18 files changed, 176 insertions, 89 deletions

diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index d3226892ad6b..9048777228e2 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -434,17 +434,14 @@ static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle)
                 return;
         index = s->packet_index;
 
+        /* this module generate empty packet for 'no data' */
         syt = calculate_syt(s, cycle);
-        if (!(s->flags & CIP_BLOCKING)) {
+        if (!(s->flags & CIP_BLOCKING))
                 data_blocks = calculate_data_blocks(s);
-        } else {
-                if (syt != 0xffff) {
-                        data_blocks = s->syt_interval;
-                } else {
-                        data_blocks = 0;
-                        syt = 0xffffff;
-                }
-        }
+        else if (syt != 0xffff)
+                data_blocks = s->syt_interval;
+        else
+                data_blocks = 0;
 
         buffer = s->buffer.packets[index].buffer;
         buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) |
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 77db69480c19..7aa9870040c1 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -698,7 +698,6 @@ struct hda_bus {
         unsigned int in_reset:1;        /* during reset operation */
         unsigned int power_keep_link_on:1; /* don't power off HDA link */
         unsigned int no_response_fallback:1; /* don't fallback at RIRB error */
-        unsigned int avoid_link_reset:1; /* don't reset link at runtime PM */
 
         int primary_dig_out_type;       /* primary digital out PCM type */
 };
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 3067ed4fe3b2..c4671d00babd 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -2506,12 +2506,8 @@ static int create_out_jack_modes(struct hda_codec *codec, int num_pins,
 
         for (i = 0; i < num_pins; i++) {
                 hda_nid_t pin = pins[i];
-                if (pin == spec->hp_mic_pin) {
-                        int ret = create_hp_mic_jack_mode(codec, pin);
-                        if (ret < 0)
-                                return ret;
+                if (pin == spec->hp_mic_pin)
                         continue;
-                }
                 if (get_out_jack_num_items(codec, pin) > 1) {
                         struct snd_kcontrol_new *knew;
                         char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
@@ -2764,7 +2760,7 @@ static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol,
                         val &= ~(AC_PINCTL_VREFEN | PIN_HP);
                         val |= get_vref_idx(vref_caps, idx) | PIN_IN;
                 } else
-                        val = snd_hda_get_default_vref(codec, nid);
+                        val = snd_hda_get_default_vref(codec, nid) | PIN_IN;
         }
         snd_hda_set_pin_ctl_cache(codec, nid, val);
         call_hp_automute(codec, NULL);
@@ -2784,9 +2780,6 @@ static int create_hp_mic_jack_mode(struct hda_codec *codec, hda_nid_t pin)
         struct hda_gen_spec *spec = codec->spec;
         struct snd_kcontrol_new *knew;
 
-        if (get_out_jack_num_items(codec, pin) <= 1 &&
-            get_in_jack_num_items(codec, pin) <= 1)
-                return 0; /* no need */
         knew = snd_hda_gen_add_kctl(spec, "Headphone Mic Jack Mode",
                                     &hp_mic_jack_mode_enum);
         if (!knew)
@@ -2815,6 +2808,42 @@ static int add_loopback_list(struct hda_gen_spec *spec, hda_nid_t mix, int idx)
         return 0;
 }
 
+/* return true if either a volume or a mute amp is found for the given
+ * aamix path; the amp has to be either in the mixer node or its direct leaf
+ */
+static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid,
+                                   hda_nid_t pin, unsigned int *mix_val,
+                                   unsigned int *mute_val)
+{
+        int idx, num_conns;
+        const hda_nid_t *list;
+        hda_nid_t nid;
+
+        idx = snd_hda_get_conn_index(codec, mix_nid, pin, true);
+        if (idx < 0)
+                return false;
+
+        *mix_val = *mute_val = 0;
+        if (nid_has_volume(codec, mix_nid, HDA_INPUT))
+                *mix_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT);
+        if (nid_has_mute(codec, mix_nid, HDA_INPUT))
+                *mute_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT);
+        if (*mix_val && *mute_val)
+                return true;
+
+        /* check leaf node */
+        num_conns = snd_hda_get_conn_list(codec, mix_nid, &list);
+        if (num_conns < idx)
+                return false;
+        nid = list[idx];
+        if (!*mix_val && nid_has_volume(codec, nid, HDA_OUTPUT))
+                *mix_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+        if (!*mute_val && nid_has_mute(codec, nid, HDA_OUTPUT))
+                *mute_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+
+        return *mix_val || *mute_val;
+}
+
 /* create input playback/capture controls for the given pin */
 static int new_analog_input(struct hda_codec *codec, int input_idx,
                             hda_nid_t pin, const char *ctlname, int ctlidx,
@@ -2822,12 +2851,11 @@ static int new_analog_input(struct hda_codec *codec, int input_idx,
 {
         struct hda_gen_spec *spec = codec->spec;
         struct nid_path *path;
-        unsigned int val;
+        unsigned int mix_val, mute_val;
         int err, idx;
 
-        if (!nid_has_volume(codec, mix_nid, HDA_INPUT) &&
-            !nid_has_mute(codec, mix_nid, HDA_INPUT))
-                return 0; /* no need for analog loopback */
+        if (!look_for_mix_leaf_ctls(codec, mix_nid, pin, &mix_val, &mute_val))
+                return 0;
 
         path = snd_hda_add_new_path(codec, pin, mix_nid, 0);
         if (!path)
@@ -2836,20 +2864,18 @@ static int new_analog_input(struct hda_codec *codec, int input_idx,
         spec->loopback_paths[input_idx] = snd_hda_get_path_idx(codec, path);
 
         idx = path->idx[path->depth - 1];
-        if (nid_has_volume(codec, mix_nid, HDA_INPUT)) {
-                val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT);
-                err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, val);
+        if (mix_val) {
+                err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, mix_val);
                 if (err < 0)
                         return err;
-                path->ctls[NID_PATH_VOL_CTL] = val;
+                path->ctls[NID_PATH_VOL_CTL] = mix_val;
         }
 
-        if (nid_has_mute(codec, mix_nid, HDA_INPUT)) {
-                val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT);
-                err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, val);
+        if (mute_val) {
+                err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, mute_val);
                 if (err < 0)
                         return err;
-                path->ctls[NID_PATH_MUTE_CTL] = val;
+                path->ctls[NID_PATH_MUTE_CTL] = mute_val;
         }
 
         path->active = true;
@@ -4383,6 +4409,17 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
         if (err < 0)
                 return err;
 
+        /* create "Headphone Mic Jack Mode" if no input selection is
+         * available (or user specifies add_jack_modes hint)
+         */
+        if (spec->hp_mic_pin &&
+            (spec->auto_mic || spec->input_mux.num_items == 1 ||
+             spec->add_jack_modes)) {
+                err = create_hp_mic_jack_mode(codec, spec->hp_mic_pin);
+                if (err < 0)
+                        return err;
+        }
+
         if (spec->add_jack_modes) {
                 if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
                         err = create_out_jack_modes(codec, cfg->line_outs,
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 7a09404579a7..c6d230193da6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2994,8 +2994,7 @@ static int azx_runtime_suspend(struct device *dev)
                   STATESTS_INT_MASK);
 
         azx_stop_chip(chip);
-        if (!chip->bus->avoid_link_reset)
-                azx_enter_link_reset(chip);
+        azx_enter_link_reset(chip);
         azx_clear_irq_pending(chip);
         if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
                 hda_display_power(false);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index c205bb1747fd..1f2717f817a0 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3244,9 +3244,29 @@ enum {
 #if IS_ENABLED(CONFIG_THINKPAD_ACPI)
 
 #include <linux/thinkpad_acpi.h>
+#include <acpi/acpi.h>
 
 static int (*led_set_func)(int, bool);
 
+static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context,
+                                 void **rv)
+{
+        bool *found = context;
+        *found = true;
+        return AE_OK;
+}
+
+static bool is_thinkpad(struct hda_codec *codec)
+{
+        bool found = false;
+        if (codec->subsystem_id >> 16 != 0x17aa)
+                return false;
+        if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found)
+                return true;
+        found = false;
+        return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found;
+}
+
 static void update_tpacpi_mute_led(void *private_data, int enabled)
 {
         struct hda_codec *codec = private_data;
@@ -3279,6 +3299,8 @@ static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec,
         bool removefunc = false;
 
         if (action == HDA_FIXUP_ACT_PROBE) {
+                if (!is_thinkpad(codec))
+                        return;
                 if (!led_set_func)
                         led_set_func = symbol_request(tpacpi_led_set);
                 if (!led_set_func) {
@@ -3494,6 +3516,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
         SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
         SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC),
         SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC),
+        SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI),
         SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004),
         SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205),
         {}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5e42059f10a1..c770bdba6531 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1782,6 +1782,8 @@ enum {
         ALC889_FIXUP_IMAC91_VREF,
         ALC882_FIXUP_INV_DMIC,
         ALC882_FIXUP_NO_PRIMARY_HP,
+        ALC887_FIXUP_ASUS_BASS,
+        ALC887_FIXUP_BASS_CHMAP,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -1915,6 +1917,9 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
         }
 }
 
+static void alc_fixup_bass_chmap(struct hda_codec *codec,
+                                 const struct hda_fixup *fix, int action);
+
 static const struct hda_fixup alc882_fixups[] = {
         [ALC882_FIXUP_ABIT_AW9D_MAX] = {
                 .type = HDA_FIXUP_PINS,
@@ -2105,6 +2110,19 @@ static const struct hda_fixup alc882_fixups[] = {
                 .type = HDA_FIXUP_FUNC,
                 .v.func = alc882_fixup_no_primary_hp,
         },
+        [ALC887_FIXUP_ASUS_BASS] = {
+                .type = HDA_FIXUP_PINS,
+                .v.pins = (const struct hda_pintbl[]) {
+                        {0x16, 0x99130130}, /* bass speaker */
+                        {}
+                },
+                .chained = true,
+                .chain_id = ALC887_FIXUP_BASS_CHMAP,
+        },
+        [ALC887_FIXUP_BASS_CHMAP] = {
+                .type = HDA_FIXUP_FUNC,
+                .v.func = alc_fixup_bass_chmap,
+        },
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -2138,6 +2156,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V),
         SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC),
         SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601),
+        SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS),
         SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT),
         SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP),
         SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP),
@@ -3798,6 +3817,7 @@ enum {
         ALC271_FIXUP_HP_GATE_MIC_JACK,
         ALC269_FIXUP_ACER_AC700,
         ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
+        ALC269VB_FIXUP_ASUS_ZENBOOK,
         ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED,
         ALC269VB_FIXUP_ORDISSIMO_EVE2,
         ALC283_FIXUP_CHROME_BOOK,
@@ -4075,6 +4095,12 @@ static const struct hda_fixup alc269_fixups[] = {
                 .chained = true,
                 .chain_id = ALC269_FIXUP_THINKPAD_ACPI,
         },
+        [ALC269VB_FIXUP_ASUS_ZENBOOK] = {
+                .type = HDA_FIXUP_FUNC,
+                .v.func = alc269_fixup_limit_int_mic_boost,
+                .chained = true,
+                .chain_id = ALC269VB_FIXUP_DMIC,
+        },
         [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = {
                 .type = HDA_FIXUP_FUNC,
                 .v.func = alc269_fixup_limit_int_mic_boost,
@@ -4189,8 +4215,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
         SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
         SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
-        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, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK),
+        SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK),
         SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
         SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
         SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC),
@@ -4715,7 +4741,7 @@ static const struct snd_pcm_chmap_elem asus_pcm_2_1_chmaps[] = {
 };
 
 /* override the 2.1 chmap */
-static void alc662_fixup_bass_chmap(struct hda_codec *codec,
+static void alc_fixup_bass_chmap(struct hda_codec *codec,
                                     const struct hda_fixup *fix, int action)
 {
         if (action == HDA_FIXUP_ACT_BUILD) {
@@ -4923,7 +4949,7 @@ static const struct hda_fixup alc662_fixups[] = {
         },
         [ALC662_FIXUP_BASS_CHMAP] = {
                 .type = HDA_FIXUP_FUNC,
-                .v.func = alc662_fixup_bass_chmap,
+                .v.func = alc_fixup_bass_chmap,
                 .chained = true,
                 .chain_id = ALC662_FIXUP_ASUS_MODE4
         },
@@ -4936,7 +4962,7 @@ static const struct hda_fixup alc662_fixups[] = {
         },
         [ALC662_FIXUP_BASS_1A_CHMAP] = {
                 .type = HDA_FIXUP_FUNC,
-                .v.func = alc662_fixup_bass_chmap,
+                .v.func = alc_fixup_bass_chmap,
                 .chained = true,
                 .chain_id = ALC662_FIXUP_BASS_1A,
         },
@@ -5118,6 +5144,7 @@ static int patch_alc662(struct hda_codec *codec)
                 case 0x10ec0272:
                 case 0x10ec0663:
                 case 0x10ec0665:
+                case 0x10ec0668:
                         set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
                         break;
                 case 0x10ec0273:
@@ -5175,6 +5202,7 @@ static int patch_alc680(struct hda_codec *codec)
  */
 static const struct hda_codec_preset snd_hda_preset_realtek[] = {
         { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 },
+        { .id = 0x10ec0231, .name = "ALC231", .patch = patch_alc269 },
         { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 },
         { .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 },
         { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d2cc0041d9d3..088a5afbd1b9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2094,7 +2094,8 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec,
 
         if (action == HDA_FIXUP_ACT_PRE_PROBE) {
                 spec->mic_mute_led_gpio = 0x08; /* GPIO3 */
-                codec->bus->avoid_link_reset = 1;
+                /* resetting controller clears GPIO, so we need to keep on */
+                codec->bus->power_keep_link_on = 1;
         }
 }
 
diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
index 992ae38d5a15..1b372283bd01 100644
--- a/sound/soc/atmel/sam9x5_wm8731.c
+++ b/sound/soc/atmel/sam9x5_wm8731.c
@@ -97,6 +97,8 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
                 goto out;
         }
 
+        snd_soc_card_set_drvdata(card, priv);
+
         card->dev = &pdev->dev;
         card->owner = THIS_MODULE;
         card->dai_link = dai;
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index c3c7396a6181..99b359e19d35 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -248,19 +248,6 @@ ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE),
 
-SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
-           ARIZONA_OUT1_OSR_SHIFT, 1, 0),
-SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
-           ARIZONA_OUT2_OSR_SHIFT, 1, 0),
-SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
-           ARIZONA_OUT3_OSR_SHIFT, 1, 0),
-SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
-           ARIZONA_OUT4_OSR_SHIFT, 1, 0),
-SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
-           ARIZONA_OUT5_OSR_SHIFT, 1, 0),
-SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L,
-           ARIZONA_OUT6_OSR_SHIFT, 1, 0),
-
 SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
              ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
 SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
@@ -293,18 +280,6 @@ SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L,
                  ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT,
                  0xbf, 0, digital_tlv),
 
-SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
-                       ARIZONA_OUTPUT_PATH_CONFIG_1R,
-                       ARIZONA_OUT1L_PGA_VOL_SHIFT,
-                       0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
-                       ARIZONA_OUTPUT_PATH_CONFIG_2R,
-                       ARIZONA_OUT2L_PGA_VOL_SHIFT,
-                       0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
-                       ARIZONA_OUTPUT_PATH_CONFIG_3R,
-                       ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
-
 SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
            ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
 SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT,
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 456bb8c6d759..bc7472c968e3 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -447,10 +447,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
                 iface |= 0x0001;
                 break;
         case SND_SOC_DAIFMT_DSP_A:
-                iface |= 0x0003;
+                iface |= 0x0013;
                 break;
         case SND_SOC_DAIFMT_DSP_B:
-                iface |= 0x0013;
+                iface |= 0x0003;
                 break;
         default:
                 return -EINVAL;
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index eb4373840bb6..3665f612819d 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -69,7 +69,6 @@ static int pcm030_fabric_probe(struct platform_device *op)
                 return -ENOMEM;
 
         card->dev = &op->dev;
-        platform_set_drvdata(op, pdata);
 
         pdata->card = card;
 
@@ -98,6 +97,8 @@ static int pcm030_fabric_probe(struct platform_device *op)
         if (ret)
                 dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret);
 
+        platform_set_drvdata(op, pdata);
+
         return ret;
 }
 
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index d34d91743e3f..0b18f654b413 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -33,6 +33,10 @@
          SNDRV_PCM_FMTBIT_S24_LE | \
          SNDRV_PCM_FMTBIT_S32_LE)
 
+#define KIRKWOOD_SPDIF_FORMATS \
+        (SNDRV_PCM_FMTBIT_S16_LE | \
+         SNDRV_PCM_FMTBIT_S24_LE)
+
 static int kirkwood_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
                 unsigned int fmt)
 {
@@ -244,15 +248,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
                                    ctl);
         }
 
-        if (dai->id == 0)
-                ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN;      /* i2s */
-        else
-                ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN;        /* spdif */
-
         switch (cmd) {
         case SNDRV_PCM_TRIGGER_START:
                 /* configure */
                 ctl = priv->ctl_play;
+                if (dai->id == 0)
+                        ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN;      /* i2s */
+                else
+                        ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN;        /* spdif */
+
                 value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
                 writel(value, priv->io + KIRKWOOD_PLAYCTL);
 
@@ -449,14 +453,14 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = {
                 .channels_max = 2,
                 .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
                                 SNDRV_PCM_RATE_96000,
-                .formats = KIRKWOOD_I2S_FORMATS,
+                .formats = KIRKWOOD_SPDIF_FORMATS,
         },
         .capture = {
                 .channels_min = 1,
                 .channels_max = 2,
                 .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
                                 SNDRV_PCM_RATE_96000,
-                .formats = KIRKWOOD_I2S_FORMATS,
+                .formats = KIRKWOOD_SPDIF_FORMATS,
         },
         .ops = &kirkwood_i2s_dai_ops,
     },
@@ -493,7 +497,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
                 .rates = SNDRV_PCM_RATE_8000_192000 |
                          SNDRV_PCM_RATE_CONTINUOUS |
                          SNDRV_PCM_RATE_KNOT,
-                .formats = KIRKWOOD_I2S_FORMATS,
+                .formats = KIRKWOOD_SPDIF_FORMATS,
         },
         .capture = {
                 .channels_min = 1,
@@ -501,7 +505,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
                 .rates = SNDRV_PCM_RATE_8000_192000 |
                          SNDRV_PCM_RATE_CONTINUOUS |
                          SNDRV_PCM_RATE_KNOT,
-                .formats = KIRKWOOD_I2S_FORMATS,
+                .formats = KIRKWOOD_SPDIF_FORMATS,
         },
         .ops = &kirkwood_i2s_dai_ops,
     },
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 6d216cb6c19b..3fde9e402710 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -100,12 +100,12 @@ static int n810_startup(struct snd_pcm_substream *substream)
                                      SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
 
         n810_ext_control(&codec->dapm);
-        return clk_enable(sys_clkout2);
+        return clk_prepare_enable(sys_clkout2);
 }
 
 static void n810_shutdown(struct snd_pcm_substream *substream)
 {
-        clk_disable(sys_clkout2);
+        clk_disable_unprepare(sys_clkout2);
 }
 
 static int n810_hw_params(struct snd_pcm_substream *substream,
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 14011d90d70a..ff60e11ecb56 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU
 config SND_SOC_RCAR
         tristate "R-Car series SRU/SCU/SSIU/SSI support"
         select SND_SIMPLE_CARD
+        select REGMAP
         help
           This option enables R-Car SUR/SCU/SSIU/SSI sound support
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4e53d87e881d..a66783e13a9c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3212,11 +3212,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
                         break;
                 case 2:
                         ((u16 *)(&ucontrol->value.bytes.data))[0]
-                                &= ~params->mask;
+                                &= cpu_to_be16(~params->mask);
                         break;
                 case 4:
                         ((u32 *)(&ucontrol->value.bytes.data))[0]
-                                &= ~params->mask;
+                                &= cpu_to_be32(~params->mask);
                         break;
                 default:
                         return -EINVAL;
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index b1d732255c02..3449c1e909ae 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -66,7 +66,7 @@ static void devm_card_release(struct device *dev, void *res)
  */
 int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
 {
-        struct device **ptr;
+        struct snd_soc_card **ptr;
         int ret;
 
         ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL);
@@ -75,7 +75,7 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
 
         ret = snd_soc_register_card(card);
         if (ret == 0) {
-                *ptr = dev;
+                *ptr = card;
                 devres_add(dev, ptr);
         } else {
                 devres_free(ptr);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 42782c01e413..11a90cd027fa 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -148,12 +148,12 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream,
         }
 }
 
-static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw,
+static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
         struct snd_soc_pcm_stream *codec_stream,
         struct snd_soc_pcm_stream *cpu_stream)
 {
-        hw->rate_min = max(codec_stream->rate_min, cpu_stream->rate_min);
-        hw->rate_max = max(codec_stream->rate_max, cpu_stream->rate_max);
+        struct snd_pcm_hardware *hw = &runtime->hw;
+
         hw->channels_min = max(codec_stream->channels_min,
                 cpu_stream->channels_min);
         hw->channels_max = min(codec_stream->channels_max,
@@ -166,6 +166,13 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw,
         if (cpu_stream->rates
                 & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
                 hw->rates |= codec_stream->rates;
+
+        snd_pcm_limit_hw_rates(runtime);
+
+        hw->rate_min = max(hw->rate_min, cpu_stream->rate_min);
+        hw->rate_min = max(hw->rate_min, codec_stream->rate_min);
+        hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max);
+        hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max);
 }
 
 /*
@@ -235,15 +242,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 
         /* Check that the codec and cpu DAIs are compatible */
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-                soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->playback,
+                soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback,
                         &cpu_dai_drv->playback);
         } else {
-                soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->capture,
+                soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture,
                         &cpu_dai_drv->capture);
         }
 
         ret = -EINVAL;
-        snd_pcm_limit_hw_rates(runtime);
         if (!runtime->hw.rates) {
                 printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n",
                         codec_dai->name, cpu_dai->name);
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index b9ba0fcc45df..83aabea259d7 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -636,8 +636,22 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep,
         if (usb_pipein(ep->pipe) ||
                         snd_usb_endpoint_implicit_feedback_sink(ep)) {
 
+                urb_packs = packs_per_ms;
+                /*
+                 * Wireless devices can poll at a max rate of once per 4ms.
+                 * For dataintervals less than 5, increase the packet count to
+                 * allow the host controller to use bursting to fill in the
+                 * gaps.
+                 */
+                if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) {
+                        int interval = ep->datainterval;
+                        while (interval < 5) {
+                                urb_packs <<= 1;
+                                ++interval;
+                        }
+                }
                 /* make capture URBs <= 1 ms and smaller than a period */
-                urb_packs = min(max_packs_per_urb, packs_per_ms);
+                urb_packs = min(max_packs_per_urb, urb_packs);
                 while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
                         urb_packs >>= 1;
                 ep->nurbs = MAX_URBS;