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

Documentation/sound/alsa/ALSA-Configuration.txt

@@ -795,6 +795,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  lg-lw		LG LW20/LW25 laptop
 	  tcl		TCL S700
 	  clevo		Clevo laptops (m520G, m665n)
+	  medion	Medion Rim 2150
 	  test		for testing/debugging purpose, almost all controls can be
 			adjusted.  Appearing only when compiled with
 			$CONFIG_SND_DEBUG=y

include/sound/mpu401.h

@@ -50,6 +50,7 @@
 #define MPU401_INFO_INTEGRATED	(1 << 2)	/* integrated h/w port */
 #define MPU401_INFO_MMIO	(1 << 3)	/* MMIO access */
 #define MPU401_INFO_TX_IRQ	(1 << 4)	/* independent TX irq */
+#define MPU401_INFO_NO_ACK	(1 << 6)	/* No ACK cmd needed */
 
 #define MPU401_MODE_BIT_INPUT		0
 #define MPU401_MODE_BIT_OUTPUT		1

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

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) {

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 },

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
 		}
 	}
 

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"

sound/soc/Makefile

 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/

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]);
-	}
 
 	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;
 	}
@@ -642,7 +639,7 @@ 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))
 				continue;
@@ -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");

sound/soc/omap/Kconfig

+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

sound/soc/omap/Makefile

+# 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

sound/soc/omap/n810.c

+/*
+ * 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");

sound/soc/omap/omap-mcbsp.h

+/*
+ * 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

sound/soc/omap/omap-pcm.c

+/*
+ * 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");

sound/soc/omap/omap-pcm.h

+/*
+ * 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