Merge branch 'master' of /pub/scm/linux/kernel/git/torvalds/linux-2.6

27 files changed, 1454 insertions, 103 deletions

diff --git a/sound/core/info.c b/sound/core/info.c
index 9977ec2eace3..cb5ead3e202d 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -544,7 +544,7 @@ int __init snd_info_init(void)
 {
 	struct proc_dir_entry *p;
 
-	p = snd_create_proc_entry("asound", S_IFDIR | S_IRUGO | S_IXUGO, &proc_root);
+	p = snd_create_proc_entry("asound", S_IFDIR | S_IRUGO | S_IXUGO, NULL);
 	if (p == NULL)
 		return -ENOMEM;
 	snd_proc_root = p;
@@ -594,7 +594,7 @@ int __exit snd_info_done(void)
 #ifdef CONFIG_SND_OSSEMUL
 		snd_info_free_entry(snd_oss_root);
 #endif
-		snd_remove_proc_entry(&proc_root, snd_proc_root);
+		snd_remove_proc_entry(NULL, snd_proc_root);
 	}
 	return 0;
 }
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 920e5780c228..23b7bc02728b 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -629,9 +629,8 @@ static const struct file_operations snd_mem_proc_fops = {
 static int __init snd_mem_init(void)
 {
 #ifdef CONFIG_PROC_FS
-	snd_mem_proc = create_proc_entry(SND_MEM_PROC_FILE, 0644, NULL);
-	if (snd_mem_proc)
-		snd_mem_proc->proc_fops = &snd_mem_proc_fops;
+	snd_mem_proc = proc_create(SND_MEM_PROC_FILE, 0644, NULL,
+				   &snd_mem_proc_fops);
 #endif
 	return 0;
 }
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/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index 59203511e77d..54a1f9036c66 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -194,6 +194,7 @@ static void pcsp_stop_beep(struct snd_pcsp *chip)
 	spin_unlock_irq(&chip->substream_lock);
 }
 
+#ifdef CONFIG_PM
 static int pcsp_suspend(struct platform_device *dev, pm_message_t state)
 {
 	struct snd_pcsp *chip = platform_get_drvdata(dev);
@@ -201,6 +202,9 @@ static int pcsp_suspend(struct platform_device *dev, pm_message_t state)
 	snd_pcm_suspend_all(chip->pcm);
 	return 0;
 }
+#else
+#define pcsp_suspend NULL
+#endif	/* CONFIG_PM */
 
 static void pcsp_shutdown(struct platform_device *dev)
 {
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 581debf37dcb..7e4742109572 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -515,19 +515,16 @@ config SND_FM801
 config SND_FM801_TEA575X_BOOL
 	bool "ForteMedia FM801 + TEA5757 tuner"
 	depends on SND_FM801
+	depends on VIDEO_V4L1=y || VIDEO_V4L1=SND_FM801
 	help
 	  Say Y here to include support for soundcards based on the ForteMedia
 	  FM801 chip with a TEA5757 tuner connected to GPIO1-3 pins (Media
 	  Forte SF256-PCS-02) into the snd-fm801 driver.
 
-	  This will enable support for the old V4L1 API.
-
 config SND_FM801_TEA575X
 	tristate
 	depends on SND_FM801_TEA575X_BOOL
 	default SND_FM801
-	select VIDEO_V4L1
-	select VIDEO_DEV
 
 config SND_HDA_INTEL
 	tristate "Intel HD Audio"
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 39198e505b12..2da89810ca10 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -3446,6 +3446,7 @@ static const struct snd_kcontrol_new snd_ac97_controls_vt1617a[] = {
 int patch_vt1617a(struct snd_ac97 * ac97)
 {
 	int err = 0;
+	int val;
 
 	/* we choose to not fail out at this point, but we tell the
 	   caller when we return */
@@ -3456,7 +3457,13 @@ int patch_vt1617a(struct snd_ac97 * ac97)
 	/* bring analog power consumption to normal by turning off the
 	 * headphone amplifier, like WinXP driver for EPIA SP
 	 */
-	snd_ac97_write_cache(ac97, 0x5c, 0x20);
+	/* We need to check the bit before writing it.
+	 * On some (many?) hardwares, setting bit actually clears it!
+	 */
+	val = snd_ac97_read(ac97, 0x5c);
+	if (!(val & 0x20))
+		snd_ac97_write_cache(ac97, 0x5c, 0x20);
+
 	ac97->ext_id |= AC97_EI_SPDIF;	/* force the detection of spdif */
 	ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000;
 	ac97->build_ops = &patch_vt1616_ops;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index cdda64b02f46..6d4df45e81e0 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 },
@@ -11816,7 +11902,10 @@ static void alc861_auto_set_output_and_unmute(struct hda_codec *codec,
 					      hda_nid_t nid,
 					      int pin_type, int dac_idx)
 {
-	alc_set_pin_output(codec, nid, pin_type);
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    pin_type);
+	snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+			    AMP_OUT_UNMUTE);
 }
 
 static void alc861_auto_init_multi_out(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index b3a15d616873..393f7fd2b1be 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4289,6 +4289,8 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = {
 	{ .id = 0x83847635, .name = "STAC9250D", .patch = patch_stac925x },
 	{ .id = 0x83847636, .name = "STAC9251", .patch = patch_stac925x },
 	{ .id = 0x83847637, .name = "STAC9250D", .patch = patch_stac925x },
+	{ .id = 0x83847645, .name = "92HD206X", .patch = patch_stac927x },
+	{ .id = 0x83847646, .name = "92HD206D", .patch = patch_stac927x },
  	/* The following does not take into account .id=0x83847661 when subsys =
  	 * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are
  	 * currently not fully supported.
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/sh/aica.c b/sound/sh/aica.c
index d49417bf78c6..9ca113326143 100644
--- a/sound/sh/aica.c
+++ b/sound/sh/aica.c
@@ -663,7 +663,7 @@ static int __init aica_init(void)
 		return err;
 	pd = platform_device_register_simple(SND_AICA_DRIVER, -1,
 					     aica_memory_space, 2);
-	if (unlikely(IS_ERR(pd))) {
+	if (IS_ERR(pd)) {
 		platform_driver_unregister(&snd_aica_driver);
 		return PTR_ERR(pd);
 	}
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/wm8753.c b/sound/soc/codecs/wm8753.c
index 76a5c7b05dfb..fb41826c4c4c 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -150,7 +150,7 @@ static int wm8753_write(struct snd_soc_codec *codec, unsigned int reg,
 	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
 	data[1] = value & 0x00ff;
 
-	wm8753_write_reg_cache (codec, reg, value);
+	wm8753_write_reg_cache(codec, reg, value);
 	if (codec->hw_write(codec->control_data, data, 2) == 2)
 		return 0;
 	else
@@ -249,7 +249,7 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
 	int mode = wm8753_read_reg_cache(codec, WM8753_IOCTL);
 
-	if (((mode &0xc) >> 2) == ucontrol->value.integer.value[0])
+	if (((mode & 0xc) >> 2) == ucontrol->value.integer.value[0])
 		return 0;
 
 	mode &= 0xfff3;
@@ -342,7 +342,8 @@ static int wm8753_add_controls(struct snd_soc_codec *codec)
 
 	for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) {
 		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8753_snd_controls[i],codec, NULL));
+				snd_soc_cnew(&wm8753_snd_controls[i],
+						codec, NULL));
 		if (err < 0)
 			return err;
 	}
@@ -722,7 +723,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
 
 	if ((Ndiv < 6) || (Ndiv > 12))
 		printk(KERN_WARNING
-			"WM8753 N value outwith recommended range! N = %d\n",Ndiv);
+			"wm8753: unsupported N = %d\n", Ndiv);
 
 	pll_div->n = Ndiv;
 	Nmod = target % source;
@@ -1300,8 +1301,9 @@ static int wm8753_dapm_event(struct snd_soc_codec *codec, int event)
 }
 
 #define WM8753_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
-		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+		SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
 #define WM8753_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
@@ -1507,9 +1509,9 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
 	struct snd_soc_codec *codec = socdev->codec;
 
 	/* we only need to suspend if we are a valid card */
-	if(!codec->card)
+	if (!codec->card)
 		return 0;
-		
+
 	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
 	return 0;
 }
@@ -1523,7 +1525,7 @@ static int wm8753_resume(struct platform_device *pdev)
 	u16 *cache = codec->reg_cache;
 
 	/* we only need to resume if we are a valid card */
-	if(!codec->card)
+	if (!codec->card)
 		return 0;
 
 	/* Sync reg_cache with the hardware */
@@ -1613,9 +1615,10 @@ static int wm8753_init(struct snd_soc_device *socdev)
 	wm8753_add_widgets(codec);
 	ret = snd_soc_register_card(socdev);
 	if (ret < 0) {
-      	printk(KERN_ERR "wm8753: failed to register card\n");
+		printk(KERN_ERR "wm8753: failed to register card\n");
 		goto card_err;
-    }
+	}
+
 	return ret;
 
 card_err:
@@ -1630,7 +1633,7 @@ pcm_err:
    around */
 static struct snd_soc_device *wm8753_socdev;
 
-#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 
 /*
  * WM8753 2 wire address is determined by GPIO5
@@ -1661,7 +1664,7 @@ 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 == NULL){
+	if (!i2c) {
 		kfree(codec);
 		return -ENOMEM;
 	}
@@ -1749,7 +1752,7 @@ static int wm8753_probe(struct platform_device *pdev)
 	wm8753_socdev = socdev;
 	INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
 
-#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	if (setup->i2c_address) {
 		normal_i2c[0] = setup->i2c_address;
 		codec->hw_write = (hw_write_t)i2c_master_send;
@@ -1793,7 +1796,7 @@ static int wm8753_remove(struct platform_device *pdev)
 	run_delayed_work(&codec->delayed_work);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
-#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	i2c_del_driver(&wm8753_i2c_driver);
 #endif
 	kfree(codec->private_data);
@@ -1808,7 +1811,6 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = {
 	.suspend = 	wm8753_suspend,
 	.resume =	wm8753_resume,
 };
-
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
 
 MODULE_DESCRIPTION("ASoC WM8753 driver");
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);
+		regs->xcr2	|= XDATDLY(1);
+		break;
+	default:
+		/* Unsupported data format */
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		/* McBSP master. Set FS and bit clocks as outputs */
+		regs->pcr0	|= FSXM | FSRM |
+				   CLKXM | CLKRM;
+		/* Sample rate generator drives the FS */
+		regs->srgr2	|= FSGM;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		/* McBSP slave */
+		break;
+	default:
+		/* Unsupported master/slave configuration */
+		return -EINVAL;
+	}
+
+	/* Set bit clock (CLKX/CLKR) and FS polarities */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		/*
+		 * Normal BCLK + FS.
+		 * FS active low. TX data driven on falling edge of bit clock
+		 * and RX data sampled on rising edge of bit clock.
+		 */
+		regs->pcr0	|= FSXP | FSRP |
+				   CLKXP | CLKRP;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		regs->pcr0	|= CLKXP | CLKRP;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		regs->pcr0	|= FSXP | FSRP;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
+				     int div_id, int div)
+{
+	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+
+	if (div_id != OMAP_MCBSP_CLKGDV)
+		return -ENODEV;
+
+	regs->srgr1	|= CLKGDV(div - 1);
+
+	return 0;
+}
+
+static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
+				       int clk_id)
+{
+	int sel_bit;
+	u16 reg;
+
+	if (cpu_class_is_omap1()) {
+		/* OMAP1's can use only external source clock */
+		if (unlikely(clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK))
+			return -EINVAL;
+		else
+			return 0;
+	}
+
+	switch (mcbsp_data->bus_id) {
+	case 0:
+		reg = OMAP2_CONTROL_DEVCONF0;
+		sel_bit = 2;
+		break;
+	case 1:
+		reg = OMAP2_CONTROL_DEVCONF0;
+		sel_bit = 6;
+		break;
+	/* TODO: Support for ports 3 - 5 in OMAP2430 and OMAP34xx */
+	default:
+		return -EINVAL;
+	}
+
+	if (cpu_class_is_omap2()) {
+		if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) {
+			omap_ctrl_writel(omap_ctrl_readl(reg) &
+					 ~(1 << sel_bit), reg);
+		} else {
+			omap_ctrl_writel(omap_ctrl_readl(reg) |
+					 (1 << sel_bit), reg);
+		}
+	}
+
+	return 0;
+}
+
+static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+					 int clk_id, unsigned int freq,
+					 int dir)
+{
+	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+	int err = 0;
+
+	switch (clk_id) {
+	case OMAP_MCBSP_SYSCLK_CLK:
+		regs->srgr2	|= CLKSM;
+		break;
+	case OMAP_MCBSP_SYSCLK_CLKS_FCLK:
+	case OMAP_MCBSP_SYSCLK_CLKS_EXT:
+		err = omap_mcbsp_dai_set_clks_src(mcbsp_data, clk_id);
+		break;
+
+	case OMAP_MCBSP_SYSCLK_CLKX_EXT:
+		regs->srgr2	|= CLKSM;
+	case OMAP_MCBSP_SYSCLK_CLKR_EXT:
+		regs->pcr0	|= SCLKME;
+		break;
+	default:
+		err = -ENODEV;
+	}
+
+	return err;
+}
+
+struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS] = {
+{
+	.name = "omap-mcbsp-dai",
+	.id = 0,
+	.type = SND_SOC_DAI_I2S,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = OMAP_MCBSP_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = OMAP_MCBSP_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = {
+		.startup = omap_mcbsp_dai_startup,
+		.shutdown = omap_mcbsp_dai_shutdown,
+		.trigger = omap_mcbsp_dai_trigger,
+		.hw_params = omap_mcbsp_dai_hw_params,
+	},
+	.dai_ops = {
+		.set_fmt = omap_mcbsp_dai_set_dai_fmt,
+		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,
+		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,
+	},
+	.private_data = &mcbsp_data[0].bus_id,
+},
+};
+EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_DESCRIPTION("OMAP I2S SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
new file mode 100644
index 000000000000..9965fd4b0427
--- /dev/null
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -0,0 +1,49 @@
+/*
+ * omap-mcbsp.h
+ *
+ * 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
+ *
+ */
+
+#ifndef __OMAP_I2S_H__
+#define __OMAP_I2S_H__
+
+/* Source clocks for McBSP sample rate generator */
+enum omap_mcbsp_clksrg_clk {
+	OMAP_MCBSP_SYSCLK_CLKS_FCLK,	/* Internal FCLK */
+	OMAP_MCBSP_SYSCLK_CLKS_EXT,	/* External CLKS pin */
+	OMAP_MCBSP_SYSCLK_CLK,		/* Internal ICLK */
+	OMAP_MCBSP_SYSCLK_CLKX_EXT,	/* External CLKX pin */
+	OMAP_MCBSP_SYSCLK_CLKR_EXT,	/* External CLKR pin */
+};
+
+/* McBSP dividers */
+enum omap_mcbsp_div {
+	OMAP_MCBSP_CLKGDV,		/* Sample rate generator divider */
+};
+
+/*
+ * REVISIT: Preparation for the ASoC v2. Let the number of available links to
+ * be same than number of McBSP ports found in OMAP(s) we are compiling for.
+ */
+#define NUM_LINKS	1
+
+extern struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS];
+
+#endif
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
new file mode 100644
index 000000000000..62370202c649
--- /dev/null
+++ b/sound/soc/omap/omap-pcm.c
@@ -0,0 +1,357 @@
+/*
+ * omap-pcm.c  --  ALSA PCM interface for the OMAP SoC
+ *
+ * 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/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/arch/dma.h>
+#include "omap-pcm.h"
+
+static const struct snd_pcm_hardware omap_pcm_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_INTERLEAVED |
+				  SNDRV_PCM_INFO_PAUSE |
+				  SNDRV_PCM_INFO_RESUME,
+	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 64 * 1024,
+	.periods_min		= 2,
+	.periods_max		= 255,
+	.buffer_bytes_max	= 128 * 1024,
+};
+
+struct omap_runtime_data {
+	spinlock_t			lock;
+	struct omap_pcm_dma_data	*dma_data;
+	int				dma_ch;
+	int				period_index;
+};
+
+static void omap_pcm_dma_irq(int ch, u16 stat, void *data)
+{
+	struct snd_pcm_substream *substream = data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct omap_runtime_data *prtd = runtime->private_data;
+	unsigned long flags;
+
+	if (cpu_is_omap1510()) {
+		/*
+		 * OMAP1510 doesn't support DMA chaining so have to restart
+		 * the transfer after all periods are transferred
+		 */
+		spin_lock_irqsave(&prtd->lock, flags);
+		if (prtd->period_index >= 0) {
+			if (++prtd->period_index == runtime->periods) {
+				prtd->period_index = 0;
+				omap_start_dma(prtd->dma_ch);
+			}
+		}
+		spin_unlock_irqrestore(&prtd->lock, flags);
+	}
+
+	snd_pcm_period_elapsed(substream);
+}
+
+/* this may get called several times by oss emulation */
+static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct omap_runtime_data *prtd = runtime->private_data;
+	struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data;
+	int err = 0;
+
+	if (!dma_data)
+		return -ENODEV;
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+	runtime->dma_bytes = params_buffer_bytes(params);
+
+	if (prtd->dma_data)
+		return 0;
+	prtd->dma_data = dma_data;
+	err = omap_request_dma(dma_data->dma_req, dma_data->name,
+			       omap_pcm_dma_irq, substream, &prtd->dma_ch);
+	if (!cpu_is_omap1510()) {
+		/*
+		 * Link channel with itself so DMA doesn't need any
+		 * reprogramming while looping the buffer
+		 */
+		omap_dma_link_lch(prtd->dma_ch, prtd->dma_ch);
+	}
+
+	return err;
+}
+
+static int omap_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct omap_runtime_data *prtd = runtime->private_data;
+
+	if (prtd->dma_data == NULL)
+		return 0;
+
+	if (!cpu_is_omap1510())
+		omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch);
+	omap_free_dma(prtd->dma_ch);
+	prtd->dma_data = NULL;
+
+	snd_pcm_set_runtime_buffer(substream, NULL);
+
+	return 0;
+}
+
+static int omap_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct omap_runtime_data *prtd = runtime->private_data;
+	struct omap_pcm_dma_data *dma_data = prtd->dma_data;
+	struct omap_dma_channel_params dma_params;
+
+	memset(&dma_params, 0, sizeof(dma_params));
+	/*
+	 * Note: Regardless of interface data formats supported by OMAP McBSP
+	 * or EAC blocks, internal representation is always fixed 16-bit/sample
+	 */
+	dma_params.data_type			= OMAP_DMA_DATA_TYPE_S16;
+	dma_params.trigger			= dma_data->dma_req;
+	dma_params.sync_mode			= OMAP_DMA_SYNC_ELEMENT;
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		dma_params.src_amode		= OMAP_DMA_AMODE_POST_INC;
+		dma_params.dst_amode		= OMAP_DMA_AMODE_CONSTANT;
+		dma_params.src_or_dst_synch	= OMAP_DMA_DST_SYNC;
+		dma_params.src_start		= runtime->dma_addr;
+		dma_params.dst_start		= dma_data->port_addr;
+	} else {
+		dma_params.src_amode		= OMAP_DMA_AMODE_CONSTANT;
+		dma_params.dst_amode		= OMAP_DMA_AMODE_POST_INC;
+		dma_params.src_or_dst_synch	= OMAP_DMA_SRC_SYNC;
+		dma_params.src_start		= dma_data->port_addr;
+		dma_params.dst_start		= runtime->dma_addr;
+	}
+	/*
+	 * Set DMA transfer frame size equal to ALSA period size and frame
+	 * count as no. of ALSA periods. Then with DMA frame interrupt enabled,
+	 * we can transfer the whole ALSA buffer with single DMA transfer but
+	 * still can get an interrupt at each period bounary
+	 */
+	dma_params.elem_count	= snd_pcm_lib_period_bytes(substream) / 2;
+	dma_params.frame_count	= runtime->periods;
+	omap_set_dma_params(prtd->dma_ch, &dma_params);
+
+	omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
+
+	return 0;
+}
+
+static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct omap_runtime_data *prtd = runtime->private_data;
+	int ret = 0;
+
+	spin_lock_irq(&prtd->lock);
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		prtd->period_index = 0;
+		omap_start_dma(prtd->dma_ch);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		prtd->period_index = -1;
+		omap_stop_dma(prtd->dma_ch);
+		break;
+	default:
+		ret = -EINVAL;
+	}
+	spin_unlock_irq(&prtd->lock);
+
+	return ret;
+}
+
+static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct omap_runtime_data *prtd = runtime->private_data;
+	dma_addr_t ptr;
+	snd_pcm_uframes_t offset;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		ptr = omap_get_dma_src_pos(prtd->dma_ch);
+	else
+		ptr = omap_get_dma_dst_pos(prtd->dma_ch);
+
+	offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+	if (offset >= runtime->buffer_size)
+		offset = 0;
+
+	return offset;
+}
+
+static int omap_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct omap_runtime_data *prtd;
+	int ret;
+
+	snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware);
+
+	/* Ensure that buffer size is a multiple of period size */
+	ret = snd_pcm_hw_constraint_integer(runtime,
+					    SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0)
+		goto out;
+
+	prtd = kzalloc(sizeof(prtd), GFP_KERNEL);
+	if (prtd == NULL) {
+		ret = -ENOMEM;
+		goto out;
+	}
+	spin_lock_init(&prtd->lock);
+	runtime->private_data = prtd;
+
+out:
+	return ret;
+}
+
+static int omap_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	kfree(runtime->private_data);
+	return 0;
+}
+
+static int omap_pcm_mmap(struct snd_pcm_substream *substream,
+	struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+				     runtime->dma_area,
+				     runtime->dma_addr,
+				     runtime->dma_bytes);
+}
+
+struct snd_pcm_ops omap_pcm_ops = {
+	.open		= omap_pcm_open,
+	.close		= omap_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= omap_pcm_hw_params,
+	.hw_free	= omap_pcm_hw_free,
+	.prepare	= omap_pcm_prepare,
+	.trigger	= omap_pcm_trigger,
+	.pointer	= omap_pcm_pointer,
+	.mmap		= omap_pcm_mmap,
+};
+
+static u64 omap_pcm_dmamask = DMA_BIT_MASK(32);
+
+static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
+	int stream)
+{
+	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	size_t size = omap_pcm_hardware.buffer_bytes_max;
+
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = pcm->card->dev;
+	buf->private_data = NULL;
+	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+					   &buf->addr, GFP_KERNEL);
+	if (!buf->area)
+		return -ENOMEM;
+
+	buf->bytes = size;
+	return 0;
+}
+
+static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+	int stream;
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+		if (!substream)
+			continue;
+
+		buf = &substream->dma_buffer;
+		if (!buf->area)
+			continue;
+
+		dma_free_writecombine(pcm->card->dev, buf->bytes,
+				      buf->area, buf->addr);
+		buf->area = NULL;
+	}
+}
+
+int omap_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+		 struct snd_pcm *pcm)
+{
+	int ret = 0;
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &omap_pcm_dmamask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = DMA_32BIT_MASK;
+
+	if (dai->playback.channels_min) {
+		ret = omap_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto out;
+	}
+
+	if (dai->capture.channels_min) {
+		ret = omap_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto out;
+	}
+
+out:
+	return ret;
+}
+
+struct snd_soc_platform omap_soc_platform = {
+	.name		= "omap-pcm-audio",
+	.pcm_ops 	= &omap_pcm_ops,
+	.pcm_new	= omap_pcm_new,
+	.pcm_free	= omap_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(omap_soc_platform);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_DESCRIPTION("OMAP PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
new file mode 100644
index 000000000000..e4369bdfd77d
--- /dev/null
+++ b/sound/soc/omap/omap-pcm.h
@@ -0,0 +1,35 @@
+/*
+ * omap-pcm.h
+ *
+ * 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
+ *
+ */
+
+#ifndef __OMAP_PCM_H__
+#define __OMAP_PCM_H__
+
+struct omap_pcm_dma_data {
+	char		*name;		/* stream identifier */
+	int		dma_req;	/* DMA request line */
+	unsigned long	port_addr;	/* transmit/receive register */
+};
+
+extern struct snd_soc_platform omap_soc_platform;
+
+#endif
diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c
index 9ed8f2e8da10..4eab2c19c454 100644
--- a/sound/soc/s3c24xx/ln2440sbc_alc650.c
+++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c
@@ -1,10 +1,10 @@
 /*
  * SoC audio for ln2440sbc
- * 
+ *
  * Copyright 2007 KonekTel, a.s.
  * Author: Ivan Kuten
  *         ivan.kuten@promwad.com
- * 
+ *
  * Heavily based on smdk2443_wm9710.c
  * Copyright 2007 Wolfson Microelectronics PLC.
  * Author: Graeme Gregory
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 962cc20b1af5..0e9d1c5f2484 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -33,7 +33,7 @@
 #include <asm/arch/regs-gpio.h>
 #include <asm/hardware.h>
 #include <asm/arch/audio.h>
-#include <asm/io.h>
+#include <linux/io.h>
 #include <asm/arch/spi-gpio.h>
 
 #include <asm/plat-s3c24xx/regs-iis.h>
@@ -122,7 +122,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
 
 	/* set MCLK division for sample rate */
 	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
-		S3C2410_IISMOD_32FS );
+		S3C2410_IISMOD_32FS);
 	if (ret < 0)
 		return ret;
 
@@ -133,7 +133,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
 
 	/* set prescaler division for sample rate */
 	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
-		S3C24XX_PRESCALE(4,4));
+		S3C24XX_PRESCALE(4, 4));
 	if (ret < 0)
 		return ret;
 
@@ -222,7 +222,7 @@ static struct snd_soc_ops neo1973_voice_ops = {
 	.hw_free = neo1973_voice_hw_free,
 };
 
-static int neo1973_scenario = 0;
+static int neo1973_scenario;
 
 static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
@@ -233,7 +233,7 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
 
 static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
 {
-	switch(neo1973_scenario) {
+	switch (neo1973_scenario) {
 	case NEO_AUDIO_OFF:
 		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
 		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
@@ -334,7 +334,7 @@ static void lm4857_write_regs(void)
 static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	int reg=kcontrol->private_value & 0xFF;
+	int reg = kcontrol->private_value & 0xFF;
 	int shift = (kcontrol->private_value >> 8) & 0x0F;
 	int mask = (kcontrol->private_value >> 16) & 0xFF;
 
@@ -349,11 +349,11 @@ static int lm4857_set_reg(struct snd_kcontrol *kcontrol,
 	int shift = (kcontrol->private_value >> 8) & 0x0F;
 	int mask = (kcontrol->private_value >> 16) & 0xFF;
 
-	if (((lm4857_regs[reg] >> shift ) & mask) ==
+	if (((lm4857_regs[reg] >> shift) & mask) ==
 		ucontrol->value.integer.value[0])
 		return 0;
 
-	lm4857_regs[reg] &= ~ (mask << shift);
+	lm4857_regs[reg] &= ~(mask << shift);
 	lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift;
 	lm4857_write_regs();
 	return 1;
@@ -398,7 +398,7 @@ static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
 
 
 /* example machine audio_mapnections */
-static const char* audio_map[][3] = {
+static const char *audio_map[][3] = {
 
 	/* Connections to the lm4857 amp */
 	{"Audio Out", NULL, "LOUT1"},
@@ -450,7 +450,7 @@ static const char *neo_scenarios[] = {
 };
 
 static const struct soc_enum neo_scenario_enum[] = {
-	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios),neo_scenarios),
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios),
 };
 
 static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
@@ -521,8 +521,8 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 /*
  * BT Codec DAI
  */
-static struct snd_soc_cpu_dai bt_dai =
-{	.name = "Bluetooth",
+static struct snd_soc_cpu_dai bt_dai = {
+	.name = "Bluetooth",
 	.id = 0,
 	.type = SND_SOC_DAI_PCM,
 	.playback = {
@@ -616,6 +616,35 @@ static int lm4857_i2c_attach(struct i2c_adapter *adap)
 	return i2c_probe(adap, &addr_data, lm4857_amp_probe);
 }
 
+static u8 lm4857_state;
+
+static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
+{
+	dev_dbg(&dev->dev, "lm4857_suspend\n");
+	lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf;
+	if (lm4857_state) {
+		lm4857_regs[LM4857_CTRL] &= 0xf0;
+		lm4857_write_regs();
+	}
+	return 0;
+}
+
+static int lm4857_resume(struct i2c_client *dev)
+{
+	if (lm4857_state) {
+		lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f);
+		lm4857_write_regs();
+	}
+	return 0;
+}
+
+static void lm4857_shutdown(struct i2c_client *dev)
+{
+	dev_dbg(&dev->dev, "lm4857_shutdown\n");
+	lm4857_regs[LM4857_CTRL] &= 0xf0;
+	lm4857_write_regs();
+}
+
 /* corgi i2c codec control layer */
 static struct i2c_driver lm4857_i2c_driver = {
 	.driver = {
@@ -623,6 +652,9 @@ static struct i2c_driver lm4857_i2c_driver = {
 		.owner = THIS_MODULE,
 	},
 	.id =             I2C_DRIVERID_LM4857,
+	.suspend =        lm4857_suspend,
+	.resume	=         lm4857_resume,
+	.shutdown =       lm4857_shutdown,
 	.attach_adapter = lm4857_i2c_attach,
 	.detach_client =  lm4857_i2c_detach,
 	.command =        NULL,
@@ -667,6 +699,6 @@ module_init(neo1973_init);
 module_exit(neo1973_exit);
 
 /* Module information */
-MODULE_AUTHOR("Graeme Gregory, graeme.gregory@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org, www.openmoko.org");
 MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 1c1ddbf7f3c0..e81d9a6c83da 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -19,6 +19,7 @@
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/interrupt.h>
+#include <linux/io.h>
 #include <linux/wait.h>
 #include <linux/delay.h>
 #include <linux/clk.h>
@@ -30,7 +31,6 @@
 #include <sound/soc.h>
 
 #include <asm/hardware.h>
-#include <asm/io.h>
 #include <asm/plat-s3c/regs-ac97.h>
 #include <asm/arch/regs-gpio.h>
 #include <asm/arch/regs-clock.h>
@@ -47,7 +47,7 @@ struct s3c24xx_ac97_info {
 };
 static struct s3c24xx_ac97_info s3c24xx_ac97;
 
-DECLARE_COMPLETION(ac97_completion);
+static DECLARE_COMPLETION(ac97_completion);
 static u32 codec_ready;
 static DECLARE_MUTEX(ac97_mutex);
 
@@ -290,7 +290,7 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd)
 	u32 ac_glbctrl;
 
 	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
-	switch(cmd) {
+	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
@@ -333,7 +333,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
 	u32 ac_glbctrl;
 
 	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
-	switch(cmd) {
+	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
@@ -391,7 +391,6 @@ struct snd_soc_cpu_dai s3c2443_ac97_dai[] = {
 		.trigger = s3c2443_ac97_mic_trigger,},
 },
 };
-
 EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 4ebcd6a8bf28..1ed6afd45459 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -224,6 +224,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai,
 		iismod |= S3C2410_IISMOD_SLAVE;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:
+		iismod &= ~S3C2410_IISMOD_SLAVE;
 		break;
 	default:
 		return -EINVAL;
@@ -234,6 +235,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai,
 		iismod |= S3C2410_IISMOD_MSB;
 		break;
 	case SND_SOC_DAIFMT_I2S:
+		iismod &= ~S3C2410_IISMOD_MSB;
 		break;
 	default:
 		return -EINVAL;
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 49580fb481d5..7806ae614617 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -20,6 +20,7 @@
 
 #include <linux/module.h>
 #include <linux/init.h>
+#include <linux/io.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <linux/dma-mapping.h>
@@ -30,7 +31,6 @@
 #include <sound/soc.h>
 
 #include <asm/dma.h>
-#include <asm/io.h>
 #include <asm/hardware.h>
 #include <asm/arch/dma.h>
 #include <asm/arch/audio.h>
@@ -93,7 +93,7 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
 	while (prtd->dma_loaded < prtd->dma_limit) {
 		unsigned long len = prtd->dma_period;
 
-		DBG("dma_loaded: %d\n",prtd->dma_loaded);
+		DBG("dma_loaded: %d\n", prtd->dma_loaded);
 
 		if ((pos + len) > prtd->dma_end) {
 			len  = prtd->dma_end - pos;
@@ -101,7 +101,7 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
 			       __func__, len);
 		}
 
-		ret = s3c2410_dma_enqueue(prtd->params->channel, 
+		ret = s3c2410_dma_enqueue(prtd->params->channel,
 			substream, pos, len);
 
 		if (ret == 0) {
@@ -129,7 +129,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
 		return;
 
 	prtd = substream->runtime->private_data;
-	
+
 	if (substream)
 		snd_pcm_period_elapsed(substream);
 
@@ -150,7 +150,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct s3c24xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data;
 	unsigned long totbytes = params_buffer_bytes(params);
-	int ret=0;
+	int ret = 0;
 
 	DBG("Entered %s\n", __func__);
 
@@ -171,7 +171,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
 		ret = s3c2410_dma_request(prtd->params->channel,
 					  prtd->params->client, NULL);
 
-		if (ret) {
+		if (ret < 0) {
 			DBG(KERN_ERR "failed to get dma channel\n");
 			return ret;
 		}
@@ -223,7 +223,7 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream)
 	/* return if this is a bufferless transfer e.g.
 	 * codec <--> BT codec or GSM modem -- lg FIXME */
 	if (!prtd->params)
-	 	return 0;
+		return 0;
 
 	/* channel needs configuring for mem=>device, increment memory addr,
 	 * sync to pclk, half-word transfers to the IIS-FIFO. */
@@ -293,8 +293,8 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	return ret;
 }
 
-static snd_pcm_uframes_t 
-	s3c24xx_pcm_pointer(struct snd_pcm_substream *substream)
+static snd_pcm_uframes_t
+s3c24xx_pcm_pointer(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s3c24xx_runtime_data *prtd = runtime->private_data;
@@ -313,7 +313,7 @@ static snd_pcm_uframes_t
 
 	spin_unlock(&prtd->lock);
 
-	DBG("Pointer %x %x\n",src,dst);
+	DBG("Pointer %x %x\n", src, dst);
 
 	/* we seem to be getting the odd error from the pcm library due
 	 * to out-of-bounds pointers. this is maybe due to the dma engine
@@ -355,11 +355,11 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream)
 
 	DBG("Entered %s\n", __func__);
 
-	if (prtd)
-		kfree(prtd);
-	else
+	if (!prtd)
 		DBG("s3c24xx_pcm_close called with prtd == NULL\n");
 
+	kfree(prtd);
+
 	return 0;
 }
 
@@ -371,9 +371,9 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream,
 	DBG("Entered %s\n", __func__);
 
 	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
-                                     runtime->dma_area,
-                                     runtime->dma_addr,
-                                     runtime->dma_bytes);
+				     runtime->dma_area,
+				     runtime->dma_addr,
+				     runtime->dma_bytes);
 }
 
 static struct snd_pcm_ops s3c24xx_pcm_ops = {
@@ -432,7 +432,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 
 static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK;
 
-static int s3c24xx_pcm_new(struct snd_card *card, 
+static int s3c24xx_pcm_new(struct snd_card *card,
 	struct snd_soc_codec_dai *dai, struct snd_pcm *pcm)
 {
 	int ret = 0;
@@ -467,7 +467,6 @@ struct snd_soc_platform s3c24xx_soc_platform = {
 	.pcm_new	= s3c24xx_pcm_new,
 	.pcm_free	= s3c24xx_pcm_free_dma_buffers,
 };
-
 EXPORT_SYMBOL_GPL(s3c24xx_soc_platform);
 
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");