Merge branch 'timers/posix-cpu-timers-for-tglx' of

git://git.kernel.org/pub/scm/linux/kernel/git/frederic/linux-dynticks into timers/core

Frederic sayed: "Most of these patches have been hanging around for
several month now, in -mmotm for a significant chunk. They already
missed a few releases."

Signed-off-by: Thomas Gleixner <tglx@linutronix.de>

31 files changed, 254 insertions, 137 deletions

diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index 552b97afbca5..61ab640e195f 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -113,6 +113,7 @@ MODULE_ALIAS("sound-layout-100");
 MODULE_ALIAS("aoa-device-id-14");
 MODULE_ALIAS("aoa-device-id-22");
 MODULE_ALIAS("aoa-device-id-35");
+MODULE_ALIAS("aoa-device-id-44");
 
 /* onyx with all but microphone connected */
 static struct codec_connection onyx_connections_nomic[] = {
@@ -361,6 +362,13 @@ static struct layout layouts[] = {
                 .connections = tas_connections_nolineout,
           },
         },
+        /* PowerBook6,5 */
+        { .device_id = 44,
+          .codecs[0] = {
+                .name = "tas",
+                .connections = tas_connections_all,
+          },
+        },
         /* PowerBook6,7 */
         { .layout_id = 80,
           .codecs[0] = {
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
index 010658335881..15e76131b501 100644
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -200,7 +200,8 @@ static int i2sbus_add_dev(struct macio_dev *macio,
                          * We probably cannot handle all device-id machines,
                          * so restrict to those we do handle for now.
                          */
-                        if (id && (*id == 22 || *id == 14 || *id == 35)) {
+                        if (id && (*id == 22 || *id == 14 || *id == 35 ||
+                                   *id == 44)) {
                                 snprintf(dev->sound.modalias, 32,
                                          "aoa-device-id-%d", *id);
                                 ok = 1;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index ccfa383f1fda..f92818155958 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1649,6 +1649,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
         }
         if (!snd_pcm_stream_linked(substream)) {
                 substream->group = group;
+                group = NULL;
                 spin_lock_init(&substream->group->lock);
                 INIT_LIST_HEAD(&substream->group->substreams);
                 list_add_tail(&substream->link_list, &substream->group->substreams);
@@ -1663,8 +1664,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
  _nolock:
         snd_card_unref(substream1->pcm->card);
         fput_light(file, fput_needed);
-        if (res < 0)
-                kfree(group);
+        kfree(group);
         return res;
 }
 
diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig
index 51c4ba95a32d..1a9640254433 100644
--- a/sound/oss/Kconfig
+++ b/sound/oss/Kconfig
@@ -250,7 +250,7 @@ config MSND_FIFOSIZE
 menuconfig SOUND_OSS
         tristate "OSS sound modules"
         depends on ISA_DMA_API && VIRT_TO_BUS
-        depends on !ISA_DMA_SUPPORT_BROKEN
+        depends on !GENERIC_ISA_DMA_SUPPORT_BROKEN
         help
           OSS is the Open Sound System suite of sound card drivers.  They make
           sound programming easier since they provide a common API.  Say Y or
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index ac079f93c535..4b1524a861f3 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -606,6 +606,10 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid,
         return false;
 }
 
+/* check whether the NID is referred by any active paths */
+#define is_active_nid_for_any(codec, nid) \
+        is_active_nid(codec, nid, HDA_OUTPUT, 0)
+
 /* get the default amp value for the target state */
 static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid,
                                    int dir, unsigned int caps, bool enable)
@@ -759,7 +763,8 @@ static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path)
 
         for (i = 0; i < path->depth; i++) {
                 hda_nid_t nid = path->path[i];
-                if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3)) {
+                if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3) &&
+                    !is_active_nid_for_any(codec, nid)) {
                         snd_hda_codec_write(codec, nid, 0,
                                             AC_VERB_SET_POWER_STATE,
                                             AC_PWRST_D3);
@@ -783,6 +788,8 @@ static void set_pin_eapd(struct hda_codec *codec, hda_nid_t pin, bool enable)
                 return;
         if (codec->inv_eapd)
                 enable = !enable;
+        if (spec->keep_eapd_on && !enable)
+                return;
         snd_hda_codec_update_cache(codec, pin, 0,
                                    AC_VERB_SET_EAPD_BTLENABLE,
                                    enable ? 0x02 : 0x00);
@@ -1933,17 +1940,7 @@ static int create_speaker_out_ctls(struct hda_codec *codec)
  * independent HP controls
  */
 
-/* update HP auto-mute state too */
-static void update_hp_automute_hook(struct hda_codec *codec)
-{
-        struct hda_gen_spec *spec = codec->spec;
-
-        if (spec->hp_automute_hook)
-                spec->hp_automute_hook(codec, NULL);
-        else
-                snd_hda_gen_hp_automute(codec, NULL);
-}
-
+static void call_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack);
 static int indep_hp_info(struct snd_kcontrol *kcontrol,
                          struct snd_ctl_elem_info *uinfo)
 {
@@ -2004,7 +2001,7 @@ static int indep_hp_put(struct snd_kcontrol *kcontrol,
                 else
                         *dacp = spec->alt_dac_nid;
 
-                update_hp_automute_hook(codec);
+                call_hp_automute(codec, NULL);
                 ret = 1;
         }
  unlock:
@@ -2300,7 +2297,7 @@ static void update_hp_mic(struct hda_codec *codec, int adc_mux, bool force)
                 else
                         val = PIN_HP;
                 set_pin_target(codec, pin, val, true);
-                update_hp_automute_hook(codec);
+                call_hp_automute(codec, NULL);
         }
 }
 
@@ -2709,7 +2706,7 @@ static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol,
                         val = snd_hda_get_default_vref(codec, nid);
         }
         snd_hda_set_pin_ctl_cache(codec, nid, val);
-        update_hp_automute_hook(codec);
+        call_hp_automute(codec, NULL);
 
         return 1;
 }
@@ -3854,20 +3851,42 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja
 }
 EXPORT_SYMBOL_HDA(snd_hda_gen_mic_autoswitch);
 
-/* update jack retasking */
-static void update_automute_all(struct hda_codec *codec)
+/* call appropriate hooks */
+static void call_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack)
 {
         struct hda_gen_spec *spec = codec->spec;
+        if (spec->hp_automute_hook)
+                spec->hp_automute_hook(codec, jack);
+        else
+                snd_hda_gen_hp_automute(codec, jack);
+}
 
-        update_hp_automute_hook(codec);
+static void call_line_automute(struct hda_codec *codec,
+                               struct hda_jack_tbl *jack)
+{
+        struct hda_gen_spec *spec = codec->spec;
         if (spec->line_automute_hook)
-                spec->line_automute_hook(codec, NULL);
+                spec->line_automute_hook(codec, jack);
         else
-                snd_hda_gen_line_automute(codec, NULL);
+                snd_hda_gen_line_automute(codec, jack);
+}
+
+static void call_mic_autoswitch(struct hda_codec *codec,
+                                struct hda_jack_tbl *jack)
+{
+        struct hda_gen_spec *spec = codec->spec;
         if (spec->mic_autoswitch_hook)
-                spec->mic_autoswitch_hook(codec, NULL);
+                spec->mic_autoswitch_hook(codec, jack);
         else
-                snd_hda_gen_mic_autoswitch(codec, NULL);
+                snd_hda_gen_mic_autoswitch(codec, jack);
+}
+
+/* update jack retasking */
+static void update_automute_all(struct hda_codec *codec)
+{
+        call_hp_automute(codec, NULL);
+        call_line_automute(codec, NULL);
+        call_mic_autoswitch(codec, NULL);
 }
 
 /*
@@ -4004,9 +4023,7 @@ static int check_auto_mute_availability(struct hda_codec *codec)
                 snd_printdd("hda-codec: Enable HP auto-muting on NID 0x%x\n",
                             nid);
                 snd_hda_jack_detect_enable_callback(codec, nid, HDA_GEN_HP_EVENT,
-                                                    spec->hp_automute_hook ?
-                                                    spec->hp_automute_hook :
-                                                    snd_hda_gen_hp_automute);
+                                                    call_hp_automute);
                 spec->detect_hp = 1;
         }
 
@@ -4019,9 +4036,7 @@ static int check_auto_mute_availability(struct hda_codec *codec)
                                 snd_printdd("hda-codec: Enable Line-Out auto-muting on NID 0x%x\n", nid);
                                 snd_hda_jack_detect_enable_callback(codec, nid,
                                                                     HDA_GEN_FRONT_EVENT,
-                                                                    spec->line_automute_hook ?
-                                                                    spec->line_automute_hook :
-                                                                    snd_hda_gen_line_automute);
+                                                                    call_line_automute);
                                 spec->detect_lo = 1;
                         }
                 spec->automute_lo_possible = spec->detect_hp;
@@ -4063,9 +4078,7 @@ static bool auto_mic_check_imux(struct hda_codec *codec)
                 snd_hda_jack_detect_enable_callback(codec,
                                                     spec->am_entry[i].pin,
                                                     HDA_GEN_MIC_EVENT,
-                                                    spec->mic_autoswitch_hook ?
-                                                    spec->mic_autoswitch_hook :
-                                                    snd_hda_gen_mic_autoswitch);
+                                                    call_mic_autoswitch);
         return true;
 }
 
@@ -4157,7 +4170,7 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
                 return power_state;
         if (get_wcaps_type(get_wcaps(codec, nid)) >= AC_WID_POWER)
                 return power_state;
-        if (is_active_nid(codec, nid, HDA_OUTPUT, 0))
+        if (is_active_nid_for_any(codec, nid))
                 return power_state;
         return AC_PWRST_D3;
 }
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 54e665160379..76200314ee95 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -222,6 +222,7 @@ struct hda_gen_spec {
         unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */
         unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */
         unsigned int own_eapd_ctl:1; /* set EAPD by own function */
+        unsigned int keep_eapd_on:1; /* don't turn off EAPD automatically */
         unsigned int vmaster_mute_enum:1; /* add vmaster mute mode enum */
         unsigned int indep_hp:1; /* independent HP supported */
         unsigned int prefer_hp_amp:1; /* enable HP amp for speaker if any */
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index bd8d46cca2b3..cccaf9c7a7bb 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -58,6 +58,7 @@ enum {
         CS420X_GPIO_23,
         CS420X_MBP101,
         CS420X_MBP81,
+        CS420X_MBA42,
         CS420X_AUTO,
         /* aliases */
         CS420X_IMAC27_122 = CS420X_GPIO_23,
@@ -346,6 +347,7 @@ static const struct hda_model_fixup cs420x_models[] = {
         { .id = CS420X_APPLE, .name = "apple" },
         { .id = CS420X_MBP101, .name = "mbp101" },
         { .id = CS420X_MBP81, .name = "mbp81" },
+        { .id = CS420X_MBA42, .name = "mba42" },
         {}
 };
 
@@ -361,6 +363,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = {
         SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81),
         SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122),
         SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101),
+        SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42),
         SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE),
         {} /* terminator */
 };
@@ -414,6 +417,20 @@ static const struct hda_pintbl mbp101_pincfgs[] = {
         {} /* terminator */
 };
 
+static const struct hda_pintbl mba42_pincfgs[] = {
+        { 0x09, 0x012b4030 }, /* HP */
+        { 0x0a, 0x400000f0 },
+        { 0x0b, 0x90100120 }, /* speaker */
+        { 0x0c, 0x400000f0 },
+        { 0x0d, 0x90a00110 }, /* mic */
+        { 0x0e, 0x400000f0 },
+        { 0x0f, 0x400000f0 },
+        { 0x10, 0x400000f0 },
+        { 0x12, 0x400000f0 },
+        { 0x15, 0x400000f0 },
+        {} /* terminator */
+};
+
 static void cs420x_fixup_gpio_13(struct hda_codec *codec,
                                  const struct hda_fixup *fix, int action)
 {
@@ -482,6 +499,12 @@ static const struct hda_fixup cs420x_fixups[] = {
                 .chained = true,
                 .chain_id = CS420X_GPIO_13,
         },
+        [CS420X_MBA42] = {
+                .type = HDA_FIXUP_PINS,
+                .v.pins = mba42_pincfgs,
+                .chained = true,
+                .chain_id = CS420X_GPIO_13,
+        },
 };
 
 static struct cs_spec *cs_alloc_spec(struct hda_codec *codec, int vendor_nid)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6bf47f7326ad..403010c9e82e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3482,6 +3482,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
+        SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
+        SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -3492,6 +3494,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+        SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+        SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+        SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+        SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
         SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED),
         SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
@@ -3529,6 +3535,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
         SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
+        SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
         SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
         SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
@@ -3592,6 +3599,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
         {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"},
         {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"},
         {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"},
+        {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"},
+        {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"},
         {}
 };
 
@@ -4271,6 +4280,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = {
         {.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"},
         {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"},
         {.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"},
+        {.id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE, .name = "dell-headset-multi"},
         {}
 };
 
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index e0dadcf2030d..e5245544eb52 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -136,6 +136,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec)
                 spec->codec_type = VT1708S;
         spec->no_pin_power_ctl = 1;
         spec->gen.indep_hp = 1;
+        spec->gen.keep_eapd_on = 1;
         spec->gen.pcm_playback_hook = via_playback_pcm_hook;
         return spec;
 }
@@ -231,9 +232,14 @@ static void vt1708_update_hp_work(struct hda_codec *codec)
 
 static void set_widgets_power_state(struct hda_codec *codec)
 {
+#if 0 /* FIXME: the assumed connections don't match always with the
+       * actual routes by the generic parser, so better to disable
+       * the control for safety.
+       */
         struct via_spec *spec = codec->spec;
         if (spec->set_widgets_power_state)
                 spec->set_widgets_power_state(codec);
+#endif
 }
 
 static void update_power_state(struct hda_codec *codec, hda_nid_t nid,
@@ -478,7 +484,9 @@ static int via_suspend(struct hda_codec *codec)
                 /* Fix pop noise on headphones */
                 int i;
                 for (i = 0; i < spec->gen.autocfg.hp_outs; i++)
-                        snd_hda_set_pin_ctl(codec, spec->gen.autocfg.hp_pins[i], 0);
+                        snd_hda_codec_write(codec, spec->gen.autocfg.hp_pins[i],
+                                            0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+                                            0x00);
         }
 
         return 0;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index d59abe1682c5..748e82d4d257 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1341,7 +1341,8 @@ static int sis_chip_create(struct snd_card *card,
         if (rc)
                 goto error_out;
 
-        if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0) {
+        rc = pci_set_dma_mask(pci, DMA_BIT_MASK(30));
+        if (rc < 0) {
                 dev_err(&pci->dev, "architecture does not support 30-bit PCI busmaster DMA");
                 goto error_out_enabled;
         }
diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h
index 114f69a0c629..306d0bc8455f 100644
--- a/sound/soc/codecs/ab8500-codec.h
+++ b/sound/soc/codecs/ab8500-codec.h
@@ -348,25 +348,25 @@
 
 /* AB8500_ADSLOTSELX */
 #define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD   0x00
-#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD   0x01
-#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD   0x02
-#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD   0x03
-#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD   0x04
-#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD   0x05
-#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD   0x06
-#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD   0x07
-#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD    0x08
-#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD  0x0F
+#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD   0x10
+#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD   0x20
+#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD   0x30
+#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD   0x40
+#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD   0x50
+#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD   0x60
+#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD   0x70
+#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD    0x80
+#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD  0xF0
 #define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN  0x00
-#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN  0x10
-#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN  0x20
-#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN  0x30
-#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN  0x40
-#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN  0x50
-#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN  0x60
-#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN  0x70
-#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN   0x80
-#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0xF0
+#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN  0x01
+#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN  0x02
+#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN  0x03
+#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN  0x04
+#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN  0x05
+#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN  0x06
+#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN  0x07
+#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN   0x08
+#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0x0F
 #define AB8500_ADSLOTSELX_EVEN_SHIFT            0
 #define AB8500_ADSLOTSELX_ODD_SHIFT             4
 
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 0f6f481cec09..987f728718c5 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = {
         { CS42L52_BEEP_VOL, 0x00 },     /* r1D Beep Volume off Time */
         { CS42L52_BEEP_TONE_CTL, 0x00 },        /* r1E Beep Tone Cfg. */
         { CS42L52_TONE_CTL, 0x00 },     /* r1F Tone Ctl */
-        { CS42L52_MASTERA_VOL, 0x88 },  /* r20 Master A Volume */
+        { CS42L52_MASTERA_VOL, 0x00 },  /* r20 Master A Volume */
         { CS42L52_MASTERB_VOL, 0x00 },  /* r21 Master B Volume */
         { CS42L52_HPA_VOL, 0x00 },      /* r22 Headphone A Volume */
         { CS42L52_HPB_VOL, 0x00 },      /* r23 Headphone B Volume */
@@ -193,6 +193,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0);
 
 static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0);
 
+static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0);
+
 static const unsigned int limiter_tlv[] = {
         TLV_DB_RANGE_HEAD(2),
         0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
@@ -225,7 +227,7 @@ static const char * const mic_bias_level_text[] = {
 };
 
 static const struct soc_enum mic_bias_level_enum =
-        SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0,
+        SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0,
                         ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
 
 static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
@@ -260,7 +262,7 @@ static const char * const hp_gain_num_text[] = {
 };
 
 static const struct soc_enum hp_gain_enum =
-        SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4,
+        SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5,
                 ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text);
 
 static const char * const beep_pitch_text[] = {
@@ -413,7 +415,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
         SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
 
         SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
-                              CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv),
+                              CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv),
 
         SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
                               CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
@@ -441,7 +443,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 
         SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
                             CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
-                                6, 0x7f, 0x19, hl_tlv),
+                                0, 0x7f, 0x19, mix_tlv),
         SOC_DOUBLE_R("PCM Mixer Switch",
                      CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
 
diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
index 60985c059071..4277012c4719 100644
--- a/sound/soc/codecs/cs42l52.h
+++ b/sound/soc/codecs/cs42l52.h
@@ -157,7 +157,7 @@
 #define CS42L52_PB_CTL1_INV_PCMA                (1 << 2)
 #define CS42L52_PB_CTL1_MSTB_MUTE               (1 << 1)
 #define CS42L52_PB_CTL1_MSTA_MUTE               (1 << 0)
-#define CS42L52_PB_CTL1_MUTE_MASK               0xFFFD
+#define CS42L52_PB_CTL1_MUTE_MASK               0x03
 #define CS42L52_PB_CTL1_MUTE                    3
 #define CS42L52_PB_CTL1_UNMUTE                  0
 
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 41230ad1c3e0..4a6f1daf911f 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1488,17 +1488,17 @@ static int da7213_probe(struct snd_soc_codec *codec)
                                      DA7213_DMIC_DATA_SEL_SHIFT);
                         break;
                 }
-                switch (pdata->dmic_data_sel) {
+                switch (pdata->dmic_samplephase) {
                 case DA7213_DMIC_SAMPLE_ON_CLKEDGE:
                 case DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE:
-                        dmic_cfg |= (pdata->dmic_data_sel <<
+                        dmic_cfg |= (pdata->dmic_samplephase <<
                                      DA7213_DMIC_SAMPLEPHASE_SHIFT);
                         break;
                 }
-                switch (pdata->dmic_data_sel) {
+                switch (pdata->dmic_clk_rate) {
                 case DA7213_DMIC_CLK_3_0MHZ:
                 case DA7213_DMIC_CLK_1_5MHZ:
-                        dmic_cfg |= (pdata->dmic_data_sel <<
+                        dmic_cfg |= (pdata->dmic_clk_rate <<
                                      DA7213_DMIC_CLK_RATE_SHIFT);
                         break;
                 }
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index ce0d36412c97..8d14a76c7249 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2233,7 +2233,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
         dev_dbg(codec->dev, "irq = %d\n", max98090->irq);
 
         ret = request_threaded_irq(max98090->irq, NULL,
-                max98090_interrupt, IRQF_TRIGGER_FALLING,
+                max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
                 "max98090_interrupt", codec);
         if (ret < 0) {
                 dev_err(codec->dev, "request_irq failed: %d\n",
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 65d09d60b7c6..1514bf845e4b 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -187,14 +187,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 
                         break;
                 }
-
-                if (found)
-                        snd_soc_dapm_sync(widget->dapm);
         }
 
-        ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
-
         mutex_unlock(&widget->codec->mutex);
+
+        if (found)
+                snd_soc_dapm_sync(widget->dapm);
+
+        ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val);
         return ret;
 }
 
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index 8df2b6e1a1a6..370af0cbcc9a 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -667,6 +667,7 @@ static int wm0010_boot(struct snd_soc_codec *codec)
                 /* On wm0010 only the CLKCTRL1 value is used */
                 pll_rec.clkctrl1 = wm0010->pll_clkctrl1;
 
+                ret = -ENOMEM;
                 len = pll_rec.length + 8;
                 out = kzalloc(len, GFP_KERNEL);
                 if (!out) {
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index e895d3939eef..100fdadda56a 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -1120,7 +1120,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
 ARIZONA_DSP_WIDGETS(DSP1, "DSP1"),
 
 SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
-                       ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux),
+                       ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+                       &wm5102_aec_loopback_mux),
 
 SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
                    ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 731884e04776..88ad7db52dde 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
 
 ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
@@ -503,7 +503,8 @@ SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
                  NULL, 0),
 
 SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
-                       ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5110_aec_loopback_mux),
+                       ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+                       &wm5110_aec_loopback_mux),
 
 SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
                      ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
@@ -976,6 +977,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
         if (ret != 0)
                 return ret;
 
+        arizona_init_spk(codec);
+
         snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
 
         priv->core.arizona->dapm = &codec->dapm;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 1eb152cb1097..29e95f93d482 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -383,6 +383,8 @@ static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol,
         struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
         int drc = wm8994_get_drc(kcontrol->id.name);
 
+        if (drc < 0)
+                return drc;
         ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc];
 
         return 0;
@@ -488,6 +490,9 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol,
         struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
         int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
 
+        if (block < 0)
+                return block;
+
         ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block];
 
         return 0;
@@ -1031,7 +1036,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
 {
         struct snd_soc_codec *codec = w->codec;
         struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
-        struct wm8994 *control = codec->control_data;
+        struct wm8994 *control = wm8994->wm8994;
         int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA;
         int i;
         int dac;
@@ -3831,8 +3836,14 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
                                 ret);
                 } else if (!(ret & WM1811_JACKDET_LVL)) {
                         dev_dbg(codec->dev, "Ignoring removed jack\n");
-                        return IRQ_HANDLED;
+                        goto out;
                 }
+        } else if (!(reg & WM8958_MICD_STS)) {
+                snd_soc_jack_report(wm8994->micdet[0].jack, 0,
+                                    SND_JACK_MECHANICAL | SND_JACK_HEADSET |
+                                    wm8994->btn_mask);
+                wm8994->mic_detecting = true;
+                goto out;
         }
 
         if (wm8994->mic_detecting)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 56ecfc72f2e9..81490febac6d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -631,7 +631,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
                                        int word_length)
 {
         u32 fmt;
-        u32 rotate = (word_length / 4) & 0x7;
+        u32 tx_rotate = (word_length / 4) & 0x7;
+        u32 rx_rotate = (32 - word_length) / 4;
         u32 mask = (1ULL << word_length) - 1;
 
         /*
@@ -655,9 +656,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
                 mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
                                 TXSSZ(fmt), TXSSZ(0x0F));
                 mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
-                                TXROT(rotate), TXROT(7));
+                                TXROT(tx_rotate), TXROT(7));
                 mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
-                                RXROT(rotate), RXROT(7));
+                                RXROT(rx_rotate), RXROT(7));
                 mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG,
                                 mask);
         }
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 902fab02b851..c6fa03e2114a 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -540,11 +540,6 @@ static int imx_ssi_probe(struct platform_device *pdev)
         clk_prepare_enable(ssi->clk);
 
         res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-        if (!res) {
-                ret = -ENODEV;
-                goto failed_get_resource;
-        }
-
         ssi->base = devm_ioremap_resource(&pdev->dev, res);
         if (IS_ERR(ssi->base)) {
                 ret = PTR_ERR(ssi->base);
@@ -633,7 +628,6 @@ failed_pdev_fiq_alloc:
         snd_soc_unregister_component(&pdev->dev);
 failed_register:
         release_mem_region(res->start, resource_size(res));
-failed_get_resource:
         clk_disable_unprepare(ssi->clk);
 failed_clk:
 
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index befe68f59285..4c9dad3263c5 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -471,11 +471,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
         dev_set_drvdata(&pdev->dev, priv);
 
         mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-        if (!mem) {
-                dev_err(&pdev->dev, "platform_get_resource failed\n");
-                return -ENXIO;
-        }
-
         priv->io = devm_ioremap_resource(&pdev->dev, mem);
         if (IS_ERR(priv->io))
                 return PTR_ERR(priv->io);
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 3853f7eb3f28..06a8000aa07b 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
                         goto err;
         }
 
-        snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
-                                SND_SOC_DAPM_STREAM_START);
+        if (cstream->direction == SND_COMPRESS_PLAYBACK)
+                snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+                                        SND_SOC_DAPM_STREAM_START);
+        else
+                snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE,
+                                        SND_SOC_DAPM_STREAM_START);
 
         /* cancel any delayed stream shutdown that is pending */
         rtd->pop_wait = 0;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a80c883bb8be..c7051c457b75 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -55,7 +55,8 @@ static int dapm_up_seq[] = {
         [snd_soc_dapm_clock_supply] = 1,
         [snd_soc_dapm_micbias] = 2,
         [snd_soc_dapm_dai_link] = 2,
-        [snd_soc_dapm_dai] = 3,
+        [snd_soc_dapm_dai_in] = 3,
+        [snd_soc_dapm_dai_out] = 3,
         [snd_soc_dapm_aif_in] = 3,
         [snd_soc_dapm_aif_out] = 3,
         [snd_soc_dapm_mic] = 4,
@@ -92,7 +93,8 @@ static int dapm_down_seq[] = {
         [snd_soc_dapm_value_mux] = 9,
         [snd_soc_dapm_aif_in] = 10,
         [snd_soc_dapm_aif_out] = 10,
-        [snd_soc_dapm_dai] = 10,
+        [snd_soc_dapm_dai_in] = 10,
+        [snd_soc_dapm_dai_out] = 10,
         [snd_soc_dapm_dai_link] = 11,
         [snd_soc_dapm_clock_supply] = 12,
         [snd_soc_dapm_regulator_supply] = 12,
@@ -419,7 +421,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
         case snd_soc_dapm_clock_supply:
         case snd_soc_dapm_aif_in:
         case snd_soc_dapm_aif_out:
-        case snd_soc_dapm_dai:
+        case snd_soc_dapm_dai_in:
+        case snd_soc_dapm_dai_out:
         case snd_soc_dapm_hp:
         case snd_soc_dapm_mic:
         case snd_soc_dapm_spk:
@@ -820,7 +823,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
         switch (widget->id) {
         case snd_soc_dapm_adc:
         case snd_soc_dapm_aif_out:
-        case snd_soc_dapm_dai:
+        case snd_soc_dapm_dai_out:
                 if (widget->active) {
                         widget->outputs = snd_soc_dapm_suspend_check(widget);
                         return widget->outputs;
@@ -916,7 +919,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
         switch (widget->id) {
         case snd_soc_dapm_dac:
         case snd_soc_dapm_aif_in:
-        case snd_soc_dapm_dai:
+        case snd_soc_dapm_dai_in:
                 if (widget->active) {
                         widget->inputs = snd_soc_dapm_suspend_check(widget);
                         return widget->inputs;
@@ -1135,16 +1138,6 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w)
         return out != 0 && in != 0;
 }
 
-static int dapm_dai_check_power(struct snd_soc_dapm_widget *w)
-{
-        DAPM_UPDATE_STAT(w, power_checks);
-
-        if (w->active)
-                return w->active;
-
-        return dapm_generic_check_power(w);
-}
-
 /* Check to see if an ADC has power */
 static int dapm_adc_check_power(struct snd_soc_dapm_widget *w)
 {
@@ -2318,7 +2311,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
         case snd_soc_dapm_clock_supply:
         case snd_soc_dapm_aif_in:
         case snd_soc_dapm_aif_out:
-        case snd_soc_dapm_dai:
+        case snd_soc_dapm_dai_in:
+        case snd_soc_dapm_dai_out:
         case snd_soc_dapm_dai_link:
                 list_add(&path->list, &dapm->card->paths);
                 list_add(&path->list_sink, &wsink->sources);
@@ -3129,10 +3123,12 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
                 break;
         case snd_soc_dapm_adc:
         case snd_soc_dapm_aif_out:
+        case snd_soc_dapm_dai_out:
                 w->power_check = dapm_adc_check_power;
                 break;
         case snd_soc_dapm_dac:
         case snd_soc_dapm_aif_in:
+        case snd_soc_dapm_dai_in:
                 w->power_check = dapm_dac_check_power;
                 break;
         case snd_soc_dapm_pga:
@@ -3152,9 +3148,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
         case snd_soc_dapm_clock_supply:
                 w->power_check = dapm_supply_check_power;
                 break;
-        case snd_soc_dapm_dai:
-                w->power_check = dapm_dai_check_power;
-                break;
         default:
                 w->power_check = dapm_always_on_check_power;
                 break;
@@ -3375,7 +3368,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
         template.reg = SND_SOC_NOPM;
 
         if (dai->driver->playback.stream_name) {
-                template.id = snd_soc_dapm_dai;
+                template.id = snd_soc_dapm_dai_in;
                 template.name = dai->driver->playback.stream_name;
                 template.sname = dai->driver->playback.stream_name;
 
@@ -3393,7 +3386,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
         }
 
         if (dai->driver->capture.stream_name) {
-                template.id = snd_soc_dapm_dai;
+                template.id = snd_soc_dapm_dai_out;
                 template.name = dai->driver->capture.stream_name;
                 template.sname = dai->driver->capture.stream_name;
 
@@ -3423,8 +3416,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 
         /* For each DAI widget... */
         list_for_each_entry(dai_w, &card->widgets, list) {
-                if (dai_w->id != snd_soc_dapm_dai)
+                switch (dai_w->id) {
+                case snd_soc_dapm_dai_in:
+                case snd_soc_dapm_dai_out:
+                        break;
+                default:
                         continue;
+                }
 
                 dai = dai_w->priv;
 
@@ -3433,8 +3431,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
                         if (w->dapm != dai_w->dapm)
                                 continue;
 
-                        if (w->id == snd_soc_dapm_dai)
+                        switch (w->id) {
+                        case snd_soc_dapm_dai_in:
+                        case snd_soc_dapm_dai_out:
                                 continue;
+                        default:
+                                break;
+                        }
 
                         if (!w->sname)
                                 continue;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 73bb8eefa491..ccb6be4d658d 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -928,8 +928,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream,
         /* Create any new FE <--> BE connections */
         for (i = 0; i < list->num_widgets; i++) {
 
-                if (list->widgets[i]->id != snd_soc_dapm_dai)
+                switch (list->widgets[i]->id) {
+                case snd_soc_dapm_dai_in:
+                case snd_soc_dapm_dai_out:
+                        break;
+                default:
                         continue;
+                }
 
                 /* is there a valid BE rtd for this widget */
                 be = dpcm_get_be(card, list->widgets[i], stream);
@@ -2011,9 +2016,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
                 if (cpu_dai->driver->capture.channels_min)
                         capture = 1;
         } else {
-                if (codec_dai->driver->playback.channels_min)
+                if (codec_dai->driver->playback.channels_min &&
+                    cpu_dai->driver->playback.channels_min)
                         playback = 1;
-                if (codec_dai->driver->capture.channels_min)
+                if (codec_dai->driver->capture.channels_min &&
+                    cpu_dai->driver->capture.channels_min)
                         capture = 1;
         }
 
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index a1d9b0792a1e..b9defcdeb7ef 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -42,8 +42,8 @@ static const u8 ep_w_max_packet_size[] = {
         0x94, 0x01, 0x5c, 0x02  /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */
 };
 
-static const u8 known_fw_versions[][4] = {
-        { 0x03, 0x01, 0x0b, 0x00 }
+static const u8 known_fw_versions[][2] = {
+        { 0x03, 0x01 }
 };
 
 struct ihex_record {
@@ -343,7 +343,7 @@ static int usb6fire_fw_check(u8 *version)
         int i;
 
         for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++)
-                if (!memcmp(version, known_fw_versions + i, 4))
+                if (!memcmp(version, known_fw_versions + i, 2))
                         return 0;
 
         snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 1a033177b83f..64952e2d3ed1 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -147,14 +147,32 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int
                 return -EINVAL;
         }
 
+        alts = &iface->altsetting[0];
+        altsd = get_iface_desc(alts);
+
+        /*
+         * Android with both accessory and audio interfaces enabled gets the
+         * interface numbers wrong.
+         */
+        if ((chip->usb_id == USB_ID(0x18d1, 0x2d04) ||
+             chip->usb_id == USB_ID(0x18d1, 0x2d05)) &&
+            interface == 0 &&
+            altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
+            altsd->bInterfaceSubClass == USB_SUBCLASS_VENDOR_SPEC) {
+                interface = 2;
+                iface = usb_ifnum_to_if(dev, interface);
+                if (!iface)
+                        return -EINVAL;
+                alts = &iface->altsetting[0];
+                altsd = get_iface_desc(alts);
+        }
+
         if (usb_interface_claimed(iface)) {
                 snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n",
                                                 dev->devnum, ctrlif, interface);
                 return -EINVAL;
         }
 
-        alts = &iface->altsetting[0];
-        altsd = get_iface_desc(alts);
         if ((altsd->bInterfaceClass == USB_CLASS_AUDIO ||
              altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) &&
             altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) {
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index ca4739c3f650..d5438083fd6a 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -885,7 +885,9 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval,
 
         case USB_ID(0x046d, 0x0808):
         case USB_ID(0x046d, 0x0809):
+        case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */
         case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */
+        case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */
         case USB_ID(0x046d, 0x0991):
         /* Most audio usb devices lie about volume resolution.
          * Most Logitech webcams have res = 384.
diff --git a/sound/usb/proc.c b/sound/usb/proc.c
index 135c76871063..5f761ab34c01 100644
--- a/sound/usb/proc.c
+++ b/sound/usb/proc.c
@@ -116,21 +116,22 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
 }
 
 static void proc_dump_ep_status(struct snd_usb_substream *subs,
-                                struct snd_usb_endpoint *ep,
+                                struct snd_usb_endpoint *data_ep,
+                                struct snd_usb_endpoint *sync_ep,
                                 struct snd_info_buffer *buffer)
 {
-        if (!ep)
+        if (!data_ep)
                 return;
-        snd_iprintf(buffer, "    Packet Size = %d\n", ep->curpacksize);
+        snd_iprintf(buffer, "    Packet Size = %d\n", data_ep->curpacksize);
         snd_iprintf(buffer, "    Momentary freq = %u Hz (%#x.%04x)\n",
                     subs->speed == USB_SPEED_FULL
-                    ? get_full_speed_hz(ep->freqm)
-                    : get_high_speed_hz(ep->freqm),
-                    ep->freqm >> 16, ep->freqm & 0xffff);
-        if (ep->freqshift != INT_MIN) {
-                int res = 16 - ep->freqshift;
+                    ? get_full_speed_hz(data_ep->freqm)
+                    : get_high_speed_hz(data_ep->freqm),
+                    data_ep->freqm >> 16, data_ep->freqm & 0xffff);
+        if (sync_ep && data_ep->freqshift != INT_MIN) {
+                int res = 16 - data_ep->freqshift;
                 snd_iprintf(buffer, "    Feedback Format = %d.%d\n",
-                            (ep->syncmaxsize > 3 ? 32 : 24) - res, res);
+                            (sync_ep->syncmaxsize > 3 ? 32 : 24) - res, res);
         }
 }
 
@@ -140,8 +141,7 @@ static void proc_dump_substream_status(struct snd_usb_substream *subs, struct sn
                 snd_iprintf(buffer, "  Status: Running\n");
                 snd_iprintf(buffer, "    Interface = %d\n", subs->interface);
                 snd_iprintf(buffer, "    Altset = %d\n", subs->altset_idx);
-                proc_dump_ep_status(subs, subs->data_endpoint, buffer);
-                proc_dump_ep_status(subs, subs->sync_endpoint, buffer);
+                proc_dump_ep_status(subs, subs->data_endpoint, subs->sync_endpoint, buffer);
         } else {
                 snd_iprintf(buffer, "  Status: Stop\n");
         }
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 7f1722f82c89..8b75bcf136f6 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -215,7 +215,13 @@
         .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
 },
 {
-        USB_DEVICE(0x046d, 0x0990),
+        .match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+                       USB_DEVICE_ID_MATCH_INT_CLASS |
+                       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+        .idVendor = 0x046d,
+        .idProduct = 0x0990,
+        .bInterfaceClass = USB_CLASS_AUDIO,
+        .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
         .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
                 .vendor_name = "Logitech, Inc.",
                 .product_name = "QuickCam Pro 9000",
@@ -1792,7 +1798,11 @@ YAMAHA_DEVICE(0x7010, "UB99"),
         USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108),
         .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
                 .ifnum = 0,
-                .type = QUIRK_MIDI_STANDARD_INTERFACE
+                .type = QUIRK_MIDI_FIXED_ENDPOINT,
+                .data = & (const struct snd_usb_midi_endpoint_info) {
+                        .out_cables = 0x0007,
+                        .in_cables  = 0x0007
+                }
         }
 },
 {