Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  [ALSA] soc - wm9712 - checkpatch fixes
  [ALSA] pcsp - Fix more dependency
  [ALSA] hda - Add support of Medion RIM 2150
  [ALSA] ASoC: Add drivers for the Texas Instruments OMAP processors
  [ALSA] ice1724 - Enable watermarks
  [ALSA] Add MPU401_INFO_NO_ACK bitflag

14 files changed, 1342 insertions, 37 deletions

diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig
index fe85af1c5693..a78a8d045175 100644
--- a/sound/drivers/Kconfig
+++ b/sound/drivers/Kconfig
@@ -8,6 +8,8 @@ config SND_PCSP
 	tristate "Internal PC speaker support"
 	depends on X86_PC && HIGH_RES_TIMERS
 	depends on INPUT
+	depends on SND
+	select SND_PCM
 	help
 	  If you don't have a sound card in your computer, you can include a
 	  driver for the PC speaker which allows it to act like a primitive
diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
index 18cca2457d44..2af09996a3d0 100644
--- a/sound/drivers/mpu401/mpu401_uart.c
+++ b/sound/drivers/mpu401/mpu401_uart.c
@@ -243,7 +243,7 @@ static int snd_mpu401_uart_cmd(struct snd_mpu401 * mpu, unsigned char cmd,
 #endif
 	}
 	mpu->write(mpu, cmd, MPU401C(mpu));
-	if (ack) {
+	if (ack && !(mpu->info_flags & MPU401_INFO_NO_ACK)) {
 		ok = 0;
 		timeout = 10000;
 		while (!ok && timeout-- > 0) {
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index cdda64b02f46..d9783a4263e0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -60,6 +60,7 @@ enum {
 	ALC880_TCL_S700,
 	ALC880_LG,
 	ALC880_LG_LW,
+	ALC880_MEDION_RIM,
 #ifdef CONFIG_SND_DEBUG
 	ALC880_TEST,
 #endif
@@ -2275,6 +2276,75 @@ static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res)
 		alc880_lg_lw_automute(codec);
 }
 
+static struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
+static struct hda_input_mux alc880_medion_rim_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x1 },
+	},
+};
+
+static struct hda_verb alc880_medion_rim_init_verbs[] = {
+	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* Mic1 (rear panel) pin widget for input and vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Mic2 (as headphone out) for HP output */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Internal Speaker */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+	{0x20, AC_VERB_SET_PROC_COEF,  0x3060},
+
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{ }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_medion_rim_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+	unsigned char bits;
+
+	present = snd_hda_codec_read(codec, 0x14, 0,
+				     AC_VERB_GET_PIN_SENSE, 0)
+		& AC_PINSENSE_PRESENCE;
+	bits = present ? HDA_AMP_MUTE : 0;
+	snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+	if (present)
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
+	else
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2);
+}
+
+static void alc880_medion_rim_unsol_event(struct hda_codec *codec,
+					  unsigned int res)
+{
+	/* Looks like the unsol event is incompatible with the standard
+	 * definition.  4bit tag is placed at 28 bit!
+	 */
+	if ((res >> 28) == ALC880_HP_EVENT)
+		alc880_medion_rim_automute(codec);
+}
+
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static struct hda_amp_list alc880_loopbacks[] = {
 	{ 0x0b, HDA_INPUT, 0 },
@@ -2882,6 +2952,7 @@ static const char *alc880_models[ALC880_MODEL_LAST] = {
 	[ALC880_F1734]		= "F1734",
 	[ALC880_LG]		= "lg",
 	[ALC880_LG_LW]		= "lg-lw",
+	[ALC880_MEDION_RIM]	= "medion",
 #ifdef CONFIG_SND_DEBUG
 	[ALC880_TEST]		= "test",
 #endif
@@ -2933,6 +3004,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL),
 	SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53),
 	SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810),
+	SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_MEDION_RIM),
 	SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
@@ -3227,6 +3299,20 @@ static struct alc_config_preset alc880_presets[] = {
 		.unsol_event = alc880_lg_lw_unsol_event,
 		.init_hook = alc880_lg_lw_automute,
 	},
+	[ALC880_MEDION_RIM] = {
+		.mixers = { alc880_medion_rim_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_medion_rim_init_verbs,
+				alc_gpio2_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc880_dac_nids),
+		.dac_nids = alc880_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+		.channel_mode = alc880_2_jack_modes,
+		.input_mux = &alc880_medion_rim_capture_source,
+		.unsol_event = alc880_medion_rim_unsol_event,
+		.init_hook = alc880_medion_rim_automute,
+	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC880_TEST] = {
 		.mixers = { alc880_test_mixer },
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 4490422fb930..67350901772c 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2429,6 +2429,7 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
 			if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
 						       ICEREG1724(ice, MPU_CTRL),
 						       (MPU401_INFO_INTEGRATED |
+							MPU401_INFO_NO_ACK |
 							MPU401_INFO_TX_IRQ),
 						       ice->irq, 0,
 						       &ice->rmidi[0])) < 0) {
@@ -2442,12 +2443,10 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
 			outb(inb(ICEREG1724(ice, IRQMASK)) &
 			     ~(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX),
 			     ICEREG1724(ice, IRQMASK));
-#if 0 /* for testing */
 			/* set watermarks */
 			outb(VT1724_MPU_RX_FIFO | 0x1,
 			     ICEREG1724(ice, MPU_FIFO_WM));
 			outb(0x1, ICEREG1724(ice, MPU_FIFO_WM));
-#endif
 		}
 	}
 
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index a3b51df2bea1..18f28ac4bfe8 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -30,6 +30,7 @@ source "sound/soc/s3c24xx/Kconfig"
 source "sound/soc/sh/Kconfig"
 source "sound/soc/fsl/Kconfig"
 source "sound/soc/davinci/Kconfig"
+source "sound/soc/omap/Kconfig"
 
 # Supported codecs
 source "sound/soc/codecs/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index e489dbdde458..782db2127108 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,4 @@
 snd-soc-core-objs := soc-core.o soc-dapm.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
-obj-$(CONFIG_SND_SOC)	+= codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/
+obj-$(CONFIG_SND_SOC)	+= codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index d2d79e182a45..76c1e2d33e7d 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -37,23 +37,23 @@ static int ac97_write(struct snd_soc_codec *codec,
  * WM9712 register cache
  */
 static const u16 wm9712_reg[] = {
-	0x6174, 0x8000, 0x8000, 0x8000, // 6
-	0x0f0f, 0xaaa0, 0xc008, 0x6808, // e
-	0xe808, 0xaaa0, 0xad00, 0x8000, // 16
-	0xe808, 0x3000, 0x8000, 0x0000, // 1e
-	0x0000, 0x0000, 0x0000, 0x000f, // 26
-	0x0405, 0x0410, 0xbb80, 0xbb80, // 2e
-	0x0000, 0xbb80, 0x0000, 0x0000, // 36
-	0x0000, 0x2000, 0x0000, 0x0000, // 3e
-	0x0000, 0x0000, 0x0000, 0x0000, // 46
-	0x0000, 0x0000, 0xf83e, 0xffff, // 4e
-	0x0000, 0x0000, 0x0000, 0xf83e, // 56
-	0x0008, 0x0000, 0x0000, 0x0000, // 5e
-	0xb032, 0x3e00, 0x0000, 0x0000, // 66
-	0x0000, 0x0000, 0x0000, 0x0000, // 6e
-	0x0000, 0x0000, 0x0000, 0x0006, // 76
-	0x0001, 0x0000, 0x574d, 0x4c12, // 7e
-	0x0000, 0x0000 // virtual hp mixers
+	0x6174, 0x8000, 0x8000, 0x8000, /*  6 */
+	0x0f0f, 0xaaa0, 0xc008, 0x6808, /*  e */
+	0xe808, 0xaaa0, 0xad00, 0x8000, /* 16 */
+	0xe808, 0x3000, 0x8000, 0x0000, /* 1e */
+	0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
+	0x0405, 0x0410, 0xbb80, 0xbb80, /* 2e */
+	0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
+	0x0000, 0x2000, 0x0000, 0x0000, /* 3e */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 46 */
+	0x0000, 0x0000, 0xf83e, 0xffff, /* 4e */
+	0x0000, 0x0000, 0x0000, 0xf83e, /* 56 */
+	0x0008, 0x0000, 0x0000, 0x0000, /* 5e */
+	0xb032, 0x3e00, 0x0000, 0x0000, /* 66 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
+	0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
+	0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */
+	0x0000, 0x0000 /* virtual hp mixers */
 };
 
 /* virtual HP mixers regs */
@@ -94,7 +94,7 @@ static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = {
 SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
 SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1),
 SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
-SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE,15, 1, 1),
+SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
 SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
 
 SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0),
@@ -165,7 +165,8 @@ static int wm9712_add_controls(struct snd_soc_codec *codec)
 
 	for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) {
 		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm9712_snd_ac97_controls[i],codec, NULL));
+				  snd_soc_cnew(&wm9712_snd_ac97_controls[i],
+					       codec, NULL));
 		if (err < 0)
 			return err;
 	}
@@ -363,7 +364,6 @@ static const char *audio_map[][3] = {
 	{"Left HP Mixer", "PCM Playback Switch",  "Left DAC"},
 	{"Left HP Mixer", "Mic Sidetone Switch",  "Mic PGA"},
 	{"Left HP Mixer", NULL,  "ALC Sidetone Mux"},
-	//{"Right HP Mixer", NULL, "HP Mixer"},
 
 	/* Right HP mixer */
 	{"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
@@ -454,15 +454,13 @@ static int wm9712_add_widgets(struct snd_soc_codec *codec)
 {
 	int i;
 
-	for(i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) {
+	for (i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++)
 		snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]);
-	}
 
-	/* set up audio path audio_mapnects */
-	for(i = 0; audio_map[i][0] != NULL; i++) {
+	/* set up audio path connects */
+	for (i = 0; audio_map[i][0] != NULL; i++)
 		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
-	}
+					   audio_map[i][1], audio_map[i][2]);
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
@@ -540,7 +538,8 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
 }
 
 #define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
+		SNDRV_PCM_RATE_48000)
 
 struct snd_soc_codec_dai wm9712_dai[] = {
 {
@@ -577,8 +576,6 @@ EXPORT_SYMBOL_GPL(wm9712_dai);
 
 static int wm9712_dapm_event(struct snd_soc_codec *codec, int event)
 {
-	u16 reg;
-
 	switch (event) {
 	case SNDRV_CTL_POWER_D0: /* full On */
 	case SNDRV_CTL_POWER_D1: /* partial On */
@@ -633,7 +630,7 @@ static int wm9712_soc_resume(struct platform_device *pdev)
 	u16 *cache = codec->reg_cache;
 
 	ret = wm9712_reset(codec, 1);
-	if (ret < 0){
+	if (ret < 0) {
 		printk(KERN_ERR "could not reset AC97 codec\n");
 		return ret;
 	}
@@ -642,9 +639,9 @@ static int wm9712_soc_resume(struct platform_device *pdev)
 
 	if (ret == 0) {
 		/* Sync reg_cache with the hardware after cold reset */
-		for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i+=2) {
+		for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i += 2) {
 			if (i == AC97_INT_PAGING || i == AC97_POWERDOWN ||
-				(i > 0x58 && i != 0x5c))
+			    (i > 0x58 && i != 0x5c))
 				continue;
 			soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
 		}
@@ -757,7 +754,6 @@ struct snd_soc_codec_device soc_codec_dev_wm9712 = {
 	.suspend =	wm9712_soc_suspend,
 	.resume =	wm9712_soc_resume,
 };
-
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712);
 
 MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver");
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
new file mode 100644
index 000000000000..0230d83e8e5e
--- /dev/null
+++ b/sound/soc/omap/Kconfig
@@ -0,0 +1,19 @@
+menu "SoC Audio for the Texas Instruments OMAP"
+
+config SND_OMAP_SOC
+	tristate "SoC Audio for the Texas Instruments OMAP chips"
+	depends on ARCH_OMAP && SND_SOC
+
+config SND_OMAP_SOC_MCBSP
+	tristate
+	select OMAP_MCBSP
+
+config SND_OMAP_SOC_N810
+	tristate "SoC Audio support for Nokia N810"
+	depends on SND_OMAP_SOC && MACH_NOKIA_N810
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC3X
+	help
+	  Say Y if you want to add support for SoC audio on Nokia N810.
+
+endmenu
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
new file mode 100644
index 000000000000..d8d8d58075e3
--- /dev/null
+++ b/sound/soc/omap/Makefile
@@ -0,0 +1,11 @@
+# OMAP Platform Support
+snd-soc-omap-objs := omap-pcm.o
+snd-soc-omap-mcbsp-objs := omap-mcbsp.o
+
+obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o
+obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
+
+# OMAP Machine Support
+snd-soc-n810-objs := n810.o
+
+obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
new file mode 100644
index 000000000000..83b1eb4e40f3
--- /dev/null
+++ b/sound/soc/omap/n810.c
@@ -0,0 +1,336 @@
+/*
+ * n810.c  --  SoC audio for Nokia N810
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * 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 program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/arch/hardware.h>
+#include <asm/arch/gpio.h>
+#include <asm/arch/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic3x.h"
+
+#define RX44_HEADSET_AMP_GPIO	10
+#define RX44_SPEAKER_AMP_GPIO	101
+
+static struct clk *sys_clkout2;
+static struct clk *sys_clkout2_src;
+static struct clk *func96m_clk;
+
+static int n810_spk_func;
+static int n810_jack_func;
+
+static void n810_ext_control(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func);
+	snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func);
+
+	snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int n810_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->codec;
+
+	n810_ext_control(codec);
+	return clk_enable(sys_clkout2);
+}
+
+static void n810_shutdown(struct snd_pcm_substream *substream)
+{
+	clk_disable(sys_clkout2);
+}
+
+static int n810_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_codec_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	int err;
+
+	/* Set codec DAI configuration */
+	err = codec_dai->dai_ops.set_fmt(codec_dai,
+					 SND_SOC_DAIFMT_I2S |
+					 SND_SOC_DAIFMT_NB_NF |
+					 SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0)
+		return err;
+
+	/* Set cpu DAI configuration */
+	err = cpu_dai->dai_ops.set_fmt(cpu_dai,
+				       SND_SOC_DAIFMT_I2S |
+				       SND_SOC_DAIFMT_NB_NF |
+				       SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0)
+		return err;
+
+	/* Set the codec system clock for DAC and ADC */
+	err = codec_dai->dai_ops.set_sysclk(codec_dai, 0, 12000000,
+					    SND_SOC_CLOCK_IN);
+
+	return err;
+}
+
+static struct snd_soc_ops n810_ops = {
+	.startup = n810_startup,
+	.hw_params = n810_hw_params,
+	.shutdown = n810_shutdown,
+};
+
+static int n810_get_spk(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = n810_spk_func;
+
+	return 0;
+}
+
+static int n810_set_spk(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (n810_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	n810_spk_func = ucontrol->value.integer.value[0];
+	n810_ext_control(codec);
+
+	return 1;
+}
+
+static int n810_get_jack(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = n810_jack_func;
+
+	return 0;
+}
+
+static int n810_set_jack(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (n810_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	n810_jack_func = ucontrol->value.integer.value[0];
+	n810_ext_control(codec);
+
+	return 1;
+}
+
+static int n810_spk_event(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1);
+	else
+		omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0);
+
+	return 0;
+}
+
+static int n810_jack_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1);
+	else
+		omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0);
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event),
+	SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
+};
+
+static const char *audio_map[][3] = {
+	{"Headphone Jack", NULL, "HPLOUT"},
+	{"Headphone Jack", NULL, "HPROUT"},
+
+	{"Ext Spk", NULL, "LLOUT"},
+	{"Ext Spk", NULL, "RLOUT"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *jack_function[] = {"Off", "Headphone"};
+static const struct soc_enum n810_enum[] = {
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+	SOC_ENUM_SINGLE_EXT(3, jack_function),
+};
+
+static const struct snd_kcontrol_new aic33_n810_controls[] = {
+	SOC_ENUM_EXT("Speaker Function", n810_enum[0],
+		     n810_get_spk, n810_set_spk),
+	SOC_ENUM_EXT("Jack Function", n810_enum[1],
+		     n810_get_jack, n810_set_jack),
+};
+
+static int n810_aic33_init(struct snd_soc_codec *codec)
+{
+	int i, err;
+
+	/* Not connected */
+	snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0);
+	snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0);
+	snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0);
+
+	/* Add N810 specific controls */
+	for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {
+		err = snd_ctl_add(codec->card,
+			snd_soc_cnew(&aic33_n810_controls[i], codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* Add N810 specific widgets */
+	for (i = 0; i < ARRAY_SIZE(aic33_dapm_widgets); i++)
+		snd_soc_dapm_new_control(codec, &aic33_dapm_widgets[i]);
+
+	/* Set up N810 specific audio path audio_map */
+	for (i = 0; i < ARRAY_SIZE(audio_map); i++)
+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
+			audio_map[i][1], audio_map[i][2]);
+
+	snd_soc_dapm_sync_endpoints(codec);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link n810_dai = {
+	.name = "TLV320AIC33",
+	.stream_name = "AIC33",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &aic3x_dai,
+	.init = n810_aic33_init,
+	.ops = &n810_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_n810 = {
+	.name = "N810",
+	.dai_link = &n810_dai,
+	.num_links = 1,
+};
+
+/* Audio private data */
+static struct aic3x_setup_data n810_aic33_setup = {
+	.i2c_address = 0x18,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device n810_snd_devdata = {
+	.machine = &snd_soc_machine_n810,
+	.platform = &omap_soc_platform,
+	.codec_dev = &soc_codec_dev_aic3x,
+	.codec_data = &n810_aic33_setup,
+};
+
+static struct platform_device *n810_snd_device;
+
+static int __init n810_soc_init(void)
+{
+	int err;
+	struct device *dev;
+
+	if (!machine_is_nokia_n810())
+		return -ENODEV;
+
+	n810_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!n810_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(n810_snd_device, &n810_snd_devdata);
+	n810_snd_devdata.dev = &n810_snd_device->dev;
+	*(unsigned int *)n810_dai.cpu_dai->private_data = 1; /* McBSP2 */
+	err = platform_device_add(n810_snd_device);
+	if (err)
+		goto err1;
+
+	dev = &n810_snd_device->dev;
+
+	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;
+	}
+	sys_clkout2 = clk_get(dev, "sys_clkout2");
+	if (IS_ERR(sys_clkout2)) {
+		dev_err(dev, "Could not get sys_clkout2\n");
+		goto err1;
+	}
+	/*
+	 * Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use
+	 * 96 MHz as its parent in order to get 12 MHz
+	 */
+	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;
+	}
+	clk_set_parent(sys_clkout2_src, func96m_clk);
+	clk_set_rate(sys_clkout2, 12000000);
+
+	if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0)
+		BUG();
+	if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0)
+		BUG();
+	omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0);
+	omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0);
+
+	return 0;
+err2:
+	clk_put(sys_clkout2);
+	platform_device_del(n810_snd_device);
+err1:
+	platform_device_put(n810_snd_device);
+
+	return err;
+
+}
+
+static void __exit n810_soc_exit(void)
+{
+	platform_device_unregister(n810_snd_device);
+}
+
+module_init(n810_soc_init);
+module_exit(n810_soc_exit);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_DESCRIPTION("ALSA SoC Nokia N810");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
new file mode 100644
index 000000000000..40d87e6d0de8
--- /dev/null
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -0,0 +1,414 @@
+/*
+ * omap-mcbsp.c  --  OMAP ALSA SoC DAI driver using McBSP port
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * 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 program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <asm/arch/control.h>
+#include <asm/arch/dma.h>
+#include <asm/arch/mcbsp.h>
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+#define OMAP_MCBSP_RATES	(SNDRV_PCM_RATE_44100 | \
+				 SNDRV_PCM_RATE_48000 | \
+				 SNDRV_PCM_RATE_KNOT)
+
+struct omap_mcbsp_data {
+	unsigned int			bus_id;
+	struct omap_mcbsp_reg_cfg	regs;
+	/*
+	 * Flags indicating is the bus already activated and configured by
+	 * another substream
+	 */
+	int				active;
+	int				configured;
+};
+
+#define to_mcbsp(priv)	container_of((priv), struct omap_mcbsp_data, bus_id)
+
+static struct omap_mcbsp_data mcbsp_data[NUM_LINKS];
+
+/*
+ * Stream DMA parameters. DMA request line and port address are set runtime
+ * since they are different between OMAP1 and later OMAPs
+ */
+static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2] = {
+{
+	{ .name		= "I2S PCM Stereo out", },
+	{ .name		= "I2S PCM Stereo in", },
+},
+};
+
+#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
+static const int omap1_dma_reqs[][2] = {
+	{ OMAP_DMA_MCBSP1_TX, OMAP_DMA_MCBSP1_RX },
+	{ OMAP_DMA_MCBSP2_TX, OMAP_DMA_MCBSP2_RX },
+	{ OMAP_DMA_MCBSP3_TX, OMAP_DMA_MCBSP3_RX },
+};
+static const unsigned long omap1_mcbsp_port[][2] = {
+	{ OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
+	  OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
+	{ OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1,
+	  OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
+	{ OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DXR1,
+	  OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DRR1 },
+};
+#else
+static const int omap1_dma_reqs[][2] = {};
+static const unsigned long omap1_mcbsp_port[][2] = {};
+#endif
+#if defined(CONFIG_ARCH_OMAP2420)
+static const int omap2420_dma_reqs[][2] = {
+	{ OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX },
+	{ OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX },
+};
+static const unsigned long omap2420_mcbsp_port[][2] = {
+	{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
+	  OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
+	{ OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1,
+	  OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
+};
+#else
+static const int omap2420_dma_reqs[][2] = {};
+static const unsigned long omap2420_mcbsp_port[][2] = {};
+#endif
+
+static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+	int err = 0;
+
+	if (!cpu_dai->active)
+		err = omap_mcbsp_request(mcbsp_data->bus_id);
+
+	return err;
+}
+
+static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+
+	if (!cpu_dai->active) {
+		omap_mcbsp_free(mcbsp_data->bus_id);
+		mcbsp_data->configured = 0;
+	}
+}
+
+static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+	int err = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (!mcbsp_data->active++)
+			omap_mcbsp_start(mcbsp_data->bus_id);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (!--mcbsp_data->active)
+			omap_mcbsp_stop(mcbsp_data->bus_id);
+		break;
+	default:
+		err = -EINVAL;
+	}
+
+	return err;
+}
+
+static int omap_mcbsp_dai_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_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+	int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
+	unsigned long port;
+
+	if (cpu_class_is_omap1()) {
+		dma = omap1_dma_reqs[bus_id][substream->stream];
+		port = omap1_mcbsp_port[bus_id][substream->stream];
+	} else if (cpu_is_omap2420()) {
+		dma = omap2420_dma_reqs[bus_id][substream->stream];
+		port = omap2420_mcbsp_port[bus_id][substream->stream];
+	} else {
+		/*
+		 * TODO: Add support for 2430 and 3430
+		 */
+		return -ENODEV;
+	}
+	omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
+	omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
+	cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
+
+	if (mcbsp_data->configured) {
+		/* McBSP already configured by another stream */
+		return 0;
+	}
+
+	switch (params_channels(params)) {
+	case 2:
+		/* Set 1 word per (McBPSP) frame and use dual-phase frames */
+		regs->rcr2	|= RFRLEN2(1 - 1) | RPHASE;
+		regs->rcr1	|= RFRLEN1(1 - 1);
+		regs->xcr2	|= XFRLEN2(1 - 1) | XPHASE;
+		regs->xcr1	|= XFRLEN1(1 - 1);
+		break;
+	default:
+		/* Unsupported number of channels */
+		return -EINVAL;
+	}
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		/* Set word lengths */
+		regs->rcr2	|= RWDLEN2(OMAP_MCBSP_WORD_16);
+		regs->rcr1	|= RWDLEN1(OMAP_MCBSP_WORD_16);
+		regs->xcr2	|= XWDLEN2(OMAP_MCBSP_WORD_16);
+		regs->xcr1	|= XWDLEN1(OMAP_MCBSP_WORD_16);
+		/* Set FS period and length in terms of bit clock periods */
+		regs->srgr2	|= FPER(16 * 2 - 1);
+		regs->srgr1	|= FWID(16 - 1);
+		break;
+	default:
+		/* Unsupported PCM format */
+		return -EINVAL;
+	}
+
+	omap_mcbsp_config(bus_id, &mcbsp_data->regs);
+	mcbsp_data->configured = 1;
+
+	return 0;
+}
+
+/*
+ * This must be called before _set_clkdiv and _set_sysclk since McBSP register
+ * cache is initialized here
+ */
+static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+				      unsigned int fmt)
+{
+	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+
+	if (mcbsp_data->configured)
+		return 0;
+
+	memset(regs, 0, sizeof(*regs));
+	/* Generic McBSP register settings */
+	regs->spcr2	|= XINTM(3) | FREE;
+	regs->spcr1	|= RINTM(3);
+	regs->rcr2	|= RFIG;
+	regs->xcr2	|= XFIG;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		/* 1-bit data delay */
+		regs->rcr2	|= RDATDLY(1);