Merge git://git.kernel.org/pub/scm/linux/kernel/git/davem/net

Conflicts:
	Documentation/networking/ip-sysctl.txt
	drivers/net/ethernet/broadcom/bnx2x/bnx2x_cmn.c

Both conflicts were simply overlapping context.

A build fix for qlcnic is in here too, simply removing the added
devinit annotations which no longer exist.

Signed-off-by: David S. Miller <davem@davemloft.net>

28 files changed, 310 insertions, 163 deletions

diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 6fc0ae90e5b1..fff7753e35c1 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -18,6 +18,7 @@
 #include <linux/delay.h>
 #include <linux/module.h>
 #include <linux/io.h>
+#include <linux/gpio.h>
 
 #include <sound/ac97_codec.h>
 #include <sound/pxa2xx-lib.h>
@@ -148,6 +149,8 @@ static inline void pxa_ac97_warm_pxa27x(void)
 
 static inline void pxa_ac97_cold_pxa27x(void)
 {
+        unsigned int timeout;
+
         GCR &=  GCR_COLD_RST;  /* clear everything but nCRST */
         GCR &= ~GCR_COLD_RST;  /* then assert nCRST */
 
@@ -157,8 +160,10 @@ static inline void pxa_ac97_cold_pxa27x(void)
         clk_enable(ac97conf_clk);
         udelay(5);
         clk_disable(ac97conf_clk);
-        GCR = GCR_COLD_RST;
-        udelay(50);
+        GCR = GCR_COLD_RST | GCR_WARM_RST;
+        timeout = 100;     /* wait for the codec-ready bit to be set */
+        while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--)
+                mdelay(1);
 }
 #endif
 
@@ -340,8 +345,21 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev)
         }
 
         if (cpu_is_pxa27x()) {
-                /* Use GPIO 113 as AC97 Reset on Bulverde */
+                /*
+                 * This gpio is needed for a work-around to a bug in the ac97
+                 * controller during warm reset.  The direction and level is set
+                 * here so that it is an output driven high when switching from
+                 * AC97_nRESET alt function to generic gpio.
+                 */
+                ret = gpio_request_one(reset_gpio, GPIOF_OUT_INIT_HIGH,
+                                       "pxa27x ac97 reset");
+                if (ret < 0) {
+                        pr_err("%s: gpio_request_one() failed: %d\n",
+                               __func__, ret);
+                        goto err_conf;
+                }
                 pxa27x_assert_ac97reset(reset_gpio, 0);
+
                 ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK");
                 if (IS_ERR(ac97conf_clk)) {
                         ret = PTR_ERR(ac97conf_clk);
@@ -384,6 +402,8 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_probe);
 
 void pxa2xx_ac97_hw_remove(struct platform_device *dev)
 {
+        if (cpu_is_pxa27x())
+                gpio_free(reset_gpio);
         GCR |= GCR_ACLINK_OFF;
         free_irq(IRQ_AC97, NULL);
         if (ac97conf_clk) {
diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c
index dabf8a871dcc..7004e24d209f 100644
--- a/sound/oss/pas2_card.c
+++ b/sound/oss/pas2_card.c
@@ -333,6 +333,11 @@ static void __init attach_pas_card(struct address_info *hw_config)
                 {
                         char            temp[100];
 
+                        if (pas_model < 0 ||
+                            pas_model >= ARRAY_SIZE(pas_model_names)) {
+                                printk(KERN_ERR "pas2 unrecognized model.\n");
+                                return;
+                        }
                         sprintf(temp,
                             "%s rev %d", pas_model_names[(int) pas_model],
                                     pas_read(0x2789));
diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c
index 2805e34bd41d..8bef47311e45 100644
--- a/sound/pci/au88x0/au88x0_synth.c
+++ b/sound/pci/au88x0/au88x0_synth.c
@@ -58,7 +58,7 @@ static void vortex_wt_setdsout(vortex_t * vortex, u32 wt, int en)
         if (en)
                 temp |= (1 << (wt & 0x1f));
         else
-                temp &= (1 << ~(wt & 0x1f));
+                temp &= ~(1 << (wt & 0x1f));
         hwwrite(vortex->mmio, WT_DSREG((wt >= 0x20) ? 1 : 0), temp);
 }
 
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8353c77536ac..b8fb0a5adb9b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2531,7 +2531,7 @@ static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol,
                                   struct snd_ctl_elem_info *uinfo)
 {
         static const char * const texts[] = {
-                "Off", "On", "Follow Master"
+                "On", "Off", "Follow Master"
         };
         unsigned int index;
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index cca87277baf0..0b6aebacc56b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -573,9 +573,12 @@ enum {
 #define AZX_DCAPS_PM_RUNTIME    (1 << 26)       /* runtime PM support */
 
 /* quirks for Intel PCH */
-#define AZX_DCAPS_INTEL_PCH \
+#define AZX_DCAPS_INTEL_PCH_NOPM \
         (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \
-         AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME)
+         AZX_DCAPS_COUNT_LPIB_DELAY)
+
+#define AZX_DCAPS_INTEL_PCH \
+        (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME)
 
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
@@ -3586,13 +3589,13 @@ static void azx_remove(struct pci_dev *pci)
 static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
         /* CPT */
         { PCI_DEVICE(0x8086, 0x1c20),
-          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
         /* PBG */
         { PCI_DEVICE(0x8086, 0x1d20),
-          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
         /* Panther Point */
         { PCI_DEVICE(0x8086, 0x1e20),
-          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
         /* Lynx Point */
         { PCI_DEVICE(0x8086, 0x8c20),
           .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 60890bfecc19..dd798c3196ff 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -558,24 +558,12 @@ static int conexant_build_controls(struct hda_codec *codec)
         return 0;
 }
 
-#ifdef CONFIG_PM
-static int conexant_suspend(struct hda_codec *codec)
-{
-        snd_hda_shutup_pins(codec);
-        return 0;
-}
-#endif
-
 static const struct hda_codec_ops conexant_patch_ops = {
         .build_controls = conexant_build_controls,
         .build_pcms = conexant_build_pcms,
         .init = conexant_init,
         .free = conexant_free,
         .set_power_state = conexant_set_power,
-#ifdef CONFIG_PM
-        .suspend = conexant_suspend,
-#endif
-        .reboot_notify = snd_hda_shutup_pins,
 };
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
@@ -4405,10 +4393,6 @@ static const struct hda_codec_ops cx_auto_patch_ops = {
         .init = cx_auto_init,
         .free = conexant_free,
         .unsol_event = snd_hda_jack_unsol_event,
-#ifdef CONFIG_PM
-        .suspend = conexant_suspend,
-#endif
-        .reboot_notify = snd_hda_shutup_pins,
 };
 
 /*
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6ee34593774a..f5196277b6e9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5817,6 +5817,9 @@ enum {
         ALC269_TYPE_ALC269VB,
         ALC269_TYPE_ALC269VC,
         ALC269_TYPE_ALC269VD,
+        ALC269_TYPE_ALC280,
+        ALC269_TYPE_ALC282,
+        ALC269_TYPE_ALC284,
 };
 
 /*
@@ -5833,10 +5836,13 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
         switch (spec->codec_variant) {
         case ALC269_TYPE_ALC269VA:
         case ALC269_TYPE_ALC269VC:
+        case ALC269_TYPE_ALC280:
+        case ALC269_TYPE_ALC284:
                 ssids = alc269va_ssids;
                 break;
         case ALC269_TYPE_ALC269VB:
         case ALC269_TYPE_ALC269VD:
+        case ALC269_TYPE_ALC282:
                 ssids = alc269_ssids;
                 break;
         default:
@@ -5992,6 +5998,30 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec,
         spec->automute_hook = alc269_quanta_automute;
 }
 
+/* update mute-LED according to the speaker mute state via mic1 VREF pin */
+static void alc269_fixup_mic1_mute_hook(void *private_data, int enabled)
+{
+        struct hda_codec *codec = private_data;
+        unsigned int pinval = AC_PINCTL_IN_EN + (enabled ?
+                              AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80);
+        snd_hda_set_pin_ctl_cache(codec, 0x18, pinval);
+}
+
+static void alc269_fixup_mic1_mute(struct hda_codec *codec,
+                                   const struct alc_fixup *fix, int action)
+{
+        struct alc_spec *spec = codec->spec;
+        switch (action) {
+        case ALC_FIXUP_ACT_BUILD:
+                spec->vmaster_mute.hook = alc269_fixup_mic1_mute_hook;
+                snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true);
+                /* fallthru */
+        case ALC_FIXUP_ACT_INIT:
+                snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
+                break;
+        }
+}
+
 /* update mute-LED according to the speaker mute state via mic2 VREF pin */
 static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled)
 {
@@ -6043,6 +6073,7 @@ enum {
         ALC269_FIXUP_DMIC,
         ALC269VB_FIXUP_AMIC,
         ALC269VB_FIXUP_DMIC,
+        ALC269_FIXUP_MIC1_MUTE_LED,
         ALC269_FIXUP_MIC2_MUTE_LED,
         ALC269_FIXUP_INV_DMIC,
         ALC269_FIXUP_LENOVO_DOCK,
@@ -6171,6 +6202,10 @@ static const struct alc_fixup alc269_fixups[] = {
                         { }
                 },
         },
+        [ALC269_FIXUP_MIC1_MUTE_LED] = {
+                .type = ALC_FIXUP_FUNC,
+                .v.func = alc269_fixup_mic1_mute,
+        },
         [ALC269_FIXUP_MIC2_MUTE_LED] = {
                 .type = ALC_FIXUP_FUNC,
                 .v.func = alc269_fixup_mic2_mute,
@@ -6215,6 +6250,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC),
         SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
         SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED),
+        SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED),
         SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
         SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC),
         SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
@@ -6370,7 +6406,8 @@ static int patch_alc269(struct hda_codec *codec)
 
         alc_auto_parse_customize_define(codec);
 
-        if (codec->vendor_id == 0x10ec0269) {
+        switch (codec->vendor_id) {
+        case 0x10ec0269:
                 spec->codec_variant = ALC269_TYPE_ALC269VA;
                 switch (alc_get_coef0(codec) & 0x00f0) {
                 case 0x0010:
@@ -6395,6 +6432,20 @@ static int patch_alc269(struct hda_codec *codec)
                         goto error;
                 spec->init_hook = alc269_fill_coef;
                 alc269_fill_coef(codec);
+                break;
+
+        case 0x10ec0280:
+        case 0x10ec0290:
+                spec->codec_variant = ALC269_TYPE_ALC280;
+                break;
+        case 0x10ec0282:
+        case 0x10ec0283:
+                spec->codec_variant = ALC269_TYPE_ALC282;
+                break;
+        case 0x10ec0284:
+        case 0x10ec0292:
+                spec->codec_variant = ALC269_TYPE_ALC284;
+                break;
         }
 
         /* automatic parse from the BIOS config */
@@ -7099,6 +7150,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
         { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 },
         { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 },
         { .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 },
+        { .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 },
         { .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 },
         { .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 },
         { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 6e02e064d7b4..223c3d9cc69e 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -441,6 +441,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 */
 /* status */
 #define HDSPM_AES32_wcLock      0x0200000
+#define HDSPM_AES32_wcSync      0x0100000
 #define HDSPM_AES32_wcFreq_bit  22
 /* (status >> HDSPM_AES32_wcFreq_bit) & 0xF gives WC frequency (cf function
   HDSPM_bit2freq */
@@ -3467,10 +3468,12 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm)
         switch (hdspm->io_type) {
         case AES32:
                 status = hdspm_read(hdspm, HDSPM_statusRegister);
-                if (status & HDSPM_wcSync)
-                        return 2;
-                else if (status & HDSPM_wcLock)
-                        return 1;
+                if (status & HDSPM_AES32_wcLock) {
+                        if (status & HDSPM_AES32_wcSync)
+                                return 2;
+                        else
+                                return 1;
+                }
                 return 0;
                 break;
 
@@ -4658,6 +4661,7 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
         unsigned int status;
         unsigned int status2;
         unsigned int timecode;
+        unsigned int wcLock, wcSync;
         int pref_syncref;
         char *autosync_ref;
         int x;
@@ -4751,8 +4755,11 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
 
         snd_iprintf(buffer, "--- Status:\n");
 
+        wcLock = status & HDSPM_AES32_wcLock;
+        wcSync = wcLock && (status & HDSPM_AES32_wcSync);
+
         snd_iprintf(buffer, "Word: %s  Frequency: %d\n",
-                    (status & HDSPM_AES32_wcLock) ? "Sync   " : "No Lock",
+                    (wcLock) ? (wcSync ? "Sync   " : "Lock   ") : "No Lock",
                     HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF));
 
         for (x = 0; x < 8; x++) {
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index adf397b9d0e6..1d8bb5917594 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -446,15 +446,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
         case SND_SOC_DAIFMT_DSP_A:
                 mode = 0;
                 break;
-        case SND_SOC_DAIFMT_DSP_B:
-                mode = 1;
-                break;
         case SND_SOC_DAIFMT_I2S:
                 mode = 2;
                 break;
-        case SND_SOC_DAIFMT_LEFT_J:
-                mode = 3;
-                break;
         default:
                 arizona_aif_err(dai, "Unsupported DAI format %d\n",
                                 fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -714,7 +708,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
                 snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1,
                                     ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val);
                 snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
-                                    ARIZONA_AIF1_RATE_MASK, 8);
+                                    ARIZONA_AIF1_RATE_MASK,
+                                    8 << ARIZONA_AIF1_RATE_SHIFT);
                 break;
         default:
                 arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 41dae1ed3b71..4deebeb07177 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -34,15 +34,15 @@
 
 #define ARIZONA_FLL_SRC_MCLK1      0
 #define ARIZONA_FLL_SRC_MCLK2      1
-#define ARIZONA_FLL_SRC_SLIMCLK    2
-#define ARIZONA_FLL_SRC_FLL1       3
-#define ARIZONA_FLL_SRC_FLL2       4
-#define ARIZONA_FLL_SRC_AIF1BCLK   5
-#define ARIZONA_FLL_SRC_AIF2BCLK   6
-#define ARIZONA_FLL_SRC_AIF3BCLK   7
-#define ARIZONA_FLL_SRC_AIF1LRCLK  8
-#define ARIZONA_FLL_SRC_AIF2LRCLK  9
-#define ARIZONA_FLL_SRC_AIF3LRCLK 10
+#define ARIZONA_FLL_SRC_SLIMCLK    3
+#define ARIZONA_FLL_SRC_FLL1       4
+#define ARIZONA_FLL_SRC_FLL2       5
+#define ARIZONA_FLL_SRC_AIF1BCLK   8
+#define ARIZONA_FLL_SRC_AIF2BCLK   9
+#define ARIZONA_FLL_SRC_AIF3BCLK  10
+#define ARIZONA_FLL_SRC_AIF1LRCLK 12
+#define ARIZONA_FLL_SRC_AIF2LRCLK 13
+#define ARIZONA_FLL_SRC_AIF3LRCLK 14
 
 #define ARIZONA_MIXER_VOL_MASK             0x00FE
 #define ARIZONA_MIXER_VOL_SHIFT                 1
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 4f1127935fdf..ac8742a1f25a 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -474,16 +474,16 @@ static int cs4271_probe(struct snd_soc_codec *codec)
         struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
         int ret;
         int gpio_nreset = -EINVAL;
-        int amutec_eq_bmutec = 0;
+        bool amutec_eq_bmutec = false;
 
 #ifdef CONFIG_OF
         if (of_match_device(cs4271_dt_ids, codec->dev)) {
                 gpio_nreset = of_get_named_gpio(codec->dev->of_node,
                                                 "reset-gpio", 0);
 
-                if (!of_get_property(codec->dev->of_node,
+                if (of_get_property(codec->dev->of_node,
                                      "cirrus,amutec-eq-bmutec", NULL))
-                        amutec_eq_bmutec = 1;
+                        amutec_eq_bmutec = true;
         }
 #endif
 
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 99bb1c69499e..9811a5478c87 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -737,7 +737,7 @@ static const struct cs42l52_clk_para clk_map_table[] = {
 
 static int cs42l52_get_clk(int mclk, int rate)
 {
-        int i, ret = 0;
+        int i, ret = -EINVAL;
         u_int mclk1, mclk2 = 0;
 
         for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
@@ -749,8 +749,6 @@ static int cs42l52_get_clk(int mclk, int rate)
                         }
                 }
         }
-        if (ret > ARRAY_SIZE(clk_map_table))
-                return -EINVAL;
         return ret;
 }
 
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index d75257d40a49..e19490cfb3a8 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = {
         { 101, 0x00 },
         { 102, 0x00 },
         { 103, 0x01 },
-        { 105, 0x01 },
-        { 106, 0x00 },
-        { 107, 0x01 },
+        { 104, 0x01 },
+        { 105, 0x00 },
+        { 106, 0x01 },
         { 107, 0x00 },
         { 108, 0x00 },
         { 109, 0x00 },
@@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = {
         { 184, 0x00 },
         { 185, 0x00 },
         { 186, 0x00 },
-        { 189, 0x00 },
+        { 187, 0x00 },
         { 188, 0x00 },
-        { 194, 0x00 },
-        { 195, 0x00 },
-        { 196, 0x00 },
-        { 197, 0x00 },
-        { 200, 0x00 },
-        { 201, 0x00 },
-        { 202, 0x00 },
-        { 203, 0x00 },
-        { 204, 0x00 },
-        { 205, 0x00 },
-        { 208, 0x00 },
+        { 189, 0x00 },
+        { 208, 0x06 },
         { 209, 0x00 },
-        { 210, 0x00 },
-        { 211, 0x00 },
-        { 213, 0x00 },
-        { 214, 0x00 },
-        { 215, 0x00 },
-        { 216, 0x00 },
-        { 217, 0x00 },
-        { 218, 0x00 },
-        { 219, 0x00 },
+        { 210, 0x08 },
+        { 211, 0x54 },
+        { 212, 0x14 },
+        { 213, 0x0d },
+        { 214, 0x0d },
+        { 215, 0x14 },
+        { 216, 0x60 },
         { 221, 0x00 },
         { 222, 0x00 },
+        { 223, 0x00 },
         { 224, 0x00 },
-        { 225, 0x00 },
-        { 226, 0x00 },
-        { 227, 0x00 },
-        { 228, 0x00 },
-        { 229, 0x00 },
-        { 230, 0x13 },
-        { 231, 0x00 },
-        { 232, 0x80 },
-        { 233, 0x0C },
-        { 234, 0xDD },
-        { 235, 0x00 },
-        { 236, 0x04 },
-        { 237, 0x00 },
-        { 238, 0x00 },
-        { 239, 0x00 },
-        { 240, 0x00 },
-        { 241, 0x00 },
-        { 242, 0x00 },
-        { 243, 0x00 },
-        { 244, 0x00 },
-        { 245, 0x00 },
         { 248, 0x00 },
         { 249, 0x00 },
-        { 254, 0x00 },
+        { 250, 0x00 },
         { 255, 0x00 },
 };
 
@@ -525,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0),
 };
 
 /* TLV Declarations */
-static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1);
-static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1);
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1);
+static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0);
+static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0);
 
 static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = {
 /* Sidetone supports mono only */
 SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG,
-                     0, 0x3F, 0, digital_tlv),
+                     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG,
-                     0, 0x3F, 0, digital_tlv),
+                     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG,
-                     0, 0x3F, 0, digital_tlv),
+                     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG,
-                     0, 0x3F, 0, digital_tlv),
+                     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG,
-                     0, 0x3F, 0, digital_tlv),
+                     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG,
-                     0, 0x3F, 0, digital_tlv),
+                     0, 0x3F, 0, stn_tlv),
 };
 
 static const struct snd_kcontrol_new lm49453_snd_controls[] = {
         /* mic1 and mic2 supports mono only */
-        SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6,
-                        0, digital_tlv),
-        SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6,
-                        0, digital_tlv),
+        SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv),
+        SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv),
+
+        SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63,
+                        0, adc_dac_tlv),
+        SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63,
+                        0, adc_dac_tlv),
 
         SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG,
-                          LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),
+                          LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
         SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG,
-                          LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv),
+                          LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 
         SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum),
         SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum),
@@ -569,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = {
                                           2, 1, 0),
 
         SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG,
-                          LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv),
+                          LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
         SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG,
-                          LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv),
+                          LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
         SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG,
-                          LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv),
+                          LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
         SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG,
-                          LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv),
+                          LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 
         SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG,
-                        0, 6, 0, digital_tlv),
+                        0, 63, 0, adc_dac_tlv),
 
         SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
                         0, 3, 0, port_tlv),
@@ -1218,7 +1192,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
         }
 
         snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG,
-                            LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5),
+                            LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5),
                             (aif_val | mode | clk_phase));
 
         snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index cb1675cd8e1c..92bbfec9b107 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
                         5, 1, 0),
 
         SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL,
-                        0, 4, 0, mic_gain_tlv),
+                        0, 3, 0, mic_gain_tlv),
 };
 
 /* mute the codec used by alsa core */
@@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
                         SGTL5000_HP_ZCD_EN |
                         SGTL5000_ADC_ZCD_EN);
 
-        snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0);
+        snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2);
 
         /*
          * disable DAP
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index ab355c4f0b2d..40c07be9b581 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -74,9 +74,10 @@
                                 SNDRV_PCM_FMTBIT_S32_LE)
 #define S2PC_VALUE              0x98
 #define CLOCK_OUT               0x60
-#define LEFT_J_DATA_FORMAT      0x10
-#define I2S_DATA_FORMAT         0x12
-#define RIGHT_J_DATA_FORMAT     0x14
+#define DATA_FORMAT_MSK         0x0E
+#define LEFT_J_DATA_FORMAT      0x00
+#define I2S_DATA_FORMAT         0x02
+#define RIGHT_J_DATA_FORMAT     0x04
 #define CODEC_MUTE_VAL          0x80
 
 #define POWER_CNTLMSAK          0x40
@@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
                 return -EINVAL;
         }
 
-        snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode);
+        snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode);
 
         return 0;
 }
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 1cbe88f01d63..12bcae63a7f0 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -209,9 +209,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
 
         ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY);
         if (wm2000->speech_clarity)
-                ret &= ~WM2000_SPEECH_CLARITY;
-        else
                 ret |= WM2000_SPEECH_CLARITY;
+        else
+                ret &= ~WM2000_SPEECH_CLARITY;
         wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret);
 
         wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33);
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index afcf31df77e0..e6cefe1ac677 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1566,15 +1566,9 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
         case SND_SOC_DAIFMT_DSP_A:
                 fmt_val = 0;
                 break;
-        case SND_SOC_DAIFMT_DSP_B:
-                fmt_val = 1;
-                break;
         case SND_SOC_DAIFMT_I2S:
                 fmt_val = 2;
                 break;
-        case SND_SOC_DAIFMT_LEFT_J:
-                fmt_val = 3;
-                break;
         default:
                 dev_err(codec->dev, "Unsupported DAI format %d\n",
                         fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -1626,7 +1620,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
                             WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV,
                             lrclk);
         snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5,
-                            WM2200_AIF1_FMT_MASK << 1, fmt_val << 1);
+                            WM2200_AIF1_FMT_MASK, fmt_val);
 
         return 0;
 }
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 5a5f36936235..54397a508073 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1279,15 +1279,9 @@ static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
         case SND_SOC_DAIFMT_DSP_A:
                 mask = 0;
                 break;
-        case SND_SOC_DAIFMT_DSP_B:
-                mask = 1;
-                break;
         case SND_SOC_DAIFMT_I2S:
                 mask = 2;
                 break;
-        case SND_SOC_DAIFMT_LEFT_J:
-                mask = 3;
-                break;
         default:
                 dev_err(codec->dev, "Unsupported DAI format %d\n",
                         fmt & SND_SOC_DAIFMT_FORMAT_MASK);
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 688ade080589..7a9048dad1cd 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -36,6 +36,9 @@
 struct wm5102_priv {
         struct arizona_priv core;
         struct arizona_fll fll[2];
+
+        unsigned int spk_ena:2;
+        unsigned int spk_ena_pending:1;
 };
 
 static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
@@ -787,6 +790,47 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
 };
 
+static int wm5102_spk_ev(struct snd_soc_dapm_widget *w,
+                         struct snd_kcontrol *kcontrol,
+                         int event)
+{
+        struct snd_soc_codec *codec = w->codec;
+        struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+        struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec);
+
+        if (arizona->rev < 1)
+                return 0;
+
+        switch (event) {
+        case SND_SOC_DAPM_PRE_PMU:
+                if (!wm5102->spk_ena) {
+                        snd_soc_write(codec, 0x4f5, 0x25a);
+                        wm5102->spk_ena_pending = true;
+                }
+                break;
+        case SND_SOC_DAPM_POST_PMU:
+                if (wm5102->spk_ena_pending) {
+                        msleep(75);
+                        snd_soc_write(codec, 0x4f5, 0xda);
+                        wm5102->spk_ena_pending = false;
+                        wm5102->spk_ena++;
+                }
+                break;
+        case SND_SOC_DAPM_PRE_PMD:
+                wm5102->spk_ena--;
+                if (!wm5102->spk_ena)
+                        snd_soc_write(codec, 0x4f5, 0x25a);
+                break;
+        case SND_SOC_DAPM_POST_PMD:
+                if (!wm5102->spk_ena)
+                        snd_soc_write(codec, 0x4f5, 0x0da);
+                break;
+        }
+
+        return 0;
+}
+
+
 ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
@@ -1034,10 +1078,10 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
                    ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
                    SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
-                   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+                   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
                    SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
-                   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+                   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
                    SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
                    ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index ffc89fab96fb..7b198c38f3ef 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp)
         const struct wm_adsp_region *mem;
         const char *region_name;
         char *file, *text;
+        void *buf;
         unsigned int reg;
         int regions = 0;
         int ret, offset, type, sizes;
@@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp)
                 }
 
                 if (reg) {
-                        ret = regmap_raw_write(regmap, reg, region->data,
+                        buf = kmemdup(region->data, le32_to_cpu(region->len),
+                                      GFP_KERNEL);
+                        if (!buf) {
+                                adsp_err(dsp, "Out of memory\n");
+                                return -ENOMEM;
+                        }
+
+                        ret = regmap_raw_write(regmap, reg, buf,
                                                le32_to_cpu(region->len));
+
+                        kfree(buf);
+
                         if (ret != 0) {
                                 adsp_err(dsp,
                                         "%s.%d: Failed to write %d bytes at %d in %s: %d\n",
@@ -359,6 +370,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
         const char *region_name;
         int ret, pos, blocks, type, offset, reg;
         char *file;
+        void *buf;
 
         file = kzalloc(PAGE_SIZE, GFP_KERNEL);
         if (file == NULL)
@@ -426,6 +438,13 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
                 }
 
                 if (reg) {
+                        buf = kmemdup(blk->data, le32_to_cpu(blk->len),
+                                      GFP_KERNEL);
+                        if (!buf) {
+                                adsp_err(dsp, "Out of memory\n");
+                                return -ENOMEM;
+                        }
+
                         ret = regmap_raw_write(regmap, reg, blk->data,
                                                le32_to_cpu(blk->len));
                         if (ret != 0) {
@@ -433,6 +452,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
                                         "%s.%d: Failed to write to %x in %s\n",
                                         file, blocks, reg, region_name);
                         }
+
+                        kfree(buf);
                 }
 
                 pos += le32_to_cpu(blk->len) + sizeof(*blk);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 91d592ff67b7..2370063b5824 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1255,6 +1255,8 @@ static int soc_post_component_init(struct snd_soc_card *card,
         INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
         ret = device_add(rtd->dev);
         if (ret < 0) {
+                /* calling put_device() here to free the rtd->dev */
+                put_device(rtd->dev);
                 dev_err(card->dev,
                         "ASoC: failed to register runtime device: %d\n", ret);
                 return ret;
@@ -1554,7 +1556,7 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num)
         /* unregister the rtd device */
         if (rtd->dev_registered) {
                 device_remove_file(rtd->dev, &dev_attr_codec_reg);
-                device_del(rtd->dev);
+                device_unregister(rtd->dev);
                 rtd->dev_registered = 0;
         }
 
@@ -2917,7 +2919,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol,
         platform_max = mc->platform_max;
 
         uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-        uinfo->count = 1;
+        uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1;
         uinfo->value.integer.min = 0;
         uinfo->value.integer.max = platform_max - min;
 
@@ -2941,12 +2943,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
                 (struct soc_mixer_control *)kcontrol->private_value;
         struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
         unsigned int reg = mc->reg;
+        unsigned int rreg = mc->rreg;
         unsigned int shift = mc->shift;
         int min = mc->min;
         int max = mc->max;
         unsigned int mask = (1 << fls(max)) - 1;
         unsigned int invert = mc->invert;
         unsigned int val, val_mask;
+        int ret;
 
         val = ((ucontrol->value.integer.value[0] + min) & mask);
         if (invert)
@@ -2954,7 +2958,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
         val_mask = mask << shift;
         val = val << shift;
 
-        return snd_soc_update_bits_locked(codec, reg, val_mask, val);
+        ret = snd_soc_update_bits_locked(codec, reg, val_mask, val);
+        if (ret != 0)
+                return ret;
+
+        if (snd_soc_volsw_is_stereo(mc)) {
+                val = ((ucontrol->value.integer.value[1] + min) & mask);
+                if (invert)
+                        val = max - val;
+                val_mask = mask << shift;
+                val = val << shift;
+
+                ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val);
+        }
+
+        return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range);
 
@@ -2974,6 +2992,7 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
                 (struct soc_mixer_control *)kcontrol->private_value;
         struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
         unsigned int reg = mc->reg;
+        unsigned int rreg = mc->rreg;
         unsigned int shift = mc->shift;
         int min = mc->min;
         int max = mc->max;
@@ -2988,6 +3007,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
         ucontrol->value.integer.value[0] =
                 ucontrol->value.integer.value[0] - min;
 
+        if (snd_soc_volsw_is_stereo(mc)) {
+                ucontrol->value.integer.value[1] =
+                        (snd_soc_read(codec, rreg) >> shift) & mask;
+                if (invert)
+                        ucontrol->value.integer.value[1] =
+                                max - ucontrol->value.integer.value[1];
+                ucontrol->value.integer.value[1] =
+                        ucontrol->value.integer.value[1] - min;
+        }
+
         return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index d7711fce119b..cf191e6aebbe 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1243,6 +1243,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
                 if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
                     (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
                     (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
+                    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) &&
                     (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
                         continue;
 
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index c183d34842ac..34b9bb7fe87c 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -2181,10 +2181,6 @@ int snd_usbmidi_create(struct snd_card *card,
                 umidi->usb_protocol_ops = &snd_usbmidi_novation_ops;
                 err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
                 break;
-        case QUIRK_MIDI_MBOX2:
-                umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops;
-                err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
-                break;
         case QUIRK_MIDI_RAW_BYTES:
                 umidi->usb_protocol_ops = &snd_usbmidi_raw_ops;
                 /*
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 0422b1360af3..15520de1df56 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1206,7 +1206,7 @@ static int snd_c400_create_mixer(struct usb_mixer_interface *mixer)
  * are valid they presents mono controls as L and R channels of
  * stereo. So we provide a good mixer here.
  */
-struct std_mono_table ebox44_table[] = {
+static struct std_mono_table ebox44_table[] = {
         {
                 .unitid = 4,
                 .control = 1,
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index c6593101c049..d82e378d37cb 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -511,6 +511,16 @@ static int configure_sync_endpoint(struct snd_usb_substream *subs)
         struct snd_usb_substream *sync_subs =
                 &subs->stream->substream[subs->direction ^ 1];
 
+        if (subs->sync_endpoint->type != SND_USB_ENDPOINT_TYPE_DATA ||
+            !subs->stream)
+                return snd_usb_endpoint_set_params(subs->sync_endpoint,
+                                                   subs->pcm_format,
+                                                   subs->channels,
+                                                   subs->period_bytes,
+                                                   subs->cur_rate,
+                                                   subs->cur_audiofmt,
+                                                   NULL);
+
         /* Try to find the best matching audioformat. */
         list_for_each_entry(fp, &sync_subs->fmt_list, list) {
                 int score = match_endpoint_audioformats(fp, subs->cur_audiofmt,
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index cdcf6b45e8a8..78e845ec65da 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -50,6 +50,28 @@
         }
 },
 
+{
+        /* Creative BT-D1 */
+        USB_DEVICE(0x041e, 0x0005),
+        .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+                .ifnum = 1,
+                .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+                .data = &(const struct audioformat) {
+                        .formats = SNDRV_PCM_FMTBIT_S16_LE,
+                        .channels = 2,
+                        .iface = 1,
+                        .altsetting = 1,
+                        .altset_idx = 1,
+                        .endpoint = 0x03,
+                        .ep_attr = USB_ENDPOINT_XFER_ISOC,
+                        .attributes = 0,
+                        .rates = SNDRV_PCM_RATE_CONTINUOUS,
+                        .rate_min = 48000,
+                        .rate_max = 48000,
+                }
+        }
+},
+
 /* Creative/Toshiba Multimedia Center SB-0500 */
 {
         USB_DEVICE(0x041e, 0x3048),
@@ -2993,7 +3015,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
                         },
                         {
                                 .ifnum = 6,
-                                .type = QUIRK_MIDI_MBOX2,
+                                .type = QUIRK_MIDI_MIDIMAN,
                                 .data = &(const struct snd_usb_midi_endpoint_info) {
                                         .out_ep =  0x02,
                                         .out_cables = 0x0001,
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index f104c68fe1e0..acc12f004c23 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -306,7 +306,6 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
                 [QUIRK_MIDI_YAMAHA] = create_any_midi_quirk,
                 [QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk,
                 [QUIRK_MIDI_NOVATION] = create_any_midi_quirk,
-                [QUIRK_MIDI_MBOX2] = create_any_midi_quirk,
                 [QUIRK_MIDI_RAW_BYTES] = create_any_midi_quirk,
                 [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk,
                 [QUIRK_MIDI_CME] = create_any_midi_quirk,
@@ -528,11 +527,11 @@ static void mbox2_setup_48_24_magic(struct usb_device *dev)
 #define MBOX2_BOOT_LOADING     0x01 /* Hard coded into the device */
 #define MBOX2_BOOT_READY       0x02 /* Hard coded into the device */
 
-int snd_usb_mbox2_boot_quirk(struct usb_device *dev)
+static int snd_usb_mbox2_boot_quirk(struct usb_device *dev)
 {
         struct usb_host_config *config = dev->actconfig;
         int err;
-        u8 bootresponse;
+        u8 bootresponse[12];
         int fwsize;
         int count;
 
@@ -546,20 +545,20 @@ int snd_usb_mbox2_boot_quirk(struct usb_device *dev)
         snd_printd("usb-audio: Sending Digidesign Mbox 2 boot sequence...\n");
 
         count = 0;
-        bootresponse = MBOX2_BOOT_LOADING;
-        while ((bootresponse == MBOX2_BOOT_LOADING) && (count < 10)) {
+        bootresponse[0] = MBOX2_BOOT_LOADING;
+        while ((bootresponse[0] == MBOX2_BOOT_LOADING) && (count < 10)) {
                 msleep(500); /* 0.5 second delay */
                 snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
                         /* Control magic - load onboard firmware */
                         0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012);
-                if (bootresponse == MBOX2_BOOT_READY)
+                if (bootresponse[0] == MBOX2_BOOT_READY)
                         break;
                 snd_printd("usb-audio: device not ready, resending boot sequence...\n");
                 count++;
         }
 
-        if (bootresponse != MBOX2_BOOT_READY) {
-                snd_printk(KERN_ERR "usb-audio: Unknown bootresponse=%d, or timed out, ignoring device.\n", bootresponse);
+        if (bootresponse[0] != MBOX2_BOOT_READY) {
+                snd_printk(KERN_ERR "usb-audio: Unknown bootresponse=%d, or timed out, ignoring device.\n", bootresponse[0]);
                 return -ENODEV;
         }
 
@@ -660,7 +659,6 @@ static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
         return 0; /* keep this altsetting */
 }
 
-
 static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip,
                                            int iface, int altno)
 {
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index a8172c119796..1ac3fd9cc5a6 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -76,7 +76,6 @@ enum quirk_type {
         QUIRK_MIDI_YAMAHA,
         QUIRK_MIDI_MIDIMAN,
         QUIRK_MIDI_NOVATION,
-        QUIRK_MIDI_MBOX2,
         QUIRK_MIDI_RAW_BYTES,
         QUIRK_MIDI_EMAGIC,
         QUIRK_MIDI_CME,