Merge commit 'origin/master' into next

45 files changed, 652 insertions, 292 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 656e474..91acc9a 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -863,7 +863,6 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci)
 	struct snd_ac97 *ac97;
 	int ret;
 
-	writel(0, aaci->base + AC97_POWERDOWN);
 	/*
 	 * Assert AACIRESET for 2us
 	 */
@@ -1047,7 +1046,11 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
 
 	writel(0x1fff, aaci->base + AACI_INTCLR);
 	writel(aaci->maincr, aaci->base + AACI_MAINCR);
-
+	/*
+	 * Fix: ac97 read back fail errors by reading
+	 * from any arbitrary aaci register.
+	 */
+	readl(aaci->base + AACI_CSCH1);
 	ret = aaci_probe_ac97(aaci);
 	if (ret)
 		goto out;
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 7394365..5040c7b 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -1160,6 +1160,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri,
 {
 	struct snd_timer_user *tu = timeri->callback_data;
 	struct snd_timer_tread r1;
+	unsigned long flags;
 
 	if (event >= SNDRV_TIMER_EVENT_START &&
 	    event <= SNDRV_TIMER_EVENT_PAUSE)
@@ -1169,9 +1170,9 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri,
 	r1.event = event;
 	r1.tstamp = *tstamp;
 	r1.val = resolution;
-	spin_lock(&tu->qlock);
+	spin_lock_irqsave(&tu->qlock, flags);
 	snd_timer_user_append_to_tqueue(tu, &r1);
-	spin_unlock(&tu->qlock);
+	spin_unlock_irqrestore(&tu->qlock, flags);
 	kill_fasync(&tu->fasync, SIGIO, POLL_IN);
 	wake_up(&tu->qchange_sleep);
 }
diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c
index fff62cc..971a84a 100644
--- a/sound/i2c/other/ak4113.c
+++ b/sound/i2c/other/ak4113.c
@@ -70,7 +70,7 @@ static int snd_ak4113_dev_free(struct snd_device *device)
 }
 
 int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read,
-		ak4113_write_t *write, const unsigned char pgm[5],
+		ak4113_write_t *write, const unsigned char *pgm,
 		void *private_data, struct ak4113 **r_ak4113)
 {
 	struct ak4113 *chip;
diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c
index cafc3a7..ff18286f 100644
--- a/sound/isa/sb/es968.c
+++ b/sound/isa/sb/es968.c
@@ -93,7 +93,7 @@ static int __devinit snd_card_es968_pnp(int dev, struct snd_card_es968 *acard,
 		return err;
 	}
 	port[dev] = pnp_port_start(pdev, 0);
-	dma8[dev] = pnp_dma(pdev, 1);
+	dma8[dev] = pnp_dma(pdev, 0);
 	irq[dev] = pnp_irq(pdev, 0);
 
 	return 0;
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 8dab82d..668a5ec 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2184,10 +2184,9 @@ static int __devinit snd_echo_probe(struct pci_dev *pci,
 			goto ctl_error;
 #endif
 
-	if ((err = snd_card_register(card)) < 0) {
-		snd_card_free(card);
+	err = snd_card_register(card);
+	if (err < 0)
 		goto ctl_error;
-	}
 	snd_printk(KERN_INFO "Card registered: %s\n", card->longname);
 
 	pci_set_drvdata(pci, chip);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4bb9067..cec68152 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2272,6 +2272,8 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
 	{}
 };
@@ -2362,6 +2364,7 @@ static struct snd_pci_quirk msi_black_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */
 	SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */
 	SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */
+	SND_PCI_QUIRK(0xa0a0, 0x0575, "Aopen MZ915-M", 0), /* ICH6 */
 	{}
 };
 
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e6d1bdf..e9fdfc4 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -519,14 +519,6 @@ static int ad198x_suspend(struct hda_codec *codec, pm_message_t state)
 	ad198x_power_eapd(codec);
 	return 0;
 }
-
-static int ad198x_resume(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	snd_hda_codec_resume_amp(codec);
-	snd_hda_codec_resume_cache(codec);
-	return 0;
-}
 #endif
 
 static struct hda_codec_ops ad198x_patch_ops = {
@@ -539,7 +531,6 @@ static struct hda_codec_ops ad198x_patch_ops = {
 #endif
 #ifdef SND_HDA_NEEDS_RESUME
 	.suspend = ad198x_suspend,
-	.resume = ad198x_resume,
 #endif
 	.reboot_notify = ad198x_shutup,
 };
@@ -1896,6 +1887,14 @@ static int patch_ad1981(struct hda_codec *codec)
 	case AD1981_THINKPAD:
 		spec->mixers[0] = ad1981_thinkpad_mixers;
 		spec->input_mux = &ad1981_thinkpad_capture_source;
+		/* set the upper-limit for mixer amp to 0dB for avoiding the
+		 * possible damage by overloading
+		 */
+		snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
+					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+					  (1 << AC_AMPCAP_MUTE_SHIFT));
 		break;
 	case AD1981_TOSHIBA:
 		spec->mixers[0] = ad1981_hp_mixers;
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 7de782a..350ee8a 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -766,7 +766,7 @@ static int build_input(struct hda_codec *codec)
 		for (n = 0; n < AUTO_PIN_LAST; n++) {
 			if (!spec->adc_nid[n])
 				continue;
-			err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[i]);
+			err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]);
 			if (err < 0)
 				return err;
 		}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 61682e1..56e5207 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1195,9 +1195,10 @@ static int patch_cxt5045(struct hda_codec *codec)
 
 	switch (codec->subsystem_id >> 16) {
 	case 0x103c:
+	case 0x1631:
 	case 0x1734:
-		/* HP & Fujitsu-Siemens laptops have really bad sound over 0dB
-		 * on NID 0x17. Fix max PCM level to 0 dB
+		/* HP, Packard Bell, & Fujitsu-Siemens laptops have really bad
+		 * sound over 0dB on NID 0x17. Fix max PCM level to 0 dB
 		 * (originally it has 0x2b steps with 0dB offset 0x14)
 		 */
 		snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT,
@@ -2842,6 +2843,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 		      CXT5066_DELL_LAPTOP),
 	SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
+	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
+	SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
+	SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD),
 	{}
 };
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9a23444..7404dba 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -230,6 +230,7 @@ enum {
 	ALC888_ACER_ASPIRE_7730G,
 	ALC883_MEDION,
 	ALC883_MEDION_MD2,
+	ALC883_MEDION_WIM2160,
 	ALC883_LAPTOP_EAPD,
 	ALC883_LENOVO_101E_2ch,
 	ALC883_LENOVO_NB0763,
@@ -1389,22 +1390,31 @@ struct alc_fixup {
 
 static void alc_pick_fixup(struct hda_codec *codec,
 			   const struct snd_pci_quirk *quirk,
-			   const struct alc_fixup *fix)
+			   const struct alc_fixup *fix,
+			   int pre_init)
 {
 	const struct alc_pincfg *cfg;
 
 	quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
 	if (!quirk)
 		return;
-
 	fix += quirk->value;
 	cfg = fix->pins;
-	if (cfg) {
+	if (pre_init && cfg) {
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+		snd_printdd(KERN_INFO "hda_codec: %s: Apply pincfg for %s\n",
+			    codec->chip_name, quirk->name);
+#endif
 		for (; cfg->nid; cfg++)
 			snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
 	}
-	if (fix->verbs)
+	if (!pre_init && fix->verbs) {
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+		snd_printdd(KERN_INFO "hda_codec: %s: Apply fix-verbs for %s\n",
+			    codec->chip_name, quirk->name);
+#endif
 		add_verb(codec->spec, fix->verbs);
+	}
 }
 
 static int alc_read_coef_idx(struct hda_codec *codec,
@@ -1621,6 +1631,11 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
  */
 
 static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
+/* Route to built-in subwoofer as well as speakers */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 /* Bias voltage on for external mic port */
 	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
 /* Front Mic: set to PIN_IN (empty by default) */
@@ -1632,10 +1647,12 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
 /* Enable speaker output */
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
 /* Enable headphone output */
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
 	{ }
 };
 
@@ -4126,7 +4143,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
 	SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU),
-	SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL),
+	SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734),
 	SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU),
 	SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW),
 	SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG),
@@ -4801,6 +4818,25 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec)
 	}
 }
 
+static void alc880_auto_init_input_src(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int c;
+
+	for (c = 0; c < spec->num_adc_nids; c++) {
+		unsigned int mux_idx;
+		const struct hda_input_mux *imux;
+		mux_idx = c >= spec->num_mux_defs ? 0 : c;
+		imux = &spec->input_mux[mux_idx];
+		if (!imux->num_items && mux_idx > 0)
+			imux = &spec->input_mux[0];
+		if (imux)
+			snd_hda_codec_write(codec, spec->adc_nids[c], 0,
+					    AC_VERB_SET_CONNECT_SEL,
+					    imux->items[0].index);
+	}
+}
+
 /* parse the BIOS configuration and set up the alc_spec */
 /* return 1 if successful, 0 if the proper config is not found,
  * or a negative error code
@@ -4879,6 +4915,7 @@ static void alc880_auto_init(struct hda_codec *codec)
 	alc880_auto_init_multi_out(codec);
 	alc880_auto_init_extra_out(codec);
 	alc880_auto_init_analog_input(codec);
+	alc880_auto_init_input_src(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
@@ -4984,6 +5021,70 @@ static void set_capture_mixer(struct hda_codec *codec)
 	}
 }
 
+/* fill adc_nids (and capsrc_nids) containing all active input pins */
+static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
+				 int num_nids)
+{
+	struct alc_spec *spec = codec->spec;
+	int n;
+	hda_nid_t fallback_adc = 0, fallback_cap = 0;
+
+	for (n = 0; n < num_nids; n++) {
+		hda_nid_t adc, cap;
+		hda_nid_t conn[HDA_MAX_NUM_INPUTS];
+		int nconns, i, j;
+
+		adc = nids[n];
+		if (get_wcaps_type(get_wcaps(codec, adc)) != AC_WID_AUD_IN)
+			continue;
+		cap = adc;
+		nconns = snd_hda_get_connections(codec, cap, conn,
+						 ARRAY_SIZE(conn));
+		if (nconns == 1) {
+			cap = conn[0];
+			nconns = snd_hda_get_connections(codec, cap, conn,
+							 ARRAY_SIZE(conn));
+		}
+		if (nconns <= 0)
+			continue;
+		if (!fallback_adc) {
+			fallback_adc = adc;
+			fallback_cap = cap;
+		}
+		for (i = 0; i < AUTO_PIN_LAST; i++) {
+			hda_nid_t nid = spec->autocfg.input_pins[i];
+			if (!nid)
+				continue;
+			for (j = 0; j < nconns; j++) {
+				if (conn[j] == nid)
+					break;
+			}
+			if (j >= nconns)
+				break;
+		}
+		if (i >= AUTO_PIN_LAST) {
+			int num_adcs = spec->num_adc_nids;
+			spec->private_adc_nids[num_adcs] = adc;
+			spec->private_capsrc_nids[num_adcs] = cap;
+			spec->num_adc_nids++;
+			spec->adc_nids = spec->private_adc_nids;
+			if (adc != cap)
+				spec->capsrc_nids = spec->private_capsrc_nids;
+		}
+	}
+	if (!spec->num_adc_nids) {
+		printk(KERN_WARNING "hda_codec: %s: no valid ADC found;"
+		       " using fallback 0x%x\n",
+		       codec->chip_name, fallback_adc);
+		spec->private_adc_nids[0] = fallback_adc;
+		spec->adc_nids = spec->private_adc_nids;
+		if (fallback_adc != fallback_cap) {
+			spec->private_capsrc_nids[0] = fallback_cap;
+			spec->capsrc_nids = spec->private_adc_nids;
+		}
+	}
+}
+
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 #define set_beep_amp(spec, nid, idx, dir) \
 	((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
@@ -6326,6 +6427,8 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec)
 	}
 }
 
+#define alc260_auto_init_input_src	alc880_auto_init_input_src
+
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
@@ -6412,6 +6515,7 @@ static void alc260_auto_init(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	alc260_auto_init_multi_out(codec);
 	alc260_auto_init_analog_input(codec);
+	alc260_auto_init_input_src(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
@@ -8384,6 +8488,42 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x08, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+static struct hda_verb alc883_medion_wim2160_verbs[] = {
+	/* Unmute front mixer */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+	/* Set speaker pin to front mixer */
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Init headphone pin */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+
+	{ } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_medion_wim2160_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x1a;
+	spec->autocfg.speaker_pins[0] = 0x15;
+}
+
 static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -8398,9 +8538,7 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
 
 static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
@@ -9095,6 +9233,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = {
 	[ALC888_ACER_ASPIRE_7730G]	= "acer-aspire-7730g",
 	[ALC883_MEDION]		= "medion",
 	[ALC883_MEDION_MD2]	= "medion-md2",
+	[ALC883_MEDION_WIM2160]	= "medion-wim2160",
 	[ALC883_LAPTOP_EAPD]	= "laptop-eapd",
 	[ALC883_LENOVO_101E_2ch] = "lenovo-101e",
 	[ALC883_LENOVO_NB0763]	= "lenovo-nb0763",
@@ -9211,6 +9350,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG),
 
 	SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
 	SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
 	SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R),
@@ -9749,6 +9889,21 @@ static struct alc_config_preset alc882_presets[] = {
 		.setup = alc883_medion_md2_setup,
 		.init_hook = alc_automute_amp,
 	},
+	[ALC883_MEDION_WIM2160] = {
+		.mixers = { alc883_medion_wim2160_mixer },
+		.init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+		.adc_nids = alc883_adc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+		.channel_mode = alc883_3ST_2ch_modes,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc_automute_amp_unsol_event,
+		.setup = alc883_medion_wim2160_setup,
+		.init_hook = alc_automute_amp,
+	},
 	[ALC883_LAPTOP_EAPD] = {
 		.mixers = { alc883_base_mixer },
 		.init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
@@ -10041,13 +10196,12 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
 	int idx;
 
 	alc_set_pin_output(codec, nid, pin_type);
+	if (dac_idx >= spec->multiout.num_dacs)
+		return;
 	if (spec->multiout.dac_nids[dac_idx] == 0x25)
 		idx = 4;
-	else {
-		if (spec->multiout.num_dacs >= dac_idx)
-			return;
+	else
 		idx = spec->multiout.dac_nids[dac_idx] - 2;
-	}
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
 
 }
@@ -10295,7 +10449,8 @@ static int patch_alc882(struct hda_codec *codec)
 		board_config = ALC882_AUTO;
 	}
 
-	alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups);
+	if (board_config == ALC882_AUTO)
+		alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 1);
 
 	if (board_config == ALC882_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -10368,6 +10523,9 @@ static int patch_alc882(struct hda_codec *codec)
 	set_capture_mixer(codec);
 	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
+	if (board_config == ALC882_AUTO)
+		alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 0);
+
 	spec->vmaster_nid = 0x0c;
 
 	codec->patch_ops = alc_patch_ops;
@@ -12459,11 +12617,11 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec)
 	unsigned char bits;
 
 	present = snd_hda_jack_detect(codec, 0x15);
-	bits = present ? AMP_IN_MUTE(0) : 0;
+	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 }
 
 static void alc268_acer_lc_unsol_event(struct hda_codec *codec,
@@ -12748,6 +12906,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
 		dac = 0x02;
 		break;
 	case 0x15:
+	case 0x21: /* ALC269vb has this pin, too */
 		dac = 0x03;
 		break;
 	default:
@@ -13333,9 +13492,9 @@ static hda_nid_t alc269vb_capsrc_nids[1] = {
 	0x22,
 };
 
-/* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24),
- *       not a mux!
- */
+static hda_nid_t alc269_adc_candidates[] = {
+	0x08, 0x09, 0x07,
+};
 
 #define alc269_modes		alc260_modes
 #define alc269_capture_source	alc880_lg_lw_capture_source
@@ -13482,11 +13641,11 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
 	unsigned char bits;
 
 	present = snd_hda_jack_detect(codec, 0x15);
-	bits = present ? AMP_IN_MUTE(0) : 0;
+	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-			AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-			AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 
 	snd_hda_codec_write(codec, 0x20, 0,
 			AC_VERB_SET_COEF_INDEX, 0x0c);
@@ -13511,11 +13670,11 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec)
 	/* Check port replicator headphone socket */
 	present |= snd_hda_jack_detect(codec, 0x1a);
 
-	bits = present ? AMP_IN_MUTE(0) : 0;
+	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-			AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-			AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 
 	snd_hda_codec_write(codec, 0x20, 0,
 			AC_VERB_SET_COEF_INDEX, 0x0c);
@@ -13646,11 +13805,11 @@ static void alc269_speaker_automute(struct hda_codec *codec)
 	unsigned char bits;
 
 	present = snd_hda_jack_detect(codec, nid);
-	bits = present ? AMP_IN_MUTE(0) : 0;
+	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 }
 
 /* unsolicited event for HP jack sensing */
@@ -13667,19 +13826,19 @@ static void alc269_laptop_unsol_event(struct hda_codec *codec,
 	}
 }
 
-static void alc269_laptop_dmic_setup(struct hda_codec *codec)
+static void alc269_laptop_amic_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
-	spec->int_mic.pin = 0x12;
-	spec->int_mic.mux_idx = 5;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
 	spec->auto_mic = 1;
 }
 
-static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
+static void alc269_laptop_dmic_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	spec->autocfg.hp_pins[0] = 0x15;
@@ -13687,14 +13846,14 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x12;
-	spec->int_mic.mux_idx = 6;
+	spec->int_mic.mux_idx = 5;
 	spec->auto_mic = 1;
 }
 
-static void alc269_laptop_amic_setup(struct hda_codec *codec)
+static void alc269vb_laptop_amic_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.hp_pins[0] = 0x21;
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
@@ -13703,6 +13862,18 @@ static void alc269_laptop_amic_setup(struct hda_codec *codec)
 	spec->auto_mic = 1;
 }
 
+static void alc269vb_laptop_dmic_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	spec->autocfg.hp_pins[0] = 0x21;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x12;
+	spec->int_mic.mux_idx = 6;
+	spec->auto_mic = 1;
+}
+
 static void alc269_laptop_inithook(struct hda_codec *codec)
 {
 	alc269_speaker_automute(codec);
@@ -13842,7 +14013,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	int err;
 	static hda_nid_t alc269_ignore[] = { 0x1d, 0 };
-	hda_nid_t real_capsrc_nids;
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
 					   alc269_ignore);
@@ -13866,18 +14036,19 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 
 	if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) {
 		add_verb(spec, alc269vb_init_verbs);
-		real_capsrc_nids = alc269vb_capsrc_nids[0];
 		alc_ssid_check(codec, 0, 0x1b, 0x14, 0x21);
 	} else {
 		add_verb(spec, alc269_init_verbs);
-		real_capsrc_nids = alc269_capsrc_nids[0];
 		alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0);
 	}
 
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux[0];
+	fillup_priv_adc_nids(codec, alc269_adc_candidates,
+			     sizeof(alc269_adc_candidates));
+
 	/* set default input source */
-	snd_hda_codec_write_cache(codec, real_capsrc_nids,
+	snd_hda_codec_write_cache(codec, spec->capsrc_nids[0],
 				  0, AC_VERB_SET_CONNECT_SEL,
 				  spec->input_mux->items[0].index);
 
@@ -13907,6 +14078,27 @@ static void alc269_auto_init(struct hda_codec *codec)
 		alc_inithook(codec);
 }
 
+enum {
+	ALC269_FIXUP_SONY_VAIO,
+};
+
+const static struct hda_verb alc269_sony_vaio_fixup_verbs[] = {
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD},
+	{}
+};
+
+static const struct alc_fixup alc269_fixups[] = {
+	[ALC269_FIXUP_SONY_VAIO] = {
+		.verbs = alc269_sony_vaio_fixup_verbs
+	},
+};
+
+static struct snd_pci_quirk alc269_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
+	{}
+};
+
+
 /*
  * configuration and preset
  */
@@ -13966,7 +14158,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = {
 		      ALC269_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
-	SND_PCI_QUIRK(0x104d, 0x9071, "SONY XTB", ALC269_DMIC),
+	SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
 	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
 	SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
 	SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU),
@@ -14040,7 +14232,7 @@ static struct alc_config_preset alc269_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc269_modes),
 		.channel_mode = alc269_modes,
 		.unsol_event = alc269_laptop_unsol_event,
-		.setup = alc269_laptop_amic_setup,
+		.setup = alc269vb_laptop_amic_setup,
 		.init_hook = alc269_laptop_inithook,
 	},
 	[ALC269VB_DMIC] = {
@@ -14120,6 +14312,9 @@ static int patch_alc269(struct hda_codec *codec)
 		board_config = ALC269_AUTO;
 	}
 
+	if (board_config == ALC269_AUTO)
+		alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 1);
+
 	if (board_config == ALC269_AUTO) {
 		/* automatic parse from the BIOS config */
 		err = alc269_parse_auto_config(codec);
@@ -14156,20 +14351,25 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc269_pcm_digital_playback;
 	spec->stream_digital_capture = &alc269_pcm_digital_capture;
 
-	if (!is_alc269vb) {
-		spec->adc_nids = alc269_adc_nids;
-		spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
-		spec->capsrc_nids = alc269_capsrc_nids;
-	} else {
-		spec->adc_nids = alc269vb_adc_nids;
-		spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids);
-		spec->capsrc_nids = alc269vb_capsrc_nids;
+	if (!spec->adc_nids) { /* wasn't filled automatically? use default */
+		if (!is_alc269vb) {
+			spec->adc_nids = alc269_adc_nids;
+			spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
+			spec->capsrc_nids = alc269_capsrc_nids;
+		} else {
+			spec->adc_nids = alc269vb_adc_nids;
+			spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids);
+			spec->capsrc_nids = alc269vb_capsrc_nids;
+		}
 	}
 
 	if (!spec->cap_mixer)
 		set_capture_mixer(codec);
 	set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 
+	if (board_config == ALC269_AUTO)
+		alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 0);
+
 	spec->vmaster_nid = 0x02;
 
 	codec->patch_ops = alc_patch_ops;
@@ -15258,7 +15458,8 @@ static int patch_alc861(struct hda_codec *codec)
 		board_config = ALC861_AUTO;
 	}
 
-	alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups);
+	if (board_config == ALC861_AUTO)
+		alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 1);
 
 	if (board_config == ALC861_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -15295,6 +15496,9 @@ static int patch_alc861(struct hda_codec *codec)
 
 	spec->vmaster_nid = 0x03;
 
+	if (board_config == ALC861_AUTO)
+		alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 0);
+
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC861_AUTO) {
 		spec->init_hook = alc861_auto_init;
@@ -16229,7 +16433,8 @@ static int patch_alc861vd(struct hda_codec *codec)
 		board_config = ALC861VD_AUTO;
 	}
 
-	alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups);
+	if (board_config == ALC861VD_AUTO)
+		alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 1);
 
 	if (board_config == ALC861VD_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -16277,6 +16482,9 @@ static int patch_alc861vd(struct hda_codec *codec)
 
 	spec->vmaster_nid = 0x02;
 
+	if (board_config == ALC861VD_AUTO)
+		alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 0);
+
 	codec->patch_ops = alc_patch_ops;
 
 	if (board_config == ALC861VD_AUTO)
@@ -17115,9 +17323,9 @@ static void alc663_m51va_speaker_automute(struct hda_codec *codec)
 	present = snd_hda_jack_detect(codec, 0x21);
 	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 }
 
 static void alc663_21jd_two_speaker_automute(struct hda_codec *codec)
@@ -17128,13 +17336,13 @@ static void alc663_21jd_two_speaker_automute(struct hda_codec *codec)
 	present = snd_hda_jack_detect(codec, 0x21);
 	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 }
 
 static void alc663_15jd_two_speaker_automute(struct hda_codec *codec)
@@ -17145,13 +17353,13 @@ static void alc663_15jd_two_speaker_automute(struct hda_codec *codec)
 	present = snd_hda_jack_detect(codec, 0x15);
 	bits = present ? HDA_AMP_MUTE : 0;
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 	snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), bits);
+				 HDA_AMP_MUTE, bits);
 }
 
 static void alc662_f5z_speaker_automute(struct hda_codec *codec)
@@ -17190,14 +17398,14 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
 
 	if (present1 || present2) {
 		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), AMP_IN_MUTE(0));
+					 HDA_AMP_MUTE, HDA_AMP_MUTE);
 		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), AMP_IN_MUTE(0));
+					 HDA_AMP_MUTE, HDA_AMP_MUTE);
 	} else {
 		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-				AMP_IN_MUTE(0), 0);
+					 HDA_AMP_MUTE, 0);
 		snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-				AMP_IN_MUTE(0), 0);
+					 HDA_AMP_MUTE, 0);
 	}
 }
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index c4be3fa..7fb7d01 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1607,6 +1607,10 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 				"Dell Studio 1555", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd,
 				"Dell Studio 1557", STAC_DELL_M6_DMIC),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe,
+				"Dell Studio XPS 1645", STAC_DELL_M6_BOTH),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0413,
+				"Dell Studio 1558", STAC_DELL_M6_BOTH),
 	{} /* terminator */
 };
 
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 9ddc373..7345381 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -476,7 +476,7 @@ static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec,
 	knew->name = kstrdup(tmpl->name, GFP_KERNEL);
 	if (!knew->name)
 		return NULL;
-	return 0;
+	return knew;
 }
 
 static void via_free_kctls(struct hda_codec *codec)
@@ -1215,14 +1215,13 @@ static struct snd_kcontrol_new via_hp_mixer[2] = {
 	},
 };
 
-static int via_hp_build(struct via_spec *spec)
+static int via_hp_build(struct hda_codec *codec)
 {
+	struct via_spec *spec = codec->spec;
 	struct snd_kcontrol_new *knew;
 	hda_nid_t nid;
-
-	knew = via_clone_control(spec, &via_hp_mixer[0]);
-	if (knew == NULL)
-		return -ENOMEM;
+	int nums;
+	hda_nid_t conn[HDA_MAX_CONNECTIONS];
 
 	switch (spec->codec_type) {
 	case VT1718S:
@@ -1239,6 +1238,14 @@ static int via_hp_build(struct via_spec *spec)
 		break;
 	}
 
+	nums = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS);
+	if (nums <= 1)
+		return 0;
+
+	knew = via_clone_control(spec, &via_hp_mixer[0]);
+	if (knew == NULL)
+		return -ENOMEM;
+
 	knew->subdevice = HDA_SUBDEV_NID_FLAG | nid;
 	knew->private_value = nid;
 
@@ -2561,7 +2568,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	via_smart51_build(spec);
 	return 1;
@@ -3087,7 +3094,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	via_smart51_build(spec);
 	return 1;
@@ -3654,7 +3661,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	via_smart51_build(spec);
 	return 1;
@@ -4140,7 +4147,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	via_smart51_build(spec);
 	return 1;
@@ -4510,7 +4517,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	return 1;
 }
@@ -4930,7 +4937,7 @@ static int vt1718S_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	via_smart51_build(spec);
 
@@ -5425,7 +5432,7 @@ static int vt1716S_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	via_smart51_build(spec);
 
@@ -5781,7 +5788,7 @@ static int vt2002P_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	return 1;
 }
@@ -6000,12 +6007,12 @@ static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec,
 
 	/* Line-Out: PortE */
 	err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
-			      "Master Front Playback Volume",
+			      "Front Playback Volume",
 			      HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT));
 	if (err < 0)
 		return err;
 	err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE,
-			      "Master Front Playback Switch",
+			      "Front Playback Switch",
 			      HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT));
 	if (err < 0)
 		return err;
@@ -6130,7 +6137,7 @@ static int vt1812_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux[0];
 
 	if (spec->hp_mux)
-		via_hp_build(spec);
+		via_hp_build(codec);
 
 	return 1;
 }
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index b64e781..b56e336 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -849,6 +849,7 @@ struct snd_m3 {
 	struct snd_kcontrol *master_switch;
 	struct snd_kcontrol *master_volume;
 	struct tasklet_struct hwvol_tq;
+	unsigned int in_suspend;
 
 #ifdef CONFIG_PM
 	u16 *suspend_mem;
@@ -884,6 +885,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_m3_ids) = {
 MODULE_DEVICE_TABLE(pci, snd_m3_ids);
 
 static struct snd_pci_quirk m3_amp_quirk_list[] __devinitdata = {
+	SND_PCI_QUIRK(0x0E11, 0x0094, "Compaq Evo N600c", 0x0c),
 	SND_PCI_QUIRK(0x10f7, 0x833e, "Panasonic CF-28", 0x0d),
 	SND_PCI_QUIRK(0x10f7, 0x833d, "Panasonic CF-72", 0x0d),
 	SND_PCI_QUIRK(0x1033, 0x80f1, "NEC LM800J/7", 0x03),
@@ -1613,6 +1615,11 @@ static void snd_m3_update_hw_volume(unsigned long private_data)
 	outb(0x88, chip->iobase + SHADOW_MIX_REG_MASTER);
 	outb(0x88, chip->iobase + HW_VOL_COUNTER_MASTER);
 
+	/* Ignore spurious HV interrupts during suspend / resume, this avoids
+	   mistaking them for a mute button press. */
+	if (chip->in_suspend)
+		return;
+
 	if (!chip->master_switch || !chip->master_volume)
 		return;
 
@@ -2424,6 +2431,7 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state)
 	if (chip->suspend_mem == NULL)
 		return 0;
 
+	chip->in_suspend = 1;
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
 	snd_pcm_suspend_all(chip->pcm);
 	snd_ac97_suspend(chip->ac97);
@@ -2497,6 +2505,7 @@ static int m3_resume(struct pci_dev *pci)
 	snd_m3_hv_init(chip);
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+	chip->in_suspend = 0;
 	return 0;
 }
 #endif /* CONFIG_PM */
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 55e9315..3be8f97 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1162,13 +1162,15 @@ static long snd_mixart_BA0_read(struct snd_info_entry *entry, void *file_private
 				unsigned long count, unsigned long pos)
 {
 	struct mixart_mgr *mgr = entry->private_data;
+	unsigned long maxsize;
 
-	count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
-	if(count <= 0)
+	if (pos >= MIXART_BA0_SIZE)
 		return 0;
-	if(pos + count > MIXART_BA0_SIZE)
-		count = (long)(MIXART_BA0_SIZE - pos);
-	if(copy_to_user_fromio(buf, MIXART_MEM( mgr, pos ), count))
+	maxsize = MIXART_BA0_SIZE - pos;
+	if (count > maxsize)
+		count = maxsize;
+	count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
+	if (copy_to_user_fromio(buf, MIXART_MEM(mgr, pos), count))
 		return -EFAULT;
 	return count;
 }
@@ -1181,13 +1183,15 @@ static long snd_mixart_BA1_read(struct snd_info_entry *entry, void *file_private
 				unsigned long count, unsigned long pos)
 {
 	struct mixart_mgr *mgr = entry->private_data;
+	unsigned long maxsize;
 
-	count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
-	if(count <= 0)
+	if (pos > MIXART_BA1_SIZE)
 		return 0;
-	if(pos + count > MIXART_BA1_SIZE)
-		count = (long)(MIXART_BA1_SIZE - pos);
-	if(copy_to_user_fromio(buf, MIXART_REG( mgr, pos ), count))
+	maxsize = MIXART_BA1_SIZE - pos;
+	if (count > maxsize)
+		count = maxsize;
+	count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
+	if (copy_to_user_fromio(buf, MIXART_REG(mgr, pos), count))
 		return -EFAULT;
 	return count;
 }
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 9ef6b96..3e6628c8 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -180,7 +180,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 	runtime->dma_bytes = params_buffer_bytes(params);
 
-	prtd->params = rtd->dai->cpu_dai->dma_data;
+	prtd->params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 	prtd->params->dma_intr_handler = atmel_pcm_dma_irq;
 
 	prtd->dma_buffer = runtime->dma_addr;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index e588e63..0b59806 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -363,12 +363,12 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 	ssc_p->dma_params[dir] = dma_params;
 
 	/*
-	 * The cpu_dai->dma_data field is only used to communicate the
-	 * appropriate DMA parameters to the pcm driver hw_params()
+	 * The snd_soc_pcm_stream->dma_data field is only used to communicate
+	 * the appropriate DMA parameters to the pcm driver hw_params()
 	 * function.  It should not be used for other purposes
 	 * as it is common to all substreams.
 	 */
-	rtd->dai->cpu_dai->dma_data = dma_params;
+	snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_params);
 
 	channels = params_channels(params);
 
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index fd101d4..1f5e57a 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -81,9 +81,11 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 static int ac97_soc_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_codec *codec;
 	struct snd_ac97_bus *ac97_bus;
 	struct snd_ac97_template ac97_template;
+	int i;
 	int ret = 0;
 
 	printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION);
@@ -103,12 +105,6 @@ static int ac97_soc_probe(struct platform_device *pdev)
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
-	if (ret < 0) {
-		printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n");
-		goto err;
-	}
-
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0)
@@ -124,6 +120,13 @@ static int ac97_soc_probe(struct platform_device *pdev)
 	if (ret < 0)
 		goto bus_err;
 
+	for (i = 0; i < card->num_links; i++) {
+		if (card->dai_link[i].codec_dai->ac97_control) {
+			snd_ac97_dev_add_pdata(codec->ac97,
+				card->dai_link[i].cpu_dai->ac97_pdata);
+		}
+	}
+
 	return 0;
 
 bus_err:
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index a34cbcf..002e289 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -23,7 +23,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/firmware.h>
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 8d1c637..9da0724 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3008,34 +3008,39 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_OFF:
-		/* Switch over to startup biases */
-		snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
-				    WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA |
-				    WM8994_VMID_BUF_ENA |
-				    WM8994_VMID_RAMP_MASK,
-				    WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA |
-				    WM8994_VMID_BUF_ENA |
-				    (1 << WM8994_VMID_RAMP_SHIFT));
-
-		/* Disable main biases */
-		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
-				    WM8994_BIAS_ENA | WM8994_VMID_SEL_MASK, 0);
+		if (codec->bias_level == SND_SOC_BIAS_STANDBY) {
+			/* Switch over to startup biases */
+			snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
+					    WM8994_BIAS_SRC |
+					    WM8994_STARTUP_BIAS_ENA |
+					    WM8994_VMID_BUF_ENA |
+					    WM8994_VMID_RAMP_MASK,
+					    WM8994_BIAS_SRC |
+					    WM8994_STARTUP_BIAS_ENA |
+					    WM8994_VMID_BUF_ENA |
+					    (1 << WM8994_VMID_RAMP_SHIFT));
 
-		/* Discharge line */
-		snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
-				    WM8994_LINEOUT1_DISCH |
-				    WM8994_LINEOUT2_DISCH,
-				    WM8994_LINEOUT1_DISCH |
-				    WM8994_LINEOUT2_DISCH);
+			/* Disable main biases */
+			snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
+					    WM8994_BIAS_ENA |
+					    WM8994_VMID_SEL_MASK, 0);
 
-		msleep(5);
+			/* Discharge line */
+			snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
+					    WM8994_LINEOUT1_DISCH |
+					    WM8994_LINEOUT2_DISCH,
+					    WM8994_LINEOUT1_DISCH |
+					    WM8994_LINEOUT2_DISCH);
 
-		/* Switch off startup biases */
-		snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
-				    WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA |
-				    WM8994_VMID_BUF_ENA |
-				    WM8994_VMID_RAMP_MASK, 0);
+			msleep(5);
 
+			/* Switch off startup biases */
+			snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
+					    WM8994_BIAS_SRC |
+					    WM8994_STARTUP_BIAS_ENA |
+					    WM8994_VMID_BUF_ENA |
+					    WM8994_VMID_RAMP_MASK, 0);
+		}
 		break;
 	}
 	codec->bias_level = level;
@@ -3402,7 +3407,7 @@ struct snd_soc_dai wm8994_dai[] = {
 			.rates = WM8994_RATES,
 			.formats = WM8994_FORMATS,
 		},
-		.playback = {
+		.capture = {
 			.stream_name = "AIF3 Capture",
 			.channels_min = 2,
 			.channels_max = 2,
@@ -3731,11 +3736,12 @@ static int wm8994_codec_probe(struct platform_device *pdev)
 	case 3:
 		wm8994->hubs.dcs_codes = -5;
 		wm8994->hubs.hp_startup_mode = 1;
+		wm8994->hubs.dcs_readback_mode = 1;
 		break;
 	default:
+		wm8994->hubs.dcs_readback_mode = 1;
 		break;
 	}
-			   
 
 	/* Remember if AIFnLRCLK is configured as a GPIO.  This should be
 	 * configured on init - if a system wants to do this dynamically
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 486bdd2..e1f225a 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -62,21 +62,27 @@ static const char *speaker_mode_text[] = {
 static const struct soc_enum speaker_mode =
 	SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text);
 
-static void wait_for_dc_servo(struct snd_soc_codec *codec)
+static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
 {
 	unsigned int reg;
 	int count = 0;
+	unsigned int val;
+
+	val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1;
+
+	/* Trigger the command */
+	snd_soc_write(codec, WM8993_DC_SERVO_0, val);
 
 	dev_dbg(codec->dev, "Waiting for DC servo...\n");
 
 	do {
 		count++;
 		msleep(1);
-		reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0);
+		reg = snd_soc_read(codec, WM8993_DC_SERVO_0);
 		dev_dbg(codec->dev, "DC servo: %x\n", reg);
-	} while (reg & WM8993_DCS_DATAPATH_BUSY && count < 400);
+	} while (reg & op && count < 400);
 
-	if (reg & WM8993_DCS_DATAPATH_BUSY)
+	if (reg & op)
 		dev_err(codec->dev, "Timed out waiting for DC Servo\n");
 }
 
@@ -86,51 +92,58 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec)
 static void calibrate_dc_servo(struct snd_soc_codec *codec)
 {
 	struct wm_hubs_data *hubs = codec->private_data;
-	u16 reg, dcs_cfg;
+	u16 reg, reg_l, reg_r, dcs_cfg;
 
 	/* Set for 32 series updates */
 	snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
 			    WM8993_DCS_SERIES_NO_01_MASK,
 			    32 << WM8993_DCS_SERIES_NO_01_SHIFT);
-
-	/* Enable the DC servo.  Write all bits to avoid triggering startup
-	 * or write calibration.
-	 */
-	snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
-			    0xFFFF,
-			    WM8993_DCS_ENA_CHAN_0 |
-			    WM8993_DCS_ENA_CHAN_1 |
-			    WM8993_DCS_TRIG_SERIES_1 |
-			    WM8993_DCS_TRIG_SERIES_0);
-
-	wait_for_dc_servo(codec);
+	wait_for_dc_servo(codec,
+			  WM8993_DCS_TRIG_SERIES_0 | WM8993_DCS_TRIG_SERIES_1);
 
 	/* Apply correction to DC servo result */
 	if (hubs->dcs_codes) {
 		dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
 			hubs->dcs_codes);
 
+		/* Different chips in the family support different
+		 * readback methods.
+		 */
+		switch (hubs->dcs_readback_mode) {
+		case 0:
+			reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
+				& WM8993_DCS_INTEG_CHAN_0_MASK;;
+			reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
+				& WM8993_DCS_INTEG_CHAN_1_MASK;
+			break;
+		case 1:
+			reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
+			reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
+				>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+			reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
+			break;
+		default:
+			WARN(1, "Unknown DCS readback method");
+			break;
+		}
+
 		/* HPOUT1L */
-		reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) &
-			WM8993_DCS_INTEG_CHAN_0_MASK;;
-		reg += hubs->dcs_codes;
-		dcs_cfg = reg << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+		if (reg_l + hubs->dcs_codes > 0 &&
+		    reg_l + hubs->dcs_codes < 0xff)
+			reg_l += hubs->dcs_codes;
+		dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
 
 		/* HPOUT1R */
-		reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) &
-			WM8993_DCS_INTEG_CHAN_1_MASK;
-		reg += hubs->dcs_codes;
-		dcs_cfg |= reg;
+		if (reg_r + hubs->dcs_codes > 0 &&
+		    reg_r + hubs->dcs_codes < 0xff)
+			reg_r += hubs->dcs_codes;
+		dcs_cfg |= reg_r;
 
 		/* Do it */
 		snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg);
-		snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
-				    WM8993_DCS_TRIG_DAC_WR_0 |
-				    WM8993_DCS_TRIG_DAC_WR_1,
-				    WM8993_DCS_TRIG_DAC_WR_0 |
-				    WM8993_DCS_TRIG_DAC_WR_1);
-
-		wait_for_dc_servo(codec);
+		wait_for_dc_servo(codec,
+				  WM8993_DCS_TRIG_DAC_WR_0 |
+				  WM8993_DCS_TRIG_DAC_WR_1);
 	}
 }
 
@@ -141,10 +154,16 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wm_hubs_data *hubs = codec->private_data;
 	int ret;
 
 	ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
 
+	/* If we're applying an offset correction then updating the
+	 * callibration would be likely to introduce further offsets. */
+	if (hubs->dcs_codes)
+		return ret;
+
 	/* Only need to do this if the outputs are active */
 	if (snd_soc_read(codec, WM8993_POWER_MANAGEMENT_1)
 	    & (WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA))
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 420104f..e51c166 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -21,6 +21,7 @@ extern const unsigned int wm_hubs_spkmix_tlv[];
 /* This *must* be the first element of the codec->private_data struct */
 struct wm_hubs_data {
 	int dcs_codes;
+	int dcs_readback_mode;
 	int hp_startup_mode;
 };
 
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 62af7e0..adadcd3 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -586,7 +586,8 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
 
 	davinci_i2s_dai.private_data = dev;
-	davinci_i2s_dai.dma_data = dev->dma_params;
+	davinci_i2s_dai.capture.dma_data = dev->dma_params;
+	davinci_i2s_dai.playback.dma_data = dev->dma_params;
 	ret = snd_soc_register_dai(&davinci_i2s_dai);
 	if (ret != 0)
 		goto err_free_mem;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 6c80cc3..79f0f4a 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -918,7 +918,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 
 	dma_data->channel = res->start;
 	davinci_mcasp_dai[pdata->op_mode].private_data = dev;
-	davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params;
+	davinci_mcasp_dai[pdata->op_mode].capture.dma_data = dev->dma_params;
+	davinci_mcasp_dai[pdata->op_mode].playback.dma_data = dev->dma_params;
 	davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
 	ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
 
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 80c7fdf..2dc406f 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -649,8 +649,10 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
 	struct snd_pcm_hardware *ppcm;
 	int ret = 0;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data;
+	struct davinci_pcm_dma_params *pa;
 	struct davinci_pcm_dma_params *params;
+
+	pa = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 	if (!pa)
 		return -ENODEV;
 	params = &pa[substream->stream];
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index 86668ab..2b31ac6 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -71,7 +71,12 @@ static void imx_ssi_dma_callback(int channel, void *data)
 
 static void snd_imx_dma_err_callback(int channel, void *data, int err)
 {
-	pr_err("DMA error callback called\n");
+	struct snd_pcm_substream *substream = data;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+	int ret;
 
 	pr_err("DMA timeout on channel %d -%s%s%s%s\n",
 		 channel,
@@ -79,16 +84,26 @@ static void snd_imx_dma_err_callback(int channel, void *data, int err)
 		 err & IMX_DMA_ERR_REQUEST ?  " request" : "",
 		 err & IMX_DMA_ERR_TRANSFER ? " transfer" : "",
 		 err & IMX_DMA_ERR_BUFFER ?   " buffer" : "");
+
+	imx_dma_disable(iprtd->dma);
+	ret = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count,
+			IMX_DMA_LENGTH_LOOP, dma_params->dma_addr,
+			substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+			DMA_MODE_WRITE : DMA_MODE_READ);
+	if (!ret)
+		imx_dma_enable(iprtd->dma);
 }
 
 static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
+	struct imx_pcm_dma_params *dma_params;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 	int ret;
 
+	dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream);
+
 	iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH);
 	if (iprtd->dma < 0) {
 		pr_err("Failed to claim the audio DMA\n");
@@ -193,10 +208,12 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
+	struct imx_pcm_dma_params *dma_params;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 	int err;
 
+	dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream);
+
 	iprtd->substream = substream;
 	iprtd->buf = (unsigned int *)substream->dma_buffer.area;
 	iprtd->period_cnt = 0;
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index f96a373..6b518e0 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -39,23 +39,24 @@ struct imx_pcm_runtime_data {
 	unsigned long offset;
 	unsigned long last_offset;
 	unsigned long size;
-	struct timer_list timer;
-	int poll_time;
+	struct hrtimer hrt;
+	int poll_time_ns;
+	struct snd_pcm_substream *substream;
+	atomic_t running;
 };
 
-static inline void imx_ssi_set_next_poll(struct imx_pcm_runtime_data *iprtd)
+static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
 {
-	iprtd->timer.expires = jiffies + iprtd->poll_time;
-}
-
-static void imx_ssi_timer_callback(unsigned long data)
-{
-	struct snd_pcm_substream *substream = (void *)data;
+	struct imx_pcm_runtime_data *iprtd =
+		container_of(hrt, struct imx_pcm_runtime_data, hrt);
+	struct snd_pcm_substream *substream = iprtd->substream;
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 	struct pt_regs regs;
 	unsigned long delta;
 
+	if (!atomic_read(&iprtd->running))
+		return HRTIMER_NORESTART;
+
 	get_fiq_regs(&regs);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -72,16 +73,14 @@ static void imx_ssi_timer_callback(unsigned long data)
 
 	/* If we've transferred at least a period then report it and
 	 * reset our poll time */
-	if (delta >= runtime->period_size) {
+	if (delta >= iprtd->period) {
 		snd_pcm_period_elapsed(substream);
 		iprtd->last_offset = iprtd->offset;
-
-		imx_ssi_set_next_poll(iprtd);
 	}
 
-	/* Restart the timer; if we didn't report we'll run on the next tick */
-	add_timer(&iprtd->timer);
+	hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns));
 
+	return HRTIMER_RESTART;
 }
 
 static struct fiq_handler fh = {
@@ -99,8 +98,8 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream,
 	iprtd->period = params_period_bytes(params) ;
 	iprtd->offset = 0;
 	iprtd->last_offset = 0;
-	iprtd->poll_time = HZ / (params_rate(params) / params_period_size(params));
-
+	iprtd->poll_time_ns = 1000000000 / params_rate(params) *
+				params_period_size(params);
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 
 	return 0;
@@ -135,8 +134,9 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		imx_ssi_set_next_poll(iprtd);
-		add_timer(&iprtd->timer);
+		atomic_set(&iprtd->running, 1);
+		hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns),
+		      HRTIMER_MODE_REL);
 		if (++fiq_enable == 1)
 			enable_fiq(imx_pcm_fiq);
 
@@ -145,11 +145,11 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		del_timer(&iprtd->timer);
+		atomic_set(&iprtd->running, 0);
+
 		if (--fiq_enable == 0)
 			disable_fiq(imx_pcm_fiq);
 
-
 		break;
 	default:
 		return -EINVAL;
@@ -180,7 +180,7 @@ static struct snd_pcm_hardware snd_imx_hardware = {
 	.buffer_bytes_max = IMX_SSI_DMABUF_SIZE,
 	.period_bytes_min = 128,
 	.period_bytes_max = 16 * 1024,
-	.periods_min = 2,
+	.periods_min = 4,
 	.periods_max = 255,
 	.fifo_size = 0,
 };
@@ -194,9 +194,11 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
 	iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL);
 	runtime->private_data = iprtd;
 
-	init_timer(&iprtd->timer);
-	iprtd->timer.data = (unsigned long)substream;
-	iprtd->timer.function = imx_ssi_timer_callback;
+	iprtd->substream = substream;
+
+	atomic_set(&iprtd->running, 0);
+	hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
+	iprtd->hrt.function = snd_hrtimer_callback;
 
 	ret = snd_pcm_hw_constraint_integer(substream->runtime,
 			SNDRV_PCM_HW_PARAM_PERIODS);
@@ -212,7 +214,8 @@ static int snd_imx_close(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 
-	del_timer_sync(&iprtd->timer);
+	hrtimer_cancel(&iprtd->hrt);
+
 	kfree(iprtd);
 
 	return 0;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 6546b06..80b4fee 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -235,17 +235,20 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *cpu_dai)
 {
 	struct imx_ssi *ssi = cpu_dai->private_data;
+	struct imx_pcm_dma_params *dma_data;
 	u32 reg, sccr;
 
 	/* Tx/Rx config */
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		reg = SSI_STCCR;
-		cpu_dai->dma_data = &ssi->dma_params_tx;
+		dma_data = &ssi->dma_params_tx;
 	} else {
 		reg = SSI_SRCCR;
-		cpu_dai->dma_data = &ssi->dma_params_rx;
+		dma_data = &ssi->dma_params_rx;
 	}
 
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
+
 	sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK;
 
 	/* DAI data (word) size */
@@ -653,7 +656,8 @@ static int imx_ssi_probe(struct platform_device *pdev)
 	dai->private_data = ssi;
 
 	if ((cpu_is_mx27() || cpu_is_mx21()) &&
-			!(ssi->flags & IMX_SSI_USE_AC97)) {
+			!(ssi->flags & IMX_SSI_USE_AC97) &&
+			(ssi->flags & IMX_SSI_DMA)) {
 		ssi->flags |= IMX_SSI_DMA;
 		platform = imx_ssi_dma_mx2_init(pdev, ssi);
 	} else
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index e814a95..8ad9dc9 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -297,7 +297,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
 	omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
 							OMAP_DMA_DATA_TYPE_S16;
-	cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream,
+		&omap_mcbsp_dai_dma_params[id][substream->stream]);
 
 	if (mcbsp_data->configured) {
 		/* McBSP already configured by another stream */
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 25f19e4..b7f4f7e 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -150,7 +150,8 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 	int stream = substream->stream;
 	int channels, err, link_mask = 0;
 
-	cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream];
+	snd_soc_dai_set_dma_data(cpu_dai, substream,
+				 &omap_mcpdm_dai_dma_params[stream]);
 
 	channels = params_channels(params);
 	switch (channels) {
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index ba8acbb..1e52190 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -61,12 +61,11 @@ static void omap_pcm_dma_irq(int ch, u16 stat, void *data)
 	struct omap_runtime_data *prtd = runtime->private_data;
 	unsigned long flags;
 
-	if ((cpu_is_omap1510()) &&
-			(substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) {
+	if ((cpu_is_omap1510())) {
 		/*
 		 * OMAP1510 doesn't fully support DMA progress counter
 		 * and there is no software emulation implemented yet,
-		 * so have to maintain our own playback progress counter
+		 * so have to maintain our own progress counters
 		 * that can be used by omap_pcm_pointer() instead.
 		 */
 		spin_lock_irqsave(&prtd->lock, flags);
@@ -101,9 +100,11 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct omap_runtime_data *prtd = runtime->private_data;
-	struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data;
+	struct omap_pcm_dma_data *dma_data;
 	int err = 0;
 
+	dma_data = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+
 	/* return if this is a bufferless transfer e.g.
 	 * codec <--> BT codec or GSM modem -- lg FIXME */
 	if (!dma_data)
@@ -190,8 +191,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 	dma_params.frame_count	= runtime->periods;
 	omap_set_dma_params(prtd->dma_ch, &dma_params);
 
-	if ((cpu_is_omap1510()) &&
-			(substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
+	if ((cpu_is_omap1510()))
 		omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ |
 			      OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ);
 	else
@@ -249,14 +249,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
 	dma_addr_t ptr;
 	snd_pcm_uframes_t offset;
 
-	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+	if (cpu_is_omap1510()) {
+		offset = prtd->period_index * runtime->period_size;
+	} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		ptr = omap_get_dma_dst_pos(prtd->dma_ch);
 		offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
-	} else if (!(cpu_is_omap1510())) {
+	} else {
 		ptr = omap_get_dma_src_pos(prtd->dma_ch);
 		offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
-	} else
-		offset = prtd->period_index * runtime->period_size;
+	}
 
 	if (offset >= runtime->buffer_size)
 		offset = 0;
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d5fc52d..544fd95 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -122,10 +122,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 		ssp_disable(ssp);
 	}
 
-	if (cpu_dai->dma_data) {
-		kfree(cpu_dai->dma_data);
-		cpu_dai->dma_data = NULL;
-	}
+	kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
+	snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
+
 	return ret;
 }
 
@@ -142,10 +141,8 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
 		clk_disable(ssp->clk);
 	}
 
-	if (cpu_dai->dma_data) {
-		kfree(cpu_dai->dma_data);
-		cpu_dai->dma_data = NULL;
-	}
+	kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
+	snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
 }
 
 #ifdef CONFIG_PM
@@ -570,19 +567,23 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	u32 sspsp;
 	int width = snd_pcm_format_physical_width(params_format(params));
 	int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
+	struct pxa2xx_pcm_dma_params *dma_data;
+
+	dma_data = snd_soc_dai_get_dma_data(dai, substream);
 
 	/* generate correct DMA params */
-	if (cpu_dai->dma_data)
-		kfree(cpu_dai->dma_data);
+	kfree(dma_data);
 
 	/* Network mode with one active slot (ttsa == 1) can be used
 	 * to force 16-bit frame width on the wire (for S16_LE), even
 	 * with two channels. Use 16-bit DMA transfers for this case.
 	 */
-	cpu_dai->dma_data = ssp_get_dma_params(ssp,
+	dma_data = ssp_get_dma_params(ssp,
 			((chn == 2) && (ttsa != 1)) || (width == 32),
 			substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
 
+	snd_soc_dai_set_dma_data(dai, substream, dma_data);
+
 	/* we can only change the settings if the port is not in use */
 	if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
 		return 0;
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index e9ae7b3..d314115 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -122,11 +122,14 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct pxa2xx_pcm_dma_params *dma_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out;
+		dma_data = &pxa2xx_ac97_pcm_stereo_out;
 	else
-		cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in;
+		dma_data = &pxa2xx_ac97_pcm_stereo_in;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
 
 	return 0;
 }
@@ -137,11 +140,14 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct pxa2xx_pcm_dma_params *dma_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
+		dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
 	else
-		cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+		dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
 
 	return 0;
 }
@@ -156,7 +162,8 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		return -ENODEV;
 	else
-		cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in;
+		snd_soc_dai_set_dma_data(cpu_dai, substream,
+					 &pxa2xx_ac97_pcm_mic_mono_in);
 
 	return 0;
 }
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 6b8f655..c1a52757 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -164,6 +164,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct pxa2xx_pcm_dma_params *dma_data;
 
 	BUG_ON(IS_ERR(clk_i2s));
 	clk_enable(clk_i2s);
@@ -171,9 +172,11 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	pxa_i2s_wait();
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out;
+		dma_data = &pxa2xx_i2s_pcm_stereo_out;
 	else
-		cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in;
+		dma_data = &pxa2xx_i2s_pcm_stereo_in;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
 
 	/* is port used by another stream */
 	if (!(SACR0 & SACR0_ENB)) {
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index d38e395..adc7e6f 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -25,9 +25,11 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct pxa2xx_runtime_data *prtd = runtime->private_data;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct pxa2xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data;
+	struct pxa2xx_pcm_dma_params *dma;
 	int ret;
 
+	dma = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+
 	/* return if this is a bufferless transfer e.g.
 	 * codec <--> BT codec or GSM modem -- lg FIXME */
 	if (!dma)
diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c
index ee8ed9d7..ecf4fd0 100644
--- a/sound/soc/s3c24xx/s3c-ac97.c
+++ b/sound/soc/s3c24xx/s3c-ac97.c
@@ -224,11 +224,14 @@ static int s3c_ac97_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct s3c_dma_params *dma_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		cpu_dai->dma_data = &s3c_ac97_pcm_out;
+		dma_data = &s3c_ac97_pcm_out;
 	else
-		cpu_dai->dma_data = &s3c_ac97_pcm_in;
+		dma_data = &s3c_ac97_pcm_in;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
 
 	return 0;
 }
@@ -238,8 +241,8 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
 {
 	u32 ac_glbctrl;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	int channel = ((struct s3c_dma_params *)
-		  rtd->dai->cpu_dai->dma_data)->channel;
+	struct s3c_dma_params *dma_data =
+		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
 	ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL);
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
@@ -265,7 +268,7 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
 
 	writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL);
 
-	s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+	s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
 
 	return 0;
 }
@@ -280,7 +283,7 @@ static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		return -ENODEV;
 	else
-		cpu_dai->dma_data = &s3c_ac97_mic_in;
+		snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in);
 
 	return 0;
 }
@@ -290,8 +293,8 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream,
 {
 	u32 ac_glbctrl;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	int channel = ((struct s3c_dma_params *)
-		  rtd->dai->cpu_dai->dma_data)->channel;
+	struct s3c_dma_params *dma_data =
+		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
 	ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL);
 	ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK;
@@ -311,7 +314,7 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream,
 
 	writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL);
 
-	s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+	s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
 
 	return 0;
 }
diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c
index 7725e26..1b61c23 100644
--- a/sound/soc/s3c24xx/s3c-dma.c
+++ b/sound/soc/s3c24xx/s3c-dma.c
@@ -145,10 +145,12 @@ static int s3c_dma_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s3c24xx_runtime_data *prtd = runtime->private_data;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data;
 	unsigned long totbytes = params_buffer_bytes(params);
+	struct s3c_dma_params *dma =
+		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 	int ret = 0;
 
+
 	pr_debug("Entered %s\n", __func__);
 
 	/* return if this is a bufferless transfer e.g.
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index e994d83..8851594 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -339,14 +339,17 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai_link *dai = rtd->dai;
 	struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai);
+	struct s3c_dma_params *dma_data;
 	u32 iismod;
 
 	pr_debug("Entered %s\n", __func__);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		dai->cpu_dai->dma_data = i2s->dma_playback;
+		dma_data = i2s->dma_playback;
 	else
-		dai->cpu_dai->dma_data = i2s->dma_capture;
+		dma_data = i2s->dma_capture;
+
+	snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data);
 
 	/* Working copies of register */
 	iismod = readl(i2s->regs + S3C2412_IISMOD);
@@ -394,8 +397,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
 	unsigned long irqs;
 	int ret = 0;
-	int channel = ((struct s3c_dma_params *)
-		  rtd->dai->cpu_dai->dma_data)->channel;
+	struct s3c_dma_params *dma_data =
+		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
 	pr_debug("Entered %s\n", __func__);
 
@@ -431,7 +434,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 		 * of the auto reload mechanism of S3C24XX.
 		 * This call won't bother S3C64XX.
 		 */
-		s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+		s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
 
 		break;
 
diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c
index a98f40c..326f0a9 100644
--- a/sound/soc/s3c24xx/s3c-pcm.c
+++ b/sound/soc/s3c24xx/s3c-pcm.c
@@ -178,6 +178,7 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai_link *dai = rtd->dai;
 	struct s3c_pcm_info *pcm = to_info(dai->cpu_dai);
+	struct s3c_dma_params *dma_data;
 	void __iomem *regs = pcm->regs;
 	struct clk *clk;
 	int sclk_div, sync_div;
@@ -187,9 +188,11 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream,
 	dev_dbg(pcm->dev, "Entered %s\n", __func__);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		dai->cpu_dai->dma_data = pcm->dma_playback;
+		dma_data = pcm->dma_playback;
 	else
-		dai->cpu_dai->dma_data = pcm->dma_capture;
+		dma_data = pcm->dma_capture;
+
+	snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data);
 
 	/* Strictly check for sample size */
 	switch (params_format(params)) {
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 0bc5950..c3ac890 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -242,14 +242,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct s3c_dma_params *dma_data;
 	u32 iismod;
 
 	pr_debug("Entered %s\n", __func__);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out;
+		dma_data = &s3c24xx_i2s_pcm_stereo_out;
 	else
-		rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_in;
+		dma_data = &s3c24xx_i2s_pcm_stereo_in;
+
+	snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data);
 
 	/* Working copies of register */
 	iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -258,13 +261,11 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
 		iismod &= ~S3C2410_IISMOD_16BIT;
-		((struct s3c_dma_params *)
-		  rtd->dai->cpu_dai->dma_data)->dma_size = 1;
+		dma_data->dma_size = 1;
 		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		iismod |= S3C2410_IISMOD_16BIT;
-		((struct s3c_dma_params *)
-		  rtd->dai->cpu_dai->dma_data)->dma_size = 2;
+		dma_data->dma_size = 2;
 		break;
 	default:
 		return -EINVAL;
@@ -280,8 +281,8 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 {
 	int ret = 0;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	int channel = ((struct s3c_dma_params *)
-		  rtd->dai->cpu_dai->dma_data)->channel;
+	struct s3c_dma_params *dma_data =
+		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
 	pr_debug("Entered %s\n", __func__);
 
@@ -300,7 +301,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 		else
 			s3c24xx_snd_txctrl(1);
 
-		s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+		s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index 0664fac..5b9ac17 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -519,7 +519,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev)
 
 	s6000_i2s_dai.dev = &pdev->dev;
 	s6000_i2s_dai.private_data = dev;
-	s6000_i2s_dai.dma_data = &dev->dma_params;
+	s6000_i2s_dai.capture.dma_data = &dev->dma_params;
+	s6000_i2s_dai.playback.dma_data = &dev->dma_params;
 
 	dev->sifbase = sifmem->start;
 	dev->scbbase = mmio;
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 1d61109..9c7f7f0 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -58,13 +58,15 @@ static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s6000_runtime_data *prtd = runtime->private_data;
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	int channel;
 	unsigned int period_size;
 	unsigned int dma_offset;
 	dma_addr_t dma_pos;
 	dma_addr_t src, dst;
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	period_size = snd_pcm_lib_period_bytes(substream);
 	dma_offset = prtd->period * period_size;
 	dma_pos = runtime->dma_addr + dma_offset;
@@ -101,7 +103,8 @@ static irqreturn_t s6000_pcm_irq(int irq, void *data)
 {
 	struct snd_pcm *pcm = data;
 	struct snd_soc_pcm_runtime *runtime = pcm->private_data;
-	struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *params =
+		snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
 	struct s6000_runtime_data *prtd;
 	unsigned int has_xrun;
 	int i, ret = IRQ_NONE;
@@ -172,11 +175,13 @@ static int s6000_pcm_start(struct snd_pcm_substream *substream)
 {
 	struct s6000_runtime_data *prtd = substream->runtime->private_data;
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	unsigned long flags;
 	int srcinc;
 	u32 dma;
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	spin_lock_irqsave(&prtd->lock, flags);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -212,10 +217,12 @@ static int s6000_pcm_stop(struct snd_pcm_substream *substream)
 {
 	struct s6000_runtime_data *prtd = substream->runtime->private_data;
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	unsigned long flags;
 	u32 channel;
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		channel = par->dma_out;
 	else
@@ -236,9 +243,11 @@ static int s6000_pcm_stop(struct snd_pcm_substream *substream)
 static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	int ret;
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	ret = par->trigger(substream, cmd, 0);
 	if (ret < 0)
 		return ret;
@@ -275,13 +284,15 @@ static int s6000_pcm_prepare(struct snd_pcm_substream *substream)
 static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s6000_runtime_data *prtd = runtime->private_data;
 	unsigned long flags;
 	unsigned int offset;
 	dma_addr_t count;
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	spin_lock_irqsave(&prtd->lock, flags);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -305,11 +316,12 @@ static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream)
 static int s6000_pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s6000_runtime_data *prtd;
 	int ret;
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
 	snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware);
 
 	ret = snd_pcm_hw_constraint_step(runtime, 0,
@@ -364,7 +376,7 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	int ret;
 	ret = snd_pcm_lib_malloc_pages(substream,
 				       params_buffer_bytes(hw_params));
@@ -373,6 +385,8 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	if (par->same_rate) {
 		spin_lock(&par->lock);
 		if (par->rate == -1 ||
@@ -392,7 +406,8 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream,
 static int s6000_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par =
+		snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
 
 	spin_lock(&par->lock);
 	par->in_use &= ~(1 << substream->stream);
@@ -417,7 +432,8 @@ static struct snd_pcm_ops s6000_pcm_ops = {
 static void s6000_pcm_free(struct snd_pcm *pcm)
 {
 	struct snd_soc_pcm_runtime *runtime = pcm->private_data;
-	struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *params =
+		snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
 
 	free_irq(params->irq, pcm);
 	snd_pcm_lib_preallocate_free_for_all(pcm);
@@ -429,9 +445,11 @@ static int s6000_pcm_new(struct snd_card *card,
 			 struct snd_soc_dai *dai, struct snd_pcm *pcm)
 {
 	struct snd_soc_pcm_runtime *runtime = pcm->private_data;
-	struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *params;
 	int res;
 
+	params = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	if (!card->dev->dma_mask)
 		card->dev->dma_mask = &s6000_pcm_dmamask;
 	if (!card->dev->coherent_dma_mask)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2320153..ad7f952 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1549,7 +1549,8 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 			mutex_unlock(&codec->mutex);
 			return ret;
 		}
-		if (card->dai_link[i].codec_dai->ac97_control) {
+		/* Check for codec->ac97 to handle the ac97.c fun */
+		if (card->dai_link[i].codec_dai->ac97_control && codec->ac97) {
 			snd_ac97_dev_add_pdata(codec->ac97,
 				card->dai_link[i].cpu_dai->ac97_pdata);
 		}
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index 612e18b..0ec20b6 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -254,3 +254,4 @@ module_exit(txx9aclc_ac97_exit);
 MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
 MODULE_DESCRIPTION("TXx9 ACLC AC97 driver");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:txx9aclc-ac97");
diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c
index 3175de9..95b17f7 100644
--- a/sound/soc/txx9/txx9aclc-generic.c
+++ b/sound/soc/txx9/txx9aclc-generic.c
@@ -96,3 +96,4 @@ module_exit(txx9aclc_generic_exit);
 MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
 MODULE_DESCRIPTION("Generic TXx9 ACLC ALSA SoC audio driver");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:txx9aclc-generic");
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 2c59afd..9e28b20 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -986,6 +986,8 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream)
 	DEFINE_WAIT(wait);
 	long timeout = msecs_to_jiffies(50);
 
+	if (ep->umidi->disconnected)
+		return;
 	/*
 	 * The substream buffer is empty, but some data might still be in the
 	 * currently active URBs, so we have to wait for those to complete.
@@ -1123,14 +1125,21 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi,
  * Frees an output endpoint.
  * May be called when ep hasn't been initialized completely.
  */
-static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint* ep)
+static void snd_usbmidi_out_endpoint_clear(struct snd_usb_midi_out_endpoint *ep)
 {
 	unsigned int i;
 
 	for (i = 0; i < OUTPUT_URBS; ++i)
-		if (ep->urbs[i].urb)
+		if (ep->urbs[i].urb) {
 			free_urb_and_buffer(ep->umidi, ep->urbs[i].urb,
 					    ep->max_transfer);
+			ep->urbs[i].urb = NULL;
+		}
+}
+
+static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint *ep)
+{
+	snd_usbmidi_out_endpoint_clear(ep);
 	kfree(ep);
 }
 
@@ -1262,15 +1271,18 @@ void snd_usbmidi_disconnect(struct list_head* p)
 				usb_kill_urb(ep->out->urbs[j].urb);
 			if (umidi->usb_protocol_ops->finish_out_endpoint)
 				umidi->usb_protocol_ops->finish_out_endpoint(ep->out);
+			ep->out->active_urbs = 0;
+			if (ep->out->drain_urbs) {
+				ep->out->drain_urbs = 0;
+				wake_up(&ep->out->drain_wait);
+			}
 		}
 		if (ep->in)
 			for (j = 0; j < INPUT_URBS; ++j)
 				usb_kill_urb(ep->in->urbs[j]);
 		/* free endpoints here; later call can result in Oops */
-		if (ep->out) {
-			snd_usbmidi_out_endpoint_delete(ep->out);
-			ep->out = NULL;
-		}
+		if (ep->out)
+			snd_usbmidi_out_endpoint_clear(ep->out);
 		if (ep->in) {
 			snd_usbmidi_in_endpoint_delete(ep->in);
 			ep->in = NULL;