Merge tag 'v3.13-rc7' into x86/efi-kexec to resolve conflicts

Conflicts:
	arch/x86/platform/efi/efi.c
	drivers/firmware/efi/Kconfig

Signed-off-by: Ingo Molnar <mingo@kernel.org>

35 files changed, 471 insertions, 176 deletions

diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 872d59e35ee2..721d8fd45685 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -357,7 +357,8 @@ static int set_sample_rates(struct atmel_abdac *dac)
 		if (new_rate < 0)
 			break;
 		/* make sure we are below the ABDAC clock */
-		if (new_rate <= clk_get_rate(dac->pclk)) {
+		if (index < MAX_NUM_RATES &&
+		    new_rate <= clk_get_rate(dac->pclk)) {
 			dac->rates[index] = new_rate / 256;
 			index++;
 		}
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 6e03b465e44e..a2104671f51d 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1937,6 +1937,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 		case SNDRV_PCM_STATE_DISCONNECTED:
 			err = -EBADFD;
 			goto _endloop;
+		case SNDRV_PCM_STATE_PAUSED:
+			continue;
 		}
 		if (!tout) {
 			snd_printd("%s write error (DMA or IRQ trouble?)\n",
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/firewire/dice.c b/sound/firewire/dice.c
index 57bcd31fcc12..c0aa64941cee 100644
--- a/sound/firewire/dice.c
+++ b/sound/firewire/dice.c
@@ -1019,7 +1019,7 @@ static void dice_proc_read(struct snd_info_entry *entry,
 
 	if (dice_proc_read_mem(dice, &tx_rx_header, sections[2], 2) < 0)
 		return;
-	quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx));
+	quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx) / 4);
 	for (stream = 0; stream < tx_rx_header.number; ++stream) {
 		if (dice_proc_read_mem(dice, &buf.tx, sections[2] + 2 +
 				       stream * tx_rx_header.size,
@@ -1045,7 +1045,7 @@ static void dice_proc_read(struct snd_info_entry *entry,
 
 	if (dice_proc_read_mem(dice, &tx_rx_header, sections[4], 2) < 0)
 		return;
-	quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx));
+	quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx) / 4);
 	for (stream = 0; stream < tx_rx_header.number; ++stream) {
 		if (dice_proc_read_mem(dice, &buf.rx, sections[4] + 2 +
 				       stream * tx_rx_header.size,
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..c7f6d1cab606 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -474,6 +474,20 @@ static void invalidate_nid_path(struct hda_codec *codec, int idx)
 	memset(path, 0, sizeof(*path));
 }
 
+/* return a DAC if paired to the given pin by codec driver */
+static hda_nid_t get_preferred_dac(struct hda_codec *codec, hda_nid_t pin)
+{
+	struct hda_gen_spec *spec = codec->spec;
+	const hda_nid_t *list = spec->preferred_dacs;
+
+	if (!list)
+		return 0;
+	for (; *list; list += 2)
+		if (*list == pin)
+			return list[1];
+	return 0;
+}
+
 /* look for an empty DAC slot */
 static hda_nid_t look_for_dac(struct hda_codec *codec, hda_nid_t pin,
 			      bool is_digital)
@@ -1192,7 +1206,14 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs,
 			continue;
 		}
 
-		dacs[i] = look_for_dac(codec, pin, false);
+		dacs[i] = get_preferred_dac(codec, pin);
+		if (dacs[i]) {
+			if (is_dac_already_used(codec, dacs[i]))
+				badness += bad->shared_primary;
+		}
+
+		if (!dacs[i])
+			dacs[i] = look_for_dac(codec, pin, false);
 		if (!dacs[i] && !i) {
 			/* try to steal the DAC of surrounds for the front */
 			for (j = 1; j < num_outs; j++) {
@@ -2506,12 +2527,8 @@ 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 +2781,7 @@ 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 +2801,6 @@ 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 +2829,42 @@ 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 +2872,11 @@ 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 +2885,18 @@ 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;
@@ -4271,6 +4318,26 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
 	return AC_PWRST_D3;
 }
 
+/* mute all aamix inputs initially; parse up to the first leaves */
+static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix)
+{
+	int i, nums;
+	const hda_nid_t *conn;
+	bool has_amp;
+
+	nums = snd_hda_get_conn_list(codec, mix, &conn);
+	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);
+		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);
+	}
+}
 
 /*
  * Parse the given BIOS configuration and set up the hda_gen_spec
@@ -4383,6 +4450,17 @@ 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,
@@ -4398,6 +4476,10 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
 		}
 	}
 
+	/* mute all aamix input initially */
+	if (spec->mixer_nid)
+		mute_all_mixer_nid(codec, spec->mixer_nid);
+
  dig_only:
 	parse_digital(codec);
 
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 7e45cb44d151..0929a06df812 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -249,6 +249,9 @@ struct hda_gen_spec {
 	const struct badness_table *main_out_badness;
 	const struct badness_table *extra_out_badness;
 
+	/* preferred pin/DAC pairs; an array of paired NIDs */
+	const hda_nid_t *preferred_dacs;
+
 	/* loopback mixing mode */
 	bool aamix_mode;
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 7a09404579a7..956871d8b3d2 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);
@@ -3434,6 +3433,10 @@ static void check_probe_mask(struct azx *chip, int dev)
  * white/black-list for enable_msi
  */
 static struct snd_pci_quirk msi_black_list[] = {
+	SND_PCI_QUIRK(0x103c, 0x2191, "HP", 0), /* AMD Hudson */
+	SND_PCI_QUIRK(0x103c, 0x2192, "HP", 0), /* AMD Hudson */
+	SND_PCI_QUIRK(0x103c, 0x21f7, "HP", 0), /* AMD Hudson */
+	SND_PCI_QUIRK(0x103c, 0x21fa, "HP", 0), /* AMD Hudson */
 	SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
 	SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */
 	SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */
@@ -3877,7 +3880,8 @@ static int azx_probe(struct pci_dev *pci,
 	}
 
 	dev++;
-	complete_all(&chip->probe_wait);
+	if (chip->disabled)
+		complete_all(&chip->probe_wait);
 	return 0;
 
 out_free:
@@ -3954,10 +3958,10 @@ static int azx_probe_continue(struct azx *chip)
 	if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || chip->use_vga_switcheroo)
 		pm_runtime_put_noidle(&pci->dev);
 
-	return 0;
-
 out_free:
-	chip->init_failed = 1;
+	if (err < 0)
+		chip->init_failed = 1;
+	complete_all(&chip->probe_wait);
 	return err;
 }
 
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 1a83559f4cbd..699262a3e07a 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -147,6 +147,8 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled)
 
 	if (!spec->eapd_nid)
 		return;
+	if (codec->inv_eapd)
+		enabled = !enabled;
 	snd_hda_codec_update_cache(codec, spec->eapd_nid, 0,
 				   AC_VERB_SET_EAPD_BTLENABLE,
 				   enabled ? 0x02 : 0x00);
@@ -338,6 +340,14 @@ static int patch_ad1986a(struct hda_codec *codec)
 {
 	int err;
 	struct ad198x_spec *spec;
+	static hda_nid_t preferred_pairs[] = {
+		0x1a, 0x03,
+		0x1b, 0x03,
+		0x1c, 0x04,
+		0x1d, 0x05,
+		0x1e, 0x03,
+		0
+	};
 
 	err = alloc_ad_spec(codec);
 	if (err < 0)
@@ -358,6 +368,11 @@ static int patch_ad1986a(struct hda_codec *codec)
 	 * So, let's disable the shared stream.
 	 */
 	spec->gen.multiout.no_share_stream = 1;
+	/* give fixed DAC/pin pairs */
+	spec->gen.preferred_dacs = preferred_pairs;
+
+	/* AD1986A can't manage the dynamic pin on/off smoothly */
+	spec->gen.auto_mute_via_amp = 1;
 
 	snd_hda_pick_fixup(codec, ad1986a_fixup_models, ad1986a_fixup_tbl,
 			   ad1986a_fixups);
@@ -962,6 +977,7 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
 	switch (action) {
 	case HDA_FIXUP_ACT_PRE_PROBE:
 		spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook;
+		spec->gen.own_eapd_ctl = 1;
 		snd_hda_sequence_write_cache(codec, gpio_init_verbs);
 		break;
 	case HDA_FIXUP_ACT_PROBE:
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index c205bb1747fd..3fbf2883e06e 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2936,7 +2936,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD),
-	SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
 	SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
@@ -3244,9 +3243,29 @@ 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 +3298,8 @@ 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 +3515,7 @@ 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_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 08407bed093e..f281c8068557 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1142,32 +1142,34 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
 
 static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll);
 
-static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
+static void jack_callback(struct hda_codec *codec, struct hda_jack_tbl *jack)
 {
 	struct hdmi_spec *spec = codec->spec;
+	int pin_idx = pin_nid_to_pin_index(spec, jack->nid);
+	if (pin_idx < 0)
+		return;
+
+	if (hdmi_present_sense(get_pin(spec, pin_idx), 1))
+		snd_hda_jack_report_sync(codec);
+}
+
+static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
+{
 	int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
-	int pin_nid;
-	int pin_idx;
 	struct hda_jack_tbl *jack;
 	int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;
 
 	jack = snd_hda_jack_tbl_get_from_tag(codec, tag);
 	if (!jack)
 		return;
-	pin_nid = jack->nid;
 	jack->jack_dirty = 1;
 
 	_snd_printd(SND_PR_VERBOSE,
 		"HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n",
-		codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA),
+		codec->addr, jack->nid, dev_entry, !!(res & AC_UNSOL_RES_IA),
 		!!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
 
-	pin_idx = pin_nid_to_pin_index(spec, pin_nid);
-	if (pin_idx < 0)
-		return;
-
-	if (hdmi_present_sense(get_pin(spec, pin_idx), 1))
-		snd_hda_jack_report_sync(codec);
+	jack_callback(codec, jack);
 }
 
 static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res)
@@ -2095,7 +2097,8 @@ static int generic_hdmi_init(struct hda_codec *codec)
 		hda_nid_t pin_nid = per_pin->pin_nid;
 
 		hdmi_init_pin(codec, pin_nid);
-		snd_hda_jack_detect_enable(codec, pin_nid, pin_nid);
+		snd_hda_jack_detect_enable_callback(codec, pin_nid, pin_nid,
+			codec->jackpoll_interval > 0 ? jack_callback : NULL);
 	}
 	return 0;
 }
@@ -2334,8 +2337,9 @@ static int simple_playback_build_controls(struct hda_codec *codec)
 	int err;
 
 	per_cvt = get_cvt(spec, 0);
-	err = snd_hda_create_spdif_out_ctls(codec, per_cvt->cvt_nid,
-					    per_cvt->cvt_nid);
+	err = snd_hda_create_dig_out_ctls(codec, per_cvt->cvt_nid,
+					  per_cvt->cvt_nid,
+					  HDA_PCM_TYPE_HDMI);
 	if (err < 0)
 		return err;
 	return simple_hdmi_build_jack(codec, 0);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5e42059f10a1..c5646941539a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1780,8 +1780,11 @@ enum {
 	ALC889_FIXUP_DAC_ROUTE,
 	ALC889_FIXUP_MBP_VREF,
 	ALC889_FIXUP_IMAC91_VREF,
+	ALC889_FIXUP_MBA21_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,
@@ -1882,17 +1885,13 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec,
 	}
 }
 
-/* Set VREF on speaker pins on imac91 */
-static void alc889_fixup_imac91_vref(struct hda_codec *codec,
-				     const struct hda_fixup *fix, int action)
+static void alc889_fixup_mac_pins(struct hda_codec *codec,
+				  const hda_nid_t *nids, int num_nids)
 {
 	struct alc_spec *spec = codec->spec;
-	static hda_nid_t nids[2] = { 0x18, 0x1a };
 	int i;
 
-	if (action != HDA_FIXUP_ACT_INIT)
-		return;
-	for (i = 0; i < ARRAY_SIZE(nids); i++) {
+	for (i = 0; i < num_nids; i++) {
 		unsigned int val;
 		val = snd_hda_codec_get_pin_target(codec, nids[i]);
 		val |= AC_PINCTL_VREF_50;
@@ -1901,6 +1900,26 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec,
 	spec->gen.keep_vref_in_automute = 1;
 }
 
+/* Set VREF on speaker pins on imac91 */
+static void alc889_fixup_imac91_vref(struct hda_codec *codec,
+				     const struct hda_fixup *fix, int action)
+{
+	static hda_nid_t nids[2] = { 0x18, 0x1a };
+
+	if (action == HDA_FIXUP_ACT_INIT)
+		alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids));
+}
+
+/* Set VREF on speaker pins on mba21 */
+static void alc889_fixup_mba21_vref(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	static hda_nid_t nids[2] = { 0x18, 0x19 };
+
+	if (action == HDA_FIXUP_ACT_INIT)
+		alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids));
+}
+
 /* Don't take HP output as primary
  * Strangely, the speaker output doesn't work on Vaio Z and some Vaio
  * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05
@@ -1915,6 +1934,9 @@ 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,
@@ -2097,6 +2119,12 @@ static const struct hda_fixup alc882_fixups[] = {
 		.chained = true,
 		.chain_id = ALC882_FIXUP_GPIO1,
 	},
+	[ALC889_FIXUP_MBA21_VREF] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc889_fixup_mba21_vref,
+		.chained = true,
+		.chain_id = ALC889_FIXUP_MBP_VREF,
+	},
 	[ALC882_FIXUP_INV_DMIC] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_inv_dmic_0x12,
@@ -2105,6 +2133,19 @@ 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 +2179,7 @@ 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),
@@ -2153,7 +2195,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889_FIXUP_MBP_VREF),
 	SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBP_VREF),
-	SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBP_VREF),
+	SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBA21_VREF),
 	SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889_FIXUP_MBP_VREF),
 	SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF),
 	SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_FIXUP_MACPRO_GPIO),
@@ -3268,6 +3310,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec)
 		alc_write_coef_idx(codec, 0x18, 0x7388);
 		break;
 	case 0x10ec0668:
+		alc_write_coef_idx(codec, 0x11, 0x0001);
 		alc_write_coef_idx(codec, 0x15, 0x0d60);
 		alc_write_coef_idx(codec, 0xc3, 0x0000);
 		break;
@@ -3296,6 +3339,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec)
 		alc_write_coef_idx(codec, 0x18, 0x7388);
 		break;
 	case 0x10ec0668:
+		alc_write_coef_idx(codec, 0x11, 0x0001);
 		alc_write_coef_idx(codec, 0x15, 0x0d50);
 		alc_write_coef_idx(codec, 0xc3, 0x0000);
 		break;
@@ -3581,11 +3625,6 @@ static void alc283_hp_automute_hook(struct hda_codec *codec,
 			    vref);
 }
 
-static void alc283_chromebook_caps(struct hda_codec *codec)
-{
-	snd_hda_override_wcaps(codec, 0x03, 0);
-}
-
 static void alc283_fixup_chromebook(struct hda_codec *codec,
 				    const struct hda_fixup *fix, int action)
 {
@@ -3594,9 +3633,26 @@ static void alc283_fixup_chromebook(struct hda_codec *codec,
 
 	switch (action) {
 	case HDA_FIXUP_ACT_PRE_PROBE:
-		alc283_chromebook_caps(codec);
+		snd_hda_override_wcaps(codec, 0x03, 0);
 		/* Disable AA-loopback as it causes white noise */
 		spec->gen.mixer_nid = 0;
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		/* Enable Line1 input control by verb */
+		val = alc_read_coef_idx(codec, 0x1a);
+		alc_write_coef_idx(codec, 0x1a, val | (1 << 4));
+		break;
+	}
+}
+
+static void alc283_fixup_sense_combo_jack(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	int val;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
 		spec->gen.hp_automute_hook = alc283_hp_automute_hook;
 		break;
 	case HDA_FIXUP_ACT_INIT:
@@ -3604,9 +3660,6 @@ static void alc283_fixup_chromebook(struct hda_codec *codec,
 		/* Set to manual mode */
 		val = alc_read_coef_idx(codec, 0x06);
 		alc_write_coef_idx(codec, 0x06, val & ~0x000c);
-		/* Enable Line1 input control by verb */
-		val = alc_read_coef_idx(codec, 0x1a);
-		alc_write_coef_idx(codec, 0x1a, val | (1 << 4));
 		break;
 	}
 }
@@ -3796,11 +3849,14 @@ enum {
 	ALC269_FIXUP_ASUS_X101,
 	ALC271_FIXUP_AMIC_MIC2,
 	ALC271_FIXUP_HP_GATE_MIC_JACK,
+	ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572,
 	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,
+	ALC283_FIXUP_SENSE_COMBO_JACK,
 	ALC282_FIXUP_ASUS_TX300,
 	ALC283_FIXUP_INT_MIC,
 	ALC290_FIXUP_MONO_SPEAKERS,
@@ -4056,6 +4112,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC271_FIXUP_AMIC_MIC2,
 	},
+	[ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc269_fixup_limit_int_mic_boost,
+		.chained = true,
+		.chain_id = ALC271_FIXUP_HP_GATE_MIC_JACK,
+	},
 	[ALC269_FIXUP_ACER_AC700] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -4075,6 +4137,12 @@ 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,
@@ -4094,6 +4162,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc283_fixup_chromebook,
 	},
+	[ALC283_FIXUP_SENSE_COMBO_JACK] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc283_fixup_sense_combo_jack,
+		.chained = true,
+		.chain_id = ALC283_FIXUP_CHROME_BOOK,
+	},
 	[ALC282_FIXUP_ASUS_TX300] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc282_fixup_asus_tx300,
@@ -4141,6 +4215,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
 	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
 	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
@@ -4172,11 +4247,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS),
 	SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0629, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS),
+	SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -4184,13 +4264,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED),
-	SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED),
 	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),
@@ -4292,6 +4371,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"},
 	{.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"},
 	{.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"},
+	{.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-chrome"},
+	{.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"},
 	{}
 };
 
@@ -4467,6 +4548,7 @@ enum {
 	ALC861_FIXUP_AMP_VREF_0F,
 	ALC861_FIXUP_NO_JACK_DETECT,
 	ALC861_FIXUP_ASUS_A6RP,
+	ALC660_FIXUP_ASUS_W7J,
 };
 
 /* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */
@@ -4516,10 +4598,22 @@ static const struct hda_fixup alc861_fixups[] = {
 		.v.func = alc861_fixup_asus_amp_vref_0f,
 		.chained = true,
 		.chain_id = ALC861_FIXUP_NO_JACK_DETECT,
+	},
+	[ALC660_FIXUP_ASUS_W7J] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			/* ASUS W7J needs a magic pin setup on unused NID 0x10
+			 * for enabling outputs
+			 */
+			{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+			{ }
+		},
 	}
 };
 
 static const struct snd_pci_quirk alc861_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x1043, 0x1253, "ASUS W7J", ALC660_FIXUP_ASUS_W7J),
+	SND_PCI_QUIRK(0x1043, 0x1263, "ASUS Z35HL", ALC660_FIXUP_ASUS_W7J),
 	SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP),
 	SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F),
 	SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT),
@@ -4715,7 +4809,7 @@ 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 +5017,7 @@ 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 +5030,7 @@ 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,
 	},
@@ -4952,8 +5046,11 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
 	SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP),
 	SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP),
@@ -5118,6 +5215,7 @@ 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 +5273,7 @@ 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/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 8697cedccd21..1ead3c977a51 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -648,7 +648,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
 
 	dma_params = ssc_p->dma_params[dir];
 
-	ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+	ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
 	ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error);
 
 	pr_debug("%s enabled SSC_SR=0x%08x\n",
@@ -657,6 +657,33 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int atmel_ssc_trigger(struct snd_pcm_substream *substream,
+			     int cmd, struct snd_soc_dai *dai)
+{
+	struct atmel_ssc_info *ssc_p = &ssc_info[dai->id];
+	struct atmel_pcm_dma_params *dma_params;
+	int dir;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir = 0;
+	else
+		dir = 1;
+
+	dma_params = ssc_p->dma_params[dir];
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+		break;
+	default:
+		ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
+		break;
+	}
+
+	return 0;
+}
 
 #ifdef CONFIG_PM
 static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai)
@@ -731,6 +758,7 @@ static const struct snd_soc_dai_ops atmel_ssc_dai_ops = {
 	.startup	= atmel_ssc_startup,
 	.shutdown	= atmel_ssc_shutdown,
 	.prepare	= atmel_ssc_prepare,
+	.trigger	= atmel_ssc_trigger,
 	.hw_params	= atmel_ssc_hw_params,
 	.set_fmt	= atmel_ssc_set_dai_fmt,
 	.set_clkdiv	= atmel_ssc_set_dai_clkdiv,
diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
index 992ae38d5a15..7d6a9055874b 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;
@@ -107,7 +109,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 	dai->stream_name = "WM8731 PCM";
 	dai->codec_dai_name = "wm8731-hifi";
 	dai->init = sam9x5_wm8731_init;
-	dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+	dai->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF
 		| SND_SOC_DAIFMT_CBM_CFM;
 
 	ret = snd_soc_of_parse_card_name(card, "atmel,model");
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index c3c7396a6181..0ab2dc296474 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,
@@ -1037,7 +1012,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	{ "AEC Loopback", "HPOUT3L", "OUT3L" },
 	{ "AEC Loopback", "HPOUT3R", "OUT3R" },
 	{ "HPOUT3L", NULL, "OUT3L" },
-	{ "HPOUT3R", NULL, "OUT3L" },
+	{ "HPOUT3R", NULL, "OUT3R" },
 
 	{ "AEC Loopback", "SPKOUTL", "OUT4L" },
 	{ "SPKOUTLN", NULL, "OUT4L" },
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/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 3938fb1c203e..53bbfac6a83a 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1444,7 +1444,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
-		aif1 |= WM8904_AIF_LRCLK_INV;
+		aif1 |= 0x3 | WM8904_AIF_LRCLK_INV;
 	case SND_SOC_DAIFMT_DSP_A:
 		aif1 |= 0x3;
 		break;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 543c5c2631b6..0f17ed3e29f4 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2439,7 +2439,20 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, WM8962_CLOCKING_4,
 			    WM8962_SYSCLK_RATE_MASK, clocking4);
 
+	/* DSPCLK_DIV can be only generated correctly after enabling SYSCLK.
+	 * So we here provisionally enable it and then disable it afterward
+	 * if current bias_level hasn't reached SND_SOC_BIAS_ON.
+	 */
+	if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
+		snd_soc_update_bits(codec, WM8962_CLOCKING2,
+				WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA);
+
 	dspclk = snd_soc_read(codec, WM8962_CLOCKING1);
+
+	if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
+		snd_soc_update_bits(codec, WM8962_CLOCKING2,
+				WM8962_SYSCLK_ENA_MASK, 0);
+
 	if (dspclk < 0) {
 		dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk);
 		return;
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 253c88bb7a4c..4f05fb88bddf 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1259,6 +1259,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 
 		/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
 		snd_soc_write(codec, WM8990_ANTIPOP2, 0x0);
+
+		codec->cache_sync = 1;
 		break;
 	}
 
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 46ec0e9744d4..4fbcab63e61f 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1474,13 +1474,17 @@ static int wm_adsp2_ena(struct wm_adsp *dsp)
 		return ret;
 
 	/* Wait for the RAM to start, should be near instantaneous */
-	count = 0;
-	do {
+	for (count = 0; count < 10; ++count) {
 		ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1,
 				  &val);
 		if (ret != 0)
 			return ret;
-	} while (!(val & ADSP2_RAM_RDY) && ++count < 10);
+
+		if (val & ADSP2_RAM_RDY)
+			break;
+
+		msleep(1);
+	}
 
 	if (!(val & ADSP2_RAM_RDY)) {
 		adsp_err(dsp, "Failed to start DSP RAM\n");
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 61e48852b9e8..3fd76bc391de 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -130,8 +130,6 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
 		break;
 	}
 
-	dapm->bias_level = level;
-
 	return 0;
 }
 
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..3920a5e8125f 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,
     },
@@ -469,17 +473,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000 |
-			 SNDRV_PCM_RATE_CONTINUOUS |
-			 SNDRV_PCM_RATE_KNOT,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 5512,
+		.rate_max = 192000,
 		.formats = KIRKWOOD_I2S_FORMATS,
 	},
 	.capture = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000 |
-			 SNDRV_PCM_RATE_CONTINUOUS |
-			 SNDRV_PCM_RATE_KNOT,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 5512,
+		.rate_max = 192000,
 		.formats = KIRKWOOD_I2S_FORMATS,
 	},
 	.ops = &kirkwood_i2s_dai_ops,
@@ -490,18 +494,18 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000 |
-			 SNDRV_PCM_RATE_CONTINUOUS |
-			 SNDRV_PCM_RATE_KNOT,
-		.formats = KIRKWOOD_I2S_FORMATS,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 5512,
+		.rate_max = 192000,
+		.formats = KIRKWOOD_SPDIF_FORMATS,
 	},
 	.capture = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000 |
-			 SNDRV_PCM_RATE_CONTINUOUS |
-			 SNDRV_PCM_RATE_KNOT,
-		.formats = KIRKWOOD_I2S_FORMATS,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 5512,
+		.rate_max = 192000,
+		.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-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index cbc9c96ce1f4..41949af3baae 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -305,6 +305,20 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
 	}
 }
 
+static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm)
+{
+	unsigned int i;
+
+	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
+	     i++) {
+		if (!pcm->chan[i])
+			continue;
+		dma_release_channel(pcm->chan[i]);
+		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
+			break;
+	}
+}
+
 /**
  * snd_dmaengine_pcm_register - Register a dmaengine based PCM device
  * @dev: The parent device for the PCM device
@@ -315,6 +329,7 @@ int snd_dmaengine_pcm_register(struct device *dev,
 	const struct snd_dmaengine_pcm_config *config, unsigned int flags)
 {
 	struct dmaengine_pcm *pcm;
+	int ret;
 
 	pcm = kzalloc(sizeof(*pcm), GFP_KERNEL);
 	if (!pcm)
@@ -326,11 +341,20 @@ int snd_dmaengine_pcm_register(struct device *dev,
 	dmaengine_pcm_request_chan_of(pcm, dev);
 
 	if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE)
-		return snd_soc_add_platform(dev, &pcm->platform,
+		ret = snd_soc_add_platform(dev, &pcm->platform,
 				&dmaengine_no_residue_pcm_platform);
 	else
-		return snd_soc_add_platform(dev, &pcm->platform,
+		ret = snd_soc_add_platform(dev, &pcm->platform,
 				&dmaengine_pcm_platform);
+	if (ret)
+		goto err_free_dma;
+
+	return 0;
+
+err_free_dma:
+	dmaengine_pcm_release_chan(pcm);
+	kfree(pcm);
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register);
 
@@ -345,7 +369,6 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
 {
 	struct snd_soc_platform *platform;
 	struct dmaengine_pcm *pcm;
-	unsigned int i;
 
 	platform = snd_soc_lookup_platform(dev);
 	if (!platform)
@@ -353,15 +376,8 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
 
 	pcm = soc_platform_to_pcm(platform);
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
-		if (pcm->chan[i]) {
-			dma_release_channel(pcm->chan[i]);
-			if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
-				break;
-		}
-	}
-
 	snd_soc_remove_platform(platform);
+	dmaengine_pcm_release_chan(pcm);
 	kfree(pcm);
 }
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 42782c01e413..891b9a9bcbf8 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);
@@ -594,12 +600,13 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
+	bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
 
 	/* apply codec digital mute */
-	if (!codec->active)
+	if ((playback && codec_dai->playback_active == 1) ||
+	    (!playback && codec_dai->capture_active == 1))
 		snd_soc_dai_digital_mute(codec_dai, 1, substream->stream);
 
 	/* free any machine hw params */
diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index 364bf6a907e1..8c819f811470 100644
--- a/sound/soc/tegra/tegra20_i2s.c
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -74,7 +74,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
 				unsigned int fmt)
 {
 	struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	unsigned int mask, val;
+	unsigned int mask = 0, val = 0;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
@@ -83,10 +83,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	mask = TEGRA20_I2S_CTRL_MASTER_ENABLE;
+	mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
-		val = TEGRA20_I2S_CTRL_MASTER_ENABLE;
+		val |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		break;
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index 08bc6931c7c7..8c7c1028e579 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -67,15 +67,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream,
 {
 	struct device *dev = dai->dev;
 	struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai);
-	unsigned int mask, val;
+	unsigned int mask = 0, val = 0;
 	int ret, spdifclock;
 
-	mask = TEGRA20_SPDIF_CTRL_PACK |
-	       TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
+	mask |= TEGRA20_SPDIF_CTRL_PACK |
+		TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-		val = TEGRA20_SPDIF_CTRL_PACK |
-		      TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
+		val |= TEGRA20_SPDIF_CTRL_PACK |
+		       TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
 		break;
 	default:
 		return -EINVAL;
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index 231a785b3921..02247fee1cf7 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -118,7 +118,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
 				unsigned int fmt)
 {
 	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	unsigned int mask, val;
+	unsigned int mask = 0, val = 0;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
@@ -127,10 +127,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	mask = TEGRA30_I2S_CTRL_MASTER_ENABLE;
+	mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
-		val = TEGRA30_I2S_CTRL_MASTER_ENABLE;
+		val |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		break;
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;
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 3454262358b3..f4b12c216f1c 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1603,7 +1603,7 @@ static int snd_microii_controls_create(struct usb_mixer_interface *mixer)
 			return err;
 	}
 
-	return err;
+	return 0;
 }
 
 int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)