26 files changed, 289 insertions, 115 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 24d3013c0231..7c1fc64cb53d 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -50,7 +50,11 @@ static void aaci_ac97_select_codec(struct aaci *aaci, struct snd_ac97 *ac97)
         if (v & SLFR_1RXV)
                 readl(aaci->base + AACI_SL1RX);
 
-        writel(maincr, aaci->base + AACI_MAINCR);
+        if (maincr != readl(aaci->base + AACI_MAINCR)) {
+                writel(maincr, aaci->base + AACI_MAINCR);
+                readl(aaci->base + AACI_MAINCR);
+                udelay(1);
+        }
 }
 
 /*
@@ -993,6 +997,8 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
          * disabling the channel doesn't clear the FIFO.
          */
         writel(aaci->maincr & ~MAINCR_IE, aaci->base + AACI_MAINCR);
+        readl(aaci->base + AACI_MAINCR);
+        udelay(1);
         writel(aaci->maincr, aaci->base + AACI_MAINCR);
 
         /*
diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index 7730575bfadd..b8b31c433d64 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -45,12 +45,13 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
 {
         struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt);
         struct snd_timer *t = stime->timer;
+        unsigned long oruns;
 
         if (!atomic_read(&stime->running))
                 return HRTIMER_NORESTART;
 
-        hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
-        snd_timer_interrupt(stime->timer, t->sticks);
+        oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
+        snd_timer_interrupt(stime->timer, t->sticks * oruns);
 
         if (!atomic_read(&stime->running))
                 return HRTIMER_NORESTART;
@@ -104,7 +105,7 @@ static int snd_hrtimer_stop(struct snd_timer *t)
 }
 
 static struct snd_timer_hardware hrtimer_hw = {
-        .flags =        SNDRV_TIMER_HW_AUTO,
+        .flags =        SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET,
         .open =         snd_hrtimer_open,
         .close =        snd_hrtimer_close,
         .start =        snd_hrtimer_start,
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index da03597fc893..5c426df87678 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -55,14 +55,13 @@
 #include <linux/err.h>
 #include <linux/platform_device.h>
 #include <linux/ioport.h>
+#include <linux/io.h>
 #include <linux/moduleparam.h>
 #include <sound/core.h>
 #include <sound/initval.h>
 #include <sound/rawmidi.h>
 #include <linux/delay.h>
 
-#include <asm/io.h>
-
 /*
  *      globals
  */
diff --git a/sound/oss/Makefile b/sound/oss/Makefile
index 96f14dcd0cd1..90ffb99c6b17 100644
--- a/sound/oss/Makefile
+++ b/sound/oss/Makefile
@@ -87,7 +87,7 @@ ifeq ($(CONFIG_PSS_HAVE_BOOT),y)
         $(obj)/bin2hex pss_synth < $< > $@
 else
     $(obj)/pss_boot.h:
-        (                                                       \
+        $(Q)(                                                   \
             echo 'static unsigned char * pss_synth = NULL;';    \
             echo 'static int pss_synthLen = 0;';                \
         ) > $@
@@ -102,7 +102,7 @@ ifeq ($(CONFIG_TRIX_HAVE_BOOT),y)
         $(obj)/hex2hex -i trix_boot < $< > $@
 else
     $(obj)/trix_boot.h:
-        (                                                       \
+        $(Q)(                                                   \
             echo 'static unsigned char * trix_boot = NULL;';    \
             echo 'static int trix_boot_len = 0;';               \
         ) > $@
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 23f49f356e0f..16c0bdfbb164 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1252,11 +1252,19 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) {
 static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma)
 {
         stream_t *dma = &vortex->dma_adb[adbdma];
-        int temp;
+        int temp, page, delta;
 
         temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2));
-        temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1));
-        return temp;
+        page = (temp & ADB_SUBBUF_MASK) >> ADB_SUBBUF_SHIFT;
+        if (dma->nr_periods >= 4)
+                delta = (page - dma->period_real) & 3;
+        else {
+                delta = (page - dma->period_real);
+                if (delta < 0)
+                        delta += dma->nr_periods;
+        }
+        return (dma->period_virt + delta) * dma->period_bytes
+                + (temp & (dma->period_bytes - 1));
 }
 
 static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2e91a991eb15..fcedad9a5fef 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2308,6 +2308,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
         SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+        SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
@@ -2703,7 +2704,7 @@ static int __devinit azx_probe(struct pci_dev *pci,
         if (err < 0)
                 goto out_free;
 #ifdef CONFIG_SND_HDA_PATCH_LOADER
-        if (patch[dev]) {
+        if (patch[dev] && *patch[dev]) {
                 snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n",
                            patch[dev]);
                 err = snd_hda_load_patch(chip->bus, patch[dev]);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 9bb030a469cd..dd7c5c12225d 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -85,6 +85,7 @@ struct conexant_spec {
         unsigned int auto_mic;
         int auto_mic_ext;               /* autocfg.inputs[] index for ext mic */
         unsigned int need_dac_fix;
+        hda_nid_t slave_dig_outs[2];
 
         /* capture */
         unsigned int num_adc_nids;
@@ -127,6 +128,7 @@ struct conexant_spec {
         unsigned int ideapad:1;
         unsigned int thinkpad:1;
         unsigned int hp_laptop:1;
+        unsigned int asus:1;
 
         unsigned int ext_mic_present;
         unsigned int recording;
@@ -352,6 +354,8 @@ static int conexant_build_pcms(struct hda_codec *codec)
                         info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
                                 spec->dig_in_nid;
                 }
+                if (spec->slave_dig_outs[0])
+                        codec->slave_dig_outs = spec->slave_dig_outs;
         }
 
         return 0;
@@ -403,10 +407,16 @@ static int conexant_add_jack(struct hda_codec *codec,
         struct conexant_spec *spec;
         struct conexant_jack *jack;
         const char *name;
-        int err;
+        int i, err;
 
         spec = codec->spec;
         snd_array_init(&spec->jacks, sizeof(*jack), 32);
+
+        jack = spec->jacks.list;
+        for (i = 0; i < spec->jacks.used; i++, jack++)
+                if (jack->nid == nid)
+                        return 0 ; /* already present */
+
         jack = snd_array_new(&spec->jacks);
         name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ;
 
@@ -2100,7 +2110,7 @@ static int patch_cxt5051(struct hda_codec *codec)
 static hda_nid_t cxt5066_dac_nids[1] = { 0x10 };
 static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
 static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
-#define CXT5066_SPDIF_OUT       0x21
+static hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 };
 
 /* OLPC's microphone port is DC coupled for use with external sensors,
  * therefore we use a 50% mic bias in order to center the input signal with
@@ -2312,6 +2322,19 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec)
         }
 }
 
+
+/* toggle input of built-in digital mic and mic jack appropriately */
+static void cxt5066_asus_automic(struct hda_codec *codec)
+{
+        unsigned int present;
+
+        present = snd_hda_jack_detect(codec, 0x1b);
+        snd_printdd("CXT5066: external microphone present=%d\n", present);
+        snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL,
+                            present ? 1 : 0);
+}
+
+
 /* toggle input of built-in digital mic and mic jack appropriately */
 static void cxt5066_hp_laptop_automic(struct hda_codec *codec)
 {
@@ -2387,79 +2410,55 @@ static void cxt5066_hp_automute(struct hda_codec *codec)
         cxt5066_update_speaker(codec);
 }
 
-/* unsolicited event for jack sensing */
-static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
+/* Dispatch the right mic autoswitch function */
+static void cxt5066_automic(struct hda_codec *codec)
 {
         struct conexant_spec *spec = codec->spec;
-        snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
-        switch (res >> 26) {
-        case CONEXANT_HP_EVENT:
-                cxt5066_hp_automute(codec);
-                break;
-        case CONEXANT_MIC_EVENT:
-                /* ignore mic events in DC mode; we're always using the jack */
-                if (!spec->dc_enable)
-                        cxt5066_olpc_automic(codec);
-                break;
-        }
-}
 
-/* unsolicited event for jack sensing */
-static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res)
-{
-        snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26);
-        switch (res >> 26) {
-        case CONEXANT_HP_EVENT:
-                cxt5066_hp_automute(codec);
-                break;
-        case CONEXANT_MIC_EVENT:
+        if (spec->dell_vostro)
                 cxt5066_vostro_automic(codec);
-                break;
-        }
-}
-
-/* unsolicited event for jack sensing */
-static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res)
-{
-        snd_printdd("CXT5066_ideapad: unsol event %x (%x)\n", res, res >> 26);
-        switch (res >> 26) {
-        case CONEXANT_HP_EVENT:
-                cxt5066_hp_automute(codec);
-                break;
-        case CONEXANT_MIC_EVENT:
+        else if (spec->ideapad)
                 cxt5066_ideapad_automic(codec);
-                break;
-        }
+        else if (spec->thinkpad)
+                cxt5066_thinkpad_automic(codec);
+        else if (spec->hp_laptop)
+                cxt5066_hp_laptop_automic(codec);
+        else if (spec->asus)
+                cxt5066_asus_automic(codec);
 }
 
 /* unsolicited event for jack sensing */
-static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
 {
-        snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26);
+        struct conexant_spec *spec = codec->spec;
+        snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
         switch (res >> 26) {
         case CONEXANT_HP_EVENT:
                 cxt5066_hp_automute(codec);
                 break;
         case CONEXANT_MIC_EVENT:
-                cxt5066_hp_laptop_automic(codec);
+                /* ignore mic events in DC mode; we're always using the jack */
+                if (!spec->dc_enable)
+                        cxt5066_olpc_automic(codec);
                 break;
         }
 }
 
 /* unsolicited event for jack sensing */
-static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
 {
-        snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26);
+        snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
         switch (res >> 26) {
         case CONEXANT_HP_EVENT:
                 cxt5066_hp_automute(codec);
                 break;
         case CONEXANT_MIC_EVENT:
-                cxt5066_thinkpad_automic(codec);
+                cxt5066_automic(codec);
                 break;
         }
 }
 
+
 static const struct hda_input_mux cxt5066_analog_mic_boost = {
         .num_items = 5,
         .items = {
@@ -2633,6 +2632,27 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec)
         spec->recording = 0;
 }
 
+static void conexant_check_dig_outs(struct hda_codec *codec,
+                                    hda_nid_t *dig_pins,
+                                    int num_pins)
+{
+        struct conexant_spec *spec = codec->spec;
+        hda_nid_t *nid_loc = &spec->multiout.dig_out_nid;
+        int i;
+
+        for (i = 0; i < num_pins; i++, dig_pins++) {
+                unsigned int cfg = snd_hda_codec_get_pincfg(codec, *dig_pins);
+                if (get_defcfg_connect(cfg) == AC_JACK_PORT_NONE)
+                        continue;
+                if (snd_hda_get_connections(codec, *dig_pins, nid_loc, 1) != 1)
+                        continue;
+                if (spec->slave_dig_outs[0])
+                        nid_loc++;
+                else
+                        nid_loc = spec->slave_dig_outs;
+        }
+}
+
 static struct hda_input_mux cxt5066_capture_source = {
         .num_items = 4,
         .items = {
@@ -3039,20 +3059,11 @@ static struct hda_verb cxt5066_init_verbs_hp_laptop[] = {
 /* initialize jack-sensing, too */
 static int cxt5066_init(struct hda_codec *codec)
 {
-        struct conexant_spec *spec = codec->spec;
-
         snd_printdd("CXT5066: init\n");
         conexant_init(codec);
         if (codec->patch_ops.unsol_event) {
                 cxt5066_hp_automute(codec);
-                if (spec->dell_vostro)
-                        cxt5066_vostro_automic(codec);
-                else if (spec->ideapad)
-                        cxt5066_ideapad_automic(codec);
-                else if (spec->thinkpad)
-                        cxt5066_thinkpad_automic(codec);
-                else if (spec->hp_laptop)
-                        cxt5066_hp_laptop_automic(codec);
+                cxt5066_automic(codec);
         }
         cxt5066_set_mic_boost(codec);
         return 0;
@@ -3080,6 +3091,7 @@ enum {
         CXT5066_DELL_VOSTRO,    /* Dell Vostro 1015i */
         CXT5066_IDEAPAD,        /* Lenovo IdeaPad U150 */
         CXT5066_THINKPAD,       /* Lenovo ThinkPad T410s, others? */
+        CXT5066_ASUS,           /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */
         CXT5066_HP_LAPTOP,      /* HP Laptop */
         CXT5066_MODELS
 };
@@ -3091,6 +3103,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
         [CXT5066_DELL_VOSTRO]   = "dell-vostro",
         [CXT5066_IDEAPAD]       = "ideapad",
         [CXT5066_THINKPAD]      = "thinkpad",
+        [CXT5066_ASUS]          = "asus",
         [CXT5066_HP_LAPTOP]     = "hp-laptop",
 };
 
@@ -3102,7 +3115,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
         SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
         SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
-        SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP),
+        SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
+        SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
+        SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS),
         SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
         SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
         SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
@@ -3111,7 +3126,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
         SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
         SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
         SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
+        SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
         SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+        SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
         SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
         {}
 };
@@ -3133,7 +3150,8 @@ static int patch_cxt5066(struct hda_codec *codec)
         spec->multiout.max_channels = 2;
         spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids);
         spec->multiout.dac_nids = cxt5066_dac_nids;
-        spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT;
+        conexant_check_dig_outs(codec, cxt5066_digout_pin_nids,
+            ARRAY_SIZE(cxt5066_digout_pin_nids));
         spec->num_adc_nids = 1;
         spec->adc_nids = cxt5066_adc_nids;
         spec->capsrc_nids = cxt5066_capsrc_nids;
@@ -3167,17 +3185,20 @@ static int patch_cxt5066(struct hda_codec *codec)
                 spec->num_init_verbs++;
                 spec->dell_automute = 1;
                 break;
+        case CXT5066_ASUS:
         case CXT5066_HP_LAPTOP:
                 codec->patch_ops.init = cxt5066_init;
-                codec->patch_ops.unsol_event = cxt5066_hp_laptop_event;
+                codec->patch_ops.unsol_event = cxt5066_unsol_event;
                 spec->init_verbs[spec->num_init_verbs] =
                         cxt5066_init_verbs_hp_laptop;
                 spec->num_init_verbs++;
-                spec->hp_laptop = 1;
+                spec->hp_laptop = board_config == CXT5066_HP_LAPTOP;
+                spec->asus = board_config == CXT5066_ASUS;
                 spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
                 spec->mixers[spec->num_mixers++] = cxt5066_mixers;
                 /* no S/PDIF out */
-                spec->multiout.dig_out_nid = 0;
+                if (board_config == CXT5066_HP_LAPTOP)
+                        spec->multiout.dig_out_nid = 0;
                 /* input source automatically selected */
                 spec->input_mux = NULL;
                 spec->port_d_mode = 0;
@@ -3207,7 +3228,7 @@ static int patch_cxt5066(struct hda_codec *codec)
                 break;
         case CXT5066_DELL_VOSTRO:
                 codec->patch_ops.init = cxt5066_init;
-                codec->patch_ops.unsol_event = cxt5066_vostro_event;
+                codec->patch_ops.unsol_event = cxt5066_unsol_event;
                 spec->init_verbs[0] = cxt5066_init_verbs_vostro;
                 spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
                 spec->mixers[spec->num_mixers++] = cxt5066_mixers;
@@ -3224,7 +3245,7 @@ static int patch_cxt5066(struct hda_codec *codec)
                 break;
         case CXT5066_IDEAPAD:
                 codec->patch_ops.init = cxt5066_init;
-                codec->patch_ops.unsol_event = cxt5066_ideapad_event;
+                codec->patch_ops.unsol_event = cxt5066_unsol_event;
                 spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
                 spec->mixers[spec->num_mixers++] = cxt5066_mixers;
                 spec->init_verbs[0] = cxt5066_init_verbs_ideapad;
@@ -3240,7 +3261,7 @@ static int patch_cxt5066(struct hda_codec *codec)
                 break;
         case CXT5066_THINKPAD:
                 codec->patch_ops.init = cxt5066_init;
-                codec->patch_ops.unsol_event = cxt5066_thinkpad_event;
+                codec->patch_ops.unsol_event = cxt5066_unsol_event;
                 spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
                 spec->mixers[spec->num_mixers++] = cxt5066_mixers;
                 spec->init_verbs[0] = cxt5066_init_verbs_thinkpad;
@@ -3389,7 +3410,7 @@ static void cx_auto_parse_output(struct hda_codec *codec)
                 }
         }
         spec->multiout.dac_nids = spec->private_dac_nids;
-        spec->multiout.max_channels = nums * 2;
+        spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
         if (cfg->hp_outs > 0)
                 spec->auto_mute = 1;
@@ -3708,9 +3729,9 @@ static int cx_auto_init(struct hda_codec *codec)
         return 0;
 }
 
-static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
+static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
                               const char *dir, int cidx,
-                              hda_nid_t nid, int hda_dir)
+                              hda_nid_t nid, int hda_dir, int amp_idx)
 {
         static char name[32];
         static struct snd_kcontrol_new knew[] = {
@@ -3722,7 +3743,8 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
 
         for (i = 0; i < 2; i++) {
                 struct snd_kcontrol *kctl;
-                knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, hda_dir);
+                knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx,
+                                                            hda_dir);
                 knew[i].subdevice = HDA_SUBDEV_AMP_FLAG;
                 knew[i].index = cidx;
                 snprintf(name, sizeof(name), "%s%s %s", basename, dir, sfx[i]);
@@ -3738,6 +3760,9 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
         return 0;
 }
 
+#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir)         \
+        cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0)
+
 #define cx_auto_add_pb_volume(codec, nid, str, idx)                     \
         cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT)
 
@@ -3787,29 +3812,60 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
         struct conexant_spec *spec = codec->spec;
         struct auto_pin_cfg *cfg = &spec->autocfg;
         static const char *prev_label;
-        int i, err, cidx;
+        int i, err, cidx, conn_len;
+        hda_nid_t conn[HDA_MAX_CONNECTIONS];
+
+        int multi_adc_volume = 0; /* If the ADC nid has several input volumes */
+        int adc_nid = spec->adc_nids[0];
+
+        conn_len = snd_hda_get_connections(codec, adc_nid, conn,
+                                           HDA_MAX_CONNECTIONS);
+        if (conn_len < 0)
+                return conn_len;
+
+        multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1;
+        if (!multi_adc_volume) {
+                err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid,
+                                         HDA_INPUT);
+                if (err < 0)
+                        return err;
+        }
 
-        err = cx_auto_add_volume(codec, "Capture", "", 0, spec->adc_nids[0],
-                                 HDA_INPUT);
-        if (err < 0)
-                return err;
         prev_label = NULL;
         cidx = 0;
         for (i = 0; i < cfg->num_inputs; i++) {
                 hda_nid_t nid = cfg->inputs[i].pin;
                 const char *label;
-                if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP))
+                int j;
+                int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP;
+                if (!pin_amp && !multi_adc_volume)
                         continue;
+
                 label = hda_get_autocfg_input_label(codec, cfg, i);
                 if (label == prev_label)
                         cidx++;
                 else
                         cidx = 0;
                 prev_label = label;
-                err = cx_auto_add_volume(codec, label, " Capture", cidx,
-                                         nid, HDA_INPUT);
-                if (err < 0)
-                        return err;
+
+                if (pin_amp) {
+                        err = cx_auto_add_volume(codec, label, " Boost", cidx,
+                                                 nid, HDA_INPUT);
+                        if (err < 0)
+                                return err;
+                }
+
+                if (!multi_adc_volume)
+                        continue;
+                for (j = 0; j < conn_len; j++) {
+                        if (conn[j] == nid) {
+                                err = cx_auto_add_volume_idx(codec, label,
+                                    " Capture", cidx, adc_nid, HDA_INPUT, j);
+                                if (err < 0)
+                                        return err;
+                                break;
+                        }
+                }
         }
         return 0;
 }
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 2d5b83fa8d24..a58767736727 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -642,6 +642,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
                         hdmi_ai->ver            = 0x01;
                         hdmi_ai->len            = 0x0a;
                         hdmi_ai->CC02_CT47      = channels - 1;
+                        hdmi_ai->CA             = ca;
                         hdmi_checksum_audio_infoframe(hdmi_ai);
                 } else if (spec->sink_eld[i].conn_type == 1) { /* DisplayPort */
                         struct dp_audio_infoframe *dp_ai;
@@ -651,6 +652,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
                         dp_ai->len              = 0x1b;
                         dp_ai->ver              = 0x11 << 2;
                         dp_ai->CC02_CT47        = channels - 1;
+                        dp_ai->CA               = ca;
                 } else {
                         snd_printd("HDMI: unknown connection type at pin %d\n",
                                    pin_nid);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2fa9ed99c32f..3328a259a242 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2290,6 +2290,29 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
         { } /* end */
 };
 
+static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = {
+        HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+        HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+        HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0f, 2, 0x0,
+                HDA_OUTPUT),
+        HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0f, 2, 2, HDA_INPUT),
+        HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT),
+        HDA_CODEC_VOLUME("Side Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE("Side Playback Switch", 0x0e, 2, HDA_INPUT),
+        HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+        HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+        HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+        HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+        HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+        HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+        { } /* end */
+};
+
+
 static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = {
         HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
         HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -10359,7 +10382,7 @@ static struct alc_config_preset alc882_presets[] = {
                 .init_hook = alc_automute_amp,
         },
         [ALC888_ACER_ASPIRE_4930G] = {
-                .mixers = { alc888_base_mixer,
+                .mixers = { alc888_acer_aspire_4930g_mixer,
                                 alc883_chmode_mixer },
                 .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
                                 alc888_acer_aspire_4930g_verbs },
@@ -18802,6 +18825,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
                                         ALC662_3ST_6ch_DIG),
         SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
                            ALC663_ASUS_H13),
+        SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E),
         {}
 };
 
@@ -19494,6 +19518,7 @@ static const struct alc_fixup alc662_fixups[] = {
 };
 
 static struct snd_pci_quirk alc662_fixup_tbl[] = {
+        SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
         SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
         SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
         SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index c2ae63d17cd2..f53897a708b4 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -92,6 +92,8 @@ struct oxygen_model {
         void (*update_dac_volume)(struct oxygen *chip);
         void (*update_dac_mute)(struct oxygen *chip);
         void (*update_center_lfe_mix)(struct oxygen *chip, bool mixed);
+        unsigned int (*adjust_dac_routing)(struct oxygen *chip,
+                                           unsigned int play_routing);
         void (*gpio_changed)(struct oxygen *chip);
         void (*uart_input)(struct oxygen *chip);
         void (*ac97_switch)(struct oxygen *chip,
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 9bff14d5895d..26c7e8bcb229 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -180,6 +180,8 @@ void oxygen_update_dac_routing(struct oxygen *chip)
                             (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
                             (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
                             (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT);
+        if (chip->model.adjust_dac_routing)
+                reg_value = chip->model.adjust_dac_routing(chip, reg_value);
         oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value,
                               OXYGEN_PLAY_DAC0_SOURCE_MASK |
                               OXYGEN_PLAY_DAC1_SOURCE_MASK |
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index e1fa602eba79..bc6eb58be380 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -24,6 +24,11 @@
  *
  *   SPI 0 -> CS4245
  *
+ *   I²S 1 -> CS4245
+ *   I²S 2 -> CS4361 (center/LFE)
+ *   I²S 3 -> CS4361 (surround)
+ *   I²S 4 -> CS4361 (front)
+ *
  *   GPIO 3 <- ?
  *   GPIO 4 <- headphone detect
  *   GPIO 5 -> route input jack to line-in (0) or mic-in (1)
@@ -36,6 +41,7 @@
  *   input 1 <- aux
  *   input 2 <- front mic
  *   input 4 <- line/mic
+ *   DAC out -> headphones
  *   aux out -> front panel headphones
  */
 
@@ -207,6 +213,35 @@ static void set_cs4245_adc_params(struct oxygen *chip,
         cs4245_write_cached(chip, CS4245_ADC_CTRL, value);
 }
 
+static inline unsigned int shift_bits(unsigned int value,
+                                      unsigned int shift_from,
+                                      unsigned int shift_to,
+                                      unsigned int mask)
+{
+        if (shift_from < shift_to)
+                return (value << (shift_to - shift_from)) & mask;
+        else
+                return (value >> (shift_from - shift_to)) & mask;
+}
+
+static unsigned int adjust_dg_dac_routing(struct oxygen *chip,
+                                          unsigned int play_routing)
+{
+        return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) |
+               shift_bits(play_routing,
+                          OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+                          OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+                          OXYGEN_PLAY_DAC1_SOURCE_MASK) |
+               shift_bits(play_routing,
+                          OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+                          OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+                          OXYGEN_PLAY_DAC2_SOURCE_MASK) |
+               shift_bits(play_routing,
+                          OXYGEN_PLAY_DAC0_SOURCE_SHIFT,
+                          OXYGEN_PLAY_DAC3_SOURCE_SHIFT,
+                          OXYGEN_PLAY_DAC3_SOURCE_MASK);
+}
+
 static int output_switch_info(struct snd_kcontrol *ctl,
                               struct snd_ctl_elem_info *info)
 {
@@ -557,6 +592,7 @@ struct oxygen_model model_xonar_dg = {
         .resume = dg_resume,
         .set_dac_params = set_cs4245_dac_params,
         .set_adc_params = set_cs4245_adc_params,
+        .adjust_dac_routing = adjust_dg_dac_routing,
         .dump_registers = dump_cs4245_registers,
         .model_data_size = sizeof(struct dg),
         .device_config = PLAYBACK_0_TO_I2S |
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h
index bd26e092aead..6ce9ad700290 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.h
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h
@@ -22,7 +22,7 @@
 #define __PDAUDIOCF_H
 
 #include <sound/pcm.h>
-#include <asm/io.h>
+#include <linux/io.h>
 #include <linux/interrupt.h>
 #include <pcmcia/cistpl.h>
 #include <pcmcia/ds.h>
diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c
index 989e04abb520..fe33e122e372 100644
--- a/sound/pcmcia/vx/vxp_ops.c
+++ b/sound/pcmcia/vx/vxp_ops.c
@@ -23,8 +23,8 @@
 #include <linux/delay.h>
 #include <linux/device.h>
 #include <linux/firmware.h>
+#include <linux/io.h>
 #include <sound/core.h>
-#include <asm/io.h>
 #include "vxpocket.h"
 
 
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 46dbfd067f79..347a567b01e1 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -153,7 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec)
 
 static int cq93vc_probe(struct snd_soc_codec *codec)
 {
-        struct davinci_vc *davinci_vc = codec->dev->platform_data;
+        struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec);
 
         davinci_vc->cq93vc.codec = codec;
         codec->control_data = davinci_vc;
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 03d1e860d229..bb4bf65b9e7e 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
         return 0;
 }
 
+static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC;
+
 static struct snd_soc_codec_driver cx20442_codec_dev = {
         .probe =        cx20442_codec_probe,
         .remove =       cx20442_codec_remove,
+        .reg_cache_default = &cx20442_reg,
         .reg_cache_size = 1,
         .reg_word_size = sizeof(u8),
         .read = cx20442_read_reg_cache,
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3351f77607b3..37b8aa8a680f 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0),
 
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
                      0, WM8994_POWER_MANAGEMENT_4, 9, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
                      0, WM8994_POWER_MANAGEMENT_4, 8, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
                       WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev,
@@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0,
                       WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev,
                       SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL,
                      0, WM8994_POWER_MANAGEMENT_4, 11, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL,
                      0, WM8994_POWER_MANAGEMENT_4, 10, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
                       WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev,
@@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0,
 
 SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
 
 SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux),
@@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = {
         { "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" },
         { "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" },
 
+        { "AIF1ADCDAT", NULL, "AIF1ADC1L" },
+        { "AIF1ADCDAT", NULL, "AIF1ADC1R" },
+        { "AIF1ADCDAT", NULL, "AIF1ADC2L" },
+        { "AIF1ADCDAT", NULL, "AIF1ADC2R" },
+
         { "AIF2ADCDAT", NULL, "AIF2ADC Mux" },
 
         /* AIF3 output */
@@ -1578,6 +1584,13 @@ static const struct snd_soc_dapm_route intercon[] = {
         { "Right Headphone Mux", "DAC", "DAC1R" },
 };
 
+static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
+        { "AIF1DACDAT", NULL, "AIF2DACDAT" },
+        { "AIF2DACDAT", NULL, "AIF1DACDAT" },
+        { "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
+        { "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
+};
+
 static const struct snd_soc_dapm_route wm8994_intercon[] = {
         { "AIF2DACL", NULL, "AIF2DAC Mux" },
         { "AIF2DACR", NULL, "AIF2DAC Mux" },
@@ -3129,6 +3142,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
         case WM8994:
                 snd_soc_dapm_add_routes(dapm, wm8994_intercon,
                                         ARRAY_SIZE(wm8994_intercon));
+
+                if (wm8994->revision < 4)
+                        snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
+                                                ARRAY_SIZE(wm8994_revd_intercon));
+                        
                 break;
         case WM8958:
                 snd_soc_dapm_add_routes(dapm, wm8958_intercon,
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index b36f0b39b090..fe7984221eb9 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -218,7 +218,19 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = {
                 .ops = &evm_spdif_ops,
         },
 };
-static struct snd_soc_dai_link da8xx_evm_dai = {
+
+static struct snd_soc_dai_link da830_evm_dai = {
+        .name = "TLV320AIC3X",
+        .stream_name = "AIC3X",
+        .cpu_dai_name = "davinci-mcasp.1",
+        .codec_dai_name = "tlv320aic3x-hifi",
+        .codec_name = "tlv320aic3x-codec.1-0018",
+        .platform_name = "davinci-pcm-audio",
+        .init = evm_aic3x_init,
+        .ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link da850_evm_dai = {
         .name = "TLV320AIC3X",
         .stream_name = "AIC3X",
         .cpu_dai_name= "davinci-mcasp.0",
@@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = {
 
 static struct snd_soc_card da830_snd_soc_card = {
         .name = "DA830/OMAP-L137 EVM",
-        .dai_link = &da8xx_evm_dai,
+        .dai_link = &da830_evm_dai,
         .num_links = 1,
 };
 
 static struct snd_soc_card da850_snd_soc_card = {
         .name = "DA850/OMAP-L138 EVM",
-        .dai_link = &da8xx_evm_dai,
+        .dai_link = &da850_evm_dai,
         .num_links = 1,
 };
 
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 2101bdcee21f..3167be689621 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
         /* Set up digital mute if not provided by the codec */
         if (!codec_dai->driver->ops) {
                 codec_dai->driver->ops = &ams_delta_dai_ops;
-        } else if (!codec_dai->driver->ops->digital_mute) {
-                codec_dai->driver->ops->digital_mute = ams_delta_digital_mute;
         } else {
                 ams_delta_ops.startup = ams_delta_startup;
                 ams_delta_ops.shutdown = ams_delta_shutdown;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index bac7291b6ff6..c3f6f1e72790 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1449,6 +1449,7 @@ static int soc_post_component_init(struct snd_soc_card *card,
                 rtd = &card->rtd_aux[num];
                 name = aux_dev->name;
         }
+        rtd->card = card;
 
         /* machine controls, routes and widgets are not prefixed */
         temp = codec->name_prefix;
@@ -1471,7 +1472,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
 
         /* register the rtd device */
         rtd->codec = codec;
-        rtd->card = card;
         rtd->dev.parent = card->dev;
         rtd->dev.release = rtd_release;
         rtd->dev.init_name = name;
@@ -1664,9 +1664,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
         goto out;
 
 found:
-        if (!try_module_get(codec->dev->driver->owner))
-                return -ENODEV;
-
         ret = soc_probe_codec(card, codec);
         if (ret < 0)
                 return ret;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 499730ab5638..8194f150bab7 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1742,7 +1742,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
         int max = mc->max;
         unsigned int mask = (1 << fls(max)) - 1;
         unsigned int invert = mc->invert;
-        unsigned int val, val_mask;
+        unsigned int val;
         int connect, change;
         struct snd_soc_dapm_update update;
 
@@ -1750,13 +1750,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 
         if (invert)
                 val = max - val;
-        val_mask = mask << shift;
+        mask = mask << shift;
         val = val << shift;
 
         mutex_lock(&widget->codec->mutex);
         widget->value = val;
 
-        change = snd_soc_test_bits(widget->codec, reg, val_mask, val);
+        change = snd_soc_test_bits(widget->codec, reg, mask, val);
         if (change) {
                 if (val)
                         /* new connection */
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 68b97477577b..66eabafb1c24 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -785,7 +785,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
         }
 
         dev->pcm->private_data = dev;
-        strcpy(dev->pcm->name, dev->product_name);
+        strlcpy(dev->pcm->name, dev->product_name, sizeof(dev->pcm->name));
 
         memset(dev->sub_playback, 0, sizeof(dev->sub_playback));
         memset(dev->sub_capture, 0, sizeof(dev->sub_capture));
diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c
index 2f218c77fff2..a1a47088fd0c 100644
--- a/sound/usb/caiaq/midi.c
+++ b/sound/usb/caiaq/midi.c
@@ -136,7 +136,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device)
         if (ret < 0)
                 return ret;
 
-        strcpy(rmidi->name, device->product_name);
+        strlcpy(rmidi->name, device->product_name, sizeof(rmidi->name));
 
         rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX;
         rmidi->private_data = device;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 7df89b3d7ded..85af6051b52d 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -95,7 +95,7 @@ enum {
 };
 
 
-/*E-mu 0202(0404) eXtension Unit(XU) control*/
+/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/
 enum {
         USB_XU_CLOCK_RATE               = 0xe301,
         USB_XU_CLOCK_SOURCE             = 0xe302,
@@ -1566,7 +1566,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw
                         cval->initialized = 1;
                 } else {
                         if (type == USB_XU_CLOCK_RATE) {
-                                /* E-Mu USB 0404/0202/TrackerPre
+                                /* E-Mu USB 0404/0202/TrackerPre/0204
                                  * samplerate control quirk
                                  */
                                 cval->min = 0;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 35999874d301..921a86fd9884 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -79,6 +79,13 @@
         .idProduct = 0x3f0a,
         .bInterfaceClass = USB_CLASS_AUDIO,
 },
+{
+        /* E-Mu 0204 USB */
+        .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+        .idVendor = 0x041e,
+        .idProduct = 0x3f19,
+        .bInterfaceClass = USB_CLASS_AUDIO,
+},
 
 /*
  * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index cf8bf088394b..e314cdb85003 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -532,7 +532,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat
 }
 
 /*
- * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device,
+ * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device,
  * not for interface.
  */
 
@@ -589,6 +589,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
         case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */
         case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */
         case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */
+        case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */
                 set_format_emu_quirk(subs, fmt);
                 break;
         }