Merge branch 'master' of github.com:davem330/net

Conflicts:
	MAINTAINERS
	drivers/net/Kconfig
	drivers/net/ethernet/broadcom/bnx2x/bnx2x_link.c
	drivers/net/ethernet/broadcom/tg3.c
	drivers/net/wireless/iwlwifi/iwl-pci.c
	drivers/net/wireless/iwlwifi/iwl-trans-tx-pcie.c
	drivers/net/wireless/rt2x00/rt2800usb.c
	drivers/net/wireless/wl12xx/main.c

54 files changed, 845 insertions, 170 deletions

diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index 3fd1a7e24928..552b97afbca5 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -1073,10 +1073,10 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
 	sdev->pcmid = -1;
 	list_del(&ldev->list);
 	layouts_list_items--;
+	kfree(ldev);
  outnodev:
  	of_node_put(sound);
  	layout_device = NULL;
- 	kfree(ldev);
 	return -ENODEV;
 }
 
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 86d0caf91b35..62e90b862a0d 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1761,6 +1761,10 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 	snd_pcm_uframes_t avail = 0;
 	long wait_time, tout;
 
+	init_waitqueue_entry(&wait, current);
+	set_current_state(TASK_INTERRUPTIBLE);
+	add_wait_queue(&runtime->tsleep, &wait);
+
 	if (runtime->no_period_wakeup)
 		wait_time = MAX_SCHEDULE_TIMEOUT;
 	else {
@@ -1771,16 +1775,32 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 		}
 		wait_time = msecs_to_jiffies(wait_time * 1000);
 	}
-	init_waitqueue_entry(&wait, current);
-	add_wait_queue(&runtime->tsleep, &wait);
+
 	for (;;) {
 		if (signal_pending(current)) {
 			err = -ERESTARTSYS;
 			break;
 		}
+
+		/*
+		 * We need to check if space became available already
+		 * (and thus the wakeup happened already) first to close
+		 * the race of space already having become available.
+		 * This check must happen after been added to the waitqueue
+		 * and having current state be INTERRUPTIBLE.
+		 */
+		if (is_playback)
+			avail = snd_pcm_playback_avail(runtime);
+		else
+			avail = snd_pcm_capture_avail(runtime);
+		if (avail >= runtime->twake)
+			break;
 		snd_pcm_stream_unlock_irq(substream);
-		tout = schedule_timeout_interruptible(wait_time);
+
+		tout = schedule_timeout(wait_time);
+
 		snd_pcm_stream_lock_irq(substream);
+		set_current_state(TASK_INTERRUPTIBLE);
 		switch (runtime->status->state) {
 		case SNDRV_PCM_STATE_SUSPENDED:
 			err = -ESTRPIPE;
@@ -1806,14 +1826,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 			err = -EIO;
 			break;
 		}
-		if (is_playback)
-			avail = snd_pcm_playback_avail(runtime);
-		else
-			avail = snd_pcm_capture_avail(runtime);
-		if (avail >= runtime->twake)
-			break;
 	}
  _endloop:
+	set_current_state(TASK_RUNNING);
 	remove_wait_queue(&runtime->tsleep, &wait);
 	*availp = avail;
 	return err;
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 200c9a1d48b7..a872d0a82976 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1909,6 +1909,7 @@ static unsigned int ad1981_jacks_whitelist[] = {
 	0x103c0944, /* HP nc6220 */
 	0x103c0934, /* HP nc8220 */
 	0x103c006d, /* HP nx9105 */
+	0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */
 	0x17340088, /* FSC Scenic-W */
 	0 /* end */
 };
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index e4d76a270c9f..579fc0dce128 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 	int err;
 
 	snd_azf3328_dbgcallenter();
-	if (dev >= SNDRV_CARDS)
-		return -ENODEV;
+	if (dev >= SNDRV_CARDS) {
+		err = -ENODEV;
+		goto out;
+	}
 	if (!enable[dev]) {
 		dev++;
-		return -ENOENT;
+		err = -ENOENT;
+		goto out;
 	}
 
 	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
 	if (err < 0)
-		return err;
+		goto out;
 
 	strcpy(card->driver, "AZF3328");
 	strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
diff --git a/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c
index be58bf2f3aec..2e5876ce71fe 100644
--- a/sound/pci/hda/alc268_quirks.c
+++ b/sound/pci/hda/alc268_quirks.c
@@ -476,8 +476,8 @@ static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
 
 static const struct alc_config_preset alc268_presets[] = {
 	[ALC267_QUANTA_IL1] = {
-		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
-			    alc268_capture_nosrc_mixer },
+		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_nosrc_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc267_quanta_il1_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -492,8 +492,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_3ST] = {
-		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_base_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
 		.dac_nids = alc268_dac_nids,
@@ -507,8 +507,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.input_mux = &alc268_capture_source,
 	},
 	[ALC268_TOSHIBA] = {
-		.mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_toshiba_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_toshiba_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -525,8 +525,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER] = {
-		.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_acer_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -543,8 +543,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER_DMIC] = {
-		.mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_acer_dmic_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -561,9 +561,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER_ASPIRE_ONE] = {
-		.mixers = { alc268_acer_aspire_one_mixer,
-			    alc268_beep_mixer,
-			    alc268_capture_nosrc_mixer },
+		.mixers = { alc268_acer_aspire_one_mixer, alc268_beep_mixer},
+		.cap_mixer = alc268_capture_nosrc_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_aspire_one_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -579,8 +578,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_DELL] = {
-		.mixers = { alc268_dell_mixer, alc268_beep_mixer,
-			    alc268_capture_nosrc_mixer },
+		.mixers = { alc268_dell_mixer, alc268_beep_mixer},
+		.cap_mixer = alc268_capture_nosrc_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_dell_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -596,8 +595,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ZEPTO] = {
-		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_base_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_toshiba_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -616,7 +615,8 @@ static const struct alc_config_preset alc268_presets[] = {
 	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC268_TEST] = {
-		.mixers = { alc268_test_mixer, alc268_capture_mixer },
+		.mixers = { alc268_test_mixer },
+		.cap_mixer = alc268_capture_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_volume_init_verbs,
 				alc268_beep_init_verbs },
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3e7850c238c3..f3aefef37216 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -579,9 +579,13 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 		return -1;
 	}
 	recursive++;
-	for (i = 0; i < nums; i++)
+	for (i = 0; i < nums; i++) {
+		unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i]));
+		if (type == AC_WID_PIN || type == AC_WID_AUD_OUT)
+			continue;
 		if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
 			return i;
+	}
 	return -1;
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 28ce17d09c33..c34f730f4815 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = {
 	SNDRV_PCM_RATE_192000,	/* 7: 192000Hz */
 };
 
-static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
+static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid,
 					int byte_index)
 {
 	unsigned int val;
 
 	val = snd_hda_codec_read(codec, nid, 0,
 					AC_VERB_GET_HDMI_ELDD, byte_index);
-
 #ifdef BE_PARANOID
 	printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val);
 #endif
-
-	if ((val & AC_ELDD_ELD_VALID) == 0) {
-		snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n",
-								byte_index);
-		val = 0;
-	}
-
-	return val & AC_ELDD_ELD_DATA;
+	return val;
 }
 
 #define GRAB_BITS(buf, byte, lowbit, bits) 		\
@@ -344,11 +336,26 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
 	if (!buf)
 		return -ENOMEM;
 
-	for (i = 0; i < size; i++)
-		buf[i] = hdmi_get_eld_byte(codec, nid, i);
+	for (i = 0; i < size; i++) {
+		unsigned int val = hdmi_get_eld_data(codec, nid, i);
+		if (!(val & AC_ELDD_ELD_VALID)) {
+			if (!i) {
+				snd_printd(KERN_INFO
+					   "HDMI: invalid ELD data\n");
+				ret = -EINVAL;
+				goto error;
+			}
+			snd_printd(KERN_INFO
+				  "HDMI: invalid ELD data byte %d\n", i);
+			val = 0;
+		} else
+			val &= AC_ELDD_ELD_DATA;
+		buf[i] = val;
+	}
 
 	ret = hdmi_update_eld(eld, buf, size);
 
+error:
 	kfree(buf);
 	return ret;
 }
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 47d6ffc9b5b5..c45f3e69bcf0 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
 static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 			 unsigned int *idxp)
 {
-	int i;
+	int i, idx;
 	hda_nid_t nid;
 
 	nid = codec->start_nid;
@@ -384,9 +384,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 		type = get_wcaps_type(get_wcaps(codec, nid));
 		if (type != AC_WID_AUD_IN)
 			continue;
-		*idxp = snd_hda_get_conn_index(codec, nid, pin, false);
-		if (*idxp >= 0)
+		idx = snd_hda_get_conn_index(codec, nid, pin, false);
+		if (idx >= 0) {
+			*idxp = idx;
 			return nid;
+		}
 	}
 	return 0;
 }
@@ -533,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name,
 		      int index, unsigned int pval, int dir,
 		      struct snd_kcontrol **kctlp)
 {
-	char tmp[32];
+	char tmp[44];
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
 	knew.private_value = pval;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 502fc9499453..7696d05b9356 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin,
 
 #define MAX_AUTO_DACS	5
 
+#define DAC_SLAVE_FLAG	0x8000	/* filled dac is a slave */
+
 /* fill analog DAC list from the widget tree */
 static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
 {
@@ -3370,16 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
 /* fill pin_dac_pair list from the pin and dac list */
 static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
 			      int num_pins, hda_nid_t *dacs, int *rest,
-			      struct pin_dac_pair *filled, int type)
+			      struct pin_dac_pair *filled, int nums, 
+			      int type)
 {
-	int i, nums;
+	int i, start = nums;
 
-	nums = 0;
-	for (i = 0; i < num_pins; i++) {
+	for (i = 0; i < num_pins; i++, nums++) {
 		filled[nums].pin = pins[i];
 		filled[nums].type = type;
 		filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
-		nums++;
+		if (filled[nums].dac) 
+			continue;
+		if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) {
+			filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG;
+			continue;
+		}
+		if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) {
+			filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
+			continue;
+		}
+		snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]);
 	}
 	return nums;
 }
@@ -3395,19 +3407,19 @@ static void cx_auto_parse_output(struct hda_codec *codec)
 	rest = fill_cx_auto_dacs(codec, dacs);
 	/* parse all analog output pins */
 	nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs,
-				  dacs, &rest, spec->dac_info,
-				  AUTO_PIN_LINE_OUT);
-	nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
-				  dacs, &rest, spec->dac_info + nums,
-				  AUTO_PIN_HP_OUT);
-	nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
-				  dacs, &rest, spec->dac_info + nums,
-				  AUTO_PIN_SPEAKER_OUT);
+			  dacs, &rest, spec->dac_info, 0,
+			  AUTO_PIN_LINE_OUT);
+	nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
+			  dacs, &rest, spec->dac_info, nums,
+			  AUTO_PIN_HP_OUT);
+	nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
+			  dacs, &rest, spec->dac_info, nums,
+			  AUTO_PIN_SPEAKER_OUT);
 	spec->dac_info_filled = nums;
 	/* fill multiout struct */
 	for (i = 0; i < nums; i++) {
 		hda_nid_t dac = spec->dac_info[i].dac;
-		if (!dac)
+		if (!dac || (dac & DAC_SLAVE_FLAG))
 			continue;
 		switch (spec->dac_info[i].type) {
 		case AUTO_PIN_LINE_OUT:
@@ -3862,7 +3874,7 @@ static void cx_auto_parse_input(struct hda_codec *codec)
 	}
 	if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
 		cx_auto_check_auto_mic(codec);
-	if (imux->num_items > 1 && !spec->auto_mic) {
+	if (imux->num_items > 1) {
 		for (i = 1; i < imux->num_items; i++) {
 			if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
 				spec->adc_switching = 1;
@@ -4035,6 +4047,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
 		nid = spec->dac_info[i].dac;
 		if (!nid)
 			nid = spec->multiout.dac_nids[0];
+		else if (nid & DAC_SLAVE_FLAG)
+			nid &= ~DAC_SLAVE_FLAG;
 		select_connection(codec, spec->dac_info[i].pin, nid);
 	}
 	if (spec->auto_mute) {
@@ -4167,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
 			     hda_nid_t pin, const char *name, int idx)
 {
 	unsigned int caps;
-	caps = query_amp_caps(codec, dac, HDA_OUTPUT);
-	if (caps & AC_AMPCAP_NUM_STEPS)
-		return cx_auto_add_pb_volume(codec, dac, name, idx);
+	if (dac && !(dac & DAC_SLAVE_FLAG)) {
+		caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+		if (caps & AC_AMPCAP_NUM_STEPS)
+			return cx_auto_add_pb_volume(codec, dac, name, idx);
+	}
 	caps = query_amp_caps(codec, pin, HDA_OUTPUT);
 	if (caps & AC_AMPCAP_NUM_STEPS)
 		return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4191,8 +4207,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 	for (i = 0; i < spec->dac_info_filled; i++) {
 		const char *label;
 		int idx, type;
-		if (!spec->dac_info[i].dac)
-			continue;
+		hda_nid_t dac = spec->dac_info[i].dac;
 		type = spec->dac_info[i].type;
 		if (type == AUTO_PIN_LINE_OUT)
 			type = spec->autocfg.line_out_type;
@@ -4211,7 +4226,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 			idx = num_spk++;
 			break;
 		}
-		err = try_add_pb_volume(codec, spec->dac_info[i].dac,
+		err = try_add_pb_volume(codec, dac,
 					spec->dac_info[i].pin,
 					label, idx);
 		if (err < 0)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9a1aa09f47fe..0503c999e7d3 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -168,7 +168,7 @@ struct alc_spec {
 	unsigned int auto_mic_valid_imux:1;	/* valid imux for auto-mic */
 	unsigned int automute:1;	/* HP automute enabled */
 	unsigned int detect_line:1;	/* Line-out detection enabled */
-	unsigned int automute_lines:1;	/* automute line-out as well */
+	unsigned int automute_lines:1;	/* automute line-out as well; NOP when automute_hp_lo isn't set */
 	unsigned int automute_hp_lo:1;	/* both HP and LO available */
 
 	/* other flags */
@@ -551,7 +551,7 @@ static void update_speakers(struct hda_codec *codec)
 	if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] ||
 	    spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0])
 		return;
-	if (!spec->automute_lines || !spec->automute)
+	if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines))
 		on = 0;
 	else
 		on = spec->jack_present;
@@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (!spec->automute)
-		return;
 	spec->jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
 			     spec->autocfg.hp_pins);
+	if (!spec->automute)
+		return;
 	update_speakers(codec);
 }
 
@@ -578,11 +578,11 @@ static void alc_line_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (!spec->automute || !spec->detect_line)
-		return;
 	spec->line_jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
 			     spec->autocfg.line_out_pins);
+	if (!spec->automute || !spec->detect_line)
+		return;
 	update_speakers(codec);
 }
 
@@ -803,7 +803,7 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol,
 	unsigned int val;
 	if (!spec->automute)
 		val = 0;
-	else if (!spec->automute_lines)
+	else if (!spec->automute_hp_lo || !spec->automute_lines)
 		val = 1;
 	else
 		val = 2;
@@ -824,7 +824,8 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol,
 		spec->automute = 0;
 		break;
 	case 1:
-		if (spec->automute && !spec->automute_lines)
+		if (spec->automute &&
+		    (!spec->automute_hp_lo || !spec->automute_lines))
 			return 0;
 		spec->automute = 1;
 		spec->automute_lines = 0;
@@ -1784,6 +1785,7 @@ static const char * const alc_slave_vols[] = {
 	"Speaker Playback Volume",
 	"Mono Playback Volume",
 	"Line-Out Playback Volume",
+	"PCM Playback Volume",
 	NULL,
 };
 
@@ -1798,6 +1800,7 @@ static const char * const alc_slave_sws[] = {
 	"Mono Playback Switch",
 	"IEC958 Playback Switch",
 	"Line-Out Playback Switch",
+	"PCM Playback Switch",
 	NULL,
 };
 
@@ -3081,16 +3084,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec)
 static void alc_auto_init_extra_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	hda_nid_t pin;
+	hda_nid_t pin, dac;
 
 	pin = spec->autocfg.hp_pins[0];
-	if (pin)
-		alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
-						  spec->multiout.hp_nid);
+	if (pin) {
+		dac = spec->multiout.hp_nid;
+		if (!dac)
+			dac = spec->multiout.dac_nids[0];
+		alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
+	}
 	pin = spec->autocfg.speaker_pins[0];
-	if (pin)
-		alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
-					spec->multiout.extra_out_nid[0]);
+	if (pin) {
+		dac = spec->multiout.extra_out_nid[0];
+		if (!dac)
+			dac = spec->multiout.dac_nids[0];
+		alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
+	}
 }
 
 /*
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index aa376b59c006..1b7c11432aa7 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -673,6 +673,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
 static int stac_vrefout_set(struct hda_codec *codec,
 					hda_nid_t nid, unsigned int new_vref)
 {
@@ -696,6 +697,7 @@ static int stac_vrefout_set(struct hda_codec *codec,
 
 	return 1;
 }
+#endif
 
 static unsigned int stac92xx_vref_set(struct hda_codec *codec,
 					hda_nid_t nid, unsigned int new_vref)
@@ -6571,6 +6573,7 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
 	{ .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
+	{ .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c
index d6651c033cb7..5956584ea3a4 100644
--- a/sound/soc/blackfin/bf5xx-ad193x.c
+++ b/sound/soc/blackfin/bf5xx-ad193x.c
@@ -56,7 +56,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream,
 
 	switch (params_rate(params)) {
 	case 48000:
-		clk = 12288000;
+		clk = 24576000;
 		break;
 	}
 
@@ -103,7 +103,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = {
 		.cpu_dai_name = "bfin-tdm.0",
 		.codec_dai_name ="ad193x-hifi",
 		.platform_name = "bfin-tdm-pcm-audio",
-		.codec_name = "ad193x.5",
+		.codec_name = "spi0.5",
 		.ops = &bf5xx_ad193x_ops,
 	},
 	{
@@ -112,7 +112,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = {
 		.cpu_dai_name = "bfin-tdm.1",
 		.codec_dai_name ="ad193x-hifi",
 		.platform_name = "bfin-tdm-pcm-audio",
-		.codec_name = "ad193x.5",
+		.codec_name = "spi0.5",
 		.ops = &bf5xx_ad193x_ops,
 	},
 };
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 2374ca5ffe68..eedb6f5e5823 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -27,11 +27,6 @@ struct ad193x_priv {
 	int sysclk;
 };
 
-/* ad193x register cache & default register settings */
-static const u8 ad193x_reg[AD193X_NUM_REGS] = {
-	0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0,
-};
-
 /*
  * AD193X volume/mute/de-emphasis etc. controls
  */
@@ -307,7 +302,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
 	snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
 
 	reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
-	reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
+	reg = (reg & (~AD193X_DAC_WORD_LEN_MASK))
+		| (word_len << AD193X_DAC_WORD_LEN_SHFT);
 	snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
 
 	reg = snd_soc_read(codec, AD193X_ADC_CTRL1);
@@ -389,9 +385,6 @@ static int ad193x_probe(struct snd_soc_codec *codec)
 
 static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
 	.probe = 	ad193x_probe,
-	.reg_cache_default = ad193x_reg,
-	.reg_cache_size = AD193X_NUM_REGS,
-	.reg_word_size = sizeof(u16),
 };
 
 #if defined(CONFIG_SPI_MASTER)
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index 9747b5497877..cccc2e8e5fbd 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -34,7 +34,8 @@
 #define AD193X_DAC_LEFT_HIGH    (1 << 3)
 #define AD193X_DAC_BCLK_INV     (1 << 7)
 #define AD193X_DAC_CTRL2        0x804
-#define AD193X_DAC_WORD_LEN_MASK	0xC
+#define AD193X_DAC_WORD_LEN_SHFT        3
+#define AD193X_DAC_WORD_LEN_MASK        0x18
 #define AD193X_DAC_MASTER_MUTE  1
 #define AD193X_DAC_CHNL_MUTE    0x805
 #define AD193X_DACL1_MUTE       0
@@ -63,7 +64,7 @@
 #define AD193X_ADC_CTRL1        0x80f
 #define AD193X_ADC_SERFMT_MASK		0x60
 #define AD193X_ADC_SERFMT_STEREO	(0 << 5)
-#define AD193X_ADC_SERFMT_TDM		(1 << 2)
+#define AD193X_ADC_SERFMT_TDM		(1 << 5)
 #define AD193X_ADC_SERFMT_AUX		(2 << 5)
 #define AD193X_ADC_WORD_LEN_MASK	0x3
 #define AD193X_ADC_CTRL2        0x810
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 409d89d1f34c..fbd7eb9e61ce 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -857,6 +857,7 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c,
 	ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1);
 	if (ret != 0) {
 		dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret);
+		kfree(sta32x);
 		return ret;
 	}
 
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 38f38fddd190..d0003cc3bcd6 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -778,11 +778,19 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi)
 	return 0;
 }
 
+static const struct spi_device_id wm8750_spi_ids[] = {
+	{ "wm8750", 0 },
+	{ "wm8987", 0 },
+	{ },
+};
+MODULE_DEVICE_TABLE(spi, wm8750_spi_ids);
+
 static struct spi_driver wm8750_spi_driver = {
 	.driver = {
 		.name	= "wm8750-codec",
 		.owner	= THIS_MODULE,
 	},
+	.id_table	= wm8750_spi_ids,
 	.probe		= wm8750_spi_probe,
 	.remove		= __devexit_p(wm8750_spi_remove),
 };
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 43e3d760766f..4ad8ebd290e3 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -2046,8 +2046,13 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 /* power down chip */
 static int wm8903_remove(struct snd_soc_codec *codec)
 {
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
+
 	wm8903_free_gpio(codec);
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	if (wm8903->irq)
+		free_irq(wm8903->irq, codec);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 60d740ebeb5b..1725550c293e 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2221,6 +2221,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w,
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
 		if (fll) {
+			try_wait_for_completion(&wm8962->fll_lock);
+
 			snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
 					    WM8962_FLL_ENA, WM8962_FLL_ENA);
 			if (wm8962->irq) {
@@ -2927,10 +2929,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
 					    WM8962_BIAS_ENA | 0x180);
 
 			msleep(5);
-
-			snd_soc_update_bits(codec, WM8962_CLOCKING2,
-					    WM8962_CLKREG_OVD,
-					    WM8962_CLKREG_OVD);
 		}
 
 		/* VMID 2*250k */
@@ -3288,6 +3286,8 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda);
 	snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n);
 
+	try_wait_for_completion(&wm8962->fll_lock);
+
 	snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
 			    WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK |
 			    WM8962_FLL_ENA, fll1);
@@ -3868,6 +3868,10 @@ static int wm8962_probe(struct snd_soc_codec *codec)
 	 */
 	snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0);
 
+	/* Ensure we have soft control over all registers */
+	snd_soc_update_bits(codec, WM8962_CLOCKING2,
+			    WM8962_CLKREG_OVD, WM8962_CLKREG_OVD);
+
 	regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
 
 	if (pdata) {
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 09e680ae88b2..b393f9fac97a 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2981,6 +2981,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 			wm8994->hubs.dcs_readback_mode = 1;
 			break;
 		}
+		break;
 
 	case WM8958:
 		wm8994->hubs.dcs_readback_mode = 1;
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index ab8e9d1aaff0..0cdb9d105671 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -420,7 +420,7 @@ static const char *sidetone_hpf_text[] = {
 };
 
 static const struct soc_enum sidetone_hpf =
-	SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 6, sidetone_hpf_text);
+	SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text);
 
 static const char *hpf_mode_text[] = {
 	"HiFi", "Custom", "Voice"
@@ -988,15 +988,10 @@ SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0),
 SND_SOC_DAPM_PGA("IN1L PGA", WM8996_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
 SND_SOC_DAPM_PGA("IN1R PGA", WM8996_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
 
-SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-
-SND_SOC_DAPM_PGA("IN1L", WM8996_POWER_MANAGEMENT_7, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN1R", WM8996_POWER_MANAGEMENT_7, 3, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2L", WM8996_POWER_MANAGEMENT_7, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2R", WM8996_POWER_MANAGEMENT_7, 7, 0, NULL, 0),
+SND_SOC_DAPM_MUX("IN1L Mux", WM8996_POWER_MANAGEMENT_7, 2, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN1R Mux", WM8996_POWER_MANAGEMENT_7, 3, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN2L Mux", WM8996_POWER_MANAGEMENT_7, 6, 0, &in2_mux),
+SND_SOC_DAPM_MUX("IN2R Mux", WM8996_POWER_MANAGEMENT_7, 7, 0, &in2_mux),
 
 SND_SOC_DAPM_SUPPLY("DMIC2", WM8996_POWER_MANAGEMENT_7, 9, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("DMIC1", WM8996_POWER_MANAGEMENT_7, 8, 0, NULL, 0),
@@ -1213,6 +1208,16 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = {
 	{ "AIF2RX0", NULL, "AIFCLK" },
 	{ "AIF2RX1", NULL, "AIFCLK" },
 
+	{ "AIF1TX0", NULL, "AIFCLK" },
+	{ "AIF1TX1", NULL, "AIFCLK" },
+	{ "AIF1TX2", NULL, "AIFCLK" },
+	{ "AIF1TX3", NULL, "AIFCLK" },
+	{ "AIF1TX4", NULL, "AIFCLK" },
+	{ "AIF1TX5", NULL, "AIFCLK" },
+
+	{ "AIF2TX0", NULL, "AIFCLK" },
+	{ "AIF2TX1", NULL, "AIFCLK" },
+
 	{ "DSP1RXL", NULL, "SYSDSPCLK" },
 	{ "DSP1RXR", NULL, "SYSDSPCLK" },
 	{ "DSP2RXL", NULL, "SYSDSPCLK" },
@@ -2106,6 +2111,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 
 	snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda);
 
+	/* Clear any pending completions (eg, from failed startups) */
+	try_wait_for_completion(&wm8996->fll_lock);
+
 	snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1,
 			    WM8996_FLL_ENA, WM8996_FLL_ENA);
 
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index 56efa0c1c9a9..099614e16651 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -385,14 +385,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (!res) {
 		err = -ENODEV;
-		goto fail;
+		goto fail_free_info;
 	}
 
 	info->mem = request_mem_region(res->start, resource_size(res),
 				       pdev->name);
 	if (!info->mem) {
 		err = -EBUSY;
-		goto fail;
+		goto fail_free_info;
 	}
 
 	info->regs = ioremap(info->mem->start, resource_size(info->mem));
@@ -435,6 +435,7 @@ fail_unmap_mem:
 	iounmap(info->regs);
 fail_release_mem:
 	release_mem_region(info->mem->start, resource_size(info->mem));
+fail_free_info:
 	kfree(info);
 fail:
 	return err;
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 732208c8c0b4..cb50598338e9 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -879,10 +879,12 @@ static struct device_node *find_ssi_node(struct device_node *dma_channel_np)
 		 * assume that device_node pointers are a valid comparison.
 		 */
 		np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0);
+		of_node_put(np);
 		if (np == dma_channel_np)
 			return ssi_np;
 
 		np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0);
+		of_node_put(np);
 		if (np == dma_channel_np)
 			return ssi_np;
 	}
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index fd0dc46afc34..5c6c2457386e 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -369,7 +369,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = {
 	.pcm_free	= &psc_dma_free,
 };
 
-static int mpc5200_hpcd_probe(struct of_device *op)
+static int mpc5200_hpcd_probe(struct platform_device *op)
 {
 	phys_addr_t fifo;
 	struct psc_dma *psc_dma;
@@ -487,7 +487,7 @@ out_unmap:
 	return ret;
 }
 
-static int mpc5200_hpcd_remove(struct of_device *op)
+static int mpc5200_hpcd_remove(struct platform_device *op)
 {
 	struct psc_dma *psc_dma = dev_get_drvdata(&op->dev);
 
@@ -519,7 +519,7 @@ MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match);
 static struct platform_driver mpc5200_hpcd_of_driver = {
 	.probe		= mpc5200_hpcd_probe,
 	.remove		= mpc5200_hpcd_remove,
-	.dev = {
+	.driver = {
 		.owner		= THIS_MODULE,
 		.name		= "mpc5200-pcm-audio",
 		.of_match_table    = mpc5200_hpcd_match,
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index a19297959587..358f0baaf71b 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -345,8 +345,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 	}
 
 	machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL);
-	if (!machine_data)
-		return -ENOMEM;
+	if (!machine_data) {
+		ret = -ENOMEM;
+		goto error_alloc;
+	}
 
 	machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
 	machine_data->dai[0].ops = &mpc8610_hpcd_ops;
@@ -494,7 +496,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 	ret = platform_device_add(sound_device);
 	if (ret) {
 		dev_err(&pdev->dev, "platform device add failed\n");
-		goto error;
+		goto error_sound;
 	}
 	dev_set_drvdata(&pdev->dev, sound_device);
 
@@ -502,14 +504,12 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 
 	return 0;
 
+error_sound:
+	platform_device_unregister(sound_device);
 error:
-	of_node_put(codec_np);
-
-	if (sound_device)
-		platform_device_unregister(sound_device);
-
 	kfree(machine_data);
-
+error_alloc:
+	of_node_put(codec_np);
 	return ret;
 }
 
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 8fa4d5f8eda1..fcb862eb0c73 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -297,8 +297,10 @@ static int get_dma_channel(struct device_node *ssi_np,
 	 * dai->platform name should already point to an allocated buffer.
 	 */
 	ret = of_address_to_resource(dma_channel_np, 0, &res);
-	if (ret)
+	if (ret) {
+		of_node_put(dma_channel_np);
 		return ret;
+	}
 	snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
 		 (unsigned long long) res.start, dma_channel_np->name);
 
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index 309c59e6fb6c..7945625e0e08 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -240,7 +240,6 @@ static int ssi_irq = 0;
 
 static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_card *card = rtd->card->snd_card;
 	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	int ret;
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index a33fc51f363b..d0bcf3fcea01 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	if (!priv->mem) {
 		dev_err(&pdev->dev, "request_mem_region failed\n");
 		err = -EBUSY;
-		goto error;
+		goto err_alloc;
 	}
 
 	priv->io = ioremap(priv->mem->start, SZ_16K);
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 30fe0d0efe1c..0aa475f92efa 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -514,7 +514,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
 	}
 
 	/* Set codec bias level */
-	ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
+	ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
 
 	/* Add hook switch - can be used to control the codec from userspace
 	 * even if line discipline fails */
@@ -649,7 +649,9 @@ static void __exit ams_delta_module_exit(void)
 			ams_delta_hook_switch_gpios);
 
 	/* Keep modem power on */
-	ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
+	ams_delta_set_bias_level(&ams_delta_audio_card,
+				 &ams_delta_audio_card.rtd[0].codec->dapm,
+				 SND_SOC_BIAS_STANDBY);
 
 	platform_device_unregister(cx20442_platform_device);
 	platform_device_unregister(ams_delta_audio_platform_device);
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 83d213bfd3d1..62e292f49313 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -3,7 +3,7 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
@@ -402,6 +402,6 @@ static void __exit n810_soc_exit(void)
 module_init(n810_soc_init);
 module_exit(n810_soc_exit);
 
-MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
 MODULE_DESCRIPTION("ALSA SoC Nokia N810");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 07b772357244..ebcc2d4d2b18 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -3,7 +3,7 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
  *          Peter Ujfalusi <peter.ujfalusi@ti.com>
  *
  * This program is free software; you can redistribute it and/or
@@ -780,6 +780,6 @@ static void __exit snd_omap_mcbsp_exit(void)
 }
 module_exit(snd_omap_mcbsp_exit);
 
-MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
 MODULE_DESCRIPTION("OMAP I2S SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index 9a7dedd6f5a9..65cde9d3807b 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -3,7 +3,7 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
  *          Peter Ujfalusi <peter.ujfalusi@ti.com>
  *
  * This program is free software; you can redistribute it and/or
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index b2f5751edae3..9b5c88ac35b9 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -3,7 +3,7 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
  *          Peter Ujfalusi <peter.ujfalusi@ti.com>
  *
  * This program is free software; you can redistribute it and/or
@@ -436,6 +436,6 @@ static void __exit snd_omap_pcm_exit(void)
 }
 module_exit(snd_omap_pcm_exit);
 
-MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
 MODULE_DESCRIPTION("OMAP PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index a0ed1dbb52d6..f95fe3064172 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -3,7 +3,7 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
  *          Peter Ujfalusi <peter.ujfalusi@ti.com>
  *
  * This program is free software; you can redistribute it and/or
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 0aae998b6540..893300a53bab 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -5,7 +5,7 @@
  *
  * Contact: Peter Ujfalusi <peter.ujfalusi@ti.com>
  *          Eduardo Valentin <eduardo.valentin@nokia.com>
- *          Jarkko Nikula <jhnikula@gmail.com>
+ *          Jarkko Nikula <jarkko.nikula@bitmer.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index b99091fc34eb..65f980ef2870 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -185,6 +185,7 @@ config SND_SOC_SPEYSIDE
 	select SND_SAMSUNG_I2S
 	select SND_SOC_WM8996
 	select SND_SOC_WM9081
+	select SND_SOC_WM1250_EV1
 
 config SND_SOC_SPEYSIDE_WM8962
 	tristate "Audio support for Wolfson Speyside with WM8962"
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 9eb3b12eb72f..8509d3c4366e 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -1,5 +1,6 @@
 # S3c24XX Platform Support
 snd-soc-s3c24xx-objs := dma.o
+snd-soc-idma-objs := idma.o
 snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
 snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o
 snd-soc-ac97-objs := ac97.o
@@ -16,6 +17,7 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o
 obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o
 obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o
 obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o
+obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o
 
 # S3C24XX Machine Support
 snd-soc-jive-wm8750-objs := jive_wm8750.o
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index 241f55d00660..c6c65892294e 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -13,6 +13,7 @@
  *
  */
 
+#include <linux/types.h>
 #include <linux/gpio.h>
 
 #include <sound/soc.h>
diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c
new file mode 100644
index 000000000000..ebde0740ab19
--- /dev/null
+++ b/sound/soc/samsung/idma.c
@@ -0,0 +1,453 @@
+/*
+ * sound/soc/samsung/idma.c
+ *
+ * Copyright (c) 2011 Samsung Electronics Co., Ltd.
+ *		http://www.samsung.com
+ *
+ * I2S0's Internal DMA driver
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "i2s.h"
+#include "idma.h"
+#include "dma.h"
+#include "i2s-regs.h"
+
+#define ST_RUNNING		(1<<0)
+#define ST_OPENED		(1<<1)
+
+static const struct snd_pcm_hardware idma_hardware = {
+	.info = SNDRV_PCM_INFO_INTERLEAVED |
+		    SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		    SNDRV_PCM_INFO_MMAP |
+		    SNDRV_PCM_INFO_MMAP_VALID |
+		    SNDRV_PCM_INFO_PAUSE |
+		    SNDRV_PCM_INFO_RESUME,
+	.formats = SNDRV_PCM_FMTBIT_S16_LE |
+		    SNDRV_PCM_FMTBIT_U16_LE |
+		    SNDRV_PCM_FMTBIT_S24_LE |
+		    SNDRV_PCM_FMTBIT_U24_LE |
+		    SNDRV_PCM_FMTBIT_U8 |
+		    SNDRV_PCM_FMTBIT_S8,
+	.channels_min = 2,
+	.channels_max = 2,
+	.buffer_bytes_max = MAX_IDMA_BUFFER,
+	.period_bytes_min = 128,
+	.period_bytes_max = MAX_IDMA_PERIOD,
+	.periods_min = 1,
+	.periods_max = 2,
+};
+
+struct idma_ctrl {
+	spinlock_t	lock;
+	int		state;
+	dma_addr_t	start;
+	dma_addr_t	pos;
+	dma_addr_t	end;
+	dma_addr_t	period;
+	dma_addr_t	periodsz;
+	void		*token;
+	void		(*cb)(void *dt, int bytes_xfer);
+};
+
+static struct idma_info {
+	spinlock_t	lock;
+	void		 __iomem  *regs;
+	dma_addr_t	lp_tx_addr;
+} idma;
+
+static void idma_getpos(dma_addr_t *src)
+{
+	*src = idma.lp_tx_addr +
+		(readl(idma.regs + I2STRNCNT) & 0xffffff) * 4;
+}
+
+static int idma_enqueue(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct idma_ctrl *prtd = substream->runtime->private_data;
+	u32 val;
+
+	spin_lock(&prtd->lock);
+	prtd->token = (void *) substream;
+	spin_unlock(&prtd->lock);
+
+	/* Internal DMA Level0 Interrupt Address */
+	val = idma.lp_tx_addr + prtd->periodsz;
+	writel(val, idma.regs + I2SLVL0ADDR);
+
+	/* Start address0 of I2S internal DMA operation. */
+	val = idma.lp_tx_addr;
+	writel(val, idma.regs + I2SSTR0);
+
+	/*
+	 * Transfer block size for I2S internal DMA.
+	 * Should decide transfer size before start dma operation
+	 */
+	val = readl(idma.regs + I2SSIZE);
+	val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT);
+	val |= (((runtime->dma_bytes >> 2) &
+			I2SSIZE_TRNMSK) << I2SSIZE_SHIFT);
+	writel(val, idma.regs + I2SSIZE);
+
+	val = readl(idma.regs + I2SAHB);
+	val |= AHB_INTENLVL0;
+	writel(val, idma.regs + I2SAHB);
+
+	return 0;
+}
+
+static void idma_setcallbk(struct snd_pcm_substream *substream,
+				void (*cb)(void *, int))
+{
+	struct idma_ctrl *prtd = substream->runtime->private_data;
+
+	spin_lock(&prtd->lock);
+	prtd->cb = cb;
+	spin_unlock(&prtd->lock);
+}
+
+static void idma_control(int op)
+{
+	u32 val = readl(idma.regs + I2SAHB);
+
+	spin_lock(&idma.lock);
+
+	switch (op) {
+	case LPAM_DMA_START:
+		val |= (AHB_INTENLVL0 | AHB_DMAEN);
+		break;
+	case LPAM_DMA_STOP:
+		val &= ~(AHB_INTENLVL0 | AHB_DMAEN);
+		break;
+	default:
+		spin_unlock(&idma.lock);
+		return;
+	}
+
+	writel(val, idma.regs + I2SAHB);
+	spin_unlock(&idma.lock);
+}
+
+static void idma_done(void *id, int bytes_xfer)
+{
+	struct snd_pcm_substream *substream = id;
+	struct idma_ctrl *prtd = substream->runtime->private_data;
+
+	if (prtd && (prtd->state & ST_RUNNING))
+		snd_pcm_period_elapsed(substream);
+}
+
+static int idma_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct idma_ctrl *prtd = substream->runtime->private_data;
+	u32 mod = readl(idma.regs + I2SMOD);
+	u32 ahb = readl(idma.regs + I2SAHB);
+
+	ahb |= (AHB_DMARLD | AHB_INTMASK);
+	mod |= MOD_TXS_IDMA;
+	writel(ahb, idma.regs + I2SAHB);
+	writel(mod, idma.regs + I2SMOD);
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+	runtime->dma_bytes = params_buffer_bytes(params);
+
+	prtd->start = prtd->pos = runtime->dma_addr;
+	prtd->period = params_periods(params);
+	prtd->periodsz = params_period_bytes(params);
+	prtd->end = runtime->dma_addr + runtime->dma_bytes;
+
+	idma_setcallbk(substream, idma_done);
+
+	return 0;
+}
+
+static int idma_hw_free(struct snd_pcm_substream *substream)
+{
+	snd_pcm_set_runtime_buffer(substream, NULL);
+
+	return 0;
+}
+
+static int idma_prepare(struct snd_pcm_substream *substream)
+{
+	struct idma_ctrl *prtd = substream->runtime->private_data;
+
+	prtd->pos = prtd->start;
+
+	/* flush the DMA channel */
+	idma_control(LPAM_DMA_STOP);
+	idma_enqueue(substream);
+
+	return 0;
+}
+
+static int idma_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct idma_ctrl *prtd = substream->runtime->private_data;
+	int ret = 0;
+
+	spin_lock(&prtd->lock);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		prtd->state |= ST_RUNNING;
+		idma_control(LPAM_DMA_START);
+		break;
+
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		prtd->state &= ~ST_RUNNING;
+		idma_control(LPAM_DMA_STOP);
+		break;
+
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	spin_unlock(&prtd->lock);
+
+	return ret;
+}
+
+static snd_pcm_uframes_t
+	idma_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct idma_ctrl *prtd = runtime->private_data;
+	dma_addr_t src;
+	unsigned long res;
+
+	spin_lock(&prtd->lock);
+
+	idma_getpos(&src);
+	res = src - prtd->start;
+
+	spin_unlock(&prtd->lock);
+
+	return bytes_to_frames(substream->runtime, res);
+}
+
+static int idma_mmap(struct snd_pcm_substream *substream,
+	struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned long size, offset;
+	int ret;
+
+	/* From snd_pcm_lib_mmap_iomem */
+	vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+	vma->vm_flags |= VM_IO;
+	size = vma->vm_end - vma->vm_start;
+	offset = vma->vm_pgoff << PAGE_SHIFT;
+	ret = io_remap_pfn_range(vma, vma->vm_start,
+			(runtime->dma_addr + offset) >> PAGE_SHIFT,
+			size, vma->vm_page_prot);
+
+	return ret;
+}
+
+static irqreturn_t iis_irq(int irqno, void *dev_id)
+{
+	struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id;
+	u32 iiscon, iisahb, val, addr;
+
+	iisahb  = readl(idma.regs + I2SAHB);
+	iiscon  = readl(idma.regs + I2SCON);
+
+	val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0;
+
+	if (val) {
+		iisahb |= val;
+		writel(iisahb, idma.regs + I2SAHB);
+
+		addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr;
+		addr += prtd->periodsz;
+		addr %= (prtd->end - prtd->start);
+		addr += idma.lp_tx_addr;
+
+		writel(addr, idma.regs + I2SLVL0ADDR);
+
+		if (prtd->cb)
+			prtd->cb(prtd->token, prtd->period);
+	}
+
+	return IRQ_HANDLED;
+}
+
+static int idma_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct idma_ctrl *prtd;
+	int ret;
+
+	snd_soc_set_runtime_hwparams(substream, &idma_hardware);
+
+	prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL);
+	if (prtd == NULL)
+		return -ENOMEM;
+
+	ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd);
+	if (ret < 0) {
+		pr_err("fail to claim i2s irq , ret = %d\n", ret);
+		kfree(prtd);
+		return ret;
+	}
+
+	spin_lock_init(&prtd->lock);
+
+	runtime->private_data = prtd;
+
+	return 0;
+}
+
+static int idma_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct idma_ctrl *prtd = runtime->private_data;
+
+	free_irq(IRQ_I2S0, prtd);
+
+	if (!prtd)
+		pr_err("idma_close called with prtd == NULL\n");
+
+	kfree(prtd);
+
+	return 0;
+}
+
+static struct snd_pcm_ops idma_ops = {
+	.open		= idma_open,
+	.close		= idma_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.trigger	= idma_trigger,
+	.pointer	= idma_pointer,
+	.mmap		= idma_mmap,
+	.hw_params	= idma_hw_params,
+	.hw_free	= idma_hw_free,
+	.prepare	= idma_prepare,
+};
+
+static void idma_free(struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+
+	substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+	if (!substream)
+		return;
+
+	buf = &substream->dma_buffer;
+	if (!buf->area)
+		return;
+
+	iounmap(buf->area);
+
+	buf->area = NULL;
+	buf->addr = 0;
+}
+
+static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream)
+{
+	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+
+	buf->dev.dev = pcm->card->dev;
+	buf->private_data = NULL;
+
+	/* Assign PCM buffer pointers */
+	buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS;
+	buf->addr = idma.lp_tx_addr;
+	buf->bytes = idma_hardware.buffer_bytes_max;
+	buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes);
+
+	return 0;
+}
+
+static u64 idma_mask = DMA_BIT_MASK(32);
+
+static int idma_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct snd_pcm *pcm = rtd->pcm;
+	int ret = 0;
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &idma_mask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+	if (dai->driver->playback.channels_min)
+		ret = preallocate_idma_buffer(pcm,
+				SNDRV_PCM_STREAM_PLAYBACK);
+
+	return ret;
+}
+
+void idma_reg_addr_init(void *regs, dma_addr_t addr)
+{
+	spin_lock_init(&idma.lock);
+	idma.regs = regs;
+	idma.lp_tx_addr = addr;
+}
+
+struct snd_soc_platform_driver asoc_idma_platform = {
+	.ops = &idma_ops,
+	.pcm_new = idma_new,
+	.pcm_free = idma_free,
+};
+
+static int __devinit asoc_idma_platform_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform);
+}
+
+static int __devexit asoc_idma_platform_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver asoc_idma_driver = {
+	.driver = {
+		.name = "samsung-idma",
+		.owner = THIS_MODULE,
+	},
+
+	.probe = asoc_idma_platform_probe,
+	.remove = __devexit_p(asoc_idma_platform_remove),
+};
+
+static int __init asoc_idma_init(void)
+{
+	return platform_driver_register(&asoc_idma_driver);
+}
+module_init(asoc_idma_init);
+
+static void __exit asoc_idma_exit(void)
+{
+	platform_driver_unregister(&asoc_idma_driver);
+}
+module_exit(asoc_idma_exit);
+
+MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar@gmail.com>");
+MODULE_DESCRIPTION("Samsung ASoC IDMA Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h
new file mode 100644
index 000000000000..48273216166e
--- /dev/null
+++ b/sound/soc/samsung/idma.h
@@ -0,0 +1,26 @@
+/*
+ * sound/soc/samsung/idma.h
+ *
+ * Copyright (c) 2011 Samsung Electronics Co., Ltd
+ *		http://www.samsung.com
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#ifndef __SND_SOC_SAMSUNG_IDMA_H_
+#define __SND_SOC_SAMSUNG_IDMA_H_
+
+extern void idma_reg_addr_init(void *regs, dma_addr_t addr);
+
+/* dma_state */
+#define LPAM_DMA_STOP	0
+#define LPAM_DMA_START	1
+
+#define MAX_IDMA_PERIOD (128 * 1024)
+#define MAX_IDMA_BUFFER (160 * 1024)
+
+#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */
diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c
index 3b53ad54bc33..14eb6ea69e7c 100644
--- a/sound/soc/samsung/jive_wm8750.c
+++ b/sound/soc/samsung/jive_wm8750.c
@@ -131,7 +131,7 @@ static struct snd_soc_dai_link jive_dai = {
 	.cpu_dai_name	= "s3c2412-i2s",
 	.codec_dai_name = "wm8750-hifi",
 	.platform_name	= "samsung-audio",
-	.codec_name	= "wm8750-codec.0-0x1a",
+	.codec_name	= "wm8750-codec.0-001a",
 	.init		= jive_wm8750_init,
 	.ops		= &jive_ops,
 };
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 1e574a5d440d..bc8c1676459f 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -17,6 +17,7 @@
  *
  */
 
+#include <linux/types.h>
 #include <linux/gpio.h>
 
 #include <sound/soc.h>
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
index 8ac42bf82090..72535f2daaf2 100644
--- a/sound/soc/samsung/speyside_wm8962.c
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -23,6 +23,9 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
 	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
 	int ret;
 
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
 		if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
@@ -37,7 +40,7 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
 						     44100 * 256,
 						     SND_SOC_CLOCK_IN);
 			if (ret < 0) {
-				pr_err("Failed to set SYSCLK: %d\n");
+				pr_err("Failed to set SYSCLK: %d\n", ret);
 				return ret;
 			}
 		}
@@ -57,6 +60,9 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card,
 	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
 	int ret;
 
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
 	switch (level) {
 	case SND_SOC_BIAS_STANDBY:
 		ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index d9f8aded51f3..20b7f3b003a3 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -203,14 +203,14 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec)
 		rbnode = rb_entry(node, struct snd_soc_rbtree_node, node);
 		for (i = 0; i < rbnode->blklen; ++i) {
 			regtmp = rbnode->base_reg + i;
-			WARN_ON(codec->writable_register &&
-				codec->writable_register(codec, regtmp));
 			val = snd_soc_rbtree_get_register(rbnode, i);
 			def = snd_soc_get_cache_val(codec->reg_def_copy, i,
 						    rbnode->word_size);
 			if (val == def)
 				continue;
 
+			WARN_ON(!snd_soc_codec_writable_register(codec, regtmp));
+
 			codec->cache_bypass = 1;
 			ret = snd_soc_write(codec, regtmp, val);
 			codec->cache_bypass = 0;
@@ -563,8 +563,7 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec)
 
 	lzo_blocks = codec->reg_cache;
 	for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) {
-		WARN_ON(codec->writable_register &&
-			codec->writable_register(codec, i));
+		WARN_ON(!snd_soc_codec_writable_register(codec, i));
 		ret = snd_soc_cache_read(codec, i, &val);
 		if (ret)
 			return ret;
@@ -823,8 +822,6 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
 
 	codec_drv = codec->driver;
 	for (i = 0; i < codec_drv->reg_cache_size; ++i) {
-		WARN_ON(codec->writable_register &&
-			codec->writable_register(codec, i));
 		ret = snd_soc_cache_read(codec, i, &val);
 		if (ret)
 			return ret;
@@ -832,6 +829,9 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
 			if (snd_soc_get_cache_val(codec->reg_def_copy,
 						  i, codec_drv->reg_word_size) == val)
 				continue;
+
+		WARN_ON(!snd_soc_codec_writable_register(codec, i));
+
 		ret = snd_soc_write(codec, i, val);
 		if (ret)
 			return ret;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 83ad8ca27490..d2ef014af215 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1633,7 +1633,7 @@ int snd_soc_codec_readable_register(struct snd_soc_codec *codec,
 	if (codec->readable_register)
 		return codec->readable_register(codec, reg);
 	else
-		return 0;
+		return 1;
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_readable_register);
 
@@ -1651,7 +1651,7 @@ int snd_soc_codec_writable_register(struct snd_soc_codec *codec,
 	if (codec->writable_register)
 		return codec->writable_register(codec, reg);
 	else
-		return 0;
+		return 1;
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register);
 
@@ -1913,7 +1913,7 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 
 	if (prefix) {
 		name_len = strlen(long_name) + strlen(prefix) + 2;
-		name = kmalloc(name_len, GFP_ATOMIC);
+		name = kmalloc(name_len, GFP_KERNEL);
 		if (!name)
 			return NULL;
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7e15914b3633..d67c637557a7 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2763,7 +2763,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend);
 
 /**
  * snd_soc_dapm_free - free dapm resources
- * @card: SoC device
+ * @dapm: DAPM context
  *
  * Free all dapm widgets and resources.
  */
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index cca490c80589..a62f7dd4ba96 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -205,6 +205,25 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
 #define snd_soc_16_8_read_i2c NULL
 #endif
 
+#if defined(CONFIG_SPI_MASTER)
+static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec,
+		                          unsigned int r)
+{
+	struct spi_device *spi = codec->control_data;
+
+	const u16 reg = cpu_to_be16(r | 0x100);
+	u8 data;
+	int ret;
+
+	ret = spi_write_then_read(spi, &reg, 2, &data, 1);
+	if (ret < 0)
+		return 0;
+	return data;
+}
+#else
+#define snd_soc_16_8_read_spi NULL
+#endif
+
 static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
 			      unsigned int value)
 {
@@ -295,6 +314,7 @@ static struct {
 	int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
 	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
 	unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
+	unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int);
 } io_types[] = {
 	{
 		.addr_bits = 4, .data_bits = 12,
@@ -318,6 +338,7 @@ static struct {
 		.addr_bits = 16, .data_bits = 8,
 		.write = snd_soc_16_8_write,
 		.i2c_read = snd_soc_16_8_read_i2c,
+		.spi_read = snd_soc_16_8_read_spi,
 	},
 	{
 		.addr_bits = 16, .data_bits = 16,
@@ -383,6 +404,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 #ifdef CONFIG_SPI_MASTER
 		codec->hw_write = do_spi_write;
 #endif
+		if (io_types[i].spi_read)
+			codec->hw_read = io_types[i].spi_read;
 
 		codec->control_data = container_of(codec->dev,
 						   struct spi_device,
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7c17b98d5846..fa31d9c2abd8 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -105,7 +105,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 
 	snd_soc_dapm_sync(dapm);
 
-	snd_jack_report(jack->jack, status);
+	snd_jack_report(jack->jack, jack->status);
 
 out:
 	mutex_unlock(&codec->mutex);
@@ -327,7 +327,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
 					      IRQF_TRIGGER_FALLING,
 					      gpios[i].name,
 					      &gpios[i]);
-		if (ret)
+		if (ret < 0)
 			goto err;
 
 		if (gpios[i].wake) {
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b5759397afa3..2879c883eebc 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -290,6 +290,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	codec_dai->active--;
 	codec->active--;
 
+	if (!cpu_dai->active && !codec_dai->active)
+		rtd->rate = 0;
+
 	/* Muting the DAC suppresses artifacts caused during digital
 	 * shutdown, for example from stopping clocks.
 	 */
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index ff86e5e3db68..c7cfd96e991e 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -309,9 +309,14 @@ static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 
 static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 {
-	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
-	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+
+	substream = pcm->streams[stream].substream;
+	if (!substream)
+		return;
 
+	buf = &substream->dma_buffer;
 	if (!buf->area)
 		return;
 
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index a42e9ac30f28..be27f1d229af 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -56,6 +56,7 @@
 #define GPIO_HP_MUTE    BIT(1)
 #define GPIO_INT_MIC_EN BIT(2)
 #define GPIO_EXT_MIC_EN BIT(3)
+#define GPIO_HP_DET     BIT(4)
 
 struct tegra_wm8903 {
 	struct tegra_asoc_utils_data util_data;
@@ -304,6 +305,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 		snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack,
 					1,
 					&tegra_wm8903_hp_jack_gpio);
+		machine->gpio_requested |= GPIO_HP_DET;
 	}
 
 	snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
@@ -317,7 +319,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 	snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
 
 	/* FIXME: Calculate automatically based on DAPM routes? */
-	if (!machine_is_harmony() && !machine_is_ventana())
+	if (!machine_is_harmony())
 		snd_soc_dapm_nc_pin(dapm, "IN1L");
 	if (!machine_is_seaboard() && !machine_is_aebl())
 		snd_soc_dapm_nc_pin(dapm, "IN1R");
@@ -393,7 +395,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
-	if (machine_is_harmony() || machine_is_ventana()) {
+	if (machine_is_harmony()) {
 		card->dapm_routes = harmony_audio_map;
 		card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
 	} else if (machine_is_seaboard()) {
@@ -429,10 +431,10 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 	struct tegra_wm8903_platform_data *pdata = machine->pdata;
 
-	snd_soc_unregister_card(card);
-
-	tegra_asoc_utils_fini(&machine->util_data);
-
+	if (machine->gpio_requested & GPIO_HP_DET)
+		snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack,
+					1,
+					&tegra_wm8903_hp_jack_gpio);
 	if (machine->gpio_requested & GPIO_EXT_MIC_EN)
 		gpio_free(pdata->gpio_ext_mic_en);
 	if (machine->gpio_requested & GPIO_INT_MIC_EN)
@@ -441,6 +443,11 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
 		gpio_free(pdata->gpio_hp_mute);
 	if (machine->gpio_requested & GPIO_SPKR_EN)
 		gpio_free(pdata->gpio_spkr_en);
+	machine->gpio_requested = 0;
+
+	snd_soc_unregister_card(card);
+
+	tegra_asoc_utils_fini(&machine->util_data);
 
 	kfree(machine);
 
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index d0d493ca28ae..2cf87f5afed4 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -139,8 +139,12 @@ static void stream_stop(struct snd_usb_caiaqdev *dev)
 
 	for (i = 0; i < N_URBS; i++) {
 		usb_kill_urb(dev->data_urbs_in[i]);
-		usb_kill_urb(dev->data_urbs_out[i]);
+
+		if (test_bit(i, &dev->outurb_active_mask))
+			usb_kill_urb(dev->data_urbs_out[i]);
 	}
+
+	dev->outurb_active_mask = 0;
 }
 
 static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream)
@@ -612,8 +616,9 @@ static void read_completed(struct urb *urb)
 {
 	struct snd_usb_caiaq_cb_info *info = urb->context;
 	struct snd_usb_caiaqdev *dev;
-	struct urb *out;
-	int frame, len, send_it = 0, outframe = 0;
+	struct urb *out = NULL;
+	int i, frame, len, send_it = 0, outframe = 0;
+	size_t offset = 0;
 
 	if (urb->status || !info)
 		return;
@@ -623,7 +628,17 @@ static void read_completed(struct urb *urb)
 	if (!dev->streaming)
 		return;
 
-	out = dev->data_urbs_out[info->index];
+	/* find an unused output urb that is unused */
+	for (i = 0; i < N_URBS; i++)
+		if (test_and_set_bit(i, &dev->outurb_active_mask) == 0) {
+			out = dev->data_urbs_out[i];
+			break;
+		}
+
+	if (!out) {
+		log("Unable to find an output urb to use\n");
+		goto requeue;
+	}
 
 	/* read the recently received packet and send back one which has
 	 * the same layout */
@@ -634,7 +649,8 @@ static void read_completed(struct urb *urb)
 		len = urb->iso_frame_desc[outframe].actual_length;
 		out->iso_frame_desc[outframe].length = len;
 		out->iso_frame_desc[outframe].actual_length = 0;
-		out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame;
+		out->iso_frame_desc[outframe].offset = offset;
+		offset += len;
 
 		if (len > 0) {
 			spin_lock(&dev->spinlock);
@@ -650,11 +666,15 @@ static void read_completed(struct urb *urb)
 	}
 
 	if (send_it) {
-		out->number_of_packets = FRAMES_PER_URB;
+		out->number_of_packets = outframe;
 		out->transfer_flags = URB_ISO_ASAP;
 		usb_submit_urb(out, GFP_ATOMIC);
+	} else {
+		struct snd_usb_caiaq_cb_info *oinfo = out->context;
+		clear_bit(oinfo->index, &dev->outurb_active_mask);
 	}
 
+requeue:
 	/* re-submit inbound urb */
 	for (frame = 0; frame < FRAMES_PER_URB; frame++) {
 		urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame;
@@ -676,6 +696,8 @@ static void write_completed(struct urb *urb)
 		dev->output_running = 1;
 		wake_up(&dev->prepare_wait_queue);
 	}
+
+	clear_bit(info->index, &dev->outurb_active_mask);
 }
 
 static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret)
@@ -827,6 +849,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
 	if (!dev->data_cb_info)
 		return -ENOMEM;
 
+	dev->outurb_active_mask = 0;
+	BUILD_BUG_ON(N_URBS > (sizeof(dev->outurb_active_mask) * 8));
+
 	for (i = 0; i < N_URBS; i++) {
 		dev->data_cb_info[i].dev = dev;
 		dev->data_cb_info[i].index = i;
diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h
index b2b310194ffa..3f9c6339ae90 100644
--- a/sound/usb/caiaq/device.h
+++ b/sound/usb/caiaq/device.h
@@ -96,6 +96,7 @@ struct snd_usb_caiaqdev {
 	int input_panic, output_panic, warned;
 	char *audio_in_buf, *audio_out_buf;
 	unsigned int samplerates, bpp;
+	unsigned long outurb_active_mask;
 
 	struct snd_pcm_substream *sub_playback[MAX_STREAMS];
 	struct snd_pcm_substream *sub_capture[MAX_STREAMS];
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index c04d7c71ac88..cdd19d7fe500 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -152,6 +152,7 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p,
 	if (p && p->dB) {
 		cval->dBmin = p->dB->min;
 		cval->dBmax = p->dB->max;
+		cval->initialized = 1;
 	}
 }
 
@@ -1092,7 +1093,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 				" Switch" : " Volume");
 		if (control == UAC_FU_VOLUME) {
 			check_mapped_dB(map, cval);
-			if (cval->dBmin < cval->dBmax) {
+			if (cval->dBmin < cval->dBmax || !cval->initialized) {
 				kctl->tlv.c = mixer_vol_tlv;
 				kctl->vd[0].access |= 
 					SNDRV_CTL_ELEM_ACCESS_TLV_READ |
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 4d4f86552a23..a42e3ef3832d 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1707,6 +1707,40 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
+{
+	USB_DEVICE(0x0582, 0x0130),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		/* .vendor_name = "BOSS", */
+		/* .product_name = "MICRO BR-80", */
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_IGNORE_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 3,
+				.type = QUIRK_MIDI_FIXED_ENDPOINT,
+				.data = & (const struct snd_usb_midi_endpoint_info) {
+					.out_cables = 0x0001,
+					.in_cables  = 0x0001
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 /* Guillemot devices */
 {