* Terratec Aureon 7.1 USB */
                 return snd_usb_cm6206_boot_quirk(dev);
 
         case USB_ID(0x133e, 0x0815):