Merge branch 'omap/hwmod' into next/drivers

This is needed as a dependency for omap/ehci.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>

22 files changed, 243 insertions, 602 deletions

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 096507d..7d98240 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2508,7 +2508,6 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", 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(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index cbde019..1d07e8f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -297,6 +297,8 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
 	imux = &spec->input_mux[mux_idx];
 	if (!imux->num_items && mux_idx > 0)
 		imux = &spec->input_mux[0];
+	if (!imux->num_items)
+		return 0;
 
 	if (idx >= imux->num_items)
 		idx = imux->num_items - 1;
@@ -2629,6 +2631,8 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch,
 	case AUTO_PIN_SPEAKER_OUT:
 		if (cfg->line_outs == 1)
 			return "Speaker";
+		if (cfg->line_outs == 2)
+			return ch ? "Bass Speaker" : "Speaker";
 		break;
 	case AUTO_PIN_HP_OUT:
 		/* for multi-io case, only the primary out */
@@ -2902,7 +2906,7 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
 		if (!nid)
 			continue;
 		if (found_in_nid_list(nid, spec->multiout.dac_nids,
-				      spec->multiout.num_dacs))
+				      ARRAY_SIZE(spec->private_dac_nids)))
 			continue;
 		if (found_in_nid_list(nid, spec->multiout.hp_out_nid,
 				      ARRAY_SIZE(spec->multiout.hp_out_nid)))
@@ -2923,6 +2927,7 @@ static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin)
 	return 0;
 }
 
+/* return 0 if no possible DAC is found, 1 if one or more found */
 static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs,
 				    const hda_nid_t *pins, hda_nid_t *dacs)
 {
@@ -2940,7 +2945,7 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs,
 		if (!dacs[i])
 			dacs[i] = alc_auto_look_for_dac(codec, pins[i]);
 	}
-	return 0;
+	return 1;
 }
 
 static int alc_auto_fill_multi_ios(struct hda_codec *codec,
@@ -2950,7 +2955,7 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec,
 static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	const struct auto_pin_cfg *cfg = &spec->autocfg;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
 	bool redone = false;
 	int i;
 
@@ -2961,6 +2966,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 	spec->multiout.extra_out_nid[0] = 0;
 	memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids));
 	spec->multiout.dac_nids = spec->private_dac_nids;
+	spec->multi_ios = 0;
 
 	/* fill hard-wired DACs first */
 	if (!redone) {
@@ -2994,10 +3000,12 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 	for (i = 0; i < cfg->line_outs; i++) {
 		if (spec->private_dac_nids[i])
 			spec->multiout.num_dacs++;
-		else
+		else {
 			memmove(spec->private_dac_nids + i,
 				spec->private_dac_nids + i + 1,
 				sizeof(hda_nid_t) * (cfg->line_outs - i - 1));
+			spec->private_dac_nids[cfg->line_outs - 1] = 0;
+		}
 	}
 
 	if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
@@ -3019,9 +3027,28 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 	if (cfg->line_out_type != AUTO_PIN_HP_OUT)
 		alc_auto_fill_extra_dacs(codec, cfg->hp_outs, cfg->hp_pins,
 				 spec->multiout.hp_out_nid);
-	if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT)
-		alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, cfg->speaker_pins,
-				 spec->multiout.extra_out_nid);
+	if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+		int err = alc_auto_fill_extra_dacs(codec, cfg->speaker_outs,
+					cfg->speaker_pins,
+					spec->multiout.extra_out_nid);
+		/* if no speaker volume is assigned, try again as the primary
+		 * output
+		 */
+		if (!err && cfg->speaker_outs > 0 &&
+		    cfg->line_out_type == AUTO_PIN_HP_OUT) {
+			cfg->hp_outs = cfg->line_outs;
+			memcpy(cfg->hp_pins, cfg->line_out_pins,
+			       sizeof(cfg->hp_pins));
+			cfg->line_outs = cfg->speaker_outs;
+			memcpy(cfg->line_out_pins, cfg->speaker_pins,
+			       sizeof(cfg->speaker_pins));
+			cfg->speaker_outs = 0;
+			memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
+			cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
+			redone = false;
+			goto again;
+		}
+	}
 
 	return 0;
 }
@@ -3171,7 +3198,8 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
 }
 
 static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
-				     hda_nid_t dac, const char *pfx)
+				     hda_nid_t dac, const char *pfx,
+				     int cidx)
 {
 	struct alc_spec *spec = codec->spec;
 	hda_nid_t sw, vol;
@@ -3187,15 +3215,15 @@ static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
 		if (is_ctl_used(spec->sw_ctls, val))
 			return 0; /* already created */
 		mark_ctl_usage(spec->sw_ctls, val);
-		return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, val);
+		return __add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, cidx, val);
 	}
 
 	sw = alc_look_for_out_mute_nid(codec, pin, dac);
 	vol = alc_look_for_out_vol_nid(codec, pin, dac);
-	err = alc_auto_add_stereo_vol(codec, pfx, 0, vol);
+	err = alc_auto_add_stereo_vol(codec, pfx, cidx, vol);
 	if (err < 0)
 		return err;
-	err = alc_auto_add_stereo_sw(codec, pfx, 0, sw);
+	err = alc_auto_add_stereo_sw(codec, pfx, cidx, sw);
 	if (err < 0)
 		return err;
 	return 0;
@@ -3236,16 +3264,21 @@ static int alc_auto_create_extra_outs(struct hda_codec *codec, int num_pins,
 		hda_nid_t dac = *dacs;
 		if (!dac)
 			dac = spec->multiout.dac_nids[0];
-		return alc_auto_create_extra_out(codec, *pins, dac, pfx);
+		return alc_auto_create_extra_out(codec, *pins, dac, pfx, 0);
 	}
 
 	if (dacs[num_pins - 1]) {
 		/* OK, we have a multi-output system with individual volumes */
 		for (i = 0; i < num_pins; i++) {
-			snprintf(name, sizeof(name), "%s %s",
-				 pfx, channel_name[i]);
-			err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
-							name);
+			if (num_pins >= 3) {
+				snprintf(name, sizeof(name), "%s %s",
+					 pfx, channel_name[i]);
+				err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
+								name, 0);
+			} else {
+				err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
+								pfx, i);
+			}
 			if (err < 0)
 				return err;
 		}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index f365865..eeb25d52 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -215,6 +215,7 @@ struct sigmatel_spec {
 	unsigned int gpio_mute;
 	unsigned int gpio_led;
 	unsigned int gpio_led_polarity;
+	unsigned int vref_mute_led_nid; /* pin NID for mute-LED vref control */
 	unsigned int vref_led;
 
 	/* stream */
@@ -4318,12 +4319,10 @@ static void stac_store_hints(struct hda_codec *codec)
 		spec->eapd_switch = val;
 	get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity);
 	if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) {
-		if (spec->gpio_led <= 8) {
-			spec->gpio_mask |= spec->gpio_led;
-			spec->gpio_dir |= spec->gpio_led;
-			if (spec->gpio_led_polarity)
-				spec->gpio_data |= spec->gpio_led;
-		}
+		spec->gpio_mask |= spec->gpio_led;
+		spec->gpio_dir |= spec->gpio_led;
+		if (spec->gpio_led_polarity)
+			spec->gpio_data |= spec->gpio_led;
 	}
 }
 
@@ -4441,7 +4440,9 @@ static int stac92xx_init(struct hda_codec *codec)
 		int pinctl, def_conf;
 
 		/* power on when no jack detection is available */
-		if (!spec->hp_detect) {
+		/* or when the VREF is used for controlling LED */
+		if (!spec->hp_detect ||
+		    spec->vref_mute_led_nid == nid) {
 			stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
@@ -4913,8 +4914,14 @@ static int find_mute_led_gpio(struct hda_codec *codec, int default_polarity)
 			if (sscanf(dev->name, "HP_Mute_LED_%d_%x",
 				  &spec->gpio_led_polarity,
 				  &spec->gpio_led) == 2) {
-				if (spec->gpio_led < 4)
+				unsigned int max_gpio;
+				max_gpio = snd_hda_param_read(codec, codec->afg,
+							      AC_PAR_GPIO_CAP);
+				max_gpio &= AC_GPIO_IO_COUNT;
+				if (spec->gpio_led < max_gpio)
 					spec->gpio_led = 1 << spec->gpio_led;
+				else
+					spec->vref_mute_led_nid = spec->gpio_led;
 				return 1;
 			}
 			if (sscanf(dev->name, "HP_Mute_LED_%d",
@@ -5043,29 +5050,12 @@ static int stac92xx_pre_resume(struct hda_codec *codec)
 	struct sigmatel_spec *spec = codec->spec;
 
 	/* sync mute LED */
-	if (spec->gpio_led) {
-		if (spec->gpio_led <= 8) {
-			stac_gpio_set(codec, spec->gpio_mask,
-					spec->gpio_dir, spec->gpio_data);
-		} else {
-			stac_vrefout_set(codec,
-					spec->gpio_led, spec->vref_led);
-		}
-	}
-	return 0;
-}
-
-static int stac92xx_post_suspend(struct hda_codec *codec)
-{
-	struct sigmatel_spec *spec = codec->spec;
-	if (spec->gpio_led > 8) {
-		/* with vref-out pin used for mute led control
-		 * codec AFG is prevented from D3 state, but on
-		 * system suspend it can (and should) be used
-		 */
-		snd_hda_codec_read(codec, codec->afg, 0,
-				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	}
+	if (spec->vref_mute_led_nid)
+		stac_vrefout_set(codec, spec->vref_mute_led_nid,
+				 spec->vref_led);
+	else if (spec->gpio_led)
+		stac_gpio_set(codec, spec->gpio_mask,
+			      spec->gpio_dir, spec->gpio_data);
 	return 0;
 }
 
@@ -5076,7 +5066,7 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 	struct sigmatel_spec *spec = codec->spec;
 
 	if (power_state == AC_PWRST_D3) {
-		if (spec->gpio_led > 8) {
+		if (spec->vref_mute_led_nid) {
 			/* with vref-out pin used for mute led control
 			 * codec AFG is prevented from D3 state
 			 */
@@ -5129,7 +5119,7 @@ static int stac92xx_update_led_status(struct hda_codec *codec)
 		}
 	}
 	/*polarity defines *not* muted state level*/
-	if (spec->gpio_led <= 8) {
+	if (!spec->vref_mute_led_nid) {
 		if (muted)
 			spec->gpio_data &= ~spec->gpio_led; /* orange */
 		else
@@ -5147,7 +5137,8 @@ static int stac92xx_update_led_status(struct hda_codec *codec)
 		muted_lvl = spec->gpio_led_polarity ?
 				AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ;
 		spec->vref_led = muted ? muted_lvl : notmtd_lvl;
-		stac_vrefout_set(codec,	spec->gpio_led, spec->vref_led);
+		stac_vrefout_set(codec,	spec->vref_mute_led_nid,
+				 spec->vref_led);
 	}
 	return 0;
 }
@@ -5661,15 +5652,13 @@ again:
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (spec->gpio_led) {
-		if (spec->gpio_led <= 8) {
+		if (!spec->vref_mute_led_nid) {
 			spec->gpio_mask |= spec->gpio_led;
 			spec->gpio_dir |= spec->gpio_led;
 			spec->gpio_data |= spec->gpio_led;
 		} else {
 			codec->patch_ops.set_power_state =
 					stac92xx_set_power_state;
-			codec->patch_ops.post_suspend =
-					stac92xx_post_suspend;
 		}
 		codec->patch_ops.pre_resume = stac92xx_pre_resume;
 		codec->patch_ops.check_power_status =
@@ -5976,15 +5965,13 @@ again:
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (spec->gpio_led) {
-		if (spec->gpio_led <= 8) {
+		if (!spec->vref_mute_led_nid) {
 			spec->gpio_mask |= spec->gpio_led;
 			spec->gpio_dir |= spec->gpio_led;
 			spec->gpio_data |= spec->gpio_led;
 		} else {
 			codec->patch_ops.set_power_state =
 					stac92xx_set_power_state;
-			codec->patch_ops.post_suspend =
-					stac92xx_post_suspend;
 		}
 		codec->patch_ops.pre_resume = stac92xx_pre_resume;
 		codec->patch_ops.check_power_status =
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index a391e62..28dfafb 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -41,6 +41,7 @@ MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}");
 static int index = SNDRV_DEFAULT_IDX1;	/* Index 0-MAX */
 static char *id = SNDRV_DEFAULT_STR1;	/* ID for this card */
 static int enable = 1;
+static int codecs = 1;
 
 module_param(index, int, 0444);
 MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator.");
@@ -48,6 +49,8 @@ module_param(id, charp, 0444);
 MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator.");
 module_param(enable, bool, 0444);
 MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator.");
+module_param(codecs, int, 0444);
+MODULE_PARM_DESC(codecs, "Set bit to indicate that codec number is expected to be present (default 1)");
 
 static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = {
 	{ PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) },
@@ -140,6 +143,9 @@ struct sis7019 {
 	dma_addr_t silence_dma_addr;
 };
 
+/* These values are also used by the module param 'codecs' to indicate
+ * which codecs should be present.
+ */
 #define SIS_PRIMARY_CODEC_PRESENT	0x0001
 #define SIS_SECONDARY_CODEC_PRESENT	0x0002
 #define SIS_TERTIARY_CODEC_PRESENT	0x0004
@@ -1078,6 +1084,7 @@ static int sis_chip_init(struct sis7019 *sis)
 {
 	unsigned long io = sis->ioport;
 	void __iomem *ioaddr = sis->ioaddr;
+	unsigned long timeout;
 	u16 status;
 	int count;
 	int i;
@@ -1104,21 +1111,45 @@ static int sis_chip_init(struct sis7019 *sis)
 	while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count)
 		udelay(1);
 
+	/* Command complete, we can let go of the semaphore now.
+	 */
+	outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA);
+	if (!count)
+		return -EIO;
+
 	/* Now that we've finished the reset, find out what's attached.
+	 * There are some codec/board combinations that take an extremely
+	 * long time to come up. 350+ ms has been observed in the field,
+	 * so we'll give them up to 500ms.
 	 */
-	status = inl(io + SIS_AC97_STATUS);
-	if (status & SIS_AC97_STATUS_CODEC_READY)
-		sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT;
-	if (status & SIS_AC97_STATUS_CODEC2_READY)
-		sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT;
-	if (status & SIS_AC97_STATUS_CODEC3_READY)
-		sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT;
-
-	/* All done, let go of the semaphore, and check for errors
+	sis->codecs_present = 0;
+	timeout = msecs_to_jiffies(500) + jiffies;
+	while (time_before_eq(jiffies, timeout)) {
+		status = inl(io + SIS_AC97_STATUS);
+		if (status & SIS_AC97_STATUS_CODEC_READY)
+			sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT;
+		if (status & SIS_AC97_STATUS_CODEC2_READY)
+			sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT;
+		if (status & SIS_AC97_STATUS_CODEC3_READY)
+			sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT;
+
+		if (sis->codecs_present == codecs)
+			break;
+
+		msleep(1);
+	}
+
+	/* All done, check for errors.
 	 */
-	outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA);
-	if (!sis->codecs_present || !count)
+	if (!sis->codecs_present) {
+		printk(KERN_ERR "sis7019: could not find any codecs\n");
 		return -EIO;
+	}
+
+	if (sis->codecs_present != codecs) {
+		printk(KERN_WARNING "sis7019: missing codecs, found %0x, expected %0x\n",
+		       sis->codecs_present, codecs);
+	}
 
 	/* Let the hardware know that the audio driver is alive,
 	 * and enable PCM slots on the AC-link for L/R playback (3 & 4) and
@@ -1390,6 +1421,17 @@ static int __devinit snd_sis7019_probe(struct pci_dev *pci,
 	if (!enable)
 		goto error_out;
 
+	/* The user can specify which codecs should be present so that we
+	 * can wait for them to show up if they are slow to recover from
+	 * the AC97 cold reset. We default to a single codec, the primary.
+	 *
+	 * We assume that SIS_PRIMARY_*_PRESENT matches bits 0-2.
+	 */
+	codecs &= SIS_PRIMARY_CODEC_PRESENT | SIS_SECONDARY_CODEC_PRESENT |
+		  SIS_TERTIARY_CODEC_PRESENT;
+	if (!codecs)
+		codecs = SIS_PRIMARY_CODEC_PRESENT;
+
 	rc = snd_card_create(index, id, THIS_MODULE, sizeof(*sis), &card);
 	if (rc < 0)
 		goto error_out;
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index bee3c94..d1fcc81 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -1,6 +1,6 @@
 config SND_ATMEL_SOC
 	tristate "SoC Audio for the Atmel System-on-Chip"
-	depends on ARCH_AT91 || AVR32
+	depends on ARCH_AT91
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the ATMEL SSC interface. You will also need
@@ -24,25 +24,6 @@ config SND_AT91_SOC_SAM9G20_WM8731
 	  Say Y if you want to add support for SoC audio on WM8731-based
 	  AT91sam9g20 evaluation board.
 
-config SND_AT32_SOC_PLAYPAQ
-        tristate "SoC Audio support for PlayPaq with WM8510"
-        depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS
-        select SND_ATMEL_SOC_SSC
-        select SND_SOC_WM8510
-        help
-          Say Y or M here if you want to add support for SoC audio
-          on the LRS PlayPaq.
-
-config SND_AT32_SOC_PLAYPAQ_SLAVE
-        bool "Run CODEC on PlayPaq in slave mode"
-        depends on SND_AT32_SOC_PLAYPAQ
-        default n
-        help
-          Say Y if you want to run with the AT32 SSC generating the BCLK
-          and FRAME signals on the PlayPaq.  Unless you want to play
-          with the AT32 as the SSC master, you probably want to say N here,
-          as this will give you better sound quality.
-
 config SND_AT91_SOC_AFEB9260
 	tristate "SoC Audio support for AFEB9260 board"
 	depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index e7ea56b..a5c0bf1 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -8,9 +8,5 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
 # AT91 Machine Support
 snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
 
-# AT32 Machine Support
-snd-soc-playpaq-objs := playpaq_wm8510.o
-
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
-obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
 obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
deleted file mode 100644
index 73ae99a..0000000
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ /dev/null
@@ -1,473 +0,0 @@
-/* sound/soc/at32/playpaq_wm8510.c
- * ASoC machine driver for PlayPaq using WM8510 codec
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
- * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c
- *
- * NOTE: If you don't have the AT32 enhanced portmux configured (which
- * isn't currently in the mainline or Atmel patched kernel), you will
- * need to set the MCLK pin (PA30) to peripheral A in your board initialization
- * code.  Something like:
- *	at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0);
- *
- */
-
-/* #define DEBUG */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/kernel.h>
-#include <linux/errno.h>
-#include <linux/clk.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <mach/at32ap700x.h>
-#include <mach/portmux.h>
-
-#include "../codecs/wm8510.h"
-#include "atmel-pcm.h"
-#include "atmel_ssc_dai.h"
-
-
-/*-------------------------------------------------------------------------*\
- * constants
-\*-------------------------------------------------------------------------*/
-#define MCLK_PIN		GPIO_PIN_PA(30)
-#define MCLK_PERIPH		GPIO_PERIPH_A
-
-
-/*-------------------------------------------------------------------------*\
- * data types
-\*-------------------------------------------------------------------------*/
-/* SSC clocking data */
-struct ssc_clock_data {
-	/* CMR div */
-	unsigned int cmr_div;
-
-	/* Frame period (as needed by xCMR.PERIOD) */
-	unsigned int period;
-
-	/* The SSC clock rate these settings where calculated for */
-	unsigned long ssc_rate;
-};
-
-
-/*-------------------------------------------------------------------------*\
- * module data
-\*-------------------------------------------------------------------------*/
-static struct clk *_gclk0;
-static struct clk *_pll0;
-
-#define CODEC_CLK (_gclk0)
-
-
-/*-------------------------------------------------------------------------*\
- * Sound SOC operations
-\*-------------------------------------------------------------------------*/
-#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock(
-	struct snd_pcm_hw_params *params,
-	struct snd_soc_dai *cpu_dai)
-{
-	struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai);
-	struct ssc_device *ssc = ssc_p->ssc;
-	struct ssc_clock_data cd;
-	unsigned int rate, width_bits, channels;
-	unsigned int bitrate, ssc_div;
-	unsigned actual_rate;
-
-
-	/*
-	 * Figure out required bitrate
-	 */
-	rate = params_rate(params);
-	channels = params_channels(params);
-	width_bits = snd_pcm_format_physical_width(params_format(params));
-	bitrate = rate * width_bits * channels;
-
-
-	/*
-	 * Figure out required SSC divider and period for required bitrate
-	 */
-	cd.ssc_rate = clk_get_rate(ssc->clk);
-	ssc_div = cd.ssc_rate / bitrate;
-	cd.cmr_div = ssc_div / 2;
-	if (ssc_div & 1) {
-		/* round cmr_div up */
-		cd.cmr_div++;
-	}
-	cd.period = width_bits - 1;
-
-
-	/*
-	 * Find actual rate, compare to requested rate
-	 */
-	actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
-	pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n",
-		 rate, actual_rate);
-
-
-	return cd;
-}
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
-
-static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream,
-				    struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai);
-	struct ssc_device *ssc = ssc_p->ssc;
-	unsigned int pll_out = 0, bclk = 0, mclk_div = 0;
-	int ret;
-
-
-	/* Due to difficulties with getting the correct clocks from the AT32's
-	 * PLL0, we're going to let the CODEC be in charge of all the clocks
-	 */
-#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-	const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
-#else
-	struct ssc_clock_data cd;
-	const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBS_CFS);
-#endif
-
-	if (ssc == NULL) {
-		pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n");
-		return -EINVAL;
-	}
-
-
-	/*
-	 * Figure out PLL and BCLK dividers for WM8510
-	 */
-	switch (params_rate(params)) {
-	case 48000:
-		pll_out = 24576000;
-		mclk_div = WM8510_MCLKDIV_2;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 44100:
-		pll_out = 22579200;
-		mclk_div = WM8510_MCLKDIV_2;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 22050:
-		pll_out = 22579200;
-		mclk_div = WM8510_MCLKDIV_4;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 16000:
-		pll_out = 24576000;
-		mclk_div = WM8510_MCLKDIV_6;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 11025:
-		pll_out = 22579200;
-		mclk_div = WM8510_MCLKDIV_8;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	case 8000:
-		pll_out = 24576000;
-		mclk_div = WM8510_MCLKDIV_12;
-		bclk = WM8510_BCLKDIV_8;
-		break;
-
-	default:
-		pr_warning("playpaq_wm8510: Unsupported sample rate %d\n",
-			   params_rate(params));
-		return -EINVAL;
-	}
-
-
-	/*
-	 * set CPU and CODEC DAI configuration
-	 */
-	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: "
-			   "Failed to set CODEC DAI format (%d)\n",
-			   ret);
-		return ret;
-	}
-	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: "
-			   "Failed to set CPU DAI format (%d)\n",
-			   ret);
-		return ret;
-	}
-
-
-	/*
-	 * Set CPU clock configuration
-	 */
-#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-	cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai);
-	pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n",
-		 cd.cmr_div, cd.period);
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n",
-			   ret);
-		return ret;
-	}
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD,
-					  cd.period);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: "
-			   "Failed to set CPU transmit period (%d)\n",
-			   ret);
-		return ret;
-	}
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
-	/*
-	 * Set CODEC clock configuration
-	 */
-	pr_debug("playpaq_wm8510: "
-		 "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n",
-		 clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div);
-
-
-#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk);
-	if (ret < 0) {
-		pr_warning
-		    ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n",
-		     ret);
-		return ret;
-	}
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
-	ret = snd_soc_dai_set_pll(codec_dai, 0, 0,
-					 clk_get_rate(CODEC_CLK), pll_out);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n",
-			   ret);
-		return ret;
-	}
-
-
-	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div);
-	if (ret < 0) {
-		pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n",
-			   ret);
-		return ret;
-	}
-
-
-	return 0;
-}
-
-
-
-static struct snd_soc_ops playpaq_wm8510_ops = {
-	.hw_params = playpaq_wm8510_hw_params,
-};
-
-
-
-static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = {
-	SND_SOC_DAPM_MIC("Int Mic", NULL),
-	SND_SOC_DAPM_SPK("Ext Spk", NULL),
-};
-
-
-
-static const struct snd_soc_dapm_route intercon[] = {
-	/* speaker connected to SPKOUT */
-	{"Ext Spk", NULL, "SPKOUTP"},
-	{"Ext Spk", NULL, "SPKOUTN"},
-
-	{"Mic Bias", NULL, "Int Mic"},
-	{"MICN", NULL, "Mic Bias"},
-	{"MICP", NULL, "Mic Bias"},
-};
-
-
-
-static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int i;
-
-	/*
-	 * Add DAPM widgets
-	 */
-	for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++)
-		snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]);
-
-
-
-	/*
-	 * Setup audio path interconnects
-	 */
-	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-
-
-
-	/* always connected pins */
-	snd_soc_dapm_enable_pin(dapm, "Int Mic");
-	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
-
-
-
-	/* Make CSB show PLL rate */
-	snd_soc_dai_set_clkdiv(rtd->codec_dai, WM8510_OPCLKDIV,
-				       WM8510_OPCLKDIV_1 | 4);
-
-	return 0;
-}
-
-
-
-static struct snd_soc_dai_link playpaq_wm8510_dai = {
-	.name = "WM8510",
-	.stream_name = "WM8510 PCM",
-	.cpu_dai_name= "atmel-ssc-dai.0",
-	.platform_name = "atmel-pcm-audio",
-	.codec_name = "wm8510-codec.0-0x1a",
-	.codec_dai_name = "wm8510-hifi",
-	.init = playpaq_wm8510_init,
-	.ops = &playpaq_wm8510_ops,
-};
-
-
-
-static struct snd_soc_card snd_soc_playpaq = {
-	.name = "LRS_PlayPaq_WM8510",
-	.dai_link = &playpaq_wm8510_dai,
-	.num_links = 1,
-};
-
-static struct platform_device *playpaq_snd_device;
-
-
-static int __init playpaq_asoc_init(void)
-{
-	int ret = 0;
-
-	/*
-	 * Configure MCLK for WM8510
-	 */
-	_gclk0 = clk_get(NULL, "gclk0");
-	if (IS_ERR(_gclk0)) {
-		_gclk0 = NULL;
-		ret = PTR_ERR(_gclk0);
-		goto err_gclk0;
-	}
-	_pll0 = clk_get(NULL, "pll0");
-	if (IS_ERR(_pll0)) {
-		_pll0 = NULL;
-		ret = PTR_ERR(_pll0);
-		goto err_pll0;
-	}
-	ret = clk_set_parent(_gclk0, _pll0);
-	if (ret) {
-		pr_warning("snd-soc-playpaq: "
-			   "Failed to set PLL0 as parent for DAC clock\n");
-		goto err_set_clk;
-	}
-	clk_set_rate(CODEC_CLK, 12000000);
-	clk_enable(CODEC_CLK);
-
-#if defined CONFIG_AT32_ENHANCED_PORTMUX
-	at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0);
-#endif
-
-
-	/*
-	 * Create and register platform device
-	 */
-	playpaq_snd_device = platform_device_alloc("soc-audio", 0);
-	if (playpaq_snd_device == NULL) {
-		ret = -ENOMEM;
-		goto err_device_alloc;
-	}
-
-	platform_set_drvdata(playpaq_snd_device, &snd_soc_playpaq);
-
-	ret = platform_device_add(playpaq_snd_device);
-	if (ret) {
-		pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n",
-			   ret);
-		goto err_device_add;
-	}
-
-	return 0;
-
-
-err_device_add:
-	if (playpaq_snd_device != NULL) {
-		platform_device_put(playpaq_snd_device);
-		playpaq_snd_device = NULL;
-	}
-err_device_alloc:
-err_set_clk:
-	if (_pll0 != NULL) {
-		clk_put(_pll0);
-		_pll0 = NULL;
-	}
-err_pll0:
-	if (_gclk0 != NULL) {
-		clk_put(_gclk0);
-		_gclk0 = NULL;
-	}
-	return ret;
-}
-
-
-static void __exit playpaq_asoc_exit(void)
-{
-	if (_gclk0 != NULL) {
-		clk_put(_gclk0);
-		_gclk0 = NULL;
-	}
-	if (_pll0 != NULL) {
-		clk_put(_pll0);
-		_pll0 = NULL;
-	}
-
-#if defined CONFIG_AT32_ENHANCED_PORTMUX
-	at32_free_pin(MCLK_PIN);
-#endif
-
-	platform_device_unregister(playpaq_snd_device);
-	playpaq_snd_device = NULL;
-}
-
-module_init(playpaq_asoc_init);
-module_exit(playpaq_asoc_exit);
-
-MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
-MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 444747f..dd7be0d 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -34,7 +34,7 @@
 
 #define AD1836_ADC_CTRL2               13
 #define AD1836_ADC_WORD_LEN_MASK       0x30
-#define AD1836_ADC_WORD_OFFSET         5
+#define AD1836_ADC_WORD_OFFSET         4
 #define AD1836_ADC_SERFMT_MASK         (7 << 6)
 #define AD1836_ADC_SERFMT_PCK256       (0x4 << 6)
 #define AD1836_ADC_SERFMT_PCK128       (0x5 << 6)
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index f1f237e..73f46eb 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -601,7 +601,6 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
 static int cs4270_soc_resume(struct snd_soc_codec *codec)
 {
 	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
-	struct i2c_client *i2c_client = to_i2c_client(codec->dev);
 	int reg;
 
 	regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
@@ -612,14 +611,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec)
 	ndelay(500);
 
 	/* first restore the entire register cache ... */
-	for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
-		u8 val = snd_soc_read(codec, reg);
-
-		if (i2c_smbus_write_byte_data(i2c_client, reg, val)) {
-			dev_err(codec->dev, "i2c write failed\n");
-			return -EIO;
-		}
-	}
+	snd_soc_cache_sync(codec);
 
 	/* ... then disable the power-down bits */
 	reg = snd_soc_read(codec, CS4270_PWRCTL);
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 8c3c820..1ee66361 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -555,7 +555,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec)
 
 static struct snd_soc_codec_driver soc_codec_device_cs42l51 = {
 	.probe =	cs42l51_probe,
-	.reg_cache_size = CS42L51_NUMREGS,
+	.reg_cache_size = CS42L51_NUMREGS + 1,
 	.reg_word_size = sizeof(u8),
 };
 
diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c
index 9e7e964..dcf6f2a 100644
--- a/sound/soc/codecs/max9877.c
+++ b/sound/soc/codecs/max9877.c
@@ -106,13 +106,13 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol,
 	unsigned int mask = mc->max;
 	unsigned int val = (ucontrol->value.integer.value[0] & mask);
 	unsigned int val2 = (ucontrol->value.integer.value[1] & mask);
-	unsigned int change = 1;
+	unsigned int change = 0;
 
-	if (((max9877_regs[reg] >> shift) & mask) == val)
-		change = 0;
+	if (((max9877_regs[reg] >> shift) & mask) != val)
+		change = 1;
 
-	if (((max9877_regs[reg2] >> shift) & mask) == val2)
-		change = 0;
+	if (((max9877_regs[reg2] >> shift) & mask) != val2)
+		change = 1;
 
 	if (change) {
 		max9877_regs[reg] &= ~(mask << shift);
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index c5ca8cf..0441893 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -863,13 +863,13 @@ static struct i2c_driver uda1380_i2c_driver = {
 
 static int __init uda1380_modinit(void)
 {
-	int ret;
+	int ret = 0;
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	ret = i2c_add_driver(&uda1380_i2c_driver);
 	if (ret != 0)
 		pr_err("Failed to register UDA1380 I2C driver: %d\n", ret);
 #endif
-	return 0;
+	return ret;
 }
 module_init(uda1380_modinit);
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 9c982e4..d0c545b 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1325,15 +1325,15 @@ SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
 };
 
 static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = {
-SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
-		   adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
-SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
-		   adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_VIRT_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
+			adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_VIRT_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
+			adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
 };
 
 static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = {
-SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
-SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
+SND_SOC_DAPM_VIRT_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
+SND_SOC_DAPM_VIRT_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
 };
 
 static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = {
@@ -2357,6 +2357,11 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
 	bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT;
 
 	lrclk = bclk_rate / params_rate(params);
+	if (!lrclk) {
+		dev_err(dai->dev, "Unable to generate LRCLK from %dHz BCLK\n",
+			bclk_rate);
+		return -EINVAL;
+	}
 	dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n",
 		lrclk, bclk_rate / lrclk);
 
@@ -3178,6 +3183,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		switch (wm8994->revision) {
 		case 0:
 		case 1:
+		case 2:
+		case 3:
 			wm8994->hubs.dcs_codes_l = -9;
 			wm8994->hubs.dcs_codes_r = -5;
 			break;
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 31af405..ae49f1c 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -392,7 +392,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 	}
 
 	if (strcasecmp(sprop, "i2s-slave") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
 
@@ -409,31 +410,38 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 		}
 		machine_data->clk_frequency = be32_to_cpup(iprop);
 	} else if (strcasecmp(sprop, "i2s-master") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
 	} else if (strcasecmp(sprop, "lj-slave") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
 	} else if (strcasecmp(sprop, "lj-master") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
 	} else if (strcasecmp(sprop, "rj-slave") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
 	} else if (strcasecmp(sprop, "rj-master") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
 	} else if (strcasecmp(sprop, "ac97-slave") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
 	} else if (strcasecmp(sprop, "ac97-master") == 0) {
-		machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+		machine_data->dai_format =
+			SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS;
 		machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
 		machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
 	} else {
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index b133bfc..7383917 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -28,7 +28,7 @@ config SND_MXC_SOC_WM1133_EV1
 
 config SND_SOC_MX27VIS_AIC32X4
 	tristate "SoC audio support for Visstrim M10 boards"
-	depends on MACH_IMX27_VISSTRIM_M10
+	depends on MACH_IMX27_VISSTRIM_M10 && I2C
 	select SND_SOC_TLV320AIC32X4
 	select SND_MXC_SOC_MX2
 	help
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 8f49e16..c62d715 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -12,6 +12,7 @@ config SND_KIRKWOOD_SOC_I2S
 config SND_KIRKWOOD_SOC_OPENRD
 	tristate "SoC Audio support for Kirkwood Openrd Client"
 	depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE)
+	depends on I2C
 	select SND_KIRKWOOD_SOC_I2S
 	select SND_SOC_CS42L51
 	help
@@ -20,7 +21,7 @@ config SND_KIRKWOOD_SOC_OPENRD
 
 config SND_KIRKWOOD_SOC_T5325
 	tristate "SoC Audio support for HP t5325"
-	depends on SND_KIRKWOOD_SOC && MACH_T5325
+	depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C
 	select SND_KIRKWOOD_SOC_I2S
 	select SND_SOC_ALC5623
 	help
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index ffd2242..a0f7d3c 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -151,6 +151,7 @@ config SND_SOC_ZYLONITE
 config SND_SOC_RAUMFELD
 	tristate "SoC Audio support Raumfeld audio adapter"
 	depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR)
+	depends on I2C && SPI_MASTER
 	select SND_PXA_SOC_SSP
 	select SND_SOC_CS4270
 	select SND_SOC_AK4104
@@ -159,7 +160,7 @@ config SND_SOC_RAUMFELD
 
 config SND_PXA2XX_SOC_HX4700
 	tristate "SoC Audio support for HP iPAQ hx4700"
-	depends on SND_PXA2XX_SOC && MACH_H4700
+	depends on SND_PXA2XX_SOC && MACH_H4700 && I2C
 	select SND_PXA2XX_SOC_I2S
 	select SND_SOC_AK4641
 	help
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index f75e439..ad9ac42 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -9,6 +9,7 @@
 
 #include "../codecs/wm8994.h"
 #include <sound/pcm_params.h>
+#include <linux/module.h>
 
  /*
   * Default CFG switch settings to use this driver:
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 85bf541..4b8e354 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -191,7 +191,7 @@ static int speyside_late_probe(struct snd_soc_card *card)
 	snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic");
 	snd_soc_dapm_ignore_suspend(&card->dapm, "Main AMIC");
 	snd_soc_dapm_ignore_suspend(&card->dapm, "Main DMIC");
-	snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker");
+	snd_soc_dapm_ignore_suspend(&card->dapm, "Main Speaker");
 	snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Output");
 	snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Input");
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a5d3685..a25fa63 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -709,6 +709,12 @@ int snd_soc_resume(struct device *dev)
 	struct snd_soc_card *card = dev_get_drvdata(dev);
 	int i, ac97_control = 0;
 
+	/* If the initialization of this soc device failed, there is no codec
+	 * associated with it. Just bail out in this case.
+	 */
+	if (list_empty(&card->codec_dev_list))
+		return 0;
+
 	/* AC97 devices might have other drivers hanging off them so
 	 * need to resume immediately.  Other drivers don't have that
 	 * problem and may take a substantial amount of time to resume
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 0c12b98..4220bb0 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -58,7 +58,36 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params)
 }
 EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk);
 
-static struct snd_soc_platform_driver dummy_platform;
+static const struct snd_pcm_hardware dummy_dma_hardware = {
+	.formats		= 0xffffffff,
+	.channels_min		= 1,
+	.channels_max		= UINT_MAX,
+
+	/* Random values to keep userspace happy when checking constraints */
+	.info			= SNDRV_PCM_INFO_INTERLEAVED |
+				  SNDRV_PCM_INFO_BLOCK_TRANSFER,
+	.buffer_bytes_max	= 128*1024,
+	.period_bytes_min	= PAGE_SIZE,
+	.period_bytes_max	= PAGE_SIZE*2,
+	.periods_min		= 2,
+	.periods_max		= 128,
+};
+
+static int dummy_dma_open(struct snd_pcm_substream *substream)
+{
+	snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware);
+
+	return 0;
+}
+
+static struct snd_pcm_ops dummy_dma_ops = {
+	.open		= dummy_dma_open,
+	.ioctl		= snd_pcm_lib_ioctl,
+};
+
+static struct snd_soc_platform_driver dummy_platform = {
+	.ops = &dummy_dma_ops,
+};
 
 static __devinit int snd_soc_dummy_probe(struct platform_device *pdev)
 {
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index b61945f..32d2a21 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1633,6 +1633,37 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* Roland GAIA SH-01 */
+	USB_DEVICE(0x0582, 0x0111),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "Roland",
+		.product_name = "GAIA",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_MIDI_FIXED_ENDPOINT,
+				.data = &(const struct snd_usb_midi_endpoint_info) {
+				.out_cables = 0x0003,
+				.in_cables  = 0x0003
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
+{
 	USB_DEVICE(0x0582, 0x0113),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
 		/* .vendor_name = "BOSS", */