26 files changed, 404 insertions, 283 deletions

diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index 586965f9605f..7a437da05646 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -768,7 +768,7 @@ static int check_codec(struct aoa_codec *codec,
 				"required property %s not present\n", propname);
 			return -ENODEV;
 		}
-		if (*ref != codec->node->linux_phandle) {
+		if (*ref != codec->node->phandle) {
 			printk(KERN_INFO "snd-aoa-fabric-layout: "
 				"%s doesn't match!\n", propname);
 			return -ENODEV;
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 7872a02f6ca9..563d1967a0ad 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -468,5 +468,5 @@ static void __exit alsa_sound_exit(void)
 	unregister_chrdev(major, "alsa");
 }
 
-module_init(alsa_sound_init)
-module_exit(alsa_sound_exit)
+subsys_initcall(alsa_sound_init);
+module_exit(alsa_sound_exit);
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index 7fe12264ff80..0c164e5e4322 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -93,7 +93,7 @@ static int snd_oss_kernel_minor(int type, struct snd_card *card, int dev)
 	default:
 		return -EINVAL;
 	}
-	if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OSS_MINORS))
+	if (minor < 0 || minor >= SNDRV_OSS_MINORS)
 		return -EINVAL;
 	return minor;
 }
diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c
index 4191acccbcdb..c1070e33b32f 100644
--- a/sound/oss/au1550_ac97.c
+++ b/sound/oss/au1550_ac97.c
@@ -614,7 +614,8 @@ start_adc(struct au1550_state *s)
 	/* Put two buffers on the ring to get things started.
 	*/
 	for (i=0; i<2; i++) {
-		au1xxx_dbdma_put_dest(db->dmanr, db->nextIn, db->dma_fragsize);
+		au1xxx_dbdma_put_dest(db->dmanr, virt_to_phys(db->nextIn),
+				db->dma_fragsize, DDMA_FLAGS_IE);
 
 		db->nextIn += db->dma_fragsize;
 		if (db->nextIn >= db->rawbuf + db->dmasize)
@@ -732,8 +733,9 @@ static void dac_dma_interrupt(int irq, void *dev_id)
 	db->dma_qcount--;
 
 	if (db->count >= db->fragsize) {
-		if (au1xxx_dbdma_put_source(db->dmanr, db->nextOut,
-							db->fragsize) == 0) {
+		if (au1xxx_dbdma_put_source(db->dmanr,
+				virt_to_phys(db->nextOut), db->fragsize,
+				DDMA_FLAGS_IE) == 0) {
 			err("qcount < 2 and no ring room!");
 		}
 		db->nextOut += db->fragsize;
@@ -777,7 +779,8 @@ static void adc_dma_interrupt(int irq, void *dev_id)
 
 	/* Put a new empty buffer on the destination DMA.
 	*/
-	au1xxx_dbdma_put_dest(dp->dmanr, dp->nextIn, dp->dma_fragsize);
+	au1xxx_dbdma_put_dest(dp->dmanr, virt_to_phys(dp->nextIn),
+			      dp->dma_fragsize, DDMA_FLAGS_IE);
 
 	dp->nextIn += dp->dma_fragsize;
 	if (dp->nextIn >= dp->rawbuf + dp->dmasize)
@@ -1177,8 +1180,9 @@ au1550_write(struct file *file, const char *buffer, size_t count, loff_t * ppos)
 		 * we know the dma has stopped.
 		 */
 		while ((db->dma_qcount < 2) && (db->count >= db->fragsize)) {
-			if (au1xxx_dbdma_put_source(db->dmanr, db->nextOut,
-							db->fragsize) == 0) {
+			if (au1xxx_dbdma_put_source(db->dmanr,
+				virt_to_phys(db->nextOut), db->fragsize,
+				DDMA_FLAGS_IE) == 0) {
 				err("qcount < 2 and no ring room!");
 			}
 			db->nextOut += db->fragsize;
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index cb65bd0dd35b..459c1f62783b 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -166,18 +166,7 @@ static void ct_unmap_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm)
 
 static unsigned long atc_get_ptp_phys(struct ct_atc *atc, int index)
 {
-	struct ct_vm *vm;
-	void *kvirt_addr;
-	unsigned long phys_addr;
-
-	vm = atc->vm;
-	kvirt_addr = vm->get_ptp_virt(vm, index);
-	if (kvirt_addr == NULL)
-		phys_addr = (~0UL);
-	else
-		phys_addr = virt_to_phys(kvirt_addr);
-
-	return phys_addr;
+	return atc->vm->get_ptp_phys(atc->vm, index);
 }
 
 static unsigned int convert_format(snd_pcm_format_t snd_format)
@@ -1669,7 +1658,7 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
 	}
 
 	/* Set up device virtual memory management object */
-	err = ct_vm_create(&atc->vm);
+	err = ct_vm_create(&atc->vm, pci);
 	if (err < 0)
 		goto error1;
 
diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c
index 6b78752e9503..65da6e466f80 100644
--- a/sound/pci/ctxfi/ctvmem.c
+++ b/sound/pci/ctxfi/ctvmem.c
@@ -138,7 +138,7 @@ ct_vm_map(struct ct_vm *vm, struct snd_pcm_substream *substream, int size)
 		return NULL;
 	}
 
-	ptp = vm->ptp[0];
+	ptp = (unsigned long *)vm->ptp[0].area;
 	pte_start = (block->addr >> CT_PAGE_SHIFT);
 	pages = block->size >> CT_PAGE_SHIFT;
 	for (i = 0; i < pages; i++) {
@@ -158,25 +158,25 @@ static void ct_vm_unmap(struct ct_vm *vm, struct ct_vm_block *block)
 }
 
 /* *
- * return the host (kmalloced) addr of the @index-th device
- * page talbe page on success, or NULL on failure.
- * The first returned NULL indicates the termination.
+ * return the host physical addr of the @index-th device
+ * page table page on success, or ~0UL on failure.
+ * The first returned ~0UL indicates the termination.
  * */
-static void *
-ct_get_ptp_virt(struct ct_vm *vm, int index)
+static dma_addr_t
+ct_get_ptp_phys(struct ct_vm *vm, int index)
 {
-	void *addr;
+	dma_addr_t addr;
 
-	addr = (index >= CT_PTP_NUM) ? NULL : vm->ptp[index];
+	addr = (index >= CT_PTP_NUM) ? ~0UL : vm->ptp[index].addr;
 
 	return addr;
 }
 
-int ct_vm_create(struct ct_vm **rvm)
+int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci)
 {
 	struct ct_vm *vm;
 	struct ct_vm_block *block;
-	int i;
+	int i, err = 0;
 
 	*rvm = NULL;
 
@@ -188,23 +188,21 @@ int ct_vm_create(struct ct_vm **rvm)
 
 	/* Allocate page table pages */
 	for (i = 0; i < CT_PTP_NUM; i++) {
-		vm->ptp[i] = kmalloc(PAGE_SIZE, GFP_KERNEL);
-		if (!vm->ptp[i])
+		err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+					  snd_dma_pci_data(pci),
+					  PAGE_SIZE, &vm->ptp[i]);
+		if (err < 0)
 			break;
 	}
-	if (!i) {
+	if (err < 0) {
 		/* no page table pages are allocated */
-		kfree(vm);
+		ct_vm_destroy(vm);
 		return -ENOMEM;
 	}
 	vm->size = CT_ADDRS_PER_PAGE * i;
-	/* Initialise remaining ptps */
-	for (; i < CT_PTP_NUM; i++)
-		vm->ptp[i] = NULL;
-
 	vm->map = ct_vm_map;
 	vm->unmap = ct_vm_unmap;
-	vm->get_ptp_virt = ct_get_ptp_virt;
+	vm->get_ptp_phys = ct_get_ptp_phys;
 	INIT_LIST_HEAD(&vm->unused);
 	INIT_LIST_HEAD(&vm->used);
 	block = kzalloc(sizeof(*block), GFP_KERNEL);
@@ -242,7 +240,7 @@ void ct_vm_destroy(struct ct_vm *vm)
 
 	/* free allocated page table pages */
 	for (i = 0; i < CT_PTP_NUM; i++)
-		kfree(vm->ptp[i]);
+		snd_dma_free_pages(&vm->ptp[i]);
 
 	vm->size = 0;
 
diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h
index 01e4fd0386a3..b23adfca4de6 100644
--- a/sound/pci/ctxfi/ctvmem.h
+++ b/sound/pci/ctxfi/ctvmem.h
@@ -22,6 +22,8 @@
 
 #include <linux/mutex.h>
 #include <linux/list.h>
+#include <linux/pci.h>
+#include <sound/memalloc.h>
 
 /* The chip can handle the page table of 4k pages
  * (emu20k1 can handle even 8k pages, but we don't use it right now)
@@ -41,7 +43,7 @@ struct snd_pcm_substream;
 
 /* Virtual memory management object for card device */
 struct ct_vm {
-	void *ptp[CT_PTP_NUM];		/* Device page table pages */
+	struct snd_dma_buffer ptp[CT_PTP_NUM];	/* Device page table pages */
 	unsigned int size;		/* Available addr space in bytes */
 	struct list_head unused;	/* List of unused blocks */
 	struct list_head used;		/* List of used blocks */
@@ -52,10 +54,10 @@ struct ct_vm {
 				   int size);
 	/* Unmap device logical addr area. */
 	void (*unmap)(struct ct_vm *, struct ct_vm_block *block);
-	void *(*get_ptp_virt)(struct ct_vm *vm, int index);
+	dma_addr_t (*get_ptp_phys)(struct ct_vm *vm, int index);
 };
 
-int ct_vm_create(struct ct_vm **rvm);
+int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci);
 void ct_vm_destroy(struct ct_vm *vm);
 
 #endif /* CTVMEM_H */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ec9c348336cc..ff6da6f386d1 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -426,6 +426,7 @@ struct azx {
 
 	/* flags */
 	int position_fix;
+	int poll_count;
 	unsigned int running :1;
 	unsigned int initialized :1;
 	unsigned int single_cmd :1;
@@ -506,7 +507,7 @@ static char *driver_short_names[] __devinitdata = {
 #define get_azx_dev(substream) (substream->runtime->private_data)
 
 static int azx_acquire_irq(struct azx *chip, int do_disconnect);
-
+static int azx_send_cmd(struct hda_bus *bus, unsigned int val);
 /*
  * Interface for HD codec
  */
@@ -664,11 +665,12 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 {
 	struct azx *chip = bus->private_data;
 	unsigned long timeout;
+	int do_poll = 0;
 
  again:
 	timeout = jiffies + msecs_to_jiffies(1000);
 	for (;;) {
-		if (chip->polling_mode) {
+		if (chip->polling_mode || do_poll) {
 			spin_lock_irq(&chip->reg_lock);
 			azx_update_rirb(chip);
 			spin_unlock_irq(&chip->reg_lock);
@@ -676,6 +678,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		if (!chip->rirb.cmds[addr]) {
 			smp_rmb();
 			bus->rirb_error = 0;
+
+			if (!do_poll)
+				chip->poll_count = 0;
 			return chip->rirb.res[addr]; /* the last value */
 		}
 		if (time_after(jiffies, timeout))
@@ -688,6 +693,16 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		}
 	}
 
+	if (!chip->polling_mode && chip->poll_count < 2) {
+		snd_printdd(SFX "azx_get_response timeout, "
+			   "polling the codec once: last cmd=0x%08x\n",
+			   chip->last_cmd[addr]);
+		do_poll = 1;
+		chip->poll_count++;
+		goto again;
+	}
+
+
 	if (!chip->polling_mode) {
 		snd_printk(KERN_WARNING SFX "azx_get_response timeout, "
 			   "switching to polling mode: last cmd=0x%08x\n",
@@ -1878,6 +1893,9 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
 
 	if (!bdl_pos_adj[chip->dev_index])
 		return 1; /* no delayed ack */
+	if (WARN_ONCE(!azx_dev->period_bytes,
+		      "hda-intel: zero azx_dev->period_bytes"))
+		return 0; /* this shouldn't happen! */
 	if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
 		return 0; /* NG - it's below the period boundary */
 	return 1; /* OK, it's fine */
@@ -2043,7 +2061,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect)
 {
 	if (request_irq(chip->pci->irq, azx_interrupt,
 			chip->msi ? 0 : IRQF_SHARED,
-			"HDA Intel", chip)) {
+			"hda_intel", chip)) {
 		printk(KERN_ERR "hda-intel: unable to grab IRQ %d, "
 		       "disabling device\n", chip->pci->irq);
 		if (do_disconnect)
@@ -2332,6 +2350,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
  */
 static struct snd_pci_quirk msi_black_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
+	SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */
 	{}
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e3caa78ccd54..da34095c707f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1093,6 +1093,16 @@ static void alc889_coef_init(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010);
 }
 
+/* turn on/off EAPD control (only if available) */
+static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on)
+{
+	if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+		return;
+	if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+				    on ? 2 : 0);
+}
+
 static void alc_auto_init_amp(struct hda_codec *codec, int type)
 {
 	unsigned int tmp;
@@ -1110,25 +1120,22 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
 	case ALC_INIT_DEFAULT:
 		switch (codec->vendor_id) {
 		case 0x10ec0260:
-			snd_hda_codec_write(codec, 0x0f, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
-			snd_hda_codec_write(codec, 0x10, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
+			set_eapd(codec, 0x0f, 1);
+			set_eapd(codec, 0x10, 1);
 			break;
 		case 0x10ec0262:
 		case 0x10ec0267:
 		case 0x10ec0268:
 		case 0x10ec0269:
+		case 0x10ec0270:
 		case 0x10ec0272:
 		case 0x10ec0660:
 		case 0x10ec0662:
 		case 0x10ec0663:
 		case 0x10ec0862:
 		case 0x10ec0889:
-			snd_hda_codec_write(codec, 0x14, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
-			snd_hda_codec_write(codec, 0x15, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
+			set_eapd(codec, 0x14, 1);
+			set_eapd(codec, 0x15, 1);
 			break;
 		}
 		switch (codec->vendor_id) {
@@ -1230,6 +1237,8 @@ static void alc_init_auto_mic(struct hda_codec *codec)
 			return; /* invalid entry */
 		}
 	}
+	if (!ext || !fixed)
+		return;
 	if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
 		return; /* no unsol support */
 	snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n",
@@ -1834,10 +1843,8 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static void alc889_power_eapd(struct hda_codec *codec, int power)
 {
-	snd_hda_codec_write(codec, 0x14, 0,
-			    AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
-	snd_hda_codec_write(codec, 0x15, 0,
-			    AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
+	set_eapd(codec, 0x14, power);
+	set_eapd(codec, 0x15, power);
 }
 #endif
 
@@ -4812,6 +4819,49 @@ static void fixup_automic_adc(struct hda_codec *codec)
 	spec->auto_mic = 0; /* disable auto-mic to be sure */
 }
 
+/* choose the ADC/MUX containing the input pin and initialize the setup */
+static void fixup_single_adc(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t pin;
+	int i;
+
+	/* search for the input pin; there must be only one */
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		if (spec->autocfg.input_pins[i]) {
+			pin = spec->autocfg.input_pins[i];
+			break;
+		}
+	}
+	if (!pin)
+		return;
+
+	/* set the default connection to that pin */
+	for (i = 0; i < spec->num_adc_nids; i++) {
+		hda_nid_t cap = spec->capsrc_nids ?
+			spec->capsrc_nids[i] : spec->adc_nids[i];
+		int idx;
+
+		idx = get_connection_index(codec, cap, pin);
+		if (idx < 0)
+			continue;
+		/* use only this ADC */
+		if (spec->capsrc_nids)
+			spec->capsrc_nids += i;
+		spec->adc_nids += i;
+		spec->num_adc_nids = 1;
+		/* select or unmute this route */
+		if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
+			snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
+						 HDA_AMP_MUTE, 0);
+		} else {
+			snd_hda_codec_write_cache(codec, cap, 0,
+					  AC_VERB_SET_CONNECT_SEL, idx);
+		}
+		return;
+	}
+}
+
 static void set_capture_mixer(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -4824,14 +4874,15 @@ static void set_capture_mixer(struct hda_codec *codec)
 		  alc_capture_mixer3 },
 	};
 	if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
-		int mux;
-		if (spec->auto_mic) {
-			mux = 0;
+		int mux = 0;
+		if (spec->auto_mic)
 			fixup_automic_adc(codec);
-		} else if (spec->input_mux && spec->input_mux->num_items > 1)
-			mux = 1;
-		else
-			mux = 0;
+		else if (spec->input_mux) {
+			if (spec->input_mux->num_items > 1)
+				mux = 1;
+			else if (spec->input_mux->num_items == 1)
+				fixup_single_adc(codec);
+		}
 		spec->cap_mixer = caps[mux][spec->num_adc_nids - 1];
 	}
 }
@@ -7094,8 +7145,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
 	HDA_BIND_MUTE   ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
 	HDA_BIND_MUTE   ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
-	HDA_BIND_MUTE   ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE   ("Headphone Playback Switch", 0x0f, 0x02, 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("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
@@ -7496,6 +7547,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
 	/* Front Mic pin: input vref at 80% */
 	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
@@ -7680,6 +7732,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 }
 
+static void alc885_mb5_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x14, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+	snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+
+}
+
+static void alc885_mb5_unsol_event(struct hda_codec *codec,
+				    unsigned int res)
+{
+	/* Headphone insertion or removal. */
+	if ((res >> 26) == ALC880_HP_EVENT)
+		alc885_mb5_automute(codec);
+}
+
 static void alc885_imac91_automute(struct hda_codec *codec)
 {
  	unsigned int present;
@@ -9126,6 +9199,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.input_mux = &mb5_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
+		.unsol_event = alc885_mb5_unsol_event,
+		.init_hook = alc885_mb5_automute,
 	},
 	[ALC885_MACPRO] = {
 		.mixers = { alc882_macpro_mixer },
@@ -9403,6 +9478,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
 		.channel_mode = alc883_3ST_6ch_modes,
 		.need_dac_fix = 1,
+		.const_channel_count = 6,
 		.num_mux_defs =
 			ARRAY_SIZE(alc888_2_capture_sources),
 		.input_mux = alc888_2_capture_sources,
@@ -10307,7 +10383,7 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */
+	spec->autocfg.speaker_pins[0] = 0x14;
 }
 
 static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
@@ -11179,7 +11255,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
 }
 
 #define alc262_auto_create_input_ctls \
-	alc880_auto_create_input_ctls
+	alc882_auto_create_input_ctls
 
 /*
  * generic initialization of ADC, input mixers and output mixers
@@ -11718,9 +11794,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc262_hp_t5735_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_HP_RP5700] = {
 		.mixers = { alc262_hp_rp5700_mixer },
@@ -12471,6 +12547,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
 		dac = 0x02;
 		break;
 	case 0x15:
+	case 0x21:
 		dac = 0x03;
 		break;
 	default:
@@ -14855,6 +14932,8 @@ static int patch_alc861(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc861_pcm_digital_playback;
 	spec->stream_digital_capture = &alc861_pcm_digital_capture;
 
+	if (!spec->cap_mixer)
+		set_capture_mixer(codec);
 	set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
 
 	spec->vmaster_nid = 0x03;
@@ -17251,7 +17330,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
 	SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
 		      ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4),
+	SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
 	SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
 		      ALC662_3ST_6ch_DIG),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 2291a8396817..799ba2570902 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4730,6 +4730,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 	}
 }
 
+static int hp_blike_system(u32 subsystem_id);
+
+static void set_hp_led_gpio(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	switch (codec->vendor_id) {
+	case 0x111d7608:
+		/* GPIO 0 */
+		spec->gpio_led = 0x01;
+		break;
+	case 0x111d7600:
+	case 0x111d7601:
+	case 0x111d7602:
+	case 0x111d7603:
+		/* GPIO 3 */
+		spec->gpio_led = 0x08;
+		break;
+	}
+}
+
 /*
  * This method searches for the mute LED GPIO configuration
  * provided as OEM string in SMBIOS. The format of that string
@@ -4741,6 +4761,14 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
  *
  * So, HP B-series like systems may have HP_Mute_LED_0 (current models)
  * or  HP_Mute_LED_0_3 (future models) OEM SMBIOS strings
+ *
+ *
+ * The dv-series laptops don't seem to have the HP_Mute_LED* strings in
+ * SMBIOS - at least the ones I have seen do not have them - which include
+ * my own system (HP Pavilion dv6-1110ax) and my cousin's
+ * HP Pavilion dv9500t CTO.
+ * Need more information on whether it is true across the entire series.
+ * -- kunal
  */
 static int find_mute_led_gpio(struct hda_codec *codec)
 {
@@ -4751,28 +4779,27 @@ static int find_mute_led_gpio(struct hda_codec *codec)
 		while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
 								NULL, dev))) {
 			if (sscanf(dev->name, "HP_Mute_LED_%d_%d",
-			      &spec->gpio_led_polarity,
-			      &spec->gpio_led) == 2) {
+				  &spec->gpio_led_polarity,
+				  &spec->gpio_led) == 2) {
 				spec->gpio_led = 1 << spec->gpio_led;
 				return 1;
 			}
 			if (sscanf(dev->name, "HP_Mute_LED_%d",
-			      &spec->gpio_led_polarity) == 1) {
-				switch (codec->vendor_id) {
-				case 0x111d7608:
-					/* GPIO 0 */
-					spec->gpio_led = 0x01;
-					return 1;
-				case 0x111d7600:
-				case 0x111d7601:
-				case 0x111d7602:
-				case 0x111d7603:
-					/* GPIO 3 */
-					spec->gpio_led = 0x08;
-					return 1;
-				}
+				  &spec->gpio_led_polarity) == 1) {
+				set_hp_led_gpio(codec);
+				return 1;
 			}
 		}
+
+		/*
+		 * Fallback case - if we don't find the DMI strings,
+		 * we statically set the GPIO - if not a B-series system.
+		 */
+		if (!hp_blike_system(codec->subsystem_id)) {
+			set_hp_led_gpio(codec);
+			spec->gpio_led_polarity = 1;
+			return 1;
+		}
 	}
 	return 0;
 }
@@ -5548,6 +5575,8 @@ again:
 	spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
 	spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e);
 
+	snd_printdd("Found board config: %d\n", spec->board_config);
+
 	switch (spec->board_config) {
 	case STAC_HP_M4:
 		/* enable internal microphone */
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 765d7bd4c3d4..9e66f6d306f8 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -703,11 +703,13 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho
 {
 	unsigned char nvol;
 
-	if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE))
+	if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE)) {
 		nvol = 0;
-	else
+	} else {
 		nvol = ((vol % WM_VOL_CNT) * (master % WM_VOL_CNT)) /
 								WM_VOL_MAX;
+		nvol += 0x1b;
+	}
 
 	wm_put(ice, index, nvol);
 	wm_put_nocache(ice, index, 0x180 | nvol);
@@ -778,7 +780,7 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
 	for (ch = 0; ch < 2; ch++) {
 		unsigned int vol = ucontrol->value.integer.value[ch];
 		if (vol > WM_VOL_MAX)
-			continue;
+			vol = WM_VOL_MAX;
 		vol |= spec->master[ch] & WM_VOL_MUTE;
 		if (vol != spec->master[ch]) {
 			int dac;
@@ -834,8 +836,8 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *
 	for (i = 0; i < voices; i++) {
 		unsigned int vol = ucontrol->value.integer.value[i];
 		if (vol > WM_VOL_MAX)
-			continue;
-		vol |= spec->vol[ofs+i];
+			vol = WM_VOL_MAX;
+		vol |= spec->vol[ofs+i] & WM_VOL_MUTE;
 		if (vol != spec->vol[ofs+i]) {
 			spec->vol[ofs+i] = vol;
 			idx  = WM_DAC_ATTEN + ofs + i;
diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c
index 2e156467b814..b36679384b27 100644
--- a/sound/ppc/awacs.c
+++ b/sound/ppc/awacs.c
@@ -751,8 +751,8 @@ static void snd_pmac_awacs_suspend(struct snd_pmac *chip)
 
 static void snd_pmac_awacs_resume(struct snd_pmac *chip)
 {
-	if (machine_is_compatible("PowerBook3,1")
-	    || machine_is_compatible("PowerBook3,2")) {
+	if (of_machine_is_compatible("PowerBook3,1")
+	    || of_machine_is_compatible("PowerBook3,2")) {
 		msleep(100);
 		snd_pmac_awacs_write_reg(chip, 1,
 			chip->awacs_reg[1] & ~MASK_PAROUT);
@@ -780,16 +780,16 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip)
 }
 #endif /* CONFIG_PM */
 
-#define IS_PM7500 (machine_is_compatible("AAPL,7500") \
-		|| machine_is_compatible("AAPL,8500") \
-		|| machine_is_compatible("AAPL,9500"))
-#define IS_PM5500 (machine_is_compatible("AAPL,e411"))
-#define IS_BEIGE (machine_is_compatible("AAPL,Gossamer"))
-#define IS_IMAC1 (machine_is_compatible("PowerMac2,1"))
-#define IS_IMAC2 (machine_is_compatible("PowerMac2,2") \
-		|| machine_is_compatible("PowerMac4,1"))
-#define IS_G4AGP (machine_is_compatible("PowerMac3,1"))
-#define IS_LOMBARD (machine_is_compatible("PowerBook1,1"))
+#define IS_PM7500 (of_machine_is_compatible("AAPL,7500") \
+		|| of_machine_is_compatible("AAPL,8500") \
+		|| of_machine_is_compatible("AAPL,9500"))
+#define IS_PM5500 (of_machine_is_compatible("AAPL,e411"))
+#define IS_BEIGE (of_machine_is_compatible("AAPL,Gossamer"))
+#define IS_IMAC1 (of_machine_is_compatible("PowerMac2,1"))
+#define IS_IMAC2 (of_machine_is_compatible("PowerMac2,2") \
+		|| of_machine_is_compatible("PowerMac4,1"))
+#define IS_G4AGP (of_machine_is_compatible("PowerMac3,1"))
+#define IS_LOMBARD (of_machine_is_compatible("PowerBook1,1"))
 
 static int imac1, imac2;
 
diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c
index 0accfe49735b..1f72e1c786bf 100644
--- a/sound/ppc/burgundy.c
+++ b/sound/ppc/burgundy.c
@@ -582,7 +582,7 @@ static int snd_pmac_burgundy_detect_headphone(struct snd_pmac *chip)
 static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, int do_notify)
 {
 	if (chip->auto_mute) {
-		int imac = machine_is_compatible("iMac");
+		int imac = of_machine_is_compatible("iMac");
 		int reg, oreg;
 		reg = oreg = snd_pmac_burgundy_rcb(chip,
 				MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES);
@@ -620,7 +620,7 @@ static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, int do_noti
  */
 int __devinit snd_pmac_burgundy_init(struct snd_pmac *chip)
 {
-	int imac = machine_is_compatible("iMac");
+	int imac = of_machine_is_compatible("iMac");
 	int i, err;
 
 	/* Checks to see the chip is alive and kicking */
diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c
index 7bc492ee77ec..85081172403f 100644
--- a/sound/ppc/pmac.c
+++ b/sound/ppc/pmac.c
@@ -922,11 +922,11 @@ static void __devinit detect_byte_swap(struct snd_pmac *chip)
 	}
 
 	/* it seems the Pismo & iBook can't byte-swap in hardware. */
-	if (machine_is_compatible("PowerBook3,1") ||
-	    machine_is_compatible("PowerBook2,1"))
+	if (of_machine_is_compatible("PowerBook3,1") ||
+	    of_machine_is_compatible("PowerBook2,1"))
 		chip->can_byte_swap = 0 ;
 
-	if (machine_is_compatible("PowerBook2,1"))
+	if (of_machine_is_compatible("PowerBook2,1"))
 		chip->can_duplex = 0;
 }
 
@@ -959,11 +959,11 @@ static int __devinit snd_pmac_detect(struct snd_pmac *chip)
 	chip->control_mask = MASK_IEPC | MASK_IEE | 0x11; /* default */
 
 	/* check machine type */
-	if (machine_is_compatible("AAPL,3400/2400")
-	    || machine_is_compatible("AAPL,3500"))
+	if (of_machine_is_compatible("AAPL,3400/2400")
+	    || of_machine_is_compatible("AAPL,3500"))
 		chip->is_pbook_3400 = 1;
-	else if (machine_is_compatible("PowerBook1,1")
-		 || machine_is_compatible("AAPL,PowerBook1998"))
+	else if (of_machine_is_compatible("PowerBook1,1")
+		 || of_machine_is_compatible("AAPL,PowerBook1998"))
 		chip->is_pbook_G3 = 1;
 	chip->node = of_find_node_by_name(NULL, "awacs");
 	sound = of_node_get(chip->node);
@@ -1033,8 +1033,8 @@ static int __devinit snd_pmac_detect(struct snd_pmac *chip)
 	}
 	if (of_device_is_compatible(sound, "tumbler")) {
 		chip->model = PMAC_TUMBLER;
-		chip->can_capture = machine_is_compatible("PowerMac4,2")
-				|| machine_is_compatible("PowerBook4,1");
+		chip->can_capture = of_machine_is_compatible("PowerMac4,2")
+				|| of_machine_is_compatible("PowerBook4,1");
 		chip->can_duplex = 0;
 		// chip->can_byte_swap = 0; /* FIXME: check this */
 		chip->num_freqs = ARRAY_SIZE(tumbler_freqs);
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
index 410a893aa66b..4b67140fdec3 100644
--- a/sound/soc/au1x/Kconfig
+++ b/sound/soc/au1x/Kconfig
@@ -22,11 +22,13 @@ config SND_SOC_AU1XPSC_AC97
 ##
 ## Boards
 ##
-config SND_SOC_SAMPLE_PSC_AC97
-	tristate "Sample Au12x0/Au1550 PSC AC97 sound machine"
+config SND_SOC_DB1200
+	tristate "DB1200 AC97+I2S audio support"
 	depends on SND_SOC_AU1XPSC
 	select SND_SOC_AU1XPSC_AC97
 	select SND_SOC_AC97_CODEC
+	select SND_SOC_AU1XPSC_I2S
+	select SND_SOC_WM8731
 	help
-	  This is a sample AC97 sound machine for use in Au12x0/Au1550
-	  based systems which have audio on PSC1 (e.g. Db1200 demoboard).
+	  Select this option to enable audio (AC97 or I2S) on the
+	  Alchemy/AMD/RMI DB1200 demoboard.
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
index 6c6950b8003a..16873076e8c4 100644
--- a/sound/soc/au1x/Makefile
+++ b/sound/soc/au1x/Makefile
@@ -8,6 +8,6 @@ obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o
 obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o
 
 # Boards
-snd-soc-sample-ac97-objs := sample-ac97.o
+snd-soc-db1200-objs := db1200.o
 
-obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o
+obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
new file mode 100644
index 000000000000..cdf7be1b9b91
--- /dev/null
+++ b/sound/soc/au1x/db1200.c
@@ -0,0 +1,141 @@
+/*
+ * DB1200 ASoC audio fabric support code.
+ *
+ * (c) 2008-9 Manuel Lauss <manuel.lauss@gmail.com>
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+#include <asm/mach-au1x00/au1xxx_dbdma.h>
+#include <asm/mach-db1x00/bcsr.h>
+
+#include "../codecs/ac97.h"
+#include "../codecs/wm8731.h"
+#include "psc.h"
+
+/*-------------------------  AC97 PART  ---------------------------*/
+
+static struct snd_soc_dai_link db1200_ac97_dai = {
+	.name		= "AC97",
+	.stream_name	= "AC97 HiFi",
+	.cpu_dai	= &au1xpsc_ac97_dai,
+	.codec_dai	= &ac97_dai,
+};
+
+static struct snd_soc_card db1200_ac97_machine = {
+	.name		= "DB1200_AC97",
+	.dai_link	= &db1200_ac97_dai,
+	.num_links	= 1,
+	.platform	= &au1xpsc_soc_platform,
+};
+
+static struct snd_soc_device db1200_ac97_devdata = {
+	.card		= &db1200_ac97_machine,
+	.codec_dev	= &soc_codec_dev_ac97,
+};
+
+/*-------------------------  I2S PART  ---------------------------*/
+
+static int db1200_i2s_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* WM8731 has its own 12MHz crystal */
+	snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
+				12000000, SND_SOC_CLOCK_IN);
+
+	/* codec is bitclock and lrclk master */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		goto out;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_LEFT_J |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		goto out;
+
+	ret = 0;
+out:
+	return ret;
+}
+
+static struct snd_soc_ops db1200_i2s_wm8731_ops = {
+	.startup	= db1200_i2s_startup,
+};
+
+static struct snd_soc_dai_link db1200_i2s_dai = {
+	.name		= "WM8731",
+	.stream_name	= "WM8731 PCM",
+	.cpu_dai	= &au1xpsc_i2s_dai,
+	.codec_dai	= &wm8731_dai,
+	.ops		= &db1200_i2s_wm8731_ops,
+};
+
+static struct snd_soc_card db1200_i2s_machine = {
+	.name		= "DB1200_I2S",
+	.dai_link	= &db1200_i2s_dai,
+	.num_links	= 1,
+	.platform	= &au1xpsc_soc_platform,
+};
+
+static struct snd_soc_device db1200_i2s_devdata = {
+	.card		= &db1200_i2s_machine,
+	.codec_dev	= &soc_codec_dev_wm8731,
+};
+
+/*-------------------------  COMMON PART  ---------------------------*/
+
+static struct platform_device *db1200_asoc_dev;
+
+static int __init db1200_audio_load(void)
+{
+	int ret;
+
+	ret = -ENOMEM;
+	db1200_asoc_dev = platform_device_alloc("soc-audio", -1);
+	if (!db1200_asoc_dev)
+		goto out;
+
+	/* DB1200 board setup set PSC1MUX to preferred audio device */
+	if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX)
+		platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_devdata);
+	else
+		platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_devdata);
+
+	db1200_ac97_devdata.dev = &db1200_asoc_dev->dev;
+	db1200_i2s_devdata.dev = &db1200_asoc_dev->dev;
+	ret = platform_device_add(db1200_asoc_dev);
+
+	if (ret) {
+		platform_device_put(db1200_asoc_dev);
+		db1200_asoc_dev = NULL;
+	}
+out:
+	return ret;
+}
+
+static void __exit db1200_audio_unload(void)
+{
+	platform_device_unregister(db1200_asoc_dev);
+}
+
+module_init(db1200_audio_load);
+module_exit(db1200_audio_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("DB1200 ASoC audio support");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 19e4d37eba1c..6d9f4c624949 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -51,8 +51,8 @@ struct au1xpsc_audio_dmadata {
 	struct snd_pcm_substream *substream;
 	unsigned long curr_period;	/* current segment DDMA is working on */
 	unsigned long q_period;		/* queue period(s) */
-	unsigned long dma_area;		/* address of queued DMA area */
-	unsigned long dma_area_s;	/* start address of DMA area */
+	dma_addr_t dma_area;		/* address of queued DMA area */
+	dma_addr_t dma_area_s;		/* start address of DMA area */
 	unsigned long pos;		/* current byte position being played */
 	unsigned long periods;		/* number of SG segments in total */
 	unsigned long period_bytes;	/* size in bytes of one SG segment */
@@ -94,8 +94,7 @@ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
 
 static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd)
 {
-	au1xxx_dbdma_put_source_flags(cd->ddma_chan,
-				(void *)phys_to_virt(cd->dma_area),
+	au1xxx_dbdma_put_source(cd->ddma_chan, cd->dma_area,
 				cd->period_bytes, DDMA_FLAGS_IE);
 
 	/* update next-to-queue period */
@@ -109,9 +108,8 @@ static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd)
 
 static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd)
 {
-	au1xxx_dbdma_put_dest_flags(cd->ddma_chan,
-				(void *)phys_to_virt(cd->dma_area),
-				cd->period_bytes, DDMA_FLAGS_IE);
+	au1xxx_dbdma_put_dest(cd->ddma_chan, cd->dma_area,
+			      cd->period_bytes, DDMA_FLAGS_IE);
 
 	/* update next-to-queue period */
 	++cd->q_period;
@@ -233,7 +231,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
 	pcd->substream = substream;
 	pcd->period_bytes = params_period_bytes(params);
 	pcd->periods = params_periods(params);
-	pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr;
+	pcd->dma_area_s = pcd->dma_area = runtime->dma_addr;
 	pcd->q_period = 0;
 	pcd->curr_period = 0;
 	pcd->pos = 0;
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
deleted file mode 100644
index 27683eb7905e..000000000000
--- a/sound/soc/au1x/sample-ac97.c
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
- * Sample Au12x0/Au1550 PSC AC97 sound machine.
- *
- * Copyright (c) 2007-2008 Manuel Lauss <mano@roarinelk.homelinux.net>
- *
- *  This program is free software; you can redistribute it and/or modify
- *  it under the terms outlined in the file COPYING at the root of this
- *  source archive.
- *
- * This is a very generic AC97 sound machine driver for boards which
- * have (AC97) audio at PSC1 (e.g. DB1200 demoboards).
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-#include <asm/mach-au1x00/au1000.h>
-#include <asm/mach-au1x00/au1xxx_psc.h>
-#include <asm/mach-au1x00/au1xxx_dbdma.h>
-
-#include "../codecs/ac97.h"
-#include "psc.h"
-
-static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec)
-{
-	snd_soc_dapm_sync(codec);
-	return 0;
-}
-
-static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = {
-	.name		= "AC97",
-	.stream_name	= "AC97 HiFi",
-	.cpu_dai	= &au1xpsc_ac97_dai,	/* see psc-ac97.c */
-	.codec_dai	= &ac97_dai,		/* see codecs/ac97.c */
-	.init		= au1xpsc_sample_ac97_init,
-	.ops		= NULL,
-};
-
-static struct snd_soc_card au1xpsc_sample_ac97_machine = {
-	.name		= "Au1xxx PSC AC97 Audio",
-	.dai_link	= &au1xpsc_sample_ac97_dai,
-	.num_links	= 1,
-};
-
-static struct snd_soc_device au1xpsc_sample_ac97_devdata = {
-	.card		= &au1xpsc_sample_ac97_machine,
-	.platform	= &au1xpsc_soc_platform, /* see dbdma2.c */
-	.codec_dev	= &soc_codec_dev_ac97,
-};
-
-static struct resource au1xpsc_psc1_res[] = {
-	[0] = {
-		.start	= CPHYSADDR(PSC1_BASE_ADDR),
-		.end	= CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff,
-		.flags	= IORESOURCE_MEM,
-	},
-	[1] = {
-#ifdef CONFIG_SOC_AU1200
-		.start	= AU1200_PSC1_INT,
-		.end	= AU1200_PSC1_INT,
-#elif defined(CONFIG_SOC_AU1550)
-		.start	= AU1550_PSC1_INT,
-		.end	= AU1550_PSC1_INT,
-#endif
-		.flags	= IORESOURCE_IRQ,
-	},
-	[2] = {
-		.start	= DSCR_CMD0_PSC1_TX,
-		.end	= DSCR_CMD0_PSC1_TX,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		.start	= DSCR_CMD0_PSC1_RX,
-		.end	= DSCR_CMD0_PSC1_RX,
-		.flags	= IORESOURCE_DMA,
-	},
-};
-
-static struct platform_device *au1xpsc_sample_ac97_dev;
-
-static int __init au1xpsc_sample_ac97_load(void)
-{
-	int ret;
-
-#ifdef CONFIG_SOC_AU1200
-	unsigned long io;
-
-	/* modify sys_pinfunc for AC97 on PSC1 */
-	io = au_readl(SYS_PINFUNC);
-	io |= SYS_PINFUNC_P1C;
-	io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B);
-	au_writel(io, SYS_PINFUNC);
-	au_sync();
-#endif
-
-	ret = -ENOMEM;
-
-	/* setup PSC clock source for AC97 part: external clock provided
-	 * by codec.  The psc-ac97.c driver depends on this setting!
-	 */
-	au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET);
-	au_sync();
-
-	au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1);
-	if (!au1xpsc_sample_ac97_dev)
-		goto out;
-
-	au1xpsc_sample_ac97_dev->resource =
-		kmemdup(au1xpsc_psc1_res, sizeof(struct resource) *
-			ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL);
-	au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res);
-	au1xpsc_sample_ac97_dev->id = 1;
-
-	platform_set_drvdata(au1xpsc_sample_ac97_dev,
-			     &au1xpsc_sample_ac97_devdata);
-	au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev;
-	ret = platform_device_add(au1xpsc_sample_ac97_dev);
-
-	if (ret) {
-		platform_device_put(au1xpsc_sample_ac97_dev);
-		au1xpsc_sample_ac97_dev = NULL;
-	}
-
-out:
-	return ret;
-}
-
-static void __exit au1xpsc_sample_ac97_exit(void)
-{
-	platform_device_unregister(au1xpsc_sample_ac97_dev);
-}
-
-module_init(au1xpsc_sample_ac97_load);
-module_exit(au1xpsc_sample_ac97_exit);
-
-MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine");
-MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index a9dc5fb54774..da589d8664d0 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev)
 	u16 reg;
 
 	/* Sync reg_cache with the hardware */
-	for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
+	for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) {
 		u16 val = tlv320aic23_read_reg_cache(codec, reg);
 		tlv320aic23_write(codec, reg, val);
 	}
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ce5515e3f2b0..3595bd57c4eb 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev)
 	struct i2c_client *i2c = codec->control_data;
 	int i;
 	u16 *reg_cache = codec->reg_cache;
-	u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults),
+	u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults),
 				 GFP_KERNEL);
 
 	/* Bring the codec back up to standby first to minimise pop/clicks */
@@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev)
 		for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
 			if (tmp_cache[i] != reg_cache[i])
 				snd_soc_write(codec, i, tmp_cache[i]);
+		kfree(tmp_cache);
 	} else {
 		dev_err(&i2c->dev, "Failed to allocate temporary cache\n");
 	}
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
index 3326e2a1e863..1a5b8e0d6a34 100644
--- a/sound/soc/fsl/efika-audio-fabric.c
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -55,7 +55,7 @@ static __init int efika_fabric_init(void)
 	struct platform_device *pdev;
 	int rc;
 
-	if (!machine_is_compatible("bplan,efika"))
+	if (!of_machine_is_compatible("bplan,efika"))
 		return -ENODEV;
 
 	card.platform = &mpc5200_audio_dma_platform;
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index b928ef7d28eb..6644cba7cbf2 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -55,7 +55,7 @@ static __init int pcm030_fabric_init(void)
 	struct platform_device *pdev;
 	int rc;
 
-	if (!machine_is_compatible("phytec,pcm030"))
+	if (!of_machine_is_compatible("phytec,pcm030"))
 		return -ENODEV;
 
 	card.platform = &mpc5200_audio_dma_platform;
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 3db8a6c523f4..19283e5edfbf 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o
+obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 71b2c161158d..68980c19a3bc 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -145,6 +145,7 @@ static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+	{"PCM DAC", NULL, "APLL Enable"},
 	{"Headphone Amplifier", NULL, "PCM DAC"},
 	{"Line Out", NULL, "PCM DAC"},
 	{"Headphone Jack", NULL, "Headphone Amplifier"},
diff --git a/sound/sound_core.c b/sound/sound_core.c
index dbca7c909a31..7c2d677a2df5 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -61,7 +61,7 @@ static void __exit cleanup_soundcore(void)
 	class_destroy(sound_class);
 }
 
-module_init(init_soundcore);
+subsys_initcall(init_soundcore);
 module_exit(cleanup_soundcore);