Merge commit 'v2.6.27-rc7' into x86/microcode

55 files changed, 570 insertions, 293 deletions

diff --git a/sound/Kconfig b/sound/Kconfig
index a37bee094eb..8ebf512ced6 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -91,6 +91,9 @@ endif # SOUND_PRIME
 
 endif # !M68K
 
+endif # SOUND
+
+# AC97_BUS is used from both sound and ucb1400
 config AC97_BUS
 	tristate
 	help
@@ -99,4 +102,3 @@ config AC97_BUS
 	  sound although they're sharing the AC97 bus. Concerned drivers
 	  should "select" this.
 
-endif # SOUND
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 5b3274b465e..199cca3366d 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -26,10 +26,10 @@
 
 #include <asm/irq.h>
 #include <linux/mutex.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/pxa2xx-gpio.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 
diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c
index 0ede9e4656a..381094aab23 100644
--- a/sound/arm/pxa2xx-pcm.c
+++ b/sound/arm/pxa2xx-pcm.c
@@ -21,8 +21,8 @@
 #include <sound/pcm_params.h>
 
 #include <asm/dma.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
 
 #include "pxa2xx-pcm.h"
 
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
index faeddf3eced..b9c51bf8cd7 100644
--- a/sound/arm/sa11xx-uda1341.c
+++ b/sound/arm/sa11xx-uda1341.c
@@ -71,8 +71,8 @@
 #include <linux/pm.h>
 #endif
 
-#include <asm/hardware.h>
-#include <asm/arch/h3600.h>
+#include <mach/hardware.h>
+#include <mach/h3600.h>
 #include <asm/mach-types.h>
 #include <asm/dma.h>
 
diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c
index 558dadbf45f..e024e4588b8 100644
--- a/sound/core/seq/oss/seq_oss_synth.c
+++ b/sound/core/seq/oss/seq_oss_synth.c
@@ -604,6 +604,9 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in
 {
 	struct seq_oss_synth *rec;
 
+	if (dev < 0 || dev >= dp->max_synthdev)
+		return -ENXIO;
+
 	if (dp->synths[dev].is_midi) {
 		struct midi_info minf;
 		snd_seq_oss_midi_make_info(dp, dp->synths[dev].midi_mapped, &minf);
diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c
index ee0741f9eb5..fbef38a9604 100644
--- a/sound/mips/au1x00.c
+++ b/sound/mips/au1x00.c
@@ -38,7 +38,6 @@
 #include <linux/interrupt.h>
 #include <linux/init.h>
 #include <linux/slab.h>
-#include <linux/version.h>
 #include <sound/core.h>
 #include <sound/initval.h>
 #include <sound/pcm.h>
diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c
index bb4a0969f46..725fef0f59a 100644
--- a/sound/oss/vidc.c
+++ b/sound/oss/vidc.c
@@ -22,7 +22,7 @@
 #include <linux/kernel.h>
 #include <linux/interrupt.h>
 
-#include <asm/hardware.h>
+#include <mach/hardware.h>
 #include <asm/dma.h>
 #include <asm/io.h>
 #include <asm/hardware/iomd.h>
diff --git a/sound/oss/vidc_fill.S b/sound/oss/vidc_fill.S
index 01ccc074cc1..bed34921d17 100644
--- a/sound/oss/vidc_fill.S
+++ b/sound/oss/vidc_fill.S
@@ -11,7 +11,7 @@
  */
 #include <linux/linkage.h>
 #include <asm/assembler.h>
-#include <asm/hardware.h>
+#include <mach/hardware.h>
 #include <asm/hardware/iomd.h>
 
 		.text
diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c
index 88490418f93..c47842fad65 100644
--- a/sound/oss/waveartist.c
+++ b/sound/oss/waveartist.c
@@ -47,7 +47,7 @@
 #include "waveartist.h"
 
 #ifdef CONFIG_ARM
-#include <asm/hardware.h>
+#include <mach/hardware.h>
 #include <asm/mach-types.h>
 #endif
 
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index f7d95b224a9..31f52d3fc21 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -845,7 +845,7 @@ config SND_VIRTUOSO
 	select SND_OXYGEN_LIB
 	help
 	  Say Y here to include support for sound cards based on the
-	  Asus AV100/AV200 chips, i.e., Xonar D2, DX and D2X.
+	  Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2 and D2X.
 
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-virtuoso.
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 2f8b28add27..6abe8a3bd36 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -249,11 +249,12 @@ static struct snd_ca0106_details ca0106_chip_details[] = {
 	   .name   = "MSI K8N Diamond MB [SB0438]",
 	   .gpio_type = 2,
 	   .i2c_adc = 1 } ,
-	 /* Another MSI K8N Diamond MB, which has apprently a different SSID */
+	 /* MSI K8N Diamond PLUS MB */
 	 { .serial = 0x10091102,
 	   .name   = "MSI K8N Diamond MB",
 	   .gpio_type = 2,
-	   .i2c_adc = 1 } ,
+	   .i2c_adc = 1,
+	   .spi_dac = 2 },
 	 /* Shuttle XPC SD31P which has an onboard Creative Labs
 	  * Sound Blaster Live! 24-bit EAX
 	  * high-definition 7.1 audio processor".
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ef9f072b47f..1c53e337ecb 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -101,6 +101,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
 			 "{Intel, ICH8},"
 			 "{Intel, ICH9},"
 			 "{Intel, ICH10},"
+			 "{Intel, PCH},"
 			 "{Intel, SCH},"
 			 "{ATI, SB450},"
 			 "{ATI, SB600},"
@@ -277,6 +278,9 @@ enum {
 /* Defines for Nvidia HDA support */
 #define NVIDIA_HDA_TRANSREG_ADDR      0x4e
 #define NVIDIA_HDA_ENABLE_COHBITS     0x0f
+#define NVIDIA_HDA_ISTRM_COH          0x4d
+#define NVIDIA_HDA_OSTRM_COH          0x4c
+#define NVIDIA_HDA_ENABLE_COHBIT      0x01
 
 /* Defines for Intel SCH HDA snoop control */
 #define INTEL_SCH_HDA_DEVC      0x78
@@ -899,6 +903,12 @@ static void azx_init_pci(struct azx *chip)
 		update_pci_byte(chip->pci,
 				NVIDIA_HDA_TRANSREG_ADDR,
 				0x0f, NVIDIA_HDA_ENABLE_COHBITS);
+		update_pci_byte(chip->pci,
+				NVIDIA_HDA_ISTRM_COH,
+				0x01, NVIDIA_HDA_ENABLE_COHBIT);
+		update_pci_byte(chip->pci,
+				NVIDIA_HDA_OSTRM_COH,
+				0x01, NVIDIA_HDA_ENABLE_COHBIT);
 		break;
 	case AZX_DRIVER_SCH:
 		pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop);
@@ -2263,6 +2273,8 @@ static struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH },
 	{ PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH },
 	{ PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH },
+	/* PCH */
+	{ PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH },
 	/* SCH */
 	{ PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH },
 	/* ATI SB 450/600 */
@@ -2272,6 +2284,7 @@ static struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI },
 	{ PCI_DEVICE(0x1002, 0x7919), .driver_data = AZX_DRIVER_ATIHDMI },
 	{ PCI_DEVICE(0x1002, 0x960f), .driver_data = AZX_DRIVER_ATIHDMI },
+	{ PCI_DEVICE(0x1002, 0x970f), .driver_data = AZX_DRIVER_ATIHDMI },
 	{ PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI },
 	{ PCI_DEVICE(0x1002, 0xaa08), .driver_data = AZX_DRIVER_ATIHDMI },
 	{ PCI_DEVICE(0x1002, 0xaa10), .driver_data = AZX_DRIVER_ATIHDMI },
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index add4e87e0b2..66025161bd6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -952,7 +952,7 @@ do_sku:
 					    tmp | 0x2010);
 			break;
 		case 0x10ec0888:
-			alc888_coef_init(codec);
+			/*alc888_coef_init(codec);*/ /* called in alc_init() */
 			break;
 		case 0x10ec0267:
 		case 0x10ec0268:
@@ -2439,6 +2439,8 @@ static int alc_init(struct hda_codec *codec)
 	unsigned int i;
 
 	alc_fix_pll(codec);
+	if (codec->vendor_id == 0x10ec0888)
+		alc888_coef_init(codec);
 
 	for (i = 0; i < spec->num_init_verbs; i++)
 		snd_hda_sequence_write(codec, spec->init_verbs[i]);
@@ -6195,7 +6197,6 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
 	SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
 	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
-	SND_PCI_QUIRK(0x106b, 0x00a0, "Apple iMac 24''", ALC885_IMAC24),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8  */
 	SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
@@ -6437,6 +6438,39 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec)
 	}
 }
 
+static void alc882_auto_init_input_src(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	const struct hda_input_mux *imux = spec->input_mux;
+	int c;
+
+	for (c = 0; c < spec->num_adc_nids; c++) {
+		hda_nid_t conn_list[HDA_MAX_NUM_INPUTS];
+		hda_nid_t nid = spec->capsrc_nids[c];
+		int conns, mute, idx, item;
+
+		conns = snd_hda_get_connections(codec, nid, conn_list,
+						ARRAY_SIZE(conn_list));
+		if (conns < 0)
+			continue;
+		for (idx = 0; idx < conns; idx++) {
+			/* if the current connection is the selected one,
+			 * unmute it as default - otherwise mute it
+			 */
+			mute = AMP_IN_MUTE(idx);
+			for (item = 0; item < imux->num_items; item++) {
+				if (imux->items[item].index == idx) {
+					if (spec->cur_mux[c] == item)
+						mute = AMP_IN_UNMUTE(idx);
+					break;
+				}
+			}
+			snd_hda_codec_write(codec, nid, 0,
+					    AC_VERB_SET_AMP_GAIN_MUTE, mute);
+		}
+	}
+}
+
 /* add mic boosts if needed */
 static int alc_auto_add_mic_boost(struct hda_codec *codec)
 {
@@ -6491,6 +6525,7 @@ static void alc882_auto_init(struct hda_codec *codec)
 	alc882_auto_init_multi_out(codec);
 	alc882_auto_init_hp_out(codec);
 	alc882_auto_init_analog_input(codec);
+	alc882_auto_init_input_src(codec);
 	if (spec->unsol_event)
 		alc_sku_automute(codec);
 }
@@ -8285,6 +8320,8 @@ static void alc883_auto_init_analog_input(struct hda_codec *codec)
 	}
 }
 
+#define alc883_auto_init_input_src	alc882_auto_init_input_src
+
 /* almost identical with ALC880 parser... */
 static int alc883_parse_auto_config(struct hda_codec *codec)
 {
@@ -8315,6 +8352,7 @@ static void alc883_auto_init(struct hda_codec *codec)
 	alc883_auto_init_multi_out(codec);
 	alc883_auto_init_hp_out(codec);
 	alc883_auto_init_analog_input(codec);
+	alc883_auto_init_input_src(codec);
 	if (spec->unsol_event)
 		alc_sku_automute(codec);
 }
@@ -8389,8 +8427,6 @@ static int patch_alc883(struct hda_codec *codec)
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC883_AUTO)
 		spec->init_hook = alc883_auto_init;
-	else if (codec->vendor_id == 0x10ec0888)
-		spec->init_hook = alc888_coef_init;
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (!spec->loopback.amplist)
@@ -9663,6 +9699,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 #define alc262_auto_init_multi_out	alc882_auto_init_multi_out
 #define alc262_auto_init_hp_out		alc882_auto_init_hp_out
 #define alc262_auto_init_analog_input	alc882_auto_init_analog_input
+#define alc262_auto_init_input_src	alc882_auto_init_input_src
 
 
 /* init callback for auto-configuration model -- overriding the default init */
@@ -9672,6 +9709,7 @@ static void alc262_auto_init(struct hda_codec *codec)
 	alc262_auto_init_multi_out(codec);
 	alc262_auto_init_hp_out(codec);
 	alc262_auto_init_analog_input(codec);
+	alc262_auto_init_input_src(codec);
 	if (spec->unsol_event)
 		alc_sku_automute(codec);
 }
@@ -13330,6 +13368,8 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
 	}
 }
 
+#define alc861vd_auto_init_input_src	alc882_auto_init_input_src
+
 #define alc861vd_idx_to_mixer_vol(nid)		((nid) + 0x02)
 #define alc861vd_idx_to_mixer_switch(nid)	((nid) + 0x0c)
 
@@ -13512,6 +13552,7 @@ static void alc861vd_auto_init(struct hda_codec *codec)
 	alc861vd_auto_init_multi_out(codec);
 	alc861vd_auto_init_hp_out(codec);
 	alc861vd_auto_init_analog_input(codec);
+	alc861vd_auto_init_input_src(codec);
 	if (spec->unsol_event)
 		alc_sku_automute(codec);
 }
@@ -14025,6 +14066,13 @@ static struct hda_verb alc662_auto_init_verbs[] = {
 	{ }
 };
 
+/* additional verbs for ALC663 */
+static struct hda_verb alc663_auto_init_verbs[] = {
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{ }
+};
+
 static struct hda_verb alc663_m51va_init_verbs[] = {
 	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -14553,6 +14601,14 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
 	if (!pin)
 		return 0;
 
+	if (pin == 0x17) {
+		/* ALC663 has a mono output pin on 0x17 */
+		sprintf(name, "%s Playback Switch", pfx);
+		err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+				  HDA_COMPOSE_AMP_VAL(pin, 2, 0, HDA_OUTPUT));
+		return err;
+	}
+
 	if (alc880_is_fixed_pin(pin)) {
 		nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
                 /* printk("DAC nid=%x\n",nid); */
@@ -14677,6 +14733,8 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec)
 	}
 }
 
+#define alc662_auto_init_input_src	alc882_auto_init_input_src
+
 static int alc662_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -14721,6 +14779,14 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 	spec->input_mux = &spec->private_imux;
 	
 	spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs;
+	if (codec->vendor_id == 0x10ec0663)
+		spec->init_verbs[spec->num_init_verbs++] =
+			alc663_auto_init_verbs;
+
+	err = alc_auto_add_mic_boost(codec);
+	if (err < 0)
+		return err;
+
 	spec->mixers[spec->num_mixers] = alc662_capture_mixer;
 	spec->num_mixers++;
 	return 1;
@@ -14733,6 +14799,7 @@ static void alc662_auto_init(struct hda_codec *codec)
 	alc662_auto_init_multi_out(codec);
 	alc662_auto_init_hp_out(codec);
 	alc662_auto_init_analog_input(codec);
+	alc662_auto_init_input_src(codec);
 	if (spec->unsol_event)
 		alc_sku_automute(codec);
 }
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 7fdafcb0015..ad994fcab72 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -560,8 +560,9 @@ static struct hda_verb dell_eq_core_init[] = {
 };
 
 static struct hda_verb dell_m6_core_init[] = {
-	/* set master volume and direct control */
-	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+	/* set master volume to max value without distortion
+	 * and direct control */
+	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec},
 	/* setup audio connections */
 	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
 	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c
index 7442460583d..dad393ae040 100644
--- a/sound/pci/oxygen/hifier.c
+++ b/sound/pci/oxygen/hifier.c
@@ -17,6 +17,7 @@
  *  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
  */
 
+#include <linux/delay.h>
 #include <linux/pci.h>
 #include <sound/control.h>
 #include <sound/core.h>
@@ -107,6 +108,9 @@ static void set_ak4396_params(struct oxygen *chip,
 	else
 		value |= AK4396_DFS_QUAD;
 	data->ak4396_ctl2 = value;
+
+	msleep(1); /* wait for the new MCLK to become stable */
+
 	ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB);
 	ak4396_write(chip, AK4396_CONTROL_2, value);
 	ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index 7c8ae31eb46..c5829d30ef8 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -28,6 +28,7 @@
  * GPIO 1 -> DFS1 of AK5385
  */
 
+#include <linux/delay.h>
 #include <linux/mutex.h>
 #include <linux/pci.h>
 #include <sound/ac97_codec.h>
@@ -213,6 +214,9 @@ static void set_ak4396_params(struct oxygen *chip,
 	else
 		value |= AK4396_DFS_QUAD;
 	data->ak4396_ctl2 = value;
+
+	msleep(1); /* wait for the new MCLK to become stable */
+
 	for (i = 0; i < 4; ++i) {
 		ak4396_write(chip, i,
 			     AK4396_CONTROL_1, AK4396_DIF_24_MSB);
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 6facac5aed9..05eb8994c14 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -512,9 +512,12 @@ static int ac97_switch_get(struct snd_kcontrol *ctl,
 
 static void mute_ac97_ctl(struct oxygen *chip, unsigned int control)
 {
-	unsigned int priv_idx = chip->controls[control]->private_value & 0xff;
+	unsigned int priv_idx;
 	u16 value;
 
+	if (!chip->controls[control])
+		return;
+	priv_idx = chip->controls[control]->private_value & 0xff;
 	value = oxygen_read_ac97(chip, 0, priv_idx);
 	if (!(value & 0x8000)) {
 		oxygen_write_ac97(chip, 0, priv_idx, value | 0x8000);
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 9a2c16bf94e..01d7b75f918 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -36,15 +36,15 @@
  */
 
 /*
- * Xonar DX
- * --------
+ * Xonar D1/DX
+ * -----------
  *
  * CMI8788:
  *
  * I²C <-> CS4398 (front)
  *     <-> CS4362A (surround, center/LFE, back)
  *
- * GPI 0 <- external power present
+ * GPI 0 <- external power present (DX only)
  *
  * GPIO 0 -> enable output to speakers
  * GPIO 1 -> enable front panel I/O
@@ -96,6 +96,7 @@ MODULE_PARM_DESC(enable, "enable card");
 enum {
 	MODEL_D2,
 	MODEL_D2X,
+	MODEL_D1,
 	MODEL_DX,
 };
 
@@ -103,6 +104,7 @@ static struct pci_device_id xonar_ids[] __devinitdata = {
 	{ OXYGEN_PCI_SUBID(0x1043, 0x8269), .driver_data = MODEL_D2 },
 	{ OXYGEN_PCI_SUBID(0x1043, 0x8275), .driver_data = MODEL_DX },
 	{ OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X },
+	{ OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 },
 	{ }
 };
 MODULE_DEVICE_TABLE(pci, xonar_ids);
@@ -313,15 +315,12 @@ static void cs43xx_init(struct oxygen *chip)
 	cs4362a_write(chip, 0x01, CS4362A_CPEN);
 }
 
-static void xonar_dx_init(struct oxygen *chip)
+static void xonar_d1_init(struct oxygen *chip)
 {
 	struct xonar_data *data = chip->model_data;
 
 	data->anti_pop_delay = 800;
 	data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
-	data->ext_power_reg = OXYGEN_GPI_DATA;
-	data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
-	data->ext_power_bit = GPI_DX_EXT_POWER;
 	data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST;
 	data->cs4362a_fm = CS4362A_FM_SINGLE |
 		CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
@@ -345,6 +344,16 @@ static void xonar_dx_init(struct oxygen *chip)
 	snd_component_add(chip->card, "CS5361");
 }
 
+static void xonar_dx_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	data->ext_power_reg = OXYGEN_GPI_DATA;
+	data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
+	data->ext_power_bit = GPI_DX_EXT_POWER;
+	xonar_d1_init(chip);
+}
+
 static void xonar_cleanup(struct oxygen *chip)
 {
 	struct xonar_data *data = chip->model_data;
@@ -352,7 +361,7 @@ static void xonar_cleanup(struct oxygen *chip)
 	oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit);
 }
 
-static void xonar_dx_cleanup(struct oxygen *chip)
+static void xonar_d1_cleanup(struct oxygen *chip)
 {
 	xonar_cleanup(chip);
 	cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN);
@@ -365,7 +374,7 @@ static void xonar_d2_resume(struct oxygen *chip)
 	xonar_enable_output(chip);
 }
 
-static void xonar_dx_resume(struct oxygen *chip)
+static void xonar_d1_resume(struct oxygen *chip)
 {
 	cs43xx_init(chip);
 	xonar_enable_output(chip);
@@ -513,7 +522,7 @@ static const struct snd_kcontrol_new front_panel_switch = {
 	.put = front_panel_put,
 };
 
-static void xonar_dx_ac97_switch(struct oxygen *chip,
+static void xonar_d1_ac97_switch(struct oxygen *chip,
 				 unsigned int reg, unsigned int mute)
 {
 	if (reg == AC97_LINE) {
@@ -536,7 +545,7 @@ static int xonar_d2_control_filter(struct snd_kcontrol_new *template)
 	return 0;
 }
 
-static int xonar_dx_control_filter(struct snd_kcontrol_new *template)
+static int xonar_d1_control_filter(struct snd_kcontrol_new *template)
 {
 	if (!strncmp(template->name, "CD Capture ", 11))
 		return 1; /* no CD input */
@@ -548,7 +557,7 @@ static int xonar_mixer_init(struct oxygen *chip)
 	return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip));
 }
 
-static int xonar_dx_mixer_init(struct oxygen *chip)
+static int xonar_d1_mixer_init(struct oxygen *chip)
 {
 	return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip));
 }
@@ -615,23 +624,51 @@ static const struct oxygen_model xonar_models[] = {
 		.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 		.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 	},
+	[MODEL_D1] = {
+		.shortname = "Xonar D1",
+		.longname = "Asus Virtuoso 100",
+		.chip = "AV200",
+		.owner = THIS_MODULE,
+		.init = xonar_d1_init,
+		.control_filter = xonar_d1_control_filter,
+		.mixer_init = xonar_d1_mixer_init,
+		.cleanup = xonar_d1_cleanup,
+		.suspend = xonar_d1_cleanup,
+		.resume = xonar_d1_resume,
+		.set_dac_params = set_cs43xx_params,
+		.set_adc_params = set_cs53x1_params,
+		.update_dac_volume = update_cs43xx_volume,
+		.update_dac_mute = update_cs43xx_mute,
+		.ac97_switch = xonar_d1_ac97_switch,
+		.dac_tlv = cs4362a_db_scale,
+		.model_data_size = sizeof(struct xonar_data),
+		.pcm_dev_cfg = PLAYBACK_0_TO_I2S |
+			       PLAYBACK_1_TO_SPDIF |
+			       CAPTURE_0_FROM_I2S_2,
+		.dac_channels = 8,
+		.dac_volume_min = 0,
+		.dac_volume_max = 127,
+		.function_flags = OXYGEN_FUNCTION_2WIRE,
+		.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+		.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+	},
 	[MODEL_DX] = {
 		.shortname = "Xonar DX",
 		.longname = "Asus Virtuoso 100",
 		.chip = "AV200",
 		.owner = THIS_MODULE,
 		.init = xonar_dx_init,
-		.control_filter = xonar_dx_control_filter,
-		.mixer_init = xonar_dx_mixer_init,
-		.cleanup = xonar_dx_cleanup,
-		.suspend = xonar_dx_cleanup,
-		.resume = xonar_dx_resume,
+		.control_filter = xonar_d1_control_filter,
+		.mixer_init = xonar_d1_mixer_init,
+		.cleanup = xonar_d1_cleanup,
+		.suspend = xonar_d1_cleanup,
+		.resume = xonar_d1_resume,
 		.set_dac_params = set_cs43xx_params,
 		.set_adc_params = set_cs53x1_params,
 		.update_dac_volume = update_cs43xx_volume,
 		.update_dac_mute = update_cs43xx_mute,
 		.gpio_changed = xonar_gpio_changed,
-		.ac97_switch = xonar_dx_ac97_switch,
+		.ac97_switch = xonar_d1_ac97_switch,
 		.dac_tlv = cs4362a_db_scale,
 		.model_data_size = sizeof(struct xonar_data),
 		.pcm_dev_cfg = PLAYBACK_0_TO_I2S |
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
index 9ca11332614..54df8baf916 100644
--- a/sound/sh/aica.c
+++ b/sound/sh/aica.c
@@ -42,7 +42,7 @@
 #include <sound/info.h>
 #include <asm/io.h>
 #include <asm/dma.h>
-#include <asm/dreamcast/sysasic.h>
+#include <mach/sysasic.h>
 #include "aica.h"
 
 MODULE_AUTHOR("Adrian McMenamin <adrian@mcmen.demon.co.uk>");
diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c
index fee5f8e5895..3f326219f1e 100644
--- a/sound/soc/at32/playpaq_wm8510.c
+++ b/sound/soc/at32/playpaq_wm8510.c
@@ -36,8 +36,8 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
-#include <asm/arch/at32ap700x.h>
-#include <asm/arch/portmux.h>
+#include <mach/at32ap700x.h>
+#include <mach/portmux.h>
 
 #include "../codecs/wm8510.h"
 #include "at32-pcm.h"
diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c
index d47492b2b6e..7ab48bd25e4 100644
--- a/sound/soc/at91/at91-pcm.c
+++ b/sound/soc/at91/at91-pcm.c
@@ -28,8 +28,8 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <asm/arch/hardware.h>
-#include <asm/arch/at91_ssc.h>
+#include <mach/hardware.h>
+#include <mach/at91_ssc.h>
 
 #include "at91-pcm.h"
 
diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h
index 58d0f00a07b..e5aada2cb10 100644
--- a/sound/soc/at91/at91-pcm.h
+++ b/sound/soc/at91/at91-pcm.h
@@ -19,7 +19,7 @@
 #ifndef _AT91_PCM_H
 #define _AT91_PCM_H
 
-#include <asm/arch/hardware.h>
+#include <mach/hardware.h>
 
 struct at91_ssc_periph {
 	void __iomem	*base;
diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c
index 090e607f869..5d44515e62e 100644
--- a/sound/soc/at91/at91-ssc.c
+++ b/sound/soc/at91/at91-ssc.c
@@ -28,9 +28,9 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/arch/hardware.h>
-#include <asm/arch/at91_pmc.h>
-#include <asm/arch/at91_ssc.h>
+#include <mach/hardware.h>
+#include <mach/at91_pmc.h>
+#include <mach/at91_ssc.h>
 
 #include "at91-pcm.h"
 #include "at91-ssc.h"
diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
index d532de95424..b81d6b2cfa1 100644
--- a/sound/soc/at91/eti_b1_wm8731.c
+++ b/sound/soc/at91/eti_b1_wm8731.c
@@ -22,7 +22,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/clk.h>
 #include <linux/timer.h>
@@ -33,8 +32,8 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
-#include <asm/hardware.h>
-#include <asm/arch/gpio.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
 
 #include "../codecs/wm8731.h"
 #include "at91-pcm.h"
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index b26003c4f3e..7da9f467b7b 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -562,10 +562,9 @@ static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	client_template.addr = addr;
 
 	i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
-	if (i2c == NULL) {
-		kfree(codec);
+	if (i2c == NULL)
 		return -ENOMEM;
-	}
+
 	i2c_set_clientdata(i2c, codec);
 	codec->control_data = i2c;
 
@@ -583,7 +582,6 @@ static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	return ret;
 
 err:
-	kfree(codec);
 	kfree(i2c);
 	return ret;
 }
@@ -660,6 +658,11 @@ static int ak4535_probe(struct platform_device *pdev)
 #else
 	/* Add other interfaces here */
 #endif
+
+	if (ret != 0) {
+		kfree(codec->private_data);
+		kfree(codec);
+	}
 	return ret;
 }
 
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index b1dce5f459d..5f9abb19943 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1199,10 +1199,9 @@ static int aic3x_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	client_template.addr = addr;
 
 	i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
-	if (i2c == NULL) {
-		kfree(codec);
+	if (i2c == NULL)
 		return -ENOMEM;
-	}
+
 	i2c_set_clientdata(i2c, codec);
 	codec->control_data = i2c;
 
@@ -1221,7 +1220,6 @@ static int aic3x_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	return ret;
 
 err:
-	kfree(codec);
 	kfree(i2c);
 	return ret;
 }
@@ -1302,6 +1300,11 @@ static int aic3x_probe(struct platform_device *pdev)
 #else
 	/* Add other interfaces here */
 #endif
+
+	if (ret != 0) {
+		kfree(codec->private_data);
+		kfree(codec);
+	}
 	return ret;
 }
 
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index a52d6d9e007..807318fbdc8 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -729,10 +729,9 @@ static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	client_template.addr = addr;
 
 	i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
-	if (i2c == NULL) {
-		kfree(codec);
+	if (i2c == NULL)
 		return -ENOMEM;
-	}
+
 	i2c_set_clientdata(i2c, codec);
 	codec->control_data = i2c;
 
@@ -750,7 +749,6 @@ static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	return ret;
 
 err:
-	kfree(codec);
 	kfree(i2c);
 	return ret;
 }
@@ -817,6 +815,9 @@ static int uda1380_probe(struct platform_device *pdev)
 #else
 	/* Add other interfaces here */
 #endif
+
+	if (ret != 0)
+		kfree(codec);
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 67325fd9544..3d998e6a997 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -693,10 +693,9 @@ static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	client_template.addr = addr;
 
 	i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
-	if (i2c == NULL) {
-		kfree(codec);
+	if (i2c == NULL)
 		return -ENOMEM;
-	}
+
 	i2c_set_clientdata(i2c, codec);
 	codec->control_data = i2c;
 
@@ -714,7 +713,6 @@ static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	return ret;
 
 err:
-	kfree(codec);
 	kfree(i2c);
 	return ret;
 }
@@ -782,6 +780,9 @@ static int wm8510_probe(struct platform_device *pdev)
 #else
 	/* Add other interfaces here */
 #endif
+
+	if (ret != 0)
+		kfree(codec);
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 369d39c3f74..9402fcaf04f 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -596,10 +596,9 @@ static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	client_template.addr = addr;
 
 	i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
-	if (i2c == NULL) {
-		kfree(codec);
+	if (i2c == NULL)
 		return -ENOMEM;
-	}
+
 	i2c_set_clientdata(i2c, codec);
 	codec->control_data = i2c;
 
@@ -617,7 +616,6 @@ static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	return ret;
 
 err:
-	kfree(codec);
 	kfree(i2c);
 	return ret;
 }
@@ -693,6 +691,11 @@ static int wm8731_probe(struct platform_device *pdev)
 #else
 	/* Add other interfaces here */
 #endif
+
+	if (ret != 0) {
+		kfree(codec->private_data);
+		kfree(codec);
+	}
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index e23cb09f0d1..dd1f55404b2 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -348,8 +348,9 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("ROUT1"),
 	SND_SOC_DAPM_OUTPUT("LOUT2"),
 	SND_SOC_DAPM_OUTPUT("ROUT2"),
-	SND_SOC_DAPM_OUTPUT("MONO"),
+	SND_SOC_DAPM_OUTPUT("MONO1"),
 	SND_SOC_DAPM_OUTPUT("OUT3"),
+	SND_SOC_DAPM_OUTPUT("VREF"),
 
 	SND_SOC_DAPM_INPUT("LINPUT1"),
 	SND_SOC_DAPM_INPUT("LINPUT2"),
@@ -868,10 +869,9 @@ static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	client_template.addr = addr;
 
 	i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
-	if (i2c == NULL) {
-		kfree(codec);
+	if (i2c == NULL)
 		return -ENOMEM;
-	}
+
 	i2c_set_clientdata(i2c, codec);
 	codec->control_data = i2c;
 
@@ -889,7 +889,6 @@ static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	return ret;
 
 err:
-	kfree(codec);
 	kfree(i2c);
 	return ret;
 }
@@ -965,6 +964,10 @@ static int wm8750_probe(struct platform_device *pdev)
 		/* Add other interfaces here */
 #endif
 
+	if (ret != 0) {
+		kfree(codec->private_data);
+		kfree(codec);
+	}
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 8604809f0c3..5761164fe16 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -34,7 +34,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
@@ -1661,10 +1660,9 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	client_template.addr = addr;
 
 	i2c =  kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
-	if (!i2c) {
-		kfree(codec);
+	if (!i2c)
 		return -ENOMEM;
-	}
+
 	i2c_set_clientdata(i2c, codec);
 	codec->control_data = i2c;
 
@@ -1683,7 +1681,6 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	return ret;
 
 err:
-	kfree(codec);
 	kfree(i2c);
 	return ret;
 }
@@ -1760,6 +1757,11 @@ static int wm8753_probe(struct platform_device *pdev)
 #else
 		/* Add other interfaces here */
 #endif
+
+	if (ret != 0) {
+		kfree(codec->private_data);
+		kfree(codec);
+	}
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 3ecce5168e9..dd995ef448b 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -82,7 +82,7 @@ static const u16 wm8990_reg[] = {
 	0x0003,     /* R35 - ClassD1 */
 	0x0000,     /* R36 */
 	0x0100,     /* R37 - ClassD3 */
-	0x0000,     /* R38 */
+	0x0079,     /* R38 - ClassD4 */
 	0x0000,     /* R39 - Input Mixer1 */
 	0x0000,     /* R40 - Input Mixer2 */
 	0x0000,     /* R41 - Input Mixer3 */
@@ -311,11 +311,15 @@ SOC_SINGLE("Speaker Mode Switch", WM8990_CLASSD1,
 	WM8990_CDMODE_BIT, 1, 0),
 
 SOC_SINGLE("Speaker Output Attenuation Volume", WM8990_SPEAKER_VOLUME,
-	WM8990_SPKVOL_SHIFT, WM8990_SPKVOL_MASK, 0),
+	WM8990_SPKATTN_SHIFT, WM8990_SPKATTN_MASK, 0),
 SOC_SINGLE("Speaker DC Boost Volume", WM8990_CLASSD3,
 	WM8990_DCGAIN_SHIFT, WM8990_DCGAIN_MASK, 0),
 SOC_SINGLE("Speaker AC Boost Volume", WM8990_CLASSD3,
 	WM8990_ACGAIN_SHIFT, WM8990_ACGAIN_MASK, 0),
+SOC_SINGLE_TLV("Speaker Volume", WM8990_CLASSD4,
+	WM8990_SPKVOL_SHIFT, WM8990_SPKVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("Speaker ZC Switch", WM8990_CLASSD4,
+	WM8990_SPKZC_SHIFT, WM8990_SPKZC_MASK, 0),
 
 SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume",
 	WM8990_LEFT_DAC_DIGITAL_VOLUME,
@@ -920,7 +924,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"},
 	{"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"},
 	{"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"},
-	{"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"},
+	{"SPKMIX", "SPKMIX Left DAC Switch", "Left DAC"},
 
 	/* LONMIX */
 	{"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"},
@@ -1496,10 +1500,9 @@ static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	client_template.addr = addr;
 
 	i2c =  kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
-	if (i2c == NULL) {
-		kfree(codec);
+	if (i2c == NULL)
 		return -ENOMEM;
-	}
+
 	i2c_set_clientdata(i2c, codec);
 	codec->control_data = i2c;
 
@@ -1517,7 +1520,6 @@ static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 	return ret;
 
 err:
-	kfree(codec);
 	kfree(i2c);
 	return ret;
 }
@@ -1591,6 +1593,11 @@ static int wm8990_probe(struct platform_device *pdev)
 #else
 		/* Add other interfaces here */
 #endif
+
+	if (ret != 0) {
+		kfree(codec->private_data);
+		kfree(codec);
+	}
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h
index 6bea5748528..0a08325d544 100644
--- a/sound/soc/codecs/wm8990.h
+++ b/sound/soc/codecs/wm8990.h
@@ -54,6 +54,7 @@
 #define WM8990_SPEAKER_VOLUME                   0x22
 #define WM8990_CLASSD1                          0x23
 #define WM8990_CLASSD3                          0x25
+#define WM8990_CLASSD4                          0x26
 #define WM8990_INPUT_MIXER1                     0x27
 #define WM8990_INPUT_MIXER2                     0x28
 #define WM8990_INPUT_MIXER3                     0x29
@@ -528,8 +529,8 @@
 /*
  * R34 (0x22) - Speaker Volume
  */
-#define WM8990_SPKVOL_MASK                      0x0003  /* SPKVOL - [1:0] */
-#define WM8990_SPKVOL_SHIFT			0
+#define WM8990_SPKATTN_MASK                      0x0003  /* SPKATTN - [1:0] */
+#define WM8990_SPKATTN_SHIFT			 0
 
 /*
  * R35 (0x23) - ClassD1
@@ -544,6 +545,15 @@
 #define WM8990_DCGAIN_SHIFT			3
 #define WM8990_ACGAIN_MASK                      0x0007  /* ACGAIN - [2:0] */
 #define WM8990_ACGAIN_SHIFT			0
+
+/*
+ * R38 (0x26) - ClassD4
+ */
+#define WM8990_SPKZC_MASK                       0x0001  /* SPKZC */
+#define WM8990_SPKZC_SHIFT                           7  /* SPKZC */
+#define WM8990_SPKVOL_MASK                      0x007F  /* SPKVOL - [6:0] */
+#define WM8990_SPKVOL_SHIFT                          0  /* SPKVOL - [6:0] */
+
 /*
  * R39 (0x27) - Input Mixer1
  */
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 1fb7f9a7aec..2f1c91b1d55 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -13,7 +13,6 @@
 
 #include <linux/init.h>
 #include <linux/module.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
 #include <sound/core.h>
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 5e2c306399e..65fdbd81a37 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -19,9 +19,8 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
-#include <asm/mach-types.h>
 #include <asm/dma.h>
-#include <asm/arch/hardware.h>
+#include <mach/hardware.h>
 
 #include "../codecs/tlv320aic3x.h"
 #include "davinci-pcm.h"
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index da2bc590286..d2d3da9729f 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -132,12 +132,17 @@ struct fsl_dma_private {
  * Since each link descriptor has a 32-bit byte count field, we set
  * period_bytes_max to the largest 32-bit number.  We also have no maximum
  * number of periods.
+ *
+ * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a
+ * limitation in the SSI driver requires the sample rates for playback and
+ * capture to be the same.
  */
 static const struct snd_pcm_hardware fsl_dma_hardware = {
 
 	.info   		= SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_MMAP |
-				  SNDRV_PCM_INFO_MMAP_VALID,
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_JOINT_DUPLEX,
 	.formats		= FSLDMA_PCM_FORMATS,
 	.rates  		= FSLDMA_PCM_RATES,
 	.rate_min       	= 5512,
@@ -322,14 +327,75 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
  * fsl_dma_open: open a new substream.
  *
  * Each substream has its own DMA buffer.
+ *
+ * ALSA divides the DMA buffer into N periods.  We create NUM_DMA_LINKS link
+ * descriptors that ping-pong from one period to the next.  For example, if
+ * there are six periods and two link descriptors, this is how they look
+ * before playback starts:
+ *
+ *      	   The last link descriptor
+ *   ____________  points back to the first
+ *  |   	 |
+ *  V   	 |
+ *  ___    ___   |
+ * |   |->|   |->|
+ * |___|  |___|
+ *   |      |
+ *   |      |
+ *   V      V
+ *  _________________________________________
+ * |      |      |      |      |      |      |  The DMA buffer is
+ * |      |      |      |      |      |      |    divided into 6 parts
+ * |______|______|______|______|______|______|
+ *
+ * and here's how they look after the first period is finished playing:
+ *
+ *   ____________
+ *  |   	 |
+ *  V   	 |
+ *  ___    ___   |
+ * |   |->|   |->|
+ * |___|  |___|
+ *   |      |
+ *   |______________
+ *          |       |
+ *          V       V
+ *  _________________________________________
+ * |      |      |      |      |      |      |
+ * |      |      |      |      |      |      |
+ * |______|______|______|______|______|______|
+ *
+ * The first link descriptor now points to the third period.  The DMA
+ * controller is currently playing the second period.  When it finishes, it
+ * will jump back to the first descriptor and play the third period.
+ *
+ * There are four reasons we do this:
+ *
+ * 1. The only way to get the DMA controller to automatically restart the
+ *    transfer when it gets to the end of the buffer is to use chaining
+ *    mode.  Basic direct mode doesn't offer that feature.
+ * 2. We need to receive an interrupt at the end of every period.  The DMA
+ *    controller can generate an interrupt at the end of every link transfer
+ *    (aka segment).  Making each period into a DMA segment will give us the
+ *    interrupts we need.
+ * 3. By creating only two link descriptors, regardless of the number of
+ *    periods, we do not need to reallocate the link descriptors if the
+ *    number of periods changes.
+ * 4. All of the audio data is still stored in a single, contiguous DMA
+ *    buffer, which is what ALSA expects.  We're just dividing it into
+ *    contiguous parts, and creating a link descriptor for each one.
  */
 static int fsl_dma_open(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct fsl_dma_private *dma_private;
+	struct ccsr_dma_channel __iomem *dma_channel;
 	dma_addr_t ld_buf_phys;
+	u64 temp_link;  	/* Pointer to next link descriptor */
+	u32 mr;
 	unsigned int channel;
 	int ret = 0;
+	unsigned int i;
 
 	/*
 	 * Reject any DMA buffer whose size is not a multiple of the period
@@ -390,68 +456,74 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
 	snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware);
 	runtime->private_data = dma_private;
 
+	/* Program the fixed DMA controller parameters */
+
+	dma_channel = dma_private->dma_channel;
+
+	temp_link = dma_private->ld_buf_phys +
+		sizeof(struct fsl_dma_link_descriptor);
+
+	for (i = 0; i < NUM_DMA_LINKS; i++) {
+		struct fsl_dma_link_descriptor *link = &dma_private->link[i];
+
+		link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+		link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+		link->next = cpu_to_be64(temp_link);
+
+		temp_link += sizeof(struct fsl_dma_link_descriptor);
+	}
+	/* The last link descriptor points to the first */
+	dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys);
+
+	/* Tell the DMA controller where the first link descriptor is */
+	out_be32(&dma_channel->clndar,
+		CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys));
+	out_be32(&dma_channel->eclndar,
+		CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys));
+
+	/* The manual says the BCR must be clear before enabling EMP */
+	out_be32(&dma_channel->bcr, 0);
+
+	/*
+	 * Program the mode register for interrupts, external master control,
+	 * and source/destination hold.  Also clear the Channel Abort bit.
+	 */
+	mr = in_be32(&dma_channel->mr) &
+		~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE);
+
+	/*
+	 * We want External Master Start and External Master Pause enabled,
+	 * because the SSI is controlling the DMA controller.  We want the DMA
+	 * controller to be set up in advance, and then we signal only the SSI
+	 * to start transferring.
+	 *
+	 * We want End-Of-Segment Interrupts enabled, because this will generate
+	 * an interrupt at the end of each segment (each link descriptor
+	 * represents one segment).  Each DMA segment is the same thing as an
+	 * ALSA period, so this is how we get an interrupt at the end of every
+	 * period.
+	 *
+	 * We want Error Interrupt enabled, so that we can get an error if
+	 * the DMA controller is mis-programmed somehow.
+	 */
+	mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN |
+		CCSR_DMA_MR_EMS_EN;
+
+	/* For playback, we want the destination address to be held.  For
+	   capture, set the source address to be held. */
+	mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+		CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE;
+
+	out_be32(&dma_channel->mr, mr);
+
 	return 0;
 }
 
 /**
- * fsl_dma_hw_params: allocate the DMA buffer and the DMA link descriptors.
+ * fsl_dma_hw_params: continue initializing the DMA links
  *
- * ALSA divides the DMA buffer into N periods.  We create NUM_DMA_LINKS link
- * descriptors that ping-pong from one period to the next.  For example, if
- * there are six periods and two link descriptors, this is how they look
- * before playback starts:
- *
- *      	   The last link descriptor
- *   ____________  points back to the first
- *  |   	 |
- *  V   	 |
- *  ___    ___   |
- * |   |->|   |->|
- * |___|  |___|
- *   |      |
- *   |      |
- *   V      V
- *  _________________________________________
- * |      |      |      |      |      |      |  The DMA buffer is
- * |      |      |      |      |      |      |    divided into 6 parts
- * |______|______|______|______|______|______|
- *
- * and here's how they look after the first period is finished playing:
- *
- *   ____________
- *  |   	 |
- *  V   	 |
- *  ___    ___   |
- * |   |->|   |->|
- * |___|  |___|
- *   |      |
- *   |______________
- *          |       |
- *          V       V
- *  _________________________________________
- * |      |      |      |      |      |      |
- * |      |      |      |      |      |      |
- * |______|______|______|______|______|______|
- *
- * The first link descriptor now points to the third period.  The DMA
- * controller is currently playing the second period.  When it finishes, it
- * will jump back to the first descriptor and play the third period.
- *
- * There are four reasons we do this:
- *
- * 1. The only way to get the DMA controller to automatically restart the
- *    transfer when it gets to the end of the buffer is to use chaining
- *    mode.  Basic direct mode doesn't offer that feature.
- * 2. We need to receive an interrupt at the end of every period.  The DMA
- *    controller can generate an interrupt at the end of every link transfer
- *    (aka segment).  Making each period into a DMA segment will give us the
- *    interrupts we need.
- * 3. By creating only two link descriptors, regardless of the number of
- *    periods, we do not need to reallocate the link descriptors if the
- *    number of periods changes.
- * 4. All of the audio data is still stored in a single, contiguous DMA
- *    buffer, which is what ALSA expects.  We're just dividing it into
- *    contiguous parts, and creating a link descriptor for each one.
+ * This function obtains hardware parameters about the opened stream and
+ * programs the DMA controller accordingly.
  *
  * Note that due to a quirk of the SSI's STX register, the target address
  * for the DMA operations depends on the sample size.  So we don't program
@@ -463,11 +535,8 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct fsl_dma_private *dma_private = runtime->private_data;
-	struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
 
 	dma_addr_t temp_addr;   /* Pointer to next period */
-	u64 temp_link;  	/* Pointer to next link descriptor */
-	u32 mr; 		/* Temporary variable for MR register */
 
 	unsigned int i;
 
@@ -485,8 +554,6 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
 		dma_private->dma_buf_next = dma_private->dma_buf_phys;
 
 	/*
-	 * Initialize each link descriptor.
-	 *
 	 * The actual address in STX0 (destination for playback, source for
 	 * capture) is based on the sample size, but we don't know the sample
 	 * size in this function, so we'll have to adjust that later.  See
@@ -502,16 +569,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
 	 * buffer itself.
 	 */
 	temp_addr = substream->dma_buffer.addr;
-	temp_link = dma_private->ld_buf_phys +
-		sizeof(struct fsl_dma_link_descriptor);
 
 	for (i = 0; i < NUM_DMA_LINKS; i++) {
 		struct fsl_dma_link_descriptor *link = &dma_private->link[i];
 
 		link->count = cpu_to_be32(period_size);
-		link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
-		link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
-		link->next = cpu_to_be64(temp_link);
 
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			link->source_addr = cpu_to_be32(temp_addr);
@@ -519,51 +581,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
 			link->dest_addr = cpu_to_be32(temp_addr);
 
 		temp_addr += period_size;
-		temp_link += sizeof(struct fsl_dma_link_descriptor);
 	}
-	/* The last link descriptor points to the first */
-	dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys);
-
-	/* Tell the DMA controller where the first link descriptor is */
-	out_be32(&dma_channel->clndar,
-		CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys));
-	out_be32(&dma_channel->eclndar,
-		CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys));
-
-	/* The manual says the BCR must be clear before enabling EMP */
-	out_be32(&dma_channel->bcr, 0);
-
-	/*
-	 * Program the mode register for interrupts, external master control,
-	 * and source/destination hold.  Also clear the Channel Abort bit.
-	 */
-	mr = in_be32(&dma_channel->mr) &
-		~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE);
-
-	/*
-	 * We want External Master Start and External Master Pause enabled,
-	 * because the SSI is controlling the DMA controller.  We want the DMA
-	 * controller to be set up in advance, and then we signal only the SSI
-	 * to start transfering.
-	 *
-	 * We want End-Of-Segment Interrupts enabled, because this will generate
-	 * an interrupt at the end of each segment (each link descriptor
-	 * represents one segment).  Each DMA segment is the same thing as an
-	 * ALSA period, so this is how we get an interrupt at the end of every
-	 * period.
-	 *
-	 * We want Error Interrupt enabled, so that we can get an error if
-	 * the DMA controller is mis-programmed somehow.
-	 */
-	mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN |
-		CCSR_DMA_MR_EMS_EN;
-
-	/* For playback, we want the destination address to be held.  For
-	   capture, set the source address to be held. */
-	mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-		CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE;
-
-	out_be32(&dma_channel->mr, mr);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 71bff33f552..157a7895ffa 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -67,6 +67,8 @@
  * @ssi: pointer to the SSI's registers
  * @ssi_phys: physical address of the SSI registers
  * @irq: IRQ of this SSI
+ * @first_stream: pointer to the stream that was opened first
+ * @second_stream: pointer to second stream
  * @dev: struct device pointer
  * @playback: the number of playback streams opened
  * @capture: the number of capture streams opened
@@ -79,6 +81,8 @@ struct fsl_ssi_private {
 	struct ccsr_ssi __iomem *ssi;
 	dma_addr_t ssi_phys;
 	unsigned int irq;
+	struct snd_pcm_substream *first_stream;
+	struct snd_pcm_substream *second_stream;
 	struct device *dev;
 	unsigned int playback;
 	unsigned int capture;
@@ -342,6 +346,49 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream)
 		 */
 	}
 
+	if (!ssi_private->first_stream)
+		ssi_private->first_stream = substream;
+	else {
+		/* This is the second stream open, so we need to impose sample
+		 * rate and maybe sample size constraints.  Note that this can
+		 * cause a race condition if the second stream is opened before
+		 * the first stream is fully initialized.
+		 *
+		 * We provide some protection by checking to make sure the first
+		 * stream is initialized, but it's not perfect.  ALSA sometimes
+		 * re-initializes the driver with a different sample rate or
+		 * size.  If the second stream is opened before the first stream
+		 * has received its final parameters, then the second stream may
+		 * be constrained to the wrong sample rate or size.
+		 *
+		 * FIXME: This code does not handle opening and closing streams
+		 * repeatedly.  If you open two streams and then close the first
+		 * one, you may not be able to open another stream until you
+		 * close the second one as well.
+		 */
+		struct snd_pcm_runtime *first_runtime =
+			ssi_private->first_stream->runtime;
+
+		if (!first_runtime->rate || !first_runtime->sample_bits) {
+			dev_err(substream->pcm->card->dev,
+				"set sample rate and size in %s stream first\n",
+				substream->stream == SNDRV_PCM_STREAM_PLAYBACK
+				? "capture" : "playback");
+			return -EAGAIN;
+		}
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE,
+			first_runtime->rate, first_runtime->rate);
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+			SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+			first_runtime->sample_bits,
+			first_runtime->sample_bits);
+
+		ssi_private->second_stream = substream;
+	}
+
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		ssi_private->playback++;
 
@@ -371,18 +418,16 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
 
 	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-	u32 wl;
 
-	wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
+	if (substream == ssi_private->first_stream) {
+		u32 wl;
 
-	clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+		/* The SSI should always be disabled at this points (SSIEN=0) */
+		wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		/* In synchronous mode, the SSI uses STCCR for capture */
 		clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
-	else
-		clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
-
-	setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+	}
 
 	return 0;
 }
@@ -407,9 +452,13 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			setbits32(&ssi->scr, CCSR_SSI_SCR_TE);
+			clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+			setbits32(&ssi->scr,
+				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
 		} else {
-			setbits32(&ssi->scr, CCSR_SSI_SCR_RE);
+			clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+			setbits32(&ssi->scr,
+				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
 
 			/*
 			 * I think we need this delay to allow time for the SSI
@@ -452,6 +501,11 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
 		ssi_private->capture--;
 
+	if (ssi_private->first_stream == substream)
+		ssi_private->first_stream = ssi_private->second_stream;
+
+	ssi_private->second_stream = NULL;
+
 	/*
 	 * If this is the last active substream, disable the SSI and release
 	 * the IRQ.
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 02cec96859b..87d0ed01f65 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -29,9 +29,9 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/arch/hardware.h>
+#include <mach/hardware.h>
 #include <linux/gpio.h>
-#include <asm/arch/mcbsp.h>
+#include <mach/mcbsp.h>
 
 #include "omap-mcbsp.h"
 #include "omap-pcm.h"
@@ -329,12 +329,14 @@ static int __init n810_soc_init(void)
 	sys_clkout2_src = clk_get(dev, "sys_clkout2_src");
 	if (IS_ERR(sys_clkout2_src)) {
 		dev_err(dev, "Could not get sys_clkout2_src clock\n");
-		return -ENODEV;
+		err = PTR_ERR(sys_clkout2_src);
+		goto err2;
 	}
 	sys_clkout2 = clk_get(dev, "sys_clkout2");
 	if (IS_ERR(sys_clkout2)) {
 		dev_err(dev, "Could not get sys_clkout2\n");
-		goto err1;
+		err = PTR_ERR(sys_clkout2);
+		goto err3;
 	}
 	/*
 	 * Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use
@@ -343,7 +345,8 @@ static int __init n810_soc_init(void)
 	func96m_clk = clk_get(dev, "func_96m_ck");
 	if (IS_ERR(func96m_clk)) {
 		dev_err(dev, "Could not get func 96M clock\n");
-		goto err2;
+		err = PTR_ERR(func96m_clk);
+		goto err4;
 	}
 	clk_set_parent(sys_clkout2_src, func96m_clk);
 	clk_set_rate(sys_clkout2, 12000000);
@@ -356,20 +359,25 @@ static int __init n810_soc_init(void)
 	gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
 
 	return 0;
-err2:
+err4:
 	clk_put(sys_clkout2);
+err3:
+	clk_put(sys_clkout2_src);
+err2:
 	platform_device_del(n810_snd_device);
 err1:
 	platform_device_put(n810_snd_device);
 
 	return err;
-
 }
 
 static void __exit n810_soc_exit(void)
 {
 	gpio_free(N810_SPEAKER_AMP_GPIO);
 	gpio_free(N810_HEADSET_AMP_GPIO);
+	clk_put(sys_clkout2_src);
+	clk_put(sys_clkout2);
+	clk_put(func96m_clk);
 
 	platform_device_unregister(n810_snd_device);
 }
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 00b0c9d73cd..35310e16d7f 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -30,9 +30,9 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/arch/control.h>
-#include <asm/arch/dma.h>
-#include <asm/arch/mcbsp.h>
+#include <mach/control.h>
+#include <mach/dma.h>
+#include <mach/mcbsp.h>
 #include "omap-mcbsp.h"
 #include "omap-pcm.h"
 
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index e092f3d836d..690bfeaec4a 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -27,7 +27,7 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <asm/arch/dma.h>
+#include <mach/dma.h>
 #include "omap-pcm.h"
 
 static const struct snd_pcm_hardware omap_pcm_hardware = {
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index c0294464a23..0a53f72077f 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -25,10 +25,10 @@
 
 #include <asm/mach-types.h>
 #include <asm/hardware/scoop.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/corgi.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/corgi.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm8731.h"
 #include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 06e8afb2527..6781c5be242 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -21,9 +21,9 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
 #include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index 02dcac39cdf..d9c3f7b28be 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -30,9 +30,9 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
 #include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 65a4e9a8c39..a4697f7e292 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -26,10 +26,10 @@
 
 #include <asm/mach-types.h>
 #include <asm/hardware/locomo.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/poodle.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/poodle.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm8731.h"
 #include "pxa2xx-pcm.h"
@@ -85,17 +85,13 @@ static int poodle_startup(struct snd_pcm_substream *substream)
 }
 
 /* we need to unmute the HP at shutdown as the mute burns power on poodle */
-static int poodle_shutdown(struct snd_pcm_substream *substream)
+static void poodle_shutdown(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
-
 	/* set = unmute headphone */
 	locomo_gpio_write(&poodle_locomo_device.dev,
 		POODLE_LOCOMO_GPIO_MUTE_L, 1);
 	locomo_gpio_write(&poodle_locomo_device.dev,
 		POODLE_LOCOMO_GPIO_MUTE_R, 1);
-	return 0;
 }
 
 static int poodle_hw_params(struct snd_pcm_substream *substream,
@@ -232,7 +228,7 @@ static const struct soc_enum poodle_enum[] = {
 	SOC_ENUM_SINGLE_EXT(2, spk_function),
 };
 
-static const snd_kcontrol_new_t wm8731_poodle_controls[] = {
+static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
 	SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack,
 		poodle_set_jack),
 	SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk,
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 059af815ea0..d94a495bd6b 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -26,10 +26,10 @@
 
 #include <asm/irq.h>
 #include <linux/mutex.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/pxa2xx-gpio.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 8f96d87f7b4..c796b188277 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -16,15 +16,16 @@
 #include <linux/device.h>
 #include <linux/delay.h>
 #include <linux/clk.h>
+#include <linux/platform_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/pxa2xx-gpio.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
@@ -81,7 +82,6 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 
-	clk_i2s = clk_get(NULL, "I2SCLK");
 	if (IS_ERR(clk_i2s))
 		return PTR_ERR(clk_i2s);
 
@@ -152,6 +152,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx);
 	pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm);
 	pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk);
+	BUG_ON(IS_ERR(clk_i2s));
 	clk_enable(clk_i2s);
 	pxa_i2s_wait();
 
@@ -317,6 +318,43 @@ struct snd_soc_dai pxa_i2s_dai = {
 
 EXPORT_SYMBOL_GPL(pxa_i2s_dai);
 
+static int pxa2xx_i2s_probe(struct platform_device *dev)
+{
+	clk_i2s = clk_get(&dev->dev, "I2SCLK");
+	return IS_ERR(clk_i2s) ? PTR_ERR(clk_i2s) : 0;
+}
+
+static int __devexit pxa2xx_i2s_remove(struct platform_device *dev)
+{
+	clk_put(clk_i2s);
+	clk_i2s = ERR_PTR(-ENOENT);
+	return 0;
+}
+
+static struct platform_driver pxa2xx_i2s_driver = {
+	.probe = pxa2xx_i2s_probe,
+	.remove = __devexit_p(pxa2xx_i2s_remove),
+
+	.driver = {
+		.name = "pxa2xx-i2s",
+		.owner = THIS_MODULE,
+	},
+};
+
+static int __init pxa2xx_i2s_init(void)
+{
+	clk_i2s = ERR_PTR(-ENOENT);
+	return platform_driver_register(&pxa2xx_i2s_driver);
+}
+
+static void __exit pxa2xx_i2s_exit(void)
+{
+	platform_driver_unregister(&pxa2xx_i2s_driver);
+}
+
+module_init(pxa2xx_i2s_init);
+module_exit(pxa2xx_i2s_exit);
+
 /* Module information */
 MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
 MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 2df03ee5819..4345f387fe4 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -22,9 +22,9 @@
 #include <sound/soc.h>
 
 #include <asm/dma.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 64385797da5..37cb768fc93 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -26,10 +26,10 @@
 
 #include <asm/mach-types.h>
 #include <asm/hardware/scoop.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/akita.h>
-#include <asm/arch/spitz.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/akita.h>
+#include <mach/spitz.h>
 #include "../codecs/wm8750.h"
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
@@ -297,7 +297,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_disable_pin(codec, "LINPUT3");
 	snd_soc_dapm_disable_pin(codec, "RINPUT3");
 	snd_soc_dapm_disable_pin(codec, "OUT3");
-	snd_soc_dapm_disable_pin(codec, "MONO");
+	snd_soc_dapm_disable_pin(codec, "MONO1");
 
 	/* Add spitz specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index fe6cca9c9e7..2baaa750f12 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -29,11 +29,10 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/arch/tosa.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/audio.h>
-#include <asm/arch/tosa.h>
+#include <mach/tosa.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
 #include "pxa2xx-pcm.h"
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 4d7a9aa15f1..8089f8ee05c 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -24,14 +24,13 @@
 #include <sound/soc-dapm.h>
 #include <sound/tlv.h>
 
-#include <asm/mach-types.h>
 #include <asm/hardware/scoop.h>
-#include <asm/arch/regs-clock.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/hardware.h>
-#include <asm/arch/audio.h>
+#include <mach/regs-clock.h>
+#include <mach/regs-gpio.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
 #include <linux/io.h>
-#include <asm/arch/spi-gpio.h>
+#include <mach/spi-gpio.h>
 
 #include <asm/plat-s3c24xx/regs-iis.h>
 
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index ee4676ed128..ded7d995a92 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -28,16 +28,16 @@
 #include <sound/pcm_params.h>
 #include <sound/initval.h>
 #include <sound/soc.h>
-#include <asm/hardware.h>
+#include <mach/hardware.h>
 
 #include <linux/io.h>
 #include <asm/dma.h>
 
 #include <asm/plat-s3c24xx/regs-s3c2412-iis.h>
 
-#include <asm/arch/regs-gpio.h>
-#include <asm/arch/audio.h>
-#include <asm/arch/dma.h>
+#include <mach/regs-gpio.h>
+#include <mach/audio.h>
+#include <mach/dma.h>
 
 #include "s3c24xx-pcm.h"
 #include "s3c2412-i2s.h"
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 783349b7fed..19c5c3cf5d8 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -27,13 +27,13 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/hardware.h>
+#include <mach/hardware.h>
 #include <asm/plat-s3c/regs-ac97.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/arch/regs-clock.h>
-#include <asm/arch/audio.h>
+#include <mach/regs-gpio.h>
+#include <mach/regs-clock.h>
+#include <mach/audio.h>
 #include <asm/dma.h>
-#include <asm/arch/dma.h>
+#include <mach/dma.h>
 
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-ac97.h"
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 397524282b5..ba4476b55fb 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -27,12 +27,12 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/hardware.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/arch/regs-clock.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/regs-gpio.h>
+#include <mach/regs-clock.h>
+#include <mach/audio.h>
 #include <asm/dma.h>
-#include <asm/arch/dma.h>
+#include <mach/dma.h>
 
 #include <asm/plat-s3c24xx/regs-iis.h>
 
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index cef79b34dc6..e13e614bada 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -27,9 +27,9 @@
 #include <sound/soc.h>
 
 #include <asm/dma.h>
-#include <asm/hardware.h>
-#include <asm/arch/dma.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/dma.h>
+#include <mach/audio.h>
 
 #include "s3c24xx-pcm.h"
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 820347c9ae4..f9d100bc847 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -470,6 +470,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w,
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(dapm_reg_event);
 
 /*
  * Scan each dapm widget for complete audio path.