Merge branch 'fix/misc' into topic/misc

46 files changed, 348 insertions, 172 deletions

diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 16ae430..0caf77e 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -296,6 +296,7 @@ Conexant 5066
 =============
   laptop	Basic Laptop config (default)
   hp-laptop	HP laptops, e g G60
+  asus		Asus K52JU, Lenovo G560
   dell-laptop	Dell laptops
   dell-vostro	Dell Vostro
   olpc-xo-1_5	OLPC XO 1.5
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 10c3a87..b310702 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -33,9 +33,12 @@
 #include <linux/dw_dmac.h>
 
 #include <mach/cpu.h>
-#include <mach/hardware.h>
 #include <mach/gpio.h>
 
+#ifdef CONFIG_ARCH_AT91
+#include <mach/hardware.h>
+#endif
+
 #include "ac97c.h"
 
 enum {
diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index 7730575..b8b31c4 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -45,12 +45,13 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
 {
 	struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt);
 	struct snd_timer *t = stime->timer;
+	unsigned long oruns;
 
 	if (!atomic_read(&stime->running))
 		return HRTIMER_NORESTART;
 
-	hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
-	snd_timer_interrupt(stime->timer, t->sticks);
+	oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
+	snd_timer_interrupt(stime->timer, t->sticks * oruns);
 
 	if (!atomic_read(&stime->running))
 		return HRTIMER_NORESTART;
@@ -104,7 +105,7 @@ static int snd_hrtimer_stop(struct snd_timer *t)
 }
 
 static struct snd_timer_hardware hrtimer_hw = {
-	.flags =	SNDRV_TIMER_HW_AUTO,
+	.flags =	SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET,
 	.open =		snd_hrtimer_open,
 	.close =	snd_hrtimer_close,
 	.start =	snd_hrtimer_start,
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 4902ae5..53b53e9 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -141,6 +141,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
 
 fail_input:
 	input_free_device(jack->input_dev);
+	kfree(jack->id);
 	kfree(jack);
 	return err;
 }
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index da03597..5c426df 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -55,14 +55,13 @@
 #include <linux/err.h>
 #include <linux/platform_device.h>
 #include <linux/ioport.h>
+#include <linux/io.h>
 #include <linux/moduleparam.h>
 #include <sound/core.h>
 #include <sound/initval.h>
 #include <sound/rawmidi.h>
 #include <linux/delay.h>
 
-#include <asm/io.h>
-
 /*
  *      globals
  */
diff --git a/sound/oss/Makefile b/sound/oss/Makefile
index 96f14dc..90ffb99 100644
--- a/sound/oss/Makefile
+++ b/sound/oss/Makefile
@@ -87,7 +87,7 @@ ifeq ($(CONFIG_PSS_HAVE_BOOT),y)
 	$(obj)/bin2hex pss_synth < $< > $@
 else
     $(obj)/pss_boot.h:
-	(							\
+	$(Q)(							\
 	    echo 'static unsigned char * pss_synth = NULL;';	\
 	    echo 'static int pss_synthLen = 0;';		\
 	) > $@
@@ -102,7 +102,7 @@ ifeq ($(CONFIG_TRIX_HAVE_BOOT),y)
 	$(obj)/hex2hex -i trix_boot < $< > $@
 else
     $(obj)/trix_boot.h:
-	(							\
+	$(Q)(							\
 	    echo 'static unsigned char * trix_boot = NULL;';	\
 	    echo 'static int trix_boot_len = 0;';		\
 	) > $@
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 23f49f3..16c0bdf 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1252,11 +1252,19 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) {
 static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma)
 {
 	stream_t *dma = &vortex->dma_adb[adbdma];
-	int temp;
+	int temp, page, delta;
 
 	temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2));
-	temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1));
-	return temp;
+	page = (temp & ADB_SUBBUF_MASK) >> ADB_SUBBUF_SHIFT;
+	if (dma->nr_periods >= 4)
+		delta = (page - dma->period_real) & 3;
+	else {
+		delta = (page - dma->period_real);
+		if (delta < 0)
+			delta += dma->nr_periods;
+	}
+	return (dma->period_virt + delta) * dma->period_bytes
+		+ (temp & (dma->period_bytes - 1));
 }
 
 static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma)
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index bb84bb6..5715c4d0 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1317,31 +1317,25 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec,
 
 	snd_azf3328_dbgcallenter();
 	switch (bitrate) {
-#define AZF_FMT_XLATE(in_freq, out_bits) \
-	do { \
-		case AZF_FREQ_ ## in_freq: \
-			freq = SOUNDFORMAT_FREQ_ ## out_bits; \
-			break; \
-	} while (0);
-	AZF_FMT_XLATE(4000, SUSPECTED_4000)
-	AZF_FMT_XLATE(4800, SUSPECTED_4800)
-	/* the AZF3328 names it "5510" for some strange reason: */
-	AZF_FMT_XLATE(5512, 5510)
-	AZF_FMT_XLATE(6620, 6620)
-	AZF_FMT_XLATE(8000, 8000)
-	AZF_FMT_XLATE(9600, 9600)
-	AZF_FMT_XLATE(11025, 11025)
-	AZF_FMT_XLATE(13240, SUSPECTED_13240)
-	AZF_FMT_XLATE(16000, 16000)
-	AZF_FMT_XLATE(22050, 22050)
-	AZF_FMT_XLATE(32000, 32000)
+	case AZF_FREQ_4000:  freq = SOUNDFORMAT_FREQ_SUSPECTED_4000; break;
+	case AZF_FREQ_4800:  freq = SOUNDFORMAT_FREQ_SUSPECTED_4800; break;
+	case AZF_FREQ_5512:
+		/* the AZF3328 names it "5510" for some strange reason */
+			     freq = SOUNDFORMAT_FREQ_5510; break;
+	case AZF_FREQ_6620:  freq = SOUNDFORMAT_FREQ_6620; break;
+	case AZF_FREQ_8000:  freq = SOUNDFORMAT_FREQ_8000; break;
+	case AZF_FREQ_9600:  freq = SOUNDFORMAT_FREQ_9600; break;
+	case AZF_FREQ_11025: freq = SOUNDFORMAT_FREQ_11025; break;
+	case AZF_FREQ_13240: freq = SOUNDFORMAT_FREQ_SUSPECTED_13240; break;
+	case AZF_FREQ_16000: freq = SOUNDFORMAT_FREQ_16000; break;
+	case AZF_FREQ_22050: freq = SOUNDFORMAT_FREQ_22050; break;
+	case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break;
 	default:
 		snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate);
 		/* fall-through */
-	AZF_FMT_XLATE(44100, 44100)
-	AZF_FMT_XLATE(48000, 48000)
-	AZF_FMT_XLATE(66200, SUSPECTED_66200)
-#undef AZF_FMT_XLATE
+	case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break;
+	case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break;
+	case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
 	}
 	/* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */
 	/* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 4a66347..74b0560 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -381,7 +381,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
 	snd_print_pcm_rates(a->rates, buf, sizeof(buf));
 
 	if (a->format == AUDIO_CODING_TYPE_LPCM)
-		snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8));
+		snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8);
 	else if (a->max_bitrate)
 		snprintf(buf2, sizeof(buf2),
 				", max bitrate = %d", a->max_bitrate);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2e91a99..fcedad9 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2308,6 +2308,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
@@ -2703,7 +2704,7 @@ static int __devinit azx_probe(struct pci_dev *pci,
 	if (err < 0)
 		goto out_free;
 #ifdef CONFIG_SND_HDA_PATCH_LOADER
-	if (patch[dev]) {
+	if (patch[dev] && *patch[dev]) {
 		snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n",
 			   patch[dev]);
 		err = snd_hda_load_patch(chip->bus, patch[dev]);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 9bb030a..dd7c5c1 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -85,6 +85,7 @@ struct conexant_spec {
 	unsigned int auto_mic;
 	int auto_mic_ext;		/* autocfg.inputs[] index for ext mic */
 	unsigned int need_dac_fix;
+	hda_nid_t slave_dig_outs[2];
 
 	/* capture */
 	unsigned int num_adc_nids;
@@ -127,6 +128,7 @@ struct conexant_spec {
 	unsigned int ideapad:1;
 	unsigned int thinkpad:1;
 	unsigned int hp_laptop:1;
+	unsigned int asus:1;
 
 	unsigned int ext_mic_present;
 	unsigned int recording;
@@ -352,6 +354,8 @@ static int conexant_build_pcms(struct hda_codec *codec)
 			info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
 				spec->dig_in_nid;
 		}
+		if (spec->slave_dig_outs[0])
+			codec->slave_dig_outs = spec->slave_dig_outs;
 	}
 
 	return 0;
@@ -403,10 +407,16 @@ static int conexant_add_jack(struct hda_codec *codec,
 	struct conexant_spec *spec;
 	struct conexant_jack *jack;
 	const char *name;
-	int err;
+	int i, err;
 
 	spec = codec->spec;
 	snd_array_init(&spec->jacks, sizeof(*jack), 32);
+
+	jack = spec->jacks.list;
+	for (i = 0; i < spec->jacks.used; i++, jack++)
+		if (jack->nid == nid)
+			return 0 ; /* already present */
+
 	jack = snd_array_new(&spec->jacks);
 	name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ;
 
@@ -2100,7 +2110,7 @@ static int patch_cxt5051(struct hda_codec *codec)
 static hda_nid_t cxt5066_dac_nids[1] = { 0x10 };
 static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
 static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
-#define CXT5066_SPDIF_OUT	0x21
+static hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 };
 
 /* OLPC's microphone port is DC coupled for use with external sensors,
  * therefore we use a 50% mic bias in order to center the input signal with
@@ -2312,6 +2322,19 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec)
 	}
 }
 
+
+/* toggle input of built-in digital mic and mic jack appropriately */
+static void cxt5066_asus_automic(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_jack_detect(codec, 0x1b);
+	snd_printdd("CXT5066: external microphone present=%d\n", present);
+	snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL,
+			    present ? 1 : 0);
+}
+
+
 /* toggle input of built-in digital mic and mic jack appropriately */
 static void cxt5066_hp_laptop_automic(struct hda_codec *codec)
 {
@@ -2387,79 +2410,55 @@ static void cxt5066_hp_automute(struct hda_codec *codec)
 	cxt5066_update_speaker(codec);
 }
 
-/* unsolicited event for jack sensing */
-static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
+/* Dispatch the right mic autoswitch function */
+static void cxt5066_automic(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
-	snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
-	switch (res >> 26) {
-	case CONEXANT_HP_EVENT:
-		cxt5066_hp_automute(codec);
-		break;
-	case CONEXANT_MIC_EVENT:
-		/* ignore mic events in DC mode; we're always using the jack */
-		if (!spec->dc_enable)
-			cxt5066_olpc_automic(codec);
-		break;
-	}
-}
 
-/* unsolicited event for jack sensing */
-static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res)
-{
-	snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26);
-	switch (res >> 26) {
-	case CONEXANT_HP_EVENT:
-		cxt5066_hp_automute(codec);
-		break;
-	case CONEXANT_MIC_EVENT:
+	if (spec->dell_vostro)
 		cxt5066_vostro_automic(codec);
-		break;
-	}
-}
-
-/* unsolicited event for jack sensing */
-static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res)
-{
-	snd_printdd("CXT5066_ideapad: unsol event %x (%x)\n", res, res >> 26);
-	switch (res >> 26) {
-	case CONEXANT_HP_EVENT:
-		cxt5066_hp_automute(codec);
-		break;
-	case CONEXANT_MIC_EVENT:
+	else if (spec->ideapad)
 		cxt5066_ideapad_automic(codec);
-		break;
-	}
+	else if (spec->thinkpad)
+		cxt5066_thinkpad_automic(codec);
+	else if (spec->hp_laptop)
+		cxt5066_hp_laptop_automic(codec);
+	else if (spec->asus)
+		cxt5066_asus_automic(codec);
 }
 
 /* unsolicited event for jack sensing */
-static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
 {
-	snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26);
+	struct conexant_spec *spec = codec->spec;
+	snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
 	switch (res >> 26) {
 	case CONEXANT_HP_EVENT:
 		cxt5066_hp_automute(codec);
 		break;
 	case CONEXANT_MIC_EVENT:
-		cxt5066_hp_laptop_automic(codec);
+		/* ignore mic events in DC mode; we're always using the jack */
+		if (!spec->dc_enable)
+			cxt5066_olpc_automic(codec);
 		break;
 	}
 }
 
 /* unsolicited event for jack sensing */
-static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
 {
-	snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26);
+	snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
 	switch (res >> 26) {
 	case CONEXANT_HP_EVENT:
 		cxt5066_hp_automute(codec);
 		break;
 	case CONEXANT_MIC_EVENT:
-		cxt5066_thinkpad_automic(codec);
+		cxt5066_automic(codec);
 		break;
 	}
 }
 
+
 static const struct hda_input_mux cxt5066_analog_mic_boost = {
 	.num_items = 5,
 	.items = {
@@ -2633,6 +2632,27 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec)
 	spec->recording = 0;
 }
 
+static void conexant_check_dig_outs(struct hda_codec *codec,
+				    hda_nid_t *dig_pins,
+				    int num_pins)
+{
+	struct conexant_spec *spec = codec->spec;
+	hda_nid_t *nid_loc = &spec->multiout.dig_out_nid;
+	int i;
+
+	for (i = 0; i < num_pins; i++, dig_pins++) {
+		unsigned int cfg = snd_hda_codec_get_pincfg(codec, *dig_pins);
+		if (get_defcfg_connect(cfg) == AC_JACK_PORT_NONE)
+			continue;
+		if (snd_hda_get_connections(codec, *dig_pins, nid_loc, 1) != 1)
+			continue;
+		if (spec->slave_dig_outs[0])
+			nid_loc++;
+		else
+			nid_loc = spec->slave_dig_outs;
+	}
+}
+
 static struct hda_input_mux cxt5066_capture_source = {
 	.num_items = 4,
 	.items = {
@@ -3039,20 +3059,11 @@ static struct hda_verb cxt5066_init_verbs_hp_laptop[] = {
 /* initialize jack-sensing, too */
 static int cxt5066_init(struct hda_codec *codec)
 {
-	struct conexant_spec *spec = codec->spec;
-
 	snd_printdd("CXT5066: init\n");
 	conexant_init(codec);
 	if (codec->patch_ops.unsol_event) {
 		cxt5066_hp_automute(codec);
-		if (spec->dell_vostro)
-			cxt5066_vostro_automic(codec);
-		else if (spec->ideapad)
-			cxt5066_ideapad_automic(codec);
-		else if (spec->thinkpad)
-			cxt5066_thinkpad_automic(codec);
-		else if (spec->hp_laptop)
-			cxt5066_hp_laptop_automic(codec);
+		cxt5066_automic(codec);
 	}
 	cxt5066_set_mic_boost(codec);
 	return 0;
@@ -3080,6 +3091,7 @@ enum {
 	CXT5066_DELL_VOSTRO,	/* Dell Vostro 1015i */
 	CXT5066_IDEAPAD,	/* Lenovo IdeaPad U150 */
 	CXT5066_THINKPAD,	/* Lenovo ThinkPad T410s, others? */
+	CXT5066_ASUS,		/* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */
 	CXT5066_HP_LAPTOP,      /* HP Laptop */
 	CXT5066_MODELS
 };
@@ -3091,6 +3103,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
 	[CXT5066_DELL_VOSTRO]	= "dell-vostro",
 	[CXT5066_IDEAPAD]	= "ideapad",
 	[CXT5066_THINKPAD]	= "thinkpad",
+	[CXT5066_ASUS]		= "asus",
 	[CXT5066_HP_LAPTOP]	= "hp-laptop",
 };
 
@@ -3102,7 +3115,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
-	SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP),
+	SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
+	SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
+	SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", 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),
@@ -3111,7 +3126,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
+	SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
  	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
 	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
 	{}
 };
@@ -3133,7 +3150,8 @@ static int patch_cxt5066(struct hda_codec *codec)
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids);
 	spec->multiout.dac_nids = cxt5066_dac_nids;
-	spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT;
+	conexant_check_dig_outs(codec, cxt5066_digout_pin_nids,
+	    ARRAY_SIZE(cxt5066_digout_pin_nids));
 	spec->num_adc_nids = 1;
 	spec->adc_nids = cxt5066_adc_nids;
 	spec->capsrc_nids = cxt5066_capsrc_nids;
@@ -3167,17 +3185,20 @@ static int patch_cxt5066(struct hda_codec *codec)
 		spec->num_init_verbs++;
 		spec->dell_automute = 1;
 		break;
+	case CXT5066_ASUS:
 	case CXT5066_HP_LAPTOP:
 		codec->patch_ops.init = cxt5066_init;
-		codec->patch_ops.unsol_event = cxt5066_hp_laptop_event;
+		codec->patch_ops.unsol_event = cxt5066_unsol_event;
 		spec->init_verbs[spec->num_init_verbs] =
 			cxt5066_init_verbs_hp_laptop;
 		spec->num_init_verbs++;
-		spec->hp_laptop = 1;
+		spec->hp_laptop = board_config == CXT5066_HP_LAPTOP;
+		spec->asus = board_config == CXT5066_ASUS;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
 		/* no S/PDIF out */
-		spec->multiout.dig_out_nid = 0;
+		if (board_config == CXT5066_HP_LAPTOP)
+			spec->multiout.dig_out_nid = 0;
 		/* input source automatically selected */
 		spec->input_mux = NULL;
 		spec->port_d_mode = 0;
@@ -3207,7 +3228,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 		break;
 	case CXT5066_DELL_VOSTRO:
 		codec->patch_ops.init = cxt5066_init;
-		codec->patch_ops.unsol_event = cxt5066_vostro_event;
+		codec->patch_ops.unsol_event = cxt5066_unsol_event;
 		spec->init_verbs[0] = cxt5066_init_verbs_vostro;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
@@ -3224,7 +3245,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 		break;
 	case CXT5066_IDEAPAD:
 		codec->patch_ops.init = cxt5066_init;
-		codec->patch_ops.unsol_event = cxt5066_ideapad_event;
+		codec->patch_ops.unsol_event = cxt5066_unsol_event;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
 		spec->init_verbs[0] = cxt5066_init_verbs_ideapad;
@@ -3240,7 +3261,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 		break;
 	case CXT5066_THINKPAD:
 		codec->patch_ops.init = cxt5066_init;
-		codec->patch_ops.unsol_event = cxt5066_thinkpad_event;
+		codec->patch_ops.unsol_event = cxt5066_unsol_event;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
 		spec->init_verbs[0] = cxt5066_init_verbs_thinkpad;
@@ -3389,7 +3410,7 @@ static void cx_auto_parse_output(struct hda_codec *codec)
 		}
 	}
 	spec->multiout.dac_nids = spec->private_dac_nids;
-	spec->multiout.max_channels = nums * 2;
+	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
 	if (cfg->hp_outs > 0)
 		spec->auto_mute = 1;
@@ -3708,9 +3729,9 @@ static int cx_auto_init(struct hda_codec *codec)
 	return 0;
 }
 
-static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
+static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
 			      const char *dir, int cidx,
-			      hda_nid_t nid, int hda_dir)
+			      hda_nid_t nid, int hda_dir, int amp_idx)
 {
 	static char name[32];
 	static struct snd_kcontrol_new knew[] = {
@@ -3722,7 +3743,8 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
 
 	for (i = 0; i < 2; i++) {
 		struct snd_kcontrol *kctl;
-		knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, hda_dir);
+		knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx,
+							    hda_dir);
 		knew[i].subdevice = HDA_SUBDEV_AMP_FLAG;
 		knew[i].index = cidx;
 		snprintf(name, sizeof(name), "%s%s %s", basename, dir, sfx[i]);
@@ -3738,6 +3760,9 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
 	return 0;
 }
 
+#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir)		\
+	cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0)
+
 #define cx_auto_add_pb_volume(codec, nid, str, idx)			\
 	cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT)
 
@@ -3787,29 +3812,60 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
 	struct conexant_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	static const char *prev_label;
-	int i, err, cidx;
+	int i, err, cidx, conn_len;
+	hda_nid_t conn[HDA_MAX_CONNECTIONS];
+
+	int multi_adc_volume = 0; /* If the ADC nid has several input volumes */
+	int adc_nid = spec->adc_nids[0];
+
+	conn_len = snd_hda_get_connections(codec, adc_nid, conn,
+					   HDA_MAX_CONNECTIONS);
+	if (conn_len < 0)
+		return conn_len;
+
+	multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1;
+	if (!multi_adc_volume) {
+		err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid,
+					 HDA_INPUT);
+		if (err < 0)
+			return err;
+	}
 
-	err = cx_auto_add_volume(codec, "Capture", "", 0, spec->adc_nids[0],
-				 HDA_INPUT);
-	if (err < 0)
-		return err;
 	prev_label = NULL;
 	cidx = 0;
 	for (i = 0; i < cfg->num_inputs; i++) {
 		hda_nid_t nid = cfg->inputs[i].pin;
 		const char *label;
-		if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP))
+		int j;
+		int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP;
+		if (!pin_amp && !multi_adc_volume)
 			continue;
+
 		label = hda_get_autocfg_input_label(codec, cfg, i);
 		if (label == prev_label)
 			cidx++;
 		else
 			cidx = 0;
 		prev_label = label;
-		err = cx_auto_add_volume(codec, label, " Capture", cidx,
-					 nid, HDA_INPUT);
-		if (err < 0)
-			return err;
+
+		if (pin_amp) {
+			err = cx_auto_add_volume(codec, label, " Boost", cidx,
+						 nid, HDA_INPUT);
+			if (err < 0)
+				return err;
+		}
+
+		if (!multi_adc_volume)
+			continue;
+		for (j = 0; j < conn_len; j++) {
+			if (conn[j] == nid) {
+				err = cx_auto_add_volume_idx(codec, label,
+				    " Capture", cidx, adc_nid, HDA_INPUT, j);
+				if (err < 0)
+					return err;
+				break;
+			}
+		}
 	}
 	return 0;
 }
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 2d5b83f..a587677 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -642,6 +642,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
 			hdmi_ai->ver		= 0x01;
 			hdmi_ai->len		= 0x0a;
 			hdmi_ai->CC02_CT47	= channels - 1;
+			hdmi_ai->CA		= ca;
 			hdmi_checksum_audio_infoframe(hdmi_ai);
 		} else if (spec->sink_eld[i].conn_type == 1) { /* DisplayPort */
 			struct dp_audio_infoframe *dp_ai;
@@ -651,6 +652,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
 			dp_ai->len		= 0x1b;
 			dp_ai->ver		= 0x11 << 2;
 			dp_ai->CC02_CT47	= channels - 1;
+			dp_ai->CA		= ca;
 		} else {
 			snd_printd("HDMI: unknown connection type at pin %d\n",
 				   pin_nid);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7169c9a..c5208c9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2290,6 +2290,29 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0f, 2, 0x0,
+		HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0f, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Side Playback Switch", 0x0e, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+
 static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -10359,7 +10382,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.init_hook = alc_automute_amp,
 	},
 	[ALC888_ACER_ASPIRE_4930G] = {
-		.mixers = { alc888_base_mixer,
+		.mixers = { alc888_acer_aspire_4930g_mixer,
 				alc883_chmode_mixer },
 		.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
 				alc888_acer_aspire_4930g_verbs },
@@ -14954,9 +14977,11 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
-	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
-	SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 	{}
@@ -18800,6 +18825,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
 					ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
 			   ALC663_ASUS_H13),
+	SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E),
 	{}
 };
 
@@ -19492,6 +19518,7 @@ static const struct alc_fixup alc662_fixups[] = {
 };
 
 static struct snd_pci_quirk alc662_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index a76c326..63b0054 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -567,7 +567,7 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
 		hda_nid_t nid = cfg->inputs[i].pin;
 		if (spec->smart51_enabled && is_smart51_pins(spec, nid))
 			ctl = PIN_OUT;
-		else if (i == AUTO_PIN_MIC)
+		else if (cfg->inputs[i].type == AUTO_PIN_MIC)
 			ctl = PIN_VREF50;
 		else
 			ctl = PIN_IN;
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index c2ae63d..f53897a 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -92,6 +92,8 @@ struct oxygen_model {
 	void (*update_dac_volume)(struct oxygen *chip);
 	void (*update_dac_mute)(struct oxygen *chip);
 	void (*update_center_lfe_mix)(struct oxygen *chip, bool mixed);
+	unsigned int (*adjust_dac_routing)(struct oxygen *chip,
+					   unsigned int play_routing);
 	void (*gpio_changed)(struct oxygen *chip);
 	void (*uart_input)(struct oxygen *chip);
 	void (*ac97_switch)(struct oxygen *chip,
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 9bff14d..26c7e8b 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -180,6 +180,8 @@ void oxygen_update_dac_routing(struct oxygen *chip)
 			    (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
 			    (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
 			    (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT);
+	if (chip->model.adjust_dac_routing)
+		reg_value = chip->model.adjust_dac_routing(chip, reg_value);
 	oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value,
 			      OXYGEN_PLAY_DAC0_SOURCE_MASK |
 			      OXYGEN_PLAY_DAC1_SOURCE_MASK |
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index e1fa602..bc6eb58 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -24,6 +24,11 @@
  *
  *   SPI 0 -> CS4245
  *
+ *   I²S 1 -> CS4245
+ *   I²S 2 -> CS4361 (center/LFE)
+ *   I²S 3 -> CS4361 (surround)
+ *   I²S 4 -> CS4361 (front)
+ *
  *   GPIO 3 <- ?
  *   GPIO 4 <- headphone detect
  *   GPIO 5 -> route input jack to line-in (0) or mic-in (1)
@@ -36,6 +41,7 @@
  *   input 1 <- aux
  *   input 2 <- front mic
  *   input 4 <- line/mic
+ *   DAC out -> headphones
  *   aux out -> front panel headphones
  */
 
@@ -207,6 +213,35 @@ static void set_cs4245_adc_params(struct oxygen *chip,
 	cs4245_write_cached(chip, CS4245_ADC_CTRL, value);
 }
 
+static inline unsigned int shift_bits(unsigned int value,
+				      unsigned int shift_from,
+				      unsigned int shift_to,
+				      unsigned int mask)
+{
+	if (shift_from < shift_to)
+		return (value << (shift_to - shift_from)) & mask;
+	else
+		return (value >> (shift_from - shift_to)) & mask;
+}
+
+static unsigned int adjust_dg_dac_routing(struct oxygen *chip,
+					  unsigned int play_routing)
+{
+	return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) |
+	       shift_bits(play_routing,
+			  OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC1_SOURCE_MASK) |
+	       shift_bits(play_routing,
+			  OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC2_SOURCE_MASK) |
+	       shift_bits(play_routing,
+			  OXYGEN_PLAY_DAC0_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC3_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC3_SOURCE_MASK);
+}
+
 static int output_switch_info(struct snd_kcontrol *ctl,
 			      struct snd_ctl_elem_info *info)
 {
@@ -557,6 +592,7 @@ struct oxygen_model model_xonar_dg = {
 	.resume = dg_resume,
 	.set_dac_params = set_cs4245_dac_params,
 	.set_adc_params = set_cs4245_adc_params,
+	.adjust_dac_routing = adjust_dg_dac_routing,
 	.dump_registers = dump_cs4245_registers,
 	.model_data_size = sizeof(struct dg),
 	.device_config = PLAYBACK_0_TO_I2S |
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h
index bd26e09..6ce9ad7 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.h
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h
@@ -22,7 +22,7 @@
 #define __PDAUDIOCF_H
 
 #include <sound/pcm.h>
-#include <asm/io.h>
+#include <linux/io.h>
 #include <linux/interrupt.h>
 #include <pcmcia/cistpl.h>
 #include <pcmcia/ds.h>
diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c
index 989e04a..fe33e12 100644
--- a/sound/pcmcia/vx/vxp_ops.c
+++ b/sound/pcmcia/vx/vxp_ops.c
@@ -23,8 +23,8 @@
 #include <linux/delay.h>
 #include <linux/device.h>
 #include <linux/firmware.h>
+#include <linux/io.h>
 #include <sound/core.h>
-#include <asm/io.h>
 #include "vxpocket.h"
 
 
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index da2208e..5e4d499 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = {
 	.cpu_dai_name = "atmel-ssc-dai.0",
 	.codec_dai_name = "tlv320aic23-hifi",
 	.platform_name = "atmel_pcm-audio",
-	.codec_name = "tlv320aic23-codec.0-0x1a",
+	.codec_name = "tlv320aic23-codec.0-001a",
 	.init = afeb9260_tlv320aic23_init,
 	.ops = &afeb9260_ops,
 };
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index e902b24..ad28663 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = {
 	.cpu_dai_name = "bf5xx-i2s",
 	.codec_dai_name = "ssm2602-hifi",
 	.platform_name = "bf5xx-pcm-audio",
-	.codec_name = "ssm2602-codec.0-0x1b",
+	.codec_name = "ssm2602-codec.0-001b",
 	.ops = &bf5xx_ssm2602_ops,
 };
 
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 46dbfd0..347a567 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -153,7 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec)
 
 static int cq93vc_probe(struct snd_soc_codec *codec)
 {
-	struct davinci_vc *davinci_vc = codec->dev->platform_data;
+	struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec);
 
 	davinci_vc->cq93vc.codec = codec;
 	codec->control_data = davinci_vc;
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 03d1e86..bb4bf65 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
+static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC;
+
 static struct snd_soc_codec_driver cx20442_codec_dev = {
 	.probe = 	cx20442_codec_probe,
 	.remove = 	cx20442_codec_remove,
+	.reg_cache_default = &cx20442_reg,
 	.reg_cache_size = 1,
 	.reg_word_size = sizeof(u8),
 	.read = cx20442_read_reg_cache,
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 247a6a9..37b8aa8 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0),
 
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 9, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 8, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev,
@@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev,
 		      SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 11, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 10, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev,
@@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0,
 
 SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
 
 SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux),
@@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" },
 	{ "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" },
 
+	{ "AIF1ADCDAT", NULL, "AIF1ADC1L" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC1R" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC2L" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC2R" },
+
 	{ "AIF2ADCDAT", NULL, "AIF2ADC Mux" },
 
 	/* AIF3 output */
@@ -1578,6 +1584,13 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "Right Headphone Mux", "DAC", "DAC1R" },
 };
 
+static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
+	{ "AIF1DACDAT", NULL, "AIF2DACDAT" },
+	{ "AIF2DACDAT", NULL, "AIF1DACDAT" },
+	{ "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
+	{ "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
+};
+
 static const struct snd_soc_dapm_route wm8994_intercon[] = {
 	{ "AIF2DACL", NULL, "AIF2DAC Mux" },
 	{ "AIF2DACR", NULL, "AIF2DAC Mux" },
@@ -2386,7 +2399,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
 	else
 		val = 0;
 
-	return snd_soc_update_bits(codec, reg, mask, reg);
+	return snd_soc_update_bits(codec, reg, mask, val);
 }
 
 #define WM8994_RATES SNDRV_PCM_RATE_8000_96000
@@ -3129,6 +3142,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	case WM8994:
 		snd_soc_dapm_add_routes(dapm, wm8994_intercon,
 					ARRAY_SIZE(wm8994_intercon));
+
+		if (wm8994->revision < 4)
+			snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
+						ARRAY_SIZE(wm8994_revd_intercon));
+			
 		break;
 	case WM8958:
 		snd_soc_dapm_add_routes(dapm, wm8958_intercon,
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 6045cbd..608c84c 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1223,7 +1223,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
 	else
 		val = 0;
 
-	return snd_soc_update_bits(codec, reg, mask, reg);
+	return snd_soc_update_bits(codec, reg, mask, val);
 }
 
 /* The size in bits of the FLL divide multiplied by 10
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index c466982..613df5d 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -91,6 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
 static void calibrate_dc_servo(struct snd_soc_codec *codec)
 {
 	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+	s8 offset;
 	u16 reg, reg_l, reg_r, dcs_cfg;
 
 	/* If we're using a digital only path and have a previously
@@ -149,16 +150,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 			hubs->dcs_codes);
 
 		/* HPOUT1L */
-		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;
+		offset = reg_l;
+		offset += hubs->dcs_codes;
+		dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
 
 		/* HPOUT1R */
-		if (reg_r + hubs->dcs_codes > 0 &&
-		    reg_r + hubs->dcs_codes < 0xff)
-			reg_r += hubs->dcs_codes;
-		dcs_cfg |= reg_r;
+		offset = reg_r;
+		offset += hubs->dcs_codes;
+		dcs_cfg |= (u8)offset;
 
 		dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
 
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 0c2d6ba..fe79842 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -218,12 +218,24 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = {
 		.ops = &evm_spdif_ops,
 	},
 };
-static struct snd_soc_dai_link da8xx_evm_dai = {
+
+static struct snd_soc_dai_link da830_evm_dai = {
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.cpu_dai_name = "davinci-mcasp.1",
+	.codec_dai_name = "tlv320aic3x-hifi",
+	.codec_name = "tlv320aic3x-codec.1-0018",
+	.platform_name = "davinci-pcm-audio",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link da850_evm_dai = {
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
 	.cpu_dai_name= "davinci-mcasp.0",
 	.codec_dai_name = "tlv320aic3x-hifi",
-	.codec_name = "tlv320aic3x-codec.0-001a",
+	.codec_name = "tlv320aic3x-codec.1-0018",
 	.platform_name = "davinci-pcm-audio",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
@@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = {
 
 static struct snd_soc_card da830_snd_soc_card = {
 	.name = "DA830/OMAP-L137 EVM",
-	.dai_link = &da8xx_evm_dai,
+	.dai_link = &da830_evm_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_card da850_snd_soc_card = {
 	.name = "DA850/OMAP-L138 EVM",
-	.dai_link = &da8xx_evm_dai,
+	.dai_link = &da850_evm_dai,
 	.num_links = 1,
 };
 
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 2101bdc..3167be6 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
 	/* Set up digital mute if not provided by the codec */
 	if (!codec_dai->driver->ops) {
 		codec_dai->driver->ops = &ams_delta_dai_ops;
-	} else if (!codec_dai->driver->ops->digital_mute) {
-		codec_dai->driver->ops->digital_mute = ams_delta_digital_mute;
 	} else {
 		ams_delta_ops.startup = ams_delta_startup;
 		ams_delta_ops.shutdown = ams_delta_shutdown;
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index fc592f0..784cff5 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -307,10 +307,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link corgi_dai = {
 	.name = "WM8731",
 	.stream_name = "WM8731",
-	.cpu_dai_name = "pxa-is2-dai",
+	.cpu_dai_name = "pxa2xx-i2s",
 	.codec_dai_name = "wm8731-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name = "wm8731-codec-0.001a",
+	.codec_name = "wm8731-codec-0.001b",
 	.init = corgi_wm8731_init,
 	.ops = &corgi_ops,
 };
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 6298ee1..a7d4999 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = {
 	.cpu_dai_name = "pxa2xx-i2s",
 	.codec_dai_name = "wm8731-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name = "wm8731-codec.0-001a",
+	.codec_name = "wm8731-codec.0-001b",
 	.init = poodle_wm8731_init,
 	.ops = &poodle_ops,
 };
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index c2acb69..8e15713 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -315,10 +315,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link spitz_dai = {
 	.name = "wm8750",
 	.stream_name = "WM8750",
-	.cpu_dai_name = "pxa-is2",
+	.cpu_dai_name = "pxa2xx-i2s",
 	.codec_dai_name = "wm8750-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name = "wm8750-codec.0-001a",
+	.codec_name = "wm8750-codec.0-001b",
 	.init = spitz_wm8750_init,
 	.ops = &spitz_ops,
 };
diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c
index 3eec610..0d0ae2b 100644
--- a/sound/soc/samsung/neo1973_gta02_wm8753.c
+++ b/sound/soc/samsung/neo1973_gta02_wm8753.c
@@ -397,11 +397,11 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
 { /* Hifi Playback - for similatious use with voice below */
 	.name = "WM8753",
 	.stream_name = "WM8753 HiFi",
-	.cpu_dai_name = "s3c24xx-i2s",
+	.cpu_dai_name = "s3c24xx-iis",
 	.codec_dai_name = "wm8753-hifi",
 	.init = neo1973_gta02_wm8753_init,
 	.platform_name = "samsung-audio",
-	.codec_name = "wm8753-codec.0-0x1a",
+	.codec_name = "wm8753-codec.0-001a",
 	.ops = &neo1973_gta02_hifi_ops,
 },
 { /* Voice via BT */
@@ -410,7 +410,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
 	.cpu_dai_name = "bluetooth-dai",
 	.codec_dai_name = "wm8753-voice",
 	.ops = &neo1973_gta02_voice_ops,
-	.codec_name = "wm8753-codec.0-0x1a",
+	.codec_name = "wm8753-codec.0-001a",
 	.platform_name = "samsung-audio",
 },
 };
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index c7a2451..d20815d 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -559,9 +559,9 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 	.name = "WM8753",
 	.stream_name = "WM8753 HiFi",
 	.platform_name = "samsung-audio",
-	.cpu_dai_name = "s3c24xx-i2s",
+	.cpu_dai_name = "s3c24xx-iis",
 	.codec_dai_name = "wm8753-hifi",
-	.codec_name = "wm8753-codec.0-0x1a",
+	.codec_name = "wm8753-codec.0-001a",
 	.init = neo1973_wm8753_init,
 	.ops = &neo1973_hifi_ops,
 },
@@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 	.platform_name = "samsung-audio",
 	.cpu_dai_name = "bluetooth-dai",
 	.codec_dai_name = "wm8753-voice",
-	.codec_name = "wm8753-codec.0-0x1a",
+	.codec_name = "wm8753-codec.0-001a",
 	.ops = &neo1973_voice_ops,
 },
 };
diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c
index bb4292e..08fcaaa 100644
--- a/sound/soc/samsung/s3c24xx_simtec_hermes.c
+++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c
@@ -94,8 +94,8 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link simtec_dai_aic33 = {
 	.name		= "tlv320aic33",
 	.stream_name	= "TLV320AIC33",
-	.codec_name	= "tlv320aic3x-codec.0-0x1a",
-	.cpu_dai_name	= "s3c24xx-i2s",
+	.codec_name	= "tlv320aic3x-codec.0-001a",
+	.cpu_dai_name	= "s3c24xx-iis",
 	.codec_dai_name = "tlv320aic3x-hifi",
 	.platform_name	= "samsung-audio",
 	.init		= simtec_hermes_init,
diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
index fbba4e3..116e3e6 100644
--- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
@@ -85,8 +85,8 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link simtec_dai_aic23 = {
 	.name		= "tlv320aic23",
 	.stream_name	= "TLV320AIC23",
-	.codec_name	= "tlv320aic3x-codec.0-0x1a",
-	.cpu_dai_name	= "s3c24xx-i2s",
+	.codec_name	= "tlv320aic3x-codec.0-001a",
+	.cpu_dai_name	= "s3c24xx-iis",
 	.codec_dai_name = "tlv320aic3x-hifi",
 	.platform_name	= "samsung-audio",
 	.init		= simtec_tlv320aic23_init,
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index cdc8ecb..2c09e93 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -228,7 +228,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
 	.stream_name = "UDA134X",
 	.codec_name = "uda134x-hifi",
 	.codec_dai_name = "uda134x-hifi",
-	.cpu_dai_name = "s3c24xx-i2s",
+	.cpu_dai_name = "s3c24xx-iis",
 	.ops = &s3c24xx_uda134x_ops,
 	.platform_name	= "samsung-audio",
 };
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index bac7291..c3f6f1e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1449,6 +1449,7 @@ static int soc_post_component_init(struct snd_soc_card *card,
 		rtd = &card->rtd_aux[num];
 		name = aux_dev->name;
 	}
+	rtd->card = card;
 
 	/* machine controls, routes and widgets are not prefixed */
 	temp = codec->name_prefix;
@@ -1471,7 +1472,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
 
 	/* register the rtd device */
 	rtd->codec = codec;
-	rtd->card = card;
 	rtd->dev.parent = card->dev;
 	rtd->dev.release = rtd_release;
 	rtd->dev.init_name = name;
@@ -1664,9 +1664,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
 	goto out;
 
 found:
-	if (!try_module_get(codec->dev->driver->owner))
-		return -ENODEV;
-
 	ret = soc_probe_codec(card, codec);
 	if (ret < 0)
 		return ret;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 499730a..8194f15 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1742,7 +1742,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 	int max = mc->max;
 	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
-	unsigned int val, val_mask;
+	unsigned int val;
 	int connect, change;
 	struct snd_soc_dapm_update update;
 
@@ -1750,13 +1750,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 
 	if (invert)
 		val = max - val;
-	val_mask = mask << shift;
+	mask = mask << shift;
 	val = val << shift;
 
 	mutex_lock(&widget->codec->mutex);
 	widget->value = val;
 
-	change = snd_soc_test_bits(widget->codec, reg, val_mask, val);
+	change = snd_soc_test_bits(widget->codec, reg, mask, val);
 	if (change) {
 		if (val)
 			/* new connection */
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index e411cd3..d0d493c 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -785,7 +785,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
 	}
 
 	dev->pcm->private_data = dev;
-	strcpy(dev->pcm->name, dev->product_name);
+	strlcpy(dev->pcm->name, dev->product_name, sizeof(dev->pcm->name));
 
 	memset(dev->sub_playback, 0, sizeof(dev->sub_playback));
 	memset(dev->sub_capture, 0, sizeof(dev->sub_capture));
diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c
index 2f218c7..a1a4708 100644
--- a/sound/usb/caiaq/midi.c
+++ b/sound/usb/caiaq/midi.c
@@ -136,7 +136,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device)
 	if (ret < 0)
 		return ret;
 
-	strcpy(rmidi->name, device->product_name);
+	strlcpy(rmidi->name, device->product_name, sizeof(rmidi->name));
 
 	rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX;
 	rmidi->private_data = device;
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 800f7cb..c0f8270 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -323,6 +323,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx,
 		return -ENOMEM;
 	}
 
+	mutex_init(&chip->shutdown_mutex);
 	chip->index = idx;
 	chip->dev = dev;
 	chip->card = card;
@@ -531,6 +532,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
 	chip = ptr;
 	card = chip->card;
 	mutex_lock(&register_mutex);
+	mutex_lock(&chip->shutdown_mutex);
 	chip->shutdown = 1;
 	chip->num_interfaces--;
 	if (chip->num_interfaces <= 0) {
@@ -548,9 +550,11 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
 			snd_usb_mixer_disconnect(p);
 		}
 		usb_chip[chip->index] = NULL;
+		mutex_unlock(&chip->shutdown_mutex);
 		mutex_unlock(&register_mutex);
 		snd_card_free_when_closed(card);
 	} else {
+		mutex_unlock(&chip->shutdown_mutex);
 		mutex_unlock(&register_mutex);
 	}
 }
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 844cf78..675a4f1 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -95,7 +95,7 @@ enum {
 };
 
 
-/*E-mu 0202(0404) eXtension Unit(XU) control*/
+/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/
 enum {
 	USB_XU_CLOCK_RATE 		= 0xe301,
 	USB_XU_CLOCK_SOURCE		= 0xe302,
@@ -1583,7 +1583,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw
 			cval->initialized = 1;
 		} else {
 			if (type == USB_XU_CLOCK_RATE) {
-				/* E-Mu USB 0404/0202/TrackerPre
+				/* E-Mu USB 0404/0202/TrackerPre/0204
 				 * samplerate control quirk
 				 */
 				cval->min = 0;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 4132522..e3f6805 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -361,6 +361,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (changed) {
+		mutex_lock(&subs->stream->chip->shutdown_mutex);
 		/* format changed */
 		snd_usb_release_substream_urbs(subs, 0);
 		/* influenced: period_bytes, channels, rate, format, */
@@ -368,6 +369,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
 						  params_rate(hw_params),
 						  snd_pcm_format_physical_width(params_format(hw_params)) *
 							params_channels(hw_params));
+		mutex_unlock(&subs->stream->chip->shutdown_mutex);
 	}
 
 	return ret;
@@ -385,8 +387,9 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream)
 	subs->cur_audiofmt = NULL;
 	subs->cur_rate = 0;
 	subs->period_bytes = 0;
-	if (!subs->stream->chip->shutdown)
-		snd_usb_release_substream_urbs(subs, 0);
+	mutex_lock(&subs->stream->chip->shutdown_mutex);
+	snd_usb_release_substream_urbs(subs, 0);
+	mutex_unlock(&subs->stream->chip->shutdown_mutex);
 	return snd_pcm_lib_free_vmalloc_buffer(substream);
 }
 
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index e1e245d..c0dcfca 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -79,6 +79,13 @@
 	.idProduct = 0x3f0a,
 	.bInterfaceClass = USB_CLASS_AUDIO,
 },
+{
+	/* E-Mu 0204 USB */
+	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+	.idVendor = 0x041e,
+	.idProduct = 0x3f19,
+	.bInterfaceClass = USB_CLASS_AUDIO,
+},
 
 /*
  * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index ca860e6..355759b 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -566,7 +566,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat
 }
 
 /*
- * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device,
+ * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device,
  * not for interface.
  */
 
@@ -623,6 +623,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 	case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */
 	case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */
 	case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */
+	case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */
 		set_format_emu_quirk(subs, fmt);
 		break;
 	}
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index db3eb21..6e66fff 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -36,6 +36,7 @@ struct snd_usb_audio {
 	struct snd_card *card;
 	u32 usb_id;
 	int shutdown;
+	struct mutex shutdown_mutex;
 	unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
 	int num_interfaces;
 	int num_suspended_intf;