Merge remote branch 'origin/master' into next

28 files changed, 150 insertions, 99 deletions

diff --git a/sound/core/misc.c b/sound/core/misc.c
index 2c41825c836e..eb9fe2e1d291 100644
--- a/sound/core/misc.c
+++ b/sound/core/misc.c
@@ -58,26 +58,6 @@ static const char *sanity_file_name(const char *path)
         else
                 return path;
 }
-
-/* print file and line with a certain printk prefix */
-static int print_snd_pfx(unsigned int level, const char *path, int line,
-                         const char *format)
-{
-        const char *file = sanity_file_name(path);
-        char tmp[] = "<0>";
-        const char *pfx = level ? KERN_DEBUG : KERN_DEFAULT;
-        int ret = 0;
-
-        if (format[0] == '<' && format[2] == '>') {
-                tmp[1] = format[1];
-                pfx = tmp;
-                ret = 1;
-        }
-        printk("%sALSA %s:%d: ", pfx, file, line);
-        return ret;
-}
-#else
-#define print_snd_pfx(level, path, line, format)        0
 #endif
 
 #if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
@@ -85,15 +65,29 @@ void __snd_printk(unsigned int level, const char *path, int line,
                   const char *format, ...)
 {
         va_list args;
-        
+#ifdef CONFIG_SND_VERBOSE_PRINTK
+        struct va_format vaf;
+        char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV";
+#endif
+
 #ifdef CONFIG_SND_DEBUG 
         if (debug < level)
                 return;
 #endif
+
         va_start(args, format);
-        if (print_snd_pfx(level, path, line, format))
-                format += 3; /* skip the printk level-prefix */
+#ifdef CONFIG_SND_VERBOSE_PRINTK
+        vaf.fmt = format;
+        vaf.va = &args;
+        if (format[0] == '<' && format[2] == '>') {
+                memcpy(verbose_fmt, format, 3);
+                vaf.fmt = format + 3;
+        } else if (level)
+                memcpy(verbose_fmt, KERN_DEBUG, 3);
+        printk(verbose_fmt, sanity_file_name(path), line, &vaf);
+#else
         vprintk(format, args);
+#endif
         va_end(args);
 }
 EXPORT_SYMBOL_GPL(__snd_printk);
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 86ee16ca365e..440030818db7 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -209,6 +209,7 @@ static void isight_packet(struct fw_iso_context *context, u32 cycle,
                 isight->packet_index = -1;
                 return;
         }
+        fw_iso_context_queue_flush(isight->context);
 
         if (++index >= QUEUE_LENGTH)
                 index = 0;
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 2ca6f4f85b41..e3569bdd3b64 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -27,7 +27,6 @@
 #include "hpioctl.h"
 
 #include <linux/pci.h>
-#include <linux/version.h>
 #include <linux/init.h>
 #include <linux/jiffies.h>
 #include <linux/slab.h>
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 5e619a84da06..15f0161ce4a2 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1440,6 +1440,14 @@ static struct snd_emu_chip_details emu_chip_details[] = {
          .ca0102_chip = 1,
          .spk71 = 1,
          .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */
+        /* EMU0404 PCIe */
+        {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40051102,
+         .driver = "Audigy2", .name = "E-mu 0404 PCIe [MAEM8984]",
+         .id = "EMU0404",
+         .emu10k2_chip = 1,
+         .ca0108_chip = 1,
+         .spk71 = 1,
+         .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 PCIe ver_03 */
         /* Note that all E-mu cards require kernel 2.6 or newer. */
         {.vendor = 0x1102, .device = 0x0008,
          .driver = "Audigy2", .name = "SB Audigy 2 Value [Unknown]",
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index f1de1bac042c..55f0647458c7 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -50,7 +50,12 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable);
 int snd_hda_attach_beep_device(struct hda_codec *codec, int nid);
 void snd_hda_detach_beep_device(struct hda_codec *codec);
 #else
-#define snd_hda_attach_beep_device(...)         0
-#define snd_hda_detach_beep_device(...)
+static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
+{
+        return 0;
+}
+static inline void snd_hda_detach_beep_device(struct hda_codec *codec)
+{
+}
 #endif
 #endif
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3e6b9a8539c2..694b9daf691f 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3102,6 +3102,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
         SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
         SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
         SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
+        SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
         {}
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7a4e10002f56..d21191dcfe88 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1141,6 +1141,13 @@ static void update_speakers(struct hda_codec *codec)
         struct alc_spec *spec = codec->spec;
         int on;
 
+        /* Control HP pins/amps depending on master_mute state;
+         * in general, HP pins/amps control should be enabled in all cases,
+         * but currently set only for master_mute, just to be safe
+         */
+        do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
+                    spec->autocfg.hp_pins, spec->master_mute, true);
+
         if (!spec->automute)
                 on = 0;
         else
@@ -4876,7 +4883,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
         SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
         SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
-        SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST),
         SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
         SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
         SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
@@ -6201,11 +6207,6 @@ static const struct snd_kcontrol_new alc260_input_mixer[] = {
 /* update HP, line and mono out pins according to the master switch */
 static void alc260_hp_master_update(struct hda_codec *codec)
 {
-        struct alc_spec *spec = codec->spec;
-
-        /* change HP pins */
-        do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
-                    spec->autocfg.hp_pins, spec->master_mute, true);
         update_speakers(codec);
 }
 
@@ -11924,7 +11925,7 @@ static const struct hda_verb alc262_nec_verbs[] = {
  *  0x1b = port replicator headphone out
  */
 
-#define ALC_HP_EVENT    0x37
+#define ALC_HP_EVENT    ALC880_HP_EVENT
 
 static const struct hda_verb alc262_fujitsu_unsol_verbs[] = {
         {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
@@ -12598,6 +12599,7 @@ static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
  */
 enum {
         PINFIX_FSC_H270,
+        PINFIX_HP_Z200,
 };
 
 static const struct alc_fixup alc262_fixups[] = {
@@ -12610,9 +12612,17 @@ static const struct alc_fixup alc262_fixups[] = {
                         { }
                 }
         },
+        [PINFIX_HP_Z200] = {
+                .type = ALC_FIXUP_PINS,
+                .v.pins = (const struct alc_pincfg[]) {
+                        { 0x16, 0x99130120 }, /* internal speaker */
+                        { }
+                }
+        },
 };
 
 static const struct snd_pci_quirk alc262_fixup_tbl[] = {
+        SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", PINFIX_HP_Z200),
         SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270),
         {}
 };
@@ -12729,6 +12739,8 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = {
                            ALC262_HP_BPC),
         SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
                            ALC262_HP_BPC),
+        SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200",
+                           ALC262_AUTO),
         SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
                            ALC262_HP_BPC),
         SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
@@ -13314,9 +13326,8 @@ static void alc268_acer_lc_setup(struct hda_codec *codec)
         struct alc_spec *spec = codec->spec;
         spec->autocfg.hp_pins[0] = 0x15;
         spec->autocfg.speaker_pins[0] = 0x14;
-        spec->automute_mixer_nid[0] = 0x0f;
         spec->automute = 1;
-        spec->automute_mode = ALC_AUTOMUTE_MIXER;
+        spec->automute_mode = ALC_AUTOMUTE_AMP;
         spec->ext_mic.pin = 0x18;
         spec->ext_mic.mux_idx = 0;
         spec->int_mic.pin = 0x12;
@@ -13860,6 +13871,7 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
                                                 ALC268_ACER_ASPIRE_ONE),
         SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
+        SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO),
         SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
                         "Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
         /* almost compatible with toshiba but with optional digital outs;
@@ -13870,7 +13882,6 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
         SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
         SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
-        SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
         SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
         {}
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 605c99e1e520..f43bb0eaed8b 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -745,12 +745,23 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
         struct via_spec *spec = codec->spec;
         hda_nid_t nid = kcontrol->private_value;
         unsigned int pinsel = ucontrol->value.enumerated.item[0];
+        unsigned int parm0, parm1;
         /* Get Independent Mode index of headphone pin widget */
         spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel
                 ? 1 : 0;
-        if (spec->codec_type == VT1718S)
+        if (spec->codec_type == VT1718S) {
                 snd_hda_codec_write(codec, nid, 0,
                                     AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0);
+                /* Set correct mute switch for MW3 */
+                parm0 = spec->hp_independent_mode ?
+                               AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0);
+                parm1 = spec->hp_independent_mode ?
+                               AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1);
+                snd_hda_codec_write(codec, 0x1b, 0,
+                                    AC_VERB_SET_AMP_GAIN_MUTE, parm0);
+                snd_hda_codec_write(codec, 0x1b, 0,
+                                    AC_VERB_SET_AMP_GAIN_MUTE, parm1);
+        }
         else
                 snd_hda_codec_write(codec, nid, 0,
                                     AC_VERB_SET_CONNECT_SEL, pinsel);
@@ -832,10 +843,13 @@ static int via_hp_build(struct hda_codec *codec)
         knew->subdevice = HDA_SUBDEV_NID_FLAG | nid;
         knew->private_value = nid;
 
-        knew = via_clone_control(spec, &via_hp_mixer[1]);
-        if (knew == NULL)
-                return -ENOMEM;
-        knew->subdevice = side_mute_channel(spec);
+        nid = side_mute_channel(spec);
+        if (nid) {
+                knew = via_clone_control(spec, &via_hp_mixer[1]);
+                if (knew == NULL)
+                        return -ENOMEM;
+                knew->subdevice = nid;
+        }
 
         return 0;
 }
@@ -4280,9 +4294,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
         {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
         {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
         {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
-
-        /* Setup default input of Front HP to MW9 */
-        {0x28, AC_VERB_SET_CONNECT_SEL, 0x1},
         /* PW9 PW10 Output enable */
         {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
         {0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
@@ -4291,10 +4302,10 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
         /* Enable Boost Volume backdoor */
         {0x1, 0xf88, 0x8},
         /* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */
-        {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
         {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
         {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-        {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
         {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
         {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
         {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
@@ -4304,8 +4315,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
         /* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */
         {0x34, AC_VERB_SET_CONNECT_SEL, 0x2},
         {0x35, AC_VERB_SET_CONNECT_SEL, 0x1},
-        /* Unmute MW4's index 0 */
-        {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
         { }
 };
 
@@ -4453,6 +4462,19 @@ static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec,
                         if (err < 0)
                                 return err;
                 } else if (i == AUTO_SEQ_FRONT) {
+                        /* add control to mixer index 0 */
+                        err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+                                              "Master Front Playback Volume",
+                                              HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
+                                                                  HDA_INPUT));
+                        if (err < 0)
+                                return err;
+                        err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+                                              "Master Front Playback Switch",
+                                              HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
+                                                                  HDA_INPUT));
+                        if (err < 0)
+                                return err;
                         /* Front */
                         sprintf(name, "%s Playback Volume", chname[i]);
                         err = via_add_control(
diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
index 34b24286d279..2692e5ae5f2d 100644
--- a/sound/pci/lola/lola.c
+++ b/sound/pci/lola/lola.c
@@ -445,7 +445,7 @@ static void lola_reset_setups(struct lola *chip)
         lola_setup_all_analog_gains(chip, PLAY, false); /* output, update */
 }
 
-static int lola_parse_tree(struct lola *chip)
+static int __devinit lola_parse_tree(struct lola *chip)
 {
         unsigned int val;
         int nid, err;
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 949691a876d3..3f08afc0f0d3 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -521,6 +521,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #define HDSPM_DMA_AREA_KILOBYTES (HDSPM_DMA_AREA_BYTES/1024)
 
 /* revisions >= 230 indicate AES32 card */
+#define HDSPM_MADI_OLD_REV      207
 #define HDSPM_MADI_REV          210
 #define HDSPM_RAYDAT_REV        211
 #define HDSPM_AIO_REV           212
@@ -1143,7 +1144,7 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm)
 
                 /* if wordclock has synced freq and wordclock is valid */
                 if ((status2 & HDSPM_wcLock) != 0 &&
-                                (status & HDSPM_SelSyncRef0) == 0) {
+                                (status2 & HDSPM_SelSyncRef0) == 0) {
 
                         rate_bits = status2 & HDSPM_wcFreqMask;
 
@@ -1639,12 +1640,14 @@ static int snd_hdspm_midi_input_read (struct hdspm_midi *hmidi)
                 }
         }
         hmidi->pending = 0;
+        spin_unlock_irqrestore(&hmidi->lock, flags);
 
+        spin_lock_irqsave(&hmidi->hdspm->lock, flags);
         hmidi->hdspm->control_register |= hmidi->ie;
         hdspm_write(hmidi->hdspm, HDSPM_controlRegister,
                     hmidi->hdspm->control_register);
+        spin_unlock_irqrestore(&hmidi->hdspm->lock, flags);
 
-        spin_unlock_irqrestore (&hmidi->lock, flags);
         return snd_hdspm_midi_output_write (hmidi);
 }
 
@@ -6377,6 +6380,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
 
         switch (hdspm->firmware_rev) {
         case HDSPM_MADI_REV:
+        case HDSPM_MADI_OLD_REV:
                 hdspm->io_type = MADI;
                 hdspm->card_name = "RME MADI";
                 hdspm->midiPorts = 3;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 7fbfa051f6e1..eda955b15834 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -848,9 +848,10 @@ int atmel_ssc_set_audio(int ssc_id)
         if (IS_ERR(ssc))
                 pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n",
                         PTR_ERR(ssc));
-        else
+        else {
                 ssc_pdev->dev.parent = &(ssc->pdev->dev);
-        ssc_free(ssc);
+                ssc_free(ssc);
+        }
 
         ret = platform_device_add(ssc_pdev);
         if (ret < 0)
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index ea4951cf5526..f79d1655e035 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -75,7 +75,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
                 .cpu_dai_name = "bfin-tdm.0",
                 .codec_dai_name = "ad1836-hifi",
                 .platform_name = "bfin-tdm-pcm-audio",
-                .codec_name = "ad1836.0",
+                .codec_name = "spi0.4",
                 .ops = &bf5xx_ad1836_ops,
         },
         {
@@ -84,7 +84,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
                 .cpu_dai_name = "bfin-tdm.1",
                 .codec_dai_name = "ad1836-hifi",
                 .platform_name = "bfin-tdm-pcm-audio",
-                .codec_name = "ad1836.0",
+                .codec_name = "spi0.4",
                 .ops = &bf5xx_ad1836_ops,
         },
 };
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index ab63d52e36e1..754c496412bd 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -145,22 +145,22 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
         /* bit size */
         switch (params_format(params)) {
         case SNDRV_PCM_FORMAT_S16_LE:
-                word_len = 3;
+                word_len = AD1836_WORD_LEN_16;
                 break;
         case SNDRV_PCM_FORMAT_S20_3LE:
-                word_len = 1;
+                word_len = AD1836_WORD_LEN_20;
                 break;
         case SNDRV_PCM_FORMAT_S24_LE:
         case SNDRV_PCM_FORMAT_S32_LE:
-                word_len = 0;
+                word_len = AD1836_WORD_LEN_24;
                 break;
         }
 
-        snd_soc_update_bits(codec, AD1836_DAC_CTRL1,
-                AD1836_DAC_WORD_LEN_MASK, word_len);
+        snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK,
+                word_len << AD1836_DAC_WORD_LEN_OFFSET);
 
-        snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
-                AD1836_ADC_WORD_LEN_MASK, word_len);
+        snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK,
+                word_len << AD1836_ADC_WORD_OFFSET);
 
         return 0;
 }
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 845596717fdf..9d6a3f8f8aaf 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -25,6 +25,7 @@
 #define AD1836_DAC_SERFMT_PCK256       (0x4 << 5)
 #define AD1836_DAC_SERFMT_PCK128       (0x5 << 5)
 #define AD1836_DAC_WORD_LEN_MASK       0x18
+#define AD1836_DAC_WORD_LEN_OFFSET     3
 
 #define AD1836_DAC_CTRL2               1
 #define AD1836_DACL1_MUTE              0
@@ -51,6 +52,7 @@
 #define AD1836_ADCL2_MUTE               2
 #define AD1836_ADCR2_MUTE               3
 #define AD1836_ADC_WORD_LEN_MASK       0x30
+#define AD1836_ADC_WORD_OFFSET         5
 #define AD1836_ADC_SERFMT_MASK         (7 << 6)
 #define AD1836_ADC_SERFMT_PCK256       (0x4 << 6)
 #define AD1836_ADC_SERFMT_PCK128       (0x5 << 6)
@@ -60,4 +62,8 @@
 
 #define AD1836_NUM_REGS                16
 
+#define AD1836_WORD_LEN_24 0x0
+#define AD1836_WORD_LEN_20 0x1
+#define AD1836_WORD_LEN_16 0x2
+
 #endif
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 6785688f8806..9a5e67c5a6bd 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -680,20 +680,25 @@ static struct snd_soc_dai_ops wm8804_dai_ops = {
 #define WM8804_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
                         SNDRV_PCM_FMTBIT_S24_LE)
 
+#define WM8804_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+                      SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
+                      SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+                      SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
 static struct snd_soc_dai_driver wm8804_dai = {
         .name = "wm8804-spdif",
         .playback = {
                 .stream_name = "Playback",
                 .channels_min = 2,
                 .channels_max = 2,
-                .rates = SNDRV_PCM_RATE_8000_192000,
+                .rates = WM8804_RATES,
                 .formats = WM8804_FORMATS,
         },
         .capture = {
                 .stream_name = "Capture",
                 .channels_min = 2,
                 .channels_max = 2,
-                .rates = SNDRV_PCM_RATE_8000_192000,
+                .rates = WM8804_RATES,
                 .formats = WM8804_FORMATS,
         },
         .ops = &wm8804_dai_ops,
diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c
index a0b1a7278284..e2ab4fac2819 100644
--- a/sound/soc/codecs/wm8915.c
+++ b/sound/soc/codecs/wm8915.c
@@ -1839,7 +1839,7 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai,
         int old;
 
         /* Disable SYSCLK while we reconfigure */
-        old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1);
+        old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA;
         snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1,
                             WM8915_SYSCLK_ENA, 0);
 
@@ -2038,6 +2038,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
                 break;
         case WM8915_FLL_MCLK2:
                 reg = 1;
+                break;
         case WM8915_FLL_DACLRCLK1:
                 reg = 2;
                 break;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index f90ae427242b..5e05eed96c38 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1999,12 +1999,12 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol,
                 return 0;
 
         /* If the left PGA is enabled hit that VU bit... */
-        if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTL_PGA_ENA)
+        if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA)
                 return snd_soc_write(codec, WM8962_HPOUTL_VOLUME,
                                      reg_cache[WM8962_HPOUTL_VOLUME]);
 
         /* ...otherwise the right.  The VU is stereo. */
-        if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTR_PGA_ENA)
+        if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA)
                 return snd_soc_write(codec, WM8962_HPOUTR_VOLUME,
                                      reg_cache[WM8962_HPOUTR_VOLUME]);
 
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 3c2ee1bb73cd..6af23d06870f 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -13,7 +13,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 15dac0f20cd8..6680c0b4d203 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
          * should allocate a DMA buffer only for the streams that are valid.
          */
 
-        if (dai->driver->playback.channels_min) {
+        if (pcm->streams[0].substream) {
                 ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
                         fsl_dma_hardware.buffer_bytes_max,
                         &pcm->streams[0].substream->dma_buffer);
@@ -320,13 +320,13 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
                 }
         }
 
-        if (dai->driver->capture.channels_min) {
+        if (pcm->streams[1].substream) {
                 ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
                         fsl_dma_hardware.buffer_bytes_max,
                         &pcm->streams[1].substream->dma_buffer);
                 if (ret) {
-                        snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
                         dev_err(card->dev, "can't alloc capture dma buffer\n");
+                        snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
                         return ret;
                 }
         }
@@ -449,7 +449,8 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
         dma_private->ld_buf_phys = ld_buf_phys;
         dma_private->dma_buf_phys = substream->dma_buffer.addr;
 
-        ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private);
+        ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "fsldma-audio",
+                          dma_private);
         if (ret) {
                 dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n",
                         dma_private->irq, ret);
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index d8f130d39dd9..bb699bb55a50 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -11,9 +11,6 @@ menuconfig SND_IMX_SOC
 
 if SND_IMX_SOC
 
-config SND_MXC_SOC_SSI
-        tristate
-
 config SND_MXC_SOC_FIQ
         tristate
 
@@ -24,7 +21,6 @@ config SND_MXC_SOC_WM1133_EV1
         tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
         depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
         select SND_SOC_WM8350
-        select SND_MXC_SOC_SSI
         select SND_MXC_SOC_FIQ
         help
           Enable support for audio on the i.MX31ADS with the WM1133-EV1
@@ -34,7 +30,6 @@ config SND_SOC_MX27VIS_AIC32X4
         tristate "SoC audio support for Visstrim M10 boards"
         depends on MACH_IMX27_VISSTRIM_M10
         select SND_SOC_TVL320AIC32X4
-        select SND_MXC_SOC_SSI
         select SND_MXC_SOC_MX2
         help
           Say Y if you want to add support for SoC audio on Visstrim SM10
@@ -44,7 +39,6 @@ config SND_SOC_PHYCORE_AC97
         tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
         depends on MACH_PCM043 || MACH_PCA100
         select SND_SOC_WM9712
-        select SND_MXC_SOC_SSI
         select SND_MXC_SOC_FIQ
         help
           Say Y if you want to add support for SoC audio on Phytec phyCORE
@@ -57,7 +51,6 @@ config SND_SOC_EUKREA_TLV320
                 || MACH_EUKREA_MBIMXSD35_BASEBOARD \
                 || MACH_EUKREA_MBIMXSD51_BASEBOARD
         select SND_SOC_TLV320AIC23
-        select SND_MXC_SOC_SSI
         select SND_MXC_SOC_FIQ
         help
           Enable I2S based access to the TLV320AIC23B codec attached
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index aab7765f401a..4173b3d87f97 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -337,3 +337,5 @@ static void __exit snd_imx_pcm_exit(void)
         platform_driver_unregister(&imx_pcm_driver);
 }
 module_exit(snd_imx_pcm_exit);
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imx-pcm-audio");
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 5b13feca7537..61fceb09cdb5 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -774,4 +774,4 @@ module_exit(imx_ssi_exit);
 MODULE_AUTHOR("Sascha Hauer, <s.hauer@pengutronix.de>");
 MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface");
 MODULE_LICENSE("GPL");
-
+MODULE_ALIAS("platform:imx-ssi");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 2ce0b2d891d5..fab20a54e863 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -95,14 +95,14 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
         if (!card->dev->coherent_dma_mask)
                 card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-        if (dai->driver->playback.channels_min) {
+        if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
                 ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
                         SNDRV_PCM_STREAM_PLAYBACK);
                 if (ret)
                         goto out;
         }
 
-        if (dai->driver->capture.channels_min) {
+        if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
                 ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
                         SNDRV_PCM_STREAM_CAPTURE);
                 if (ret)
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index ffa09b3b2caa..992a732b5211 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -191,7 +191,7 @@ static inline bool tx_active(struct i2s_dai *i2s)
         if (!i2s)
                 return false;
 
-        active = readl(i2s->addr + I2SMOD);
+        active = readl(i2s->addr + I2SCON);
 
         if (is_secondary(i2s))
                 active &= CON_TXSDMA_ACTIVE;
@@ -223,7 +223,7 @@ static inline bool rx_active(struct i2s_dai *i2s)
         if (!i2s)
                 return false;
 
-        active = readl(i2s->addr + I2SMOD) & CON_RXDMA_ACTIVE;
+        active = readl(i2s->addr + I2SCON) & CON_RXDMA_ACTIVE;
 
         return active ? true : false;
 }
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 06b7b81a1601..039b9532b270 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -409,9 +409,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
         codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
 
         switch (control) {
-        case SND_SOC_CUSTOM:
-                break;
-
         case SND_SOC_I2C:
 #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
                 codec->hw_write = (hw_write_t)i2c_master_send;
@@ -466,6 +463,9 @@ static bool snd_soc_set_cache_val(void *base, unsigned int idx,
 static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx,
                 unsigned int word_size)
 {
+        if (!base)
+                return -1;
+
         switch (word_size) {
         case 1: {
                 const u8 *cache = base;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 776e6f418306..32ab7fc4579a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -350,9 +350,9 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
 }
 
 /* create new dapm mixer control */
-static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
-        struct snd_soc_dapm_widget *w)
+static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
 {
+        struct snd_soc_dapm_context *dapm = w->dapm;
         int i, ret = 0;
         size_t name_len, prefix_len;
         struct snd_soc_dapm_path *path;
@@ -450,9 +450,9 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
 }
 
 /* create new dapm mux control */
-static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
-        struct snd_soc_dapm_widget *w)
+static int dapm_new_mux(struct snd_soc_dapm_widget *w)
 {
+        struct snd_soc_dapm_context *dapm = w->dapm;
         struct snd_soc_dapm_path *path = NULL;
         struct snd_kcontrol *kcontrol;
         struct snd_card *card = dapm->card->snd_card;
@@ -535,8 +535,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
 }
 
 /* create new dapm volume control */
-static int dapm_new_pga(struct snd_soc_dapm_context *dapm,
-        struct snd_soc_dapm_widget *w)
+static int dapm_new_pga(struct snd_soc_dapm_widget *w)
 {
         if (w->num_kcontrols)
                 dev_err(w->dapm->dev,
@@ -1826,13 +1825,13 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
                 case snd_soc_dapm_mixer:
                 case snd_soc_dapm_mixer_named_ctl:
                         w->power_check = dapm_generic_check_power;
-                        dapm_new_mixer(dapm, w);
+                        dapm_new_mixer(w);
                         break;
                 case snd_soc_dapm_mux:
                 case snd_soc_dapm_virt_mux:
                 case snd_soc_dapm_value_mux:
                         w->power_check = dapm_generic_check_power;
-                        dapm_new_mux(dapm, w);
+                        dapm_new_mux(w);
                         break;
                 case snd_soc_dapm_adc:
                 case snd_soc_dapm_aif_out:
@@ -1845,7 +1844,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
                 case snd_soc_dapm_pga:
                 case snd_soc_dapm_out_drv:
                         w->power_check = dapm_generic_check_power;
-                        dapm_new_pga(dapm, w);
+                        dapm_new_pga(w);
                         break;
                 case snd_soc_dapm_input:
                 case snd_soc_dapm_output:
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index a91719d5918b..1e3ae3327dd3 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -270,7 +270,6 @@ static int usb6fire_fw_ezusb_upload(
         data = 0x00; /* resume ezusb cpu */
         ret = usb6fire_fw_ezusb_write(device, 0xa0, 0xe600, &data, 1);
         if (ret < 0) {
-                release_firmware(fw);
                 snd_printk(KERN_ERR PREFIX "unable to upload ezusb "
                                 "firmware %s: end message.\n", fwname);
                 return ret;
diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c
index b137b25865cc..d144cdb2f159 100644
--- a/sound/usb/6fire/pcm.c
+++ b/sound/usb/6fire/pcm.c
@@ -395,12 +395,12 @@ static int usb6fire_pcm_open(struct snd_pcm_substream *alsa_sub)
         alsa_rt->hw = pcm_hw;
 
         if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-                if (rt->rate >= 0)
+                if (rt->rate < ARRAY_SIZE(rates))
                         alsa_rt->hw.rates = rates_alsaid[rt->rate];
                 alsa_rt->hw.channels_max = OUT_N_CHANNELS;
                 sub = &rt->playback;
         } else if (alsa_sub->stream == SNDRV_PCM_STREAM_CAPTURE) {
-                if (rt->rate >= 0)
+                if (rt->rate < ARRAY_SIZE(rates))
                         alsa_rt->hw.rates = rates_alsaid[rt->rate];
                 alsa_rt->hw.channels_max = IN_N_CHANNELS;
                 sub = &rt->capture;