Merge 4.0-rc5 into char-misc-next

We want those fixes in here as well.

Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>

35 files changed, 250 insertions, 125 deletions

diff --git a/sound/core/control.c b/sound/core/control.c
index 35324a8e83c8..eeb691d1911f 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1170,6 +1170,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
 
         if (info->count < 1)
                 return -EINVAL;
+        if (!*info->id.name)
+                return -EINVAL;
+        if (strnlen(info->id.name, sizeof(info->id.name)) >= sizeof(info->id.name))
+                return -EINVAL;
         access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE :
                 (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE|
                                  SNDRV_CTL_ELEM_ACCESS_INACTIVE|
diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h
index de7602bd69b5..27b044f84c81 100644
--- a/sound/firewire/dice/dice-interface.h
+++ b/sound/firewire/dice/dice-interface.h
@@ -299,23 +299,23 @@
 #define RX_ISOCHRONOUS                  0x008
 
 /*
+ * Index of first quadlet to be interpreted; read/write.  If > 0, that many
+ * quadlets at the beginning of each data block will be ignored, and all the
+ * audio and MIDI quadlets will follow.
+ */
+#define RX_SEQ_START                    0x00c
+
+/*
  * The number of audio channels; read-only.  There will be one quadlet per
  * channel.
  */
-#define RX_NUMBER_AUDIO                 0x00c
+#define RX_NUMBER_AUDIO                 0x010
 
 /*
  * The number of MIDI ports, 0-8; read-only.  If > 0, there will be one
  * additional quadlet in each data block, following the audio quadlets.
  */
-#define RX_NUMBER_MIDI                  0x010
-
-/*
- * Index of first quadlet to be interpreted; read/write.  If > 0, that many
- * quadlets at the beginning of each data block will be ignored, and all the
- * audio and MIDI quadlets will follow.
- */
-#define RX_SEQ_START                    0x014
+#define RX_NUMBER_MIDI                  0x014
 
 /*
  * Names of all audio channels; read-only.  Quadlets are byte-swapped.  Names
diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c
index ecfe20fd4de5..f5c1d1bced59 100644
--- a/sound/firewire/dice/dice-proc.c
+++ b/sound/firewire/dice/dice-proc.c
@@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry,
                 } tx;
                 struct {
                         u32 iso;
+                        u32 seq_start;
                         u32 number_audio;
                         u32 number_midi;
-                        u32 seq_start;
                         char names[RX_NAMES_SIZE];
                         u32 ac3_caps;
                         u32 ac3_enable;
@@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry,
                         break;
                 snd_iprintf(buffer, "rx %u:\n", stream);
                 snd_iprintf(buffer, "  iso channel: %d\n", (int)buf.rx.iso);
+                snd_iprintf(buffer, "  sequence start: %u\n", buf.rx.seq_start);
                 snd_iprintf(buffer, "  audio channels: %u\n",
                             buf.rx.number_audio);
                 snd_iprintf(buffer, "  midi ports: %u\n", buf.rx.number_midi);
-                snd_iprintf(buffer, "  sequence start: %u\n", buf.rx.seq_start);
                 if (quadlets >= 68) {
                         dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE);
                         snd_iprintf(buffer, "  names: %s\n", buf.rx.names);
diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c
index 5f17b77ee152..f0e4d502d604 100644
--- a/sound/firewire/iso-resources.c
+++ b/sound/firewire/iso-resources.c
@@ -26,7 +26,7 @@
 int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit)
 {
         r->channels_mask = ~0uLL;
-        r->unit = fw_unit_get(unit);
+        r->unit = unit;
         mutex_init(&r->mutex);
         r->allocated = false;
 
@@ -42,7 +42,6 @@ void fw_iso_resources_destroy(struct fw_iso_resources *r)
 {
         WARN_ON(r->allocated);
         mutex_destroy(&r->mutex);
-        fw_unit_put(r->unit);
 }
 EXPORT_SYMBOL(fw_iso_resources_destroy);
 
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index a2ce773bdc62..17c2637d842c 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1164,7 +1164,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
                 }
         }
 
-        if (!bus->no_response_fallback)
+        if (bus->no_response_fallback)
                 return -1;
 
         if (!chip->polling_mode && chip->poll_count < 2) {
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index b680b4ec6331..8ec5289f8e05 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -687,12 +687,45 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid,
         return val;
 }
 
+/* is this a stereo widget or a stereo-to-mono mix? */
+static bool is_stereo_amps(struct hda_codec *codec, hda_nid_t nid, int dir)
+{
+        unsigned int wcaps = get_wcaps(codec, nid);
+        hda_nid_t conn;
+
+        if (wcaps & AC_WCAP_STEREO)
+                return true;
+        if (dir != HDA_INPUT || get_wcaps_type(wcaps) != AC_WID_AUD_MIX)
+                return false;
+        if (snd_hda_get_num_conns(codec, nid) != 1)
+                return false;
+        if (snd_hda_get_connections(codec, nid, &conn, 1) < 0)
+                return false;
+        return !!(get_wcaps(codec, conn) & AC_WCAP_STEREO);
+}
+
 /* initialize the amp value (only at the first time) */
 static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx)
 {
         unsigned int caps = query_amp_caps(codec, nid, dir);
         int val = get_amp_val_to_activate(codec, nid, dir, caps, false);
-        snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
+
+        if (is_stereo_amps(codec, nid, dir))
+                snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
+        else
+                snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val);
+}
+
+/* update the amp, doing in stereo or mono depending on NID */
+static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx,
+                      unsigned int mask, unsigned int val)
+{
+        if (is_stereo_amps(codec, nid, dir))
+                return snd_hda_codec_amp_stereo(codec, nid, dir, idx,
+                                                mask, val);
+        else
+                return snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
+                                                mask, val);
 }
 
 /* calculate amp value mask we can modify;
@@ -732,7 +765,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir,
                 return;
 
         val &= mask;
-        snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val);
+        update_amp(codec, nid, dir, idx, mask, val);
 }
 
 static void activate_amp_out(struct hda_codec *codec, struct nid_path *path,
@@ -4424,13 +4457,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix)
         has_amp = nid_has_mute(codec, mix, HDA_INPUT);
         for (i = 0; i < nums; i++) {
                 if (has_amp)
-                        snd_hda_codec_amp_stereo(codec, mix,
-                                                 HDA_INPUT, i,
-                                                 0xff, HDA_AMP_MUTE);
+                        update_amp(codec, mix, HDA_INPUT, i,
+                                   0xff, HDA_AMP_MUTE);
                 else if (nid_has_volume(codec, conn[i], HDA_OUTPUT))
-                        snd_hda_codec_amp_stereo(codec, conn[i],
-                                                 HDA_OUTPUT, 0,
-                                                 0xff, HDA_AMP_MUTE);
+                        update_amp(codec, conn[i], HDA_OUTPUT, 0,
+                                   0xff, HDA_AMP_MUTE);
         }
 }
 
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index ce5a6da83419..05e19f78b4cb 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -134,13 +134,38 @@ static void print_amp_caps(struct snd_info_buffer *buffer,
                     (caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT);
 }
 
+/* is this a stereo widget or a stereo-to-mono mix? */
+static bool is_stereo_amps(struct hda_codec *codec, hda_nid_t nid,
+                           int dir, unsigned int wcaps, int indices)
+{
+        hda_nid_t conn;
+
+        if (wcaps & AC_WCAP_STEREO)
+                return true;
+        /* check for a stereo-to-mono mix; it must be:
+         * only a single connection, only for input, and only a mixer widget
+         */
+        if (indices != 1 || dir != HDA_INPUT ||
+            get_wcaps_type(wcaps) != AC_WID_AUD_MIX)
+                return false;
+
+        if (snd_hda_get_raw_connections(codec, nid, &conn, 1) < 0)
+                return false;
+        /* the connection source is a stereo? */
+        wcaps = snd_hda_param_read(codec, conn, AC_PAR_AUDIO_WIDGET_CAP);
+        return !!(wcaps & AC_WCAP_STEREO);
+}
+
 static void print_amp_vals(struct snd_info_buffer *buffer,
                            struct hda_codec *codec, hda_nid_t nid,
-                           int dir, int stereo, int indices)
+                           int dir, unsigned int wcaps, int indices)
 {
         unsigned int val;
+        bool stereo;
         int i;
 
+        stereo = is_stereo_amps(codec, nid, dir, wcaps, indices);
+
         dir = dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
         for (i = 0; i < indices; i++) {
                 snd_iprintf(buffer, " [");
@@ -757,12 +782,10 @@ static void print_codec_info(struct snd_info_entry *entry,
                             (codec->single_adc_amp &&
                              wid_type == AC_WID_AUD_IN))
                                 print_amp_vals(buffer, codec, nid, HDA_INPUT,
-                                               wid_caps & AC_WCAP_STEREO,
-                                               1);
+                                               wid_caps, 1);
                         else
                                 print_amp_vals(buffer, codec, nid, HDA_INPUT,
-                                               wid_caps & AC_WCAP_STEREO,
-                                               conn_len);
+                                               wid_caps, conn_len);
                 }
                 if (wid_caps & AC_WCAP_OUT_AMP) {
                         snd_iprintf(buffer, "  Amp-Out caps: ");
@@ -771,11 +794,10 @@ static void print_codec_info(struct snd_info_entry *entry,
                         if (wid_type == AC_WID_PIN &&
                             codec->pin_amp_workaround)
                                 print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
-                                               wid_caps & AC_WCAP_STEREO,
-                                               conn_len);
+                                               wid_caps, conn_len);
                         else
                                 print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
-                                               wid_caps & AC_WCAP_STEREO, 1);
+                                               wid_caps, 1);
                 }
 
                 switch (wid_type) {
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 1589c9bcce3e..dd2b3d92071f 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -393,6 +393,7 @@ 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, 0x5600, "MacBookAir 5,2", CS420X_MBP81),
         SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42),
         SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE),
         {} /* terminator */
@@ -584,6 +585,7 @@ static int patch_cs420x(struct hda_codec *codec)
                 return -ENOMEM;
 
         spec->gen.automute_hook = cs_automute;
+        codec->single_adc_amp = 1;
 
         snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl,
                            cs420x_fixups);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index fd3ed18670e9..da67ea8645a6 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -223,6 +223,7 @@ enum {
         CXT_PINCFG_LENOVO_TP410,
         CXT_PINCFG_LEMOTE_A1004,
         CXT_PINCFG_LEMOTE_A1205,
+        CXT_PINCFG_COMPAQ_CQ60,
         CXT_FIXUP_STEREO_DMIC,
         CXT_FIXUP_INC_MIC_BOOST,
         CXT_FIXUP_HEADPHONE_MIC_PIN,
@@ -660,6 +661,15 @@ static const struct hda_fixup cxt_fixups[] = {
                 .type = HDA_FIXUP_PINS,
                 .v.pins = cxt_pincfg_lemote,
         },
+        [CXT_PINCFG_COMPAQ_CQ60] = {
+                .type = HDA_FIXUP_PINS,
+                .v.pins = (const struct hda_pintbl[]) {
+                        /* 0x17 was falsely set up as a mic, it should 0x1d */
+                        { 0x17, 0x400001f0 },
+                        { 0x1d, 0x97a70120 },
+                        { }
+                }
+        },
         [CXT_FIXUP_STEREO_DMIC] = {
                 .type = HDA_FIXUP_FUNC,
                 .v.func = cxt_fixup_stereo_dmic,
@@ -769,6 +779,7 @@ static const struct hda_model_fixup cxt5047_fixup_models[] = {
 };
 
 static const struct snd_pci_quirk cxt5051_fixups[] = {
+        SND_PCI_QUIRK(0x103c, 0x360b, "Compaq CQ60", CXT_PINCFG_COMPAQ_CQ60),
         SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200),
         {}
 };
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index b67480f1b1aa..4373ada95648 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -317,7 +317,7 @@ static int adav80x_put_deemph(struct snd_kcontrol *kcontrol,
 {
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
-        unsigned int deemph = ucontrol->value.enumerated.item[0];
+        unsigned int deemph = ucontrol->value.integer.value[0];
 
         if (deemph > 1)
                 return -EINVAL;
@@ -333,7 +333,7 @@ static int adav80x_get_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 
-        ucontrol->value.enumerated.item[0] = adav80x->deemph;
+        ucontrol->value.integer.value[0] = adav80x->deemph;
         return 0;
 };
 
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 70861c7b1631..81b54a270bd8 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -76,7 +76,7 @@ static int ak4641_put_deemph(struct snd_kcontrol *kcontrol,
 {
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
-        int deemph = ucontrol->value.enumerated.item[0];
+        int deemph = ucontrol->value.integer.value[0];
 
         if (deemph > 1)
                 return -EINVAL;
@@ -92,7 +92,7 @@ static int ak4641_get_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
 
-        ucontrol->value.enumerated.item[0] = ak4641->deemph;
+        ucontrol->value.integer.value[0] = ak4641->deemph;
         return 0;
 };
 
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 632e89f793a7..2a58b1dccd2f 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -343,25 +343,25 @@ static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route ak4671_intercon[] = {
-        {"DAC Left", "NULL", "PMPLL"},
-        {"DAC Right", "NULL", "PMPLL"},
-        {"ADC Left", "NULL", "PMPLL"},
-        {"ADC Right", "NULL", "PMPLL"},
+        {"DAC Left", NULL, "PMPLL"},
+        {"DAC Right", NULL, "PMPLL"},
+        {"ADC Left", NULL, "PMPLL"},
+        {"ADC Right", NULL, "PMPLL"},
 
         /* Outputs */
-        {"LOUT1", "NULL", "LOUT1 Mixer"},
-        {"ROUT1", "NULL", "ROUT1 Mixer"},
-        {"LOUT2", "NULL", "LOUT2 Mix Amp"},
-        {"ROUT2", "NULL", "ROUT2 Mix Amp"},
-        {"LOUT3", "NULL", "LOUT3 Mixer"},
-        {"ROUT3", "NULL", "ROUT3 Mixer"},
+        {"LOUT1", NULL, "LOUT1 Mixer"},
+        {"ROUT1", NULL, "ROUT1 Mixer"},
+        {"LOUT2", NULL, "LOUT2 Mix Amp"},
+        {"ROUT2", NULL, "ROUT2 Mix Amp"},
+        {"LOUT3", NULL, "LOUT3 Mixer"},
+        {"ROUT3", NULL, "ROUT3 Mixer"},
 
         {"LOUT1 Mixer", "DACL", "DAC Left"},
         {"ROUT1 Mixer", "DACR", "DAC Right"},
         {"LOUT2 Mixer", "DACHL", "DAC Left"},
         {"ROUT2 Mixer", "DACHR", "DAC Right"},
-        {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"},
-        {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"},
+        {"LOUT2 Mix Amp", NULL, "LOUT2 Mixer"},
+        {"ROUT2 Mix Amp", NULL, "ROUT2 Mixer"},
         {"LOUT3 Mixer", "DACSL", "DAC Left"},
         {"ROUT3 Mixer", "DACSR", "DAC Right"},
 
@@ -381,18 +381,18 @@ static const struct snd_soc_dapm_route ak4671_intercon[] = {
         {"LIN2", NULL, "Mic Bias"},
         {"RIN2", NULL, "Mic Bias"},
 
-        {"ADC Left", "NULL", "LIN MUX"},
-        {"ADC Right", "NULL", "RIN MUX"},
+        {"ADC Left", NULL, "LIN MUX"},
+        {"ADC Right", NULL, "RIN MUX"},
 
         /* Analog Loops */
-        {"LIN1 Mixing Circuit", "NULL", "LIN1"},
-        {"RIN1 Mixing Circuit", "NULL", "RIN1"},
-        {"LIN2 Mixing Circuit", "NULL", "LIN2"},
-        {"RIN2 Mixing Circuit", "NULL", "RIN2"},
-        {"LIN3 Mixing Circuit", "NULL", "LIN3"},
-        {"RIN3 Mixing Circuit", "NULL", "RIN3"},
-        {"LIN4 Mixing Circuit", "NULL", "LIN4"},
-        {"RIN4 Mixing Circuit", "NULL", "RIN4"},
+        {"LIN1 Mixing Circuit", NULL, "LIN1"},
+        {"RIN1 Mixing Circuit", NULL, "RIN1"},
+        {"LIN2 Mixing Circuit", NULL, "LIN2"},
+        {"RIN2 Mixing Circuit", NULL, "RIN2"},
+        {"LIN3 Mixing Circuit", NULL, "LIN3"},
+        {"RIN3 Mixing Circuit", NULL, "RIN3"},
+        {"LIN4 Mixing Circuit", NULL, "LIN4"},
+        {"RIN4 Mixing Circuit", NULL, "RIN4"},
 
         {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"},
         {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"},
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 79a4efcb894c..7d3a6accaf9a 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -286,7 +286,7 @@ static int cs4271_get_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
 
-        ucontrol->value.enumerated.item[0] = cs4271->deemph;
+        ucontrol->value.integer.value[0] = cs4271->deemph;
         return 0;
 }
 
@@ -296,7 +296,7 @@ static int cs4271_put_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
 
-        cs4271->deemph = ucontrol->value.enumerated.item[0];
+        cs4271->deemph = ucontrol->value.integer.value[0];
         return cs4271_set_deemph(codec);
 }
 
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index ffe96175a8a5..911c26c705fc 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -876,11 +876,11 @@ static const struct snd_soc_dapm_widget da732x_dapm_widgets[] = {
 
 static const struct snd_soc_dapm_route da732x_dapm_routes[] = {
         /* Inputs */
-        {"AUX1L PGA", "NULL", "AUX1L"},
-        {"AUX1R PGA", "NULL", "AUX1R"},
+        {"AUX1L PGA", NULL, "AUX1L"},
+        {"AUX1R PGA", NULL, "AUX1R"},
         {"MIC1 PGA", NULL, "MIC1"},
-        {"MIC2 PGA", "NULL", "MIC2"},
-        {"MIC3 PGA", "NULL", "MIC3"},
+        {"MIC2 PGA", NULL, "MIC2"},
+        {"MIC3 PGA", NULL, "MIC3"},
 
         /* Capture Path */
         {"ADC1 Left MUX", "MIC1", "MIC1 PGA"},
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index f27325155ace..c5f35a07e8e4 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -120,7 +120,7 @@ static int es8328_get_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
 
-        ucontrol->value.enumerated.item[0] = es8328->deemph;
+        ucontrol->value.integer.value[0] = es8328->deemph;
         return 0;
 }
 
@@ -129,7 +129,7 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol,
 {
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
-        int deemph = ucontrol->value.enumerated.item[0];
+        int deemph = ucontrol->value.integer.value[0];
         int ret;
 
         if (deemph > 1)
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index a722a023c262..477e13d30971 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -118,7 +118,7 @@ static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
 
-        ucontrol->value.enumerated.item[0] = priv->deemph;
+        ucontrol->value.integer.value[0] = priv->deemph;
 
         return 0;
 }
@@ -129,7 +129,7 @@ static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
 
-        priv->deemph = ucontrol->value.enumerated.item[0];
+        priv->deemph = ucontrol->value.integer.value[0];
 
         return pcm1681_set_deemph(codec);
 }
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index f374840a5a7c..9b541e52da8c 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -1198,7 +1198,7 @@ static struct dmi_system_id dmi_dell_dino[] = {
                 .ident = "Dell Dino",
                 .matches = {
                         DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."),
-                        DMI_MATCH(DMI_BOARD_NAME, "0144P8")
+                        DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343")
                 }
         },
         { }
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index e182e6569bbd..3593a1496056 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1151,13 +1151,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
                 /* Enable VDDC charge pump */
                 ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP;
         } else if (vddio >= 3100 && vdda >= 3100) {
-                /*
-                 * if vddio and vddd > 3.1v,
-                 * charge pump should be clean before set ana_pwr
-                 */
-                snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
-                                SGTL5000_VDDC_CHRGPMP_POWERUP, 0);
-
+                ana_pwr &= ~SGTL5000_VDDC_CHRGPMP_POWERUP;
                 /* VDDC use VDDIO rail */
                 lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
                 lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 47b257e41809..82095d6cd070 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -538,8 +538,8 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = {
         /* speaker map */
         { "IHFOUTL", NULL, "Speaker Rail"},
         { "IHFOUTR", NULL, "Speaker Rail"},
-        { "IHFOUTL", "NULL", "Speaker Left Playback"},
-        { "IHFOUTR", "NULL", "Speaker Right Playback"},
+        { "IHFOUTL", NULL, "Speaker Left Playback"},
+        { "IHFOUTR", NULL, "Speaker Right Playback"},
         { "Speaker Left Playback", NULL, "Speaker Left Filter"},
         { "Speaker Right Playback", NULL, "Speaker Right Filter"},
         { "Speaker Left Filter", NULL, "IHFDAC Left"},
diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c
index 249ef5c4c762..32942bed34b1 100644
--- a/sound/soc/codecs/tas5086.c
+++ b/sound/soc/codecs/tas5086.c
@@ -281,7 +281,7 @@ static int tas5086_get_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
 
-        ucontrol->value.enumerated.item[0] = priv->deemph;
+        ucontrol->value.integer.value[0] = priv->deemph;
 
         return 0;
 }
@@ -292,7 +292,7 @@ static int tas5086_put_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
 
-        priv->deemph = ucontrol->value.enumerated.item[0];
+        priv->deemph = ucontrol->value.integer.value[0];
 
         return tas5086_set_deemph(codec);
 }
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 8d9de49a5052..21d5402e343f 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -610,7 +610,7 @@ static int wm2000_anc_mode_get(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
 
-        ucontrol->value.enumerated.item[0] = wm2000->anc_active;
+        ucontrol->value.integer.value[0] = wm2000->anc_active;
 
         return 0;
 }
@@ -620,7 +620,7 @@ static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol,
 {
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
-        int anc_active = ucontrol->value.enumerated.item[0];
+        int anc_active = ucontrol->value.integer.value[0];
         int ret;
 
         if (anc_active > 1)
@@ -643,7 +643,7 @@ static int wm2000_speaker_get(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
 
-        ucontrol->value.enumerated.item[0] = wm2000->spk_ena;
+        ucontrol->value.integer.value[0] = wm2000->spk_ena;
 
         return 0;
 }
@@ -653,7 +653,7 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol,
 {
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
-        int val = ucontrol->value.enumerated.item[0];
+        int val = ucontrol->value.integer.value[0];
         int ret;
 
         if (val > 1)
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 098c143f44d6..c6d10533e2bd 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -125,7 +125,7 @@ static int wm8731_get_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
 
-        ucontrol->value.enumerated.item[0] = wm8731->deemph;
+        ucontrol->value.integer.value[0] = wm8731->deemph;
 
         return 0;
 }
@@ -135,7 +135,7 @@ static int wm8731_put_deemph(struct snd_kcontrol *kcontrol,
 {
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
-        int deemph = ucontrol->value.enumerated.item[0];
+        int deemph = ucontrol->value.integer.value[0];
         int ret = 0;
 
         if (deemph > 1)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index dde462c082be..04b04f8e147c 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -442,7 +442,7 @@ static int wm8903_get_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 
-        ucontrol->value.enumerated.item[0] = wm8903->deemph;
+        ucontrol->value.integer.value[0] = wm8903->deemph;
 
         return 0;
 }
@@ -452,7 +452,7 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol,
 {
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-        int deemph = ucontrol->value.enumerated.item[0];
+        int deemph = ucontrol->value.integer.value[0];
         int ret = 0;
 
         if (deemph > 1)
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index d3b3f57668cc..215e93c1ddf0 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -525,7 +525,7 @@ static int wm8904_get_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 
-        ucontrol->value.enumerated.item[0] = wm8904->deemph;
+        ucontrol->value.integer.value[0] = wm8904->deemph;
         return 0;
 }
 
@@ -534,7 +534,7 @@ static int wm8904_put_deemph(struct snd_kcontrol *kcontrol,
 {
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
-        int deemph = ucontrol->value.enumerated.item[0];
+        int deemph = ucontrol->value.integer.value[0];
 
         if (deemph > 1)
                 return -EINVAL;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 1ab2d462afad..00bec915d652 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -393,7 +393,7 @@ static int wm8955_get_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
 
-        ucontrol->value.enumerated.item[0] = wm8955->deemph;
+        ucontrol->value.integer.value[0] = wm8955->deemph;
         return 0;
 }
 
@@ -402,7 +402,7 @@ static int wm8955_put_deemph(struct snd_kcontrol *kcontrol,
 {
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
-        int deemph = ucontrol->value.enumerated.item[0];
+        int deemph = ucontrol->value.integer.value[0];
 
         if (deemph > 1)
                 return -EINVAL;
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index cf8fecf97f2c..3035d9856415 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -184,7 +184,7 @@ static int wm8960_get_deemph(struct snd_kcontrol *kcontrol,
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
 
-        ucontrol->value.enumerated.item[0] = wm8960->deemph;
+        ucontrol->value.integer.value[0] = wm8960->deemph;
         return 0;
 }
 
@@ -193,7 +193,7 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
 {
         struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
         struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
-        int deemph = ucontrol->value.enumerated.item[0];
+        int deemph = ucontrol->value.integer.value[0];
 
         if (deemph > 1)
                 return -EINVAL;
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 9517571e820d..98c9525bd751 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -180,7 +180,7 @@ static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol,
         struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
         struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
         struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
-        unsigned int val = ucontrol->value.enumerated.item[0];
+        unsigned int val = ucontrol->value.integer.value[0];
         struct soc_mixer_control *mc =
                 (struct soc_mixer_control *)kcontrol->private_value;
         unsigned int mixer, mask, shift, old;
@@ -193,7 +193,7 @@ static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol,
 
         mutex_lock(&wm9712->lock);
         old = wm9712->hp_mixer[mixer];
-        if (ucontrol->value.enumerated.item[0])
+        if (ucontrol->value.integer.value[0])
                 wm9712->hp_mixer[mixer] |= mask;
         else
                 wm9712->hp_mixer[mixer] &= ~mask;
@@ -231,7 +231,7 @@ static int wm9712_hp_mixer_get(struct snd_kcontrol *kcontrol,
         mixer = mc->shift >> 8;
         shift = mc->shift & 0xff;
 
-        ucontrol->value.enumerated.item[0] =
+        ucontrol->value.integer.value[0] =
                 (wm9712->hp_mixer[mixer] >> shift) & 1;
 
         return 0;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 68222917b396..79552953e1bd 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -255,7 +255,7 @@ static int wm9713_hp_mixer_put(struct snd_kcontrol *kcontrol,
         struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
         struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
         struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec);
-        unsigned int val = ucontrol->value.enumerated.item[0];
+        unsigned int val = ucontrol->value.integer.value[0];
         struct soc_mixer_control *mc =
                 (struct soc_mixer_control *)kcontrol->private_value;
         unsigned int mixer, mask, shift, old;
@@ -268,7 +268,7 @@ static int wm9713_hp_mixer_put(struct snd_kcontrol *kcontrol,
 
         mutex_lock(&wm9713->lock);
         old = wm9713->hp_mixer[mixer];
-        if (ucontrol->value.enumerated.item[0])
+        if (ucontrol->value.integer.value[0])
                 wm9713->hp_mixer[mixer] |= mask;
         else
                 wm9713->hp_mixer[mixer] &= ~mask;
@@ -306,7 +306,7 @@ static int wm9713_hp_mixer_get(struct snd_kcontrol *kcontrol,
         mixer = mc->shift >> 8;
         shift = mc->shift & 0xff;
 
-        ucontrol->value.enumerated.item[0] =
+        ucontrol->value.integer.value[0] =
                 (wm9713->hp_mixer[mixer] >> shift) & 1;
 
         return 0;
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 75870c0ea2c9..91eb3aef7f02 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1049,7 +1049,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
                                 enum spdif_txrate index, bool round)
 {
         const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };
-        bool is_sysclk = clk == spdif_priv->sysclk;
+        bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk);
         u64 rate_ideal, rate_actual, sub;
         u32 sysclk_dfmin, sysclk_dfmax;
         u32 txclk_df, sysclk_df, arate;
@@ -1143,7 +1143,7 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
                         spdif_priv->txclk_src[index], rate[index]);
         dev_dbg(&pdev->dev, "use txclk df %d for %dHz sample rate\n",
                         spdif_priv->txclk_df[index], rate[index]);
-        if (spdif_priv->txclk[index] == spdif_priv->sysclk)
+        if (clk_is_match(spdif_priv->txclk[index], spdif_priv->sysclk))
                 dev_dbg(&pdev->dev, "use sysclk df %d for %dHz sample rate\n",
                                 spdif_priv->sysclk_df[index], rate[index]);
         dev_dbg(&pdev->dev, "the best rate for %dHz sample rate is %dHz\n",
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index b9fabbf69db6..6b0c8f717ec2 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -603,7 +603,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
         factor = (div2 + 1) * (7 * psr + 1) * 2;
 
         for (i = 0; i < 255; i++) {
-                tmprate = freq * factor * (i + 2);
+                tmprate = freq * factor * (i + 1);
 
                 if (baudclk_is_used)
                         clkrate = clk_get_rate(ssi_private->baudclk);
@@ -1227,7 +1227,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev,
         ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0;
         ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0;
 
-        ret = !of_property_read_u32_array(np, "dmas", dmas, 4);
+        ret = of_property_read_u32_array(np, "dmas", dmas, 4);
         if (ssi_private->use_dma && !ret && dmas[2] == IMX_DMATYPE_SSI_DUAL) {
                 ssi_private->use_dual_fifo = true;
                 /* When using dual fifo mode, we need to keep watermark
diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c
index c42ffae5fe9f..402b728c0a06 100644
--- a/sound/soc/intel/sst-haswell-dsp.c
+++ b/sound/soc/intel/sst-haswell-dsp.c
@@ -207,9 +207,6 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw)
                 module = (void *)module + sizeof(*module) + module->mod_size;
         }
 
-        /* allocate scratch mem regions */
-        sst_block_alloc_scratch(dsp);
-
         return 0;
 }
 
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
index 394af5684c05..863a9ca34b8e 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -1732,6 +1732,7 @@ static void sst_hsw_drop_all(struct sst_hsw *hsw)
 int sst_hsw_dsp_load(struct sst_hsw *hsw)
 {
         struct sst_dsp *dsp = hsw->dsp;
+        struct sst_fw *sst_fw, *t;
         int ret;
 
         dev_dbg(hsw->dev, "loading audio DSP....");
@@ -1748,12 +1749,17 @@ int sst_hsw_dsp_load(struct sst_hsw *hsw)
                 return ret;
         }
 
-        ret = sst_fw_reload(hsw->sst_fw);
-        if (ret < 0) {
-                dev_err(hsw->dev, "error: SST FW reload failed\n");
-                sst_dsp_dma_put_channel(dsp);
-                return -ENOMEM;
+        list_for_each_entry_safe_reverse(sst_fw, t, &dsp->fw_list, list) {
+                ret = sst_fw_reload(sst_fw);
+                if (ret < 0) {
+                        dev_err(hsw->dev, "error: SST FW reload failed\n");
+                        sst_dsp_dma_put_channel(dsp);
+                        return -ENOMEM;
+                }
         }
+        ret = sst_block_alloc_scratch(hsw->dsp);
+        if (ret < 0)
+                return -EINVAL;
 
         sst_dsp_dma_put_channel(dsp);
         return 0;
@@ -1809,12 +1815,17 @@ int sst_hsw_dsp_runtime_suspend(struct sst_hsw *hsw)
 
 int sst_hsw_dsp_runtime_sleep(struct sst_hsw *hsw)
 {
-        sst_fw_unload(hsw->sst_fw);
-        sst_block_free_scratch(hsw->dsp);
+        struct sst_fw *sst_fw, *t;
+        struct sst_dsp *dsp = hsw->dsp;
+
+        list_for_each_entry_safe(sst_fw, t, &dsp->fw_list, list) {
+                sst_fw_unload(sst_fw);
+        }
+        sst_block_free_scratch(dsp);
 
         hsw->boot_complete = false;
 
-        sst_dsp_sleep(hsw->dsp);
+        sst_dsp_sleep(dsp);
 
         return 0;
 }
@@ -1943,6 +1954,11 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata)
                 goto fw_err;
         }
 
+        /* allocate scratch mem regions */
+        ret = sst_block_alloc_scratch(hsw->dsp);
+        if (ret < 0)
+                goto boot_err;
+
         /* wait for DSP boot completion */
         sst_dsp_boot(hsw->dsp);
         ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete,
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index def7d8260c4e..d19483081f9b 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -579,7 +579,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
                 if (PTR_ERR(priv->extclk) == -EPROBE_DEFER)
                         return -EPROBE_DEFER;
         } else {
-                if (priv->extclk == priv->clk) {
+                if (clk_is_match(priv->extclk, priv->clk)) {
                         devm_clk_put(&pdev->dev, priv->extclk);
                         priv->extclk = ERR_PTR(-EINVAL);
                 } else {
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 30579ca5bacb..e5c990889dcc 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -347,6 +347,8 @@ static ssize_t codec_list_read_file(struct file *file, char __user *user_buf,
         if (!buf)
                 return -ENOMEM;
 
+        mutex_lock(&client_mutex);
+
         list_for_each_entry(codec, &codec_list, list) {
                 len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n",
                                codec->component.name);
@@ -358,6 +360,8 @@ static ssize_t codec_list_read_file(struct file *file, char __user *user_buf,
                 }
         }
 
+        mutex_unlock(&client_mutex);
+
         if (ret >= 0)
                 ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
 
@@ -382,6 +386,8 @@ static ssize_t dai_list_read_file(struct file *file, char __user *user_buf,
         if (!buf)
                 return -ENOMEM;
 
+        mutex_lock(&client_mutex);
+
         list_for_each_entry(component, &component_list, list) {
                 list_for_each_entry(dai, &component->dai_list, list) {
                         len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n",
@@ -395,6 +401,8 @@ static ssize_t dai_list_read_file(struct file *file, char __user *user_buf,
                 }
         }
 
+        mutex_unlock(&client_mutex);
+
         ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
 
         kfree(buf);
@@ -418,6 +426,8 @@ static ssize_t platform_list_read_file(struct file *file,
         if (!buf)
                 return -ENOMEM;
 
+        mutex_lock(&client_mutex);
+
         list_for_each_entry(platform, &platform_list, list) {
                 len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n",
                                platform->component.name);
@@ -429,6 +439,8 @@ static ssize_t platform_list_read_file(struct file *file,
                 }
         }
 
+        mutex_unlock(&client_mutex);
+
         ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
 
         kfree(buf);
@@ -836,6 +848,8 @@ static struct snd_soc_component *soc_find_component(
 {
         struct snd_soc_component *component;
 
+        lockdep_assert_held(&client_mutex);
+
         list_for_each_entry(component, &component_list, list) {
                 if (of_node) {
                         if (component->dev->of_node == of_node)
@@ -854,6 +868,8 @@ static struct snd_soc_dai *snd_soc_find_dai(
         struct snd_soc_component *component;
         struct snd_soc_dai *dai;
 
+        lockdep_assert_held(&client_mutex);
+
         /* Find CPU DAI from registered DAIs*/
         list_for_each_entry(component, &component_list, list) {
                 if (dlc->of_node && component->dev->of_node != dlc->of_node)
@@ -1508,6 +1524,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
         struct snd_soc_codec *codec;
         int ret, i, order;
 
+        mutex_lock(&client_mutex);
         mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT);
 
         /* bind DAIs */
@@ -1662,6 +1679,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
         card->instantiated = 1;
         snd_soc_dapm_sync(&card->dapm);
         mutex_unlock(&card->mutex);
+        mutex_unlock(&client_mutex);
 
         return 0;
 
@@ -1680,6 +1698,7 @@ card_probe_error:
 
 base_error:
         mutex_unlock(&card->mutex);
+        mutex_unlock(&client_mutex);
 
         return ret;
 }
@@ -2713,13 +2732,6 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component)
         list_del(&component->list);
 }
 
-static void snd_soc_component_del(struct snd_soc_component *component)
-{
-        mutex_lock(&client_mutex);
-        snd_soc_component_del_unlocked(component);
-        mutex_unlock(&client_mutex);
-}
-
 int snd_soc_register_component(struct device *dev,
                                const struct snd_soc_component_driver *cmpnt_drv,
                                struct snd_soc_dai_driver *dai_drv,
@@ -2767,14 +2779,17 @@ void snd_soc_unregister_component(struct device *dev)
 {
         struct snd_soc_component *cmpnt;
 
+        mutex_lock(&client_mutex);
         list_for_each_entry(cmpnt, &component_list, list) {
                 if (dev == cmpnt->dev && cmpnt->registered_as_component)
                         goto found;
         }
+        mutex_unlock(&client_mutex);
         return;
 
 found:
-        snd_soc_component_del(cmpnt);
+        snd_soc_component_del_unlocked(cmpnt);
+        mutex_unlock(&client_mutex);
         snd_soc_component_cleanup(cmpnt);
         kfree(cmpnt);
 }
@@ -2882,10 +2897,14 @@ struct snd_soc_platform *snd_soc_lookup_platform(struct device *dev)
 {
         struct snd_soc_platform *platform;
 
+        mutex_lock(&client_mutex);
         list_for_each_entry(platform, &platform_list, list) {
-                if (dev == platform->dev)
+                if (dev == platform->dev) {
+                        mutex_unlock(&client_mutex);
                         return platform;
+                }
         }
+        mutex_unlock(&client_mutex);
 
         return NULL;
 }
@@ -3090,15 +3109,15 @@ void snd_soc_unregister_codec(struct device *dev)
 {
         struct snd_soc_codec *codec;
 
+        mutex_lock(&client_mutex);
         list_for_each_entry(codec, &codec_list, list) {
                 if (dev == codec->dev)
                         goto found;
         }
+        mutex_unlock(&client_mutex);
         return;
 
 found:
-
-        mutex_lock(&client_mutex);
         list_del(&codec->list);
         snd_soc_component_del_unlocked(&codec->component);
         mutex_unlock(&client_mutex);
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 67d476548dcf..07f984d5f516 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1773,6 +1773,36 @@ YAMAHA_DEVICE(0x7010, "UB99"),
                 }
         }
 },
+{
+        USB_DEVICE(0x0582, 0x0159),
+        .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+                /* .vendor_name = "Roland", */
+                /* .product_name = "UA-22", */
+                .ifnum = QUIRK_ANY_INTERFACE,
+                .type = QUIRK_COMPOSITE,
+                .data = (const struct snd_usb_audio_quirk[]) {
+                        {
+                                .ifnum = 0,
+                                .type = QUIRK_AUDIO_STANDARD_INTERFACE
+                        },
+                        {
+                                .ifnum = 1,
+                                .type = QUIRK_AUDIO_STANDARD_INTERFACE
+                        },
+                        {
+                                .ifnum = 2,
+                                .type = QUIRK_MIDI_FIXED_ENDPOINT,
+                                .data = & (const struct snd_usb_midi_endpoint_info) {
+                                        .out_cables = 0x0001,
+                                        .in_cables = 0x0001
+                                }
+                        },
+                        {
+                                .ifnum = -1
+                        }
+                }
+        }
+},
 /* this catches most recent vendor-specific Roland devices */
 {
         .match_flags = USB_DEVICE_ID_MATCH_VENDOR |