initial_commit

22 files changed, 6149 insertions, 0 deletions

diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
new file mode 100644
index 00000000..99054cf1
--- /dev/null
+++ b/sound/soc/omap/Kconfig
@@ -0,0 +1,135 @@
+config SND_OMAP_SOC
+	tristate "SoC Audio for the Texas Instruments OMAP chips"
+	depends on ARCH_OMAP
+
+config SND_OMAP_SOC_MCBSP
+	tristate
+	select OMAP_MCBSP
+
+config SND_OMAP_SOC_MCPDM
+	tristate
+
+config SND_OMAP_SOC_N810
+	tristate "SoC Audio support for Nokia N810"
+	depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
+	depends on OMAP_MUX
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC3X
+	help
+	  Say Y if you want to add support for SoC audio on Nokia N810.
+
+config SND_OMAP_SOC_RX51
+	tristate "SoC Audio support for Nokia RX-51"
+	depends on SND_OMAP_SOC && MACH_NOKIA_RX51
+	select OMAP_MCBSP
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC3X
+	select SND_SOC_TPA6130A2
+	help
+	  Say Y if you want to add support for SoC audio on Nokia RX-51
+	  hardware. This is also known as Nokia N900 product.
+
+config SND_OMAP_SOC_AMS_DELTA
+	tristate "SoC Audio support for Amstrad E3 (Delta) videophone"
+	depends on SND_OMAP_SOC && MACH_AMS_DELTA
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_CX20442
+	help
+	  Say Y  if you want to add support  for SoC audio device  connected to
+	  a handset and a speakerphone found on Amstrad E3 (Delta) videophone.
+
+	  Note that in order to get those devices fully supported,  you have to
+	  build  the kernel  with  standard  serial port  driver  included  and
+	  configured for at least 4 ports.  Then, from userspace, you must load
+	  a line discipline #19 on the modem (ttyS3) serial line.  The simplest
+	  way to achieve this is to install util-linux-ng  and use the included
+	  ldattach  utility.  This  can be  started  automatically  from  udev,
+	  a simple rule like this one should do the trick (it does for me):
+	  	ACTION=="add", KERNEL=="controlC0", \
+				RUN+="/usr/sbin/ldattach 19 /dev/ttyS3"
+
+config SND_OMAP_SOC_OSK5912
+	tristate "SoC Audio support for omap osk5912"
+	depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC23
+	help
+	  Say Y if you want to add support for SoC audio on osk5912.
+
+config SND_OMAP_SOC_OVERO
+	tristate "SoC Audio support for Gumstix Overo and CompuLab CM-T35"
+	depends on TWL4030_CORE && SND_OMAP_SOC && (MACH_OVERO || MACH_CM_T35)
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Gumstix Overo or CompuLab CM-T35
+
+config SND_OMAP_SOC_OMAP3EVM
+	tristate "SoC Audio support for OMAP3EVM board"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the omap3evm board.
+
+config SND_OMAP_SOC_AM3517EVM
+	tristate "SoC Audio support for OMAP3517 / AM3517 EVM"
+	depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC23
+	help
+	  Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517
+	  EVM.
+
+config SND_OMAP_SOC_SDP3430
+	tristate "SoC Audio support for Texas Instruments SDP3430"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on Texas Instruments
+	  SDP3430.
+
+config SND_OMAP_SOC_SDP4430
+	tristate "SoC Audio support for Texas Instruments SDP4430"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP
+	select SND_OMAP_SOC_MCPDM
+	select SND_SOC_TWL6040
+	help
+	  Say Y if you want to add support for SoC audio on Texas Instruments
+	  SDP4430.
+
+config SND_OMAP_SOC_OMAP3_PANDORA
+	tristate "SoC Audio support for OMAP3 Pandora"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the OMAP3 Pandora.
+
+config SND_OMAP_SOC_OMAP3_BEAGLE
+	tristate "SoC Audio support for OMAP3 Beagle and Devkit8000"
+	depends on TWL4030_CORE && SND_OMAP_SOC
+	depends on (MACH_OMAP3_BEAGLE || MACH_DEVKIT8000)
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the Beagleboard or
+	  the clone Devkit8000.
+
+config SND_OMAP_SOC_ZOOM2
+	tristate "SoC Audio support for Zoom2"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for Soc audio on Zoom2 board.
+
+config SND_OMAP_SOC_IGEP0020
+	tristate "SoC Audio support for IGEP v2"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_IGEP0020
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for Soc audio on IGEP v2 board.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
new file mode 100644
index 00000000..6c2c87ee
--- /dev/null
+++ b/sound/soc/omap/Makefile
@@ -0,0 +1,38 @@
+# OMAP Platform Support
+snd-soc-omap-objs := omap-pcm.o
+snd-soc-omap-mcbsp-objs := omap-mcbsp.o
+snd-soc-omap-mcpdm-objs := omap-mcpdm.o mcpdm.o
+
+obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o
+obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
+obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o
+
+# OMAP Machine Support
+snd-soc-n810-objs := n810.o
+snd-soc-rx51-objs := rx51.o
+snd-soc-ams-delta-objs := ams-delta.o
+snd-soc-osk5912-objs := osk5912.o
+snd-soc-overo-objs := overo.o
+snd-soc-omap3evm-objs := omap3evm.o
+snd-soc-am3517evm-objs := am3517evm.o
+snd-soc-sdp3430-objs := sdp3430.o
+snd-soc-sdp4430-objs := sdp4430.o
+snd-soc-omap3pandora-objs := omap3pandora.o
+snd-soc-omap3beagle-objs := omap3beagle.o
+snd-soc-zoom2-objs := zoom2.o
+snd-soc-igep0020-objs := igep0020.o
+
+obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o
+obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o
+obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
+obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
+obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
+obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
+obj-$(CONFIG_SND_OMAP_SOC_SDP4430) += snd-soc-sdp4430.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
+obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
+obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
new file mode 100644
index 00000000..73dde4a1
--- /dev/null
+++ b/sound/soc/omap/am3517evm.c
@@ -0,0 +1,195 @@
+/*
+ * am3517evm.c  -- ALSA SoC support for OMAP3517 / AM3517 EVM
+ *
+ * Author: Anuj Aggarwal <anuj.aggarwal@ti.com>
+ *
+ * Based on sound/soc/omap/beagle.c by Steve Sakoman
+ *
+ * Copyright (C) 2009 Texas Instruments Incorporated
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind,
+ * whether express or implied; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+#include "../codecs/tlv320aic23.h"
+
+#define CODEC_CLOCK 	12000000
+
+static int am3517evm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_DSP_B |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_DSP_B |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+			CODEC_CLOCK, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_CLKR_SRC_CLKX, 0,
+				SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_CLKR_SRC_CLKX\n");
+		return ret;
+	}
+
+	snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0,
+				SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops am3517evm_ops = {
+	.hw_params = am3517evm_hw_params,
+};
+
+/* am3517evm machine dapm widgets */
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Line Out", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+	SND_SOC_DAPM_MIC("Mic In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Line Out connected to LLOUT, RLOUT */
+	{"Line Out", NULL, "LOUT"},
+	{"Line Out", NULL, "ROUT"},
+
+	{"LLINEIN", NULL, "Line In"},
+	{"RLINEIN", NULL, "Line In"},
+
+	{"MICIN", NULL, "Mic In"},
+};
+
+static int am3517evm_aic23_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	/* Add am3517-evm specific widgets */
+	snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* Set up davinci-evm specific audio path audio_map */
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+
+	/* always connected */
+	snd_soc_dapm_enable_pin(dapm, "Line Out");
+	snd_soc_dapm_enable_pin(dapm, "Line In");
+	snd_soc_dapm_enable_pin(dapm, "Mic In");
+
+	snd_soc_dapm_sync(dapm);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link am3517evm_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai_name ="omap-mcbsp-dai.0",
+	.codec_dai_name = "tlv320aic23-hifi",
+	.platform_name = "omap-pcm-audio",
+	.codec_name = "tlv320aic23-codec.2-001a",
+	.init = am3517evm_aic23_init,
+	.ops = &am3517evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_am3517evm = {
+	.name = "am3517evm",
+	.dai_link = &am3517evm_dai,
+	.num_links = 1,
+};
+
+static struct platform_device *am3517evm_snd_device;
+
+static int __init am3517evm_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3517evm())
+		return -ENODEV;
+	pr_info("OMAP3517 / AM3517 EVM SoC init\n");
+
+	am3517evm_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!am3517evm_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(am3517evm_snd_device, &snd_soc_am3517evm);
+
+	ret = platform_device_add(am3517evm_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(am3517evm_snd_device);
+
+	return ret;
+}
+
+static void __exit am3517evm_soc_exit(void)
+{
+	platform_device_unregister(am3517evm_snd_device);
+}
+
+module_init(am3517evm_soc_init);
+module_exit(am3517evm_soc_exit);
+
+MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3517 / AM3517 EVM");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
new file mode 100644
index 00000000..462cbcbe
--- /dev/null
+++ b/sound/soc/omap/ams-delta.c
@@ -0,0 +1,660 @@
+/*
+ * ams-delta.c  --  SoC audio for Amstrad E3 (Delta) videophone
+ *
+ * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
+ *
+ * Initially based on sound/soc/omap/osk5912.x
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * 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/gpio.h>
+#include <linux/spinlock.h>
+#include <linux/tty.h>
+
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include <plat/board-ams-delta.h>
+#include <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/cx20442.h"
+
+
+/* Board specific DAPM widgets */
+static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
+	/* Handset */
+	SND_SOC_DAPM_MIC("Mouthpiece", NULL),
+	SND_SOC_DAPM_HP("Earpiece", NULL),
+	/* Handsfree/Speakerphone */
+	SND_SOC_DAPM_MIC("Microphone", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* How they are connected to codec pins */
+static const struct snd_soc_dapm_route ams_delta_audio_map[] = {
+	{"TELIN", NULL, "Mouthpiece"},
+	{"Earpiece", NULL, "TELOUT"},
+
+	{"MIC", NULL, "Microphone"},
+	{"Speaker", NULL, "SPKOUT"},
+};
+
+/*
+ * Controls, functional after the modem line discipline is activated.
+ */
+
+/* Virtual switch: audio input/output constellations */
+static const char *ams_delta_audio_mode[] =
+	{"Mixed", "Handset", "Handsfree", "Speakerphone"};
+
+/* Selection <-> pin translation */
+#define AMS_DELTA_MOUTHPIECE	0
+#define AMS_DELTA_EARPIECE	1
+#define AMS_DELTA_MICROPHONE	2
+#define AMS_DELTA_SPEAKER	3
+#define AMS_DELTA_AGC		4
+
+#define AMS_DELTA_MIXED		((1 << AMS_DELTA_EARPIECE) | \
+						(1 << AMS_DELTA_MICROPHONE))
+#define AMS_DELTA_HANDSET	((1 << AMS_DELTA_MOUTHPIECE) | \
+						(1 << AMS_DELTA_EARPIECE))
+#define AMS_DELTA_HANDSFREE	((1 << AMS_DELTA_MICROPHONE) | \
+						(1 << AMS_DELTA_SPEAKER))
+#define AMS_DELTA_SPEAKERPHONE	(AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC))
+
+static const unsigned short ams_delta_audio_mode_pins[] = {
+	AMS_DELTA_MIXED,
+	AMS_DELTA_HANDSET,
+	AMS_DELTA_HANDSFREE,
+	AMS_DELTA_SPEAKERPHONE,
+};
+
+static unsigned short ams_delta_audio_agc;
+
+static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
+	unsigned short pins;
+	int pin, changed = 0;
+
+	/* Refuse any mode changes if we are not able to control the codec. */
+	if (!codec->hw_write)
+		return -EUNATCH;
+
+	if (ucontrol->value.enumerated.item[0] >= control->max)
+		return -EINVAL;
+
+	mutex_lock(&codec->mutex);
+
+	/* Translate selection to bitmap */
+	pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];
+
+	/* Setup pins after corresponding bits if changed */
+	pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(dapm, "Mouthpiece");
+		else
+			snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(dapm, "Earpiece");
+		else
+			snd_soc_dapm_disable_pin(dapm, "Earpiece");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(dapm, "Microphone");
+		else
+			snd_soc_dapm_disable_pin(dapm, "Microphone");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(dapm, "Speaker");
+		else
+			snd_soc_dapm_disable_pin(dapm, "Speaker");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_AGC));
+	if (pin != ams_delta_audio_agc) {
+		ams_delta_audio_agc = pin;
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(dapm, "AGCIN");
+		else
+			snd_soc_dapm_disable_pin(dapm, "AGCIN");
+	}
+	if (changed)
+		snd_soc_dapm_sync(dapm);
+
+	mutex_unlock(&codec->mutex);
+
+	return changed;
+}
+
+static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	unsigned short pins, mode;
+
+	pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") <<
+							AMS_DELTA_MOUTHPIECE) |
+			(snd_soc_dapm_get_pin_status(dapm, "Earpiece") <<
+							AMS_DELTA_EARPIECE));
+	if (pins)
+		pins |= (snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
+							AMS_DELTA_MICROPHONE);
+	else
+		pins = ((snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
+							AMS_DELTA_MICROPHONE) |
+			(snd_soc_dapm_get_pin_status(dapm, "Speaker") <<
+							AMS_DELTA_SPEAKER) |
+			(ams_delta_audio_agc << AMS_DELTA_AGC));
+
+	for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++)
+		if (pins == ams_delta_audio_mode_pins[mode])
+			break;
+
+	if (mode >= ARRAY_SIZE(ams_delta_audio_mode))
+		return -EINVAL;
+
+	ucontrol->value.enumerated.item[0] = mode;
+
+	return 0;
+}
+
+static const struct soc_enum ams_delta_audio_enum[] = {
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode),
+						ams_delta_audio_mode),
+};
+
+static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
+	SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
+			ams_delta_get_audio_mode, ams_delta_set_audio_mode),
+};
+
+/* Hook switch */
+static struct snd_soc_jack ams_delta_hook_switch;
+static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = {
+	{
+		.gpio = 4,
+		.name = "hook_switch",
+		.report = SND_JACK_HEADSET,
+		.invert = 1,
+		.debounce_time = 150,
+	}
+};
+
+/* After we are able to control the codec over the modem,
+ * the hook switch can be used for dynamic DAPM reconfiguration. */
+static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
+	/* Handset */
+	{
+		.pin = "Mouthpiece",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Earpiece",
+		.mask = SND_JACK_HEADPHONE,
+	},
+	/* Handsfree */
+	{
+		.pin = "Microphone",
+		.mask = SND_JACK_MICROPHONE,
+		.invert = 1,
+	},
+	{
+		.pin = "Speaker",
+		.mask = SND_JACK_HEADPHONE,
+		.invert = 1,
+	},
+};
+
+
+/*
+ * Modem line discipline, required for making above controls functional.
+ * Activated from userspace with ldattach, possibly invoked from udev rule.
+ */
+
+/* To actually apply any modem controlled configuration changes to the codec,
+ * we must connect codec DAI pins to the modem for a moment.  Be careful not
+ * to interfere with our digital mute function that shares the same hardware. */
+static struct timer_list cx81801_timer;
+static bool cx81801_cmd_pending;
+static bool ams_delta_muted;
+static DEFINE_SPINLOCK(ams_delta_lock);
+
+static void cx81801_timeout(unsigned long data)
+{
+	int muted;
+
+	spin_lock(&ams_delta_lock);
+	cx81801_cmd_pending = 0;
+	muted = ams_delta_muted;
+	spin_unlock(&ams_delta_lock);
+
+	/* Reconnect the codec DAI back from the modem to the CPU DAI
+	 * only if digital mute still off */
+	if (!muted)
+		ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0);
+}
+
+/*
+ * Used for passing a codec structure pointer
+ * from the board initialization code to the tty line discipline.
+ */
+static struct snd_soc_codec *cx20442_codec;
+
+/* Line discipline .open() */
+static int cx81801_open(struct tty_struct *tty)
+{
+	int ret;
+
+	if (!cx20442_codec)
+		return -ENODEV;
+
+	/*
+	 * Pass the codec structure pointer for use by other ldisc callbacks,
+	 * both the card and the codec specific parts.
+	 */
+	tty->disc_data = cx20442_codec;
+
+	ret = v253_ops.open(tty);
+
+	if (ret < 0)
+		tty->disc_data = NULL;
+
+	return ret;
+}
+
+/* Line discipline .close() */
+static void cx81801_close(struct tty_struct *tty)
+{
+	struct snd_soc_codec *codec = tty->disc_data;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	del_timer_sync(&cx81801_timer);
+
+	/* Prevent the hook switch from further changing the DAPM pins */
+	INIT_LIST_HEAD(&ams_delta_hook_switch.pins);
+
+	if (!codec)
+		return;
+
+	v253_ops.close(tty);
+
+	/* Revert back to default audio input/output constellation */
+	snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
+	snd_soc_dapm_enable_pin(dapm, "Earpiece");
+	snd_soc_dapm_enable_pin(dapm, "Microphone");
+	snd_soc_dapm_disable_pin(dapm, "Speaker");
+	snd_soc_dapm_disable_pin(dapm, "AGCIN");
+	snd_soc_dapm_sync(dapm);
+}
+
+/* Line discipline .hangup() */
+static int cx81801_hangup(struct tty_struct *tty)
+{
+	cx81801_close(tty);
+	return 0;
+}
+
+/* Line discipline .recieve_buf() */
+static void cx81801_receive(struct tty_struct *tty,
+				const unsigned char *cp, char *fp, int count)
+{
+	struct snd_soc_codec *codec = tty->disc_data;
+	const unsigned char *c;
+	int apply, ret;
+
+	if (!codec)
+		return;
+
+	if (!codec->hw_write) {
+		/* First modem response, complete setup procedure */
+
+		/* Initialize timer used for config pulse generation */
+		setup_timer(&cx81801_timer, cx81801_timeout, 0);
+
+		v253_ops.receive_buf(tty, cp, fp, count);
+
+		/* Link hook switch to DAPM pins */
+		ret = snd_soc_jack_add_pins(&ams_delta_hook_switch,
+					ARRAY_SIZE(ams_delta_hook_switch_pins),
+					ams_delta_hook_switch_pins);
+		if (ret)
+			dev_warn(codec->dev,
+				"Failed to link hook switch to DAPM pins, "
+				"will continue with hook switch unlinked.\n");
+
+		return;
+	}
+
+	v253_ops.receive_buf(tty, cp, fp, count);
+
+	for (c = &cp[count - 1]; c >= cp; c--) {
+		if (*c != '\r')
+			continue;
+		/* Complete modem response received, apply config to codec */
+
+		spin_lock_bh(&ams_delta_lock);
+		mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150));
+		apply = !ams_delta_muted && !cx81801_cmd_pending;
+		cx81801_cmd_pending = 1;
+		spin_unlock_bh(&ams_delta_lock);
+
+		/* Apply config pulse by connecting the codec to the modem
+		 * if not already done */
+		if (apply)
+			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
+						AMS_DELTA_LATCH2_MODEM_CODEC);
+		break;
+	}
+}
+
+/* Line discipline .write_wakeup() */
+static void cx81801_wakeup(struct tty_struct *tty)
+{
+	v253_ops.write_wakeup(tty);
+}
+
+static struct tty_ldisc_ops cx81801_ops = {
+	.magic = TTY_LDISC_MAGIC,
+	.name = "cx81801",
+	.owner = THIS_MODULE,
+	.open = cx81801_open,
+	.close = cx81801_close,
+	.hangup = cx81801_hangup,
+	.receive_buf = cx81801_receive,
+	.write_wakeup = cx81801_wakeup,
+};
+
+
+/*
+ * Even if not very useful, the sound card can still work without any of the
+ * above functonality activated.  You can still control its audio input/output
+ * constellation and speakerphone gain from userspace by issuing AT commands
+ * over the modem port.
+ */
+
+static int ams_delta_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* Set cpu DAI configuration */
+	return snd_soc_dai_set_fmt(rtd->cpu_dai,
+				   SND_SOC_DAIFMT_DSP_A |
+				   SND_SOC_DAIFMT_NB_NF |
+				   SND_SOC_DAIFMT_CBM_CFM);
+}
+
+static struct snd_soc_ops ams_delta_ops = {
+	.hw_params = ams_delta_hw_params,
+};
+
+
+/* Board specific codec bias level control */
+static int ams_delta_set_bias_level(struct snd_soc_card *card,
+					enum snd_soc_bias_level level)
+{
+	struct snd_soc_codec *codec = card->rtd->codec;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
+						AMS_DELTA_LATCH2_MODEM_NRESET);
+		break;
+	case SND_SOC_BIAS_OFF:
+		if (codec->dapm.bias_level != SND_SOC_BIAS_OFF)
+			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
+						0);
+	}
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+/* Digital mute implemented using modem/CPU multiplexer.
+ * Shares hardware with codec config pulse generation */
+static bool ams_delta_muted = 1;
+
+static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	int apply;
+
+	if (ams_delta_muted == mute)
+		return 0;
+
+	spin_lock_bh(&ams_delta_lock);
+	ams_delta_muted = mute;
+	apply = !cx81801_cmd_pending;
+	spin_unlock_bh(&ams_delta_lock);
+
+	if (apply)
+		ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
+				mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0);
+	return 0;
+}
+
+/* Our codec DAI probably doesn't have its own .ops structure */
+static struct snd_soc_dai_ops ams_delta_dai_ops = {
+	.digital_mute = ams_delta_digital_mute,
+};
+
+/* Will be used if the codec ever has its own digital_mute function */
+static int ams_delta_startup(struct snd_pcm_substream *substream)
+{
+	return ams_delta_digital_mute(NULL, 0);
+}
+
+static void ams_delta_shutdown(struct snd_pcm_substream *substream)
+{
+	ams_delta_digital_mute(NULL, 1);
+}
+
+
+/*
+ * Card initialization
+ */
+
+static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_card *card = rtd->card;
+	int ret;
+	/* Codec is ready, now add/activate board specific controls */
+
+	/* Store a pointer to the codec structure for tty ldisc use */
+	cx20442_codec = codec;
+
+	/* Set up digital mute if not provided by the codec */
+	if (!codec_dai->driver->ops) {
+		codec_dai->driver->ops = &ams_delta_dai_ops;
+	} else {
+		ams_delta_ops.startup = ams_delta_startup;
+		ams_delta_ops.shutdown = ams_delta_shutdown;
+	}
+
+	/* Set codec bias level */
+	ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
+
+	/* Add hook switch - can be used to control the codec from userspace
+	 * even if line discipline fails */
+	ret = snd_soc_jack_new(rtd->codec, "hook_switch",
+				SND_JACK_HEADSET, &ams_delta_hook_switch);
+	if (ret)
+		dev_warn(card->dev,
+				"Failed to allocate resources for hook switch, "
+				"will continue without one.\n");
+	else {
+		ret = snd_soc_jack_add_gpios(&ams_delta_hook_switch,
+					ARRAY_SIZE(ams_delta_hook_switch_gpios),
+					ams_delta_hook_switch_gpios);
+		if (ret)
+			dev_warn(card->dev,
+				"Failed to set up hook switch GPIO line, "
+				"will continue with hook switch inactive.\n");
+	}
+
+	/* Register optional line discipline for over the modem control */
+	ret = tty_register_ldisc(N_V253, &cx81801_ops);
+	if (ret) {
+		dev_warn(card->dev,
+				"Failed to register line discipline, "
+				"will continue without any controls.\n");
+		return 0;
+	}
+
+	/* Add board specific DAPM widgets and routes */
+	ret = snd_soc_dapm_new_controls(dapm, ams_delta_dapm_widgets,
+					ARRAY_SIZE(ams_delta_dapm_widgets));
+	if (ret) {
+		dev_warn(card->dev,
+				"Failed to register DAPM controls, "
+				"will continue without any.\n");
+		return 0;
+	}
+
+	ret = snd_soc_dapm_add_routes(dapm, ams_delta_audio_map,
+					ARRAY_SIZE(ams_delta_audio_map));
+	if (ret) {
+		dev_warn(card->dev,
+				"Failed to set up DAPM routes, "
+				"will continue with codec default map.\n");
+		return 0;
+	}
+
+	/* Set up initial pin constellation */
+	snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
+	snd_soc_dapm_enable_pin(dapm, "Earpiece");
+	snd_soc_dapm_enable_pin(dapm, "Microphone");
+	snd_soc_dapm_disable_pin(dapm, "Speaker");
+	snd_soc_dapm_disable_pin(dapm, "AGCIN");
+	snd_soc_dapm_disable_pin(dapm, "AGCOUT");
+	snd_soc_dapm_sync(dapm);
+
+	/* Add virtual switch */
+	ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
+					ARRAY_SIZE(ams_delta_audio_controls));
+	if (ret)
+		dev_warn(card->dev,
+				"Failed to register audio mode control, "
+				"will continue without it.\n");
+
+	return 0;
+}
+
+/* DAI glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ams_delta_dai_link = {
+	.name = "CX20442",
+	.stream_name = "CX20442",
+	.cpu_dai_name ="omap-mcbsp-dai.0",
+	.codec_dai_name = "cx20442-voice",
+	.init = ams_delta_cx20442_init,
+	.platform_name = "omap-pcm-audio",
+	.codec_name = "cx20442-codec",
+	.ops = &ams_delta_ops,
+};
+
+/* Audio card driver */
+static struct snd_soc_card ams_delta_audio_card = {
+	.name = "AMS_DELTA",
+	.dai_link = &ams_delta_dai_link,
+	.num_links = 1,
+	.set_bias_level = ams_delta_set_bias_level,
+};
+
+/* Module init/exit */
+static struct platform_device *ams_delta_audio_platform_device;
+static struct platform_device *cx20442_platform_device;
+
+static int __init ams_delta_module_init(void)
+{
+	int ret;
+
+	if (!(machine_is_ams_delta()))
+		return -ENODEV;
+
+	ams_delta_audio_platform_device =
+			platform_device_alloc("soc-audio", -1);
+	if (!ams_delta_audio_platform_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(ams_delta_audio_platform_device,
+				&ams_delta_audio_card);
+
+	ret = platform_device_add(ams_delta_audio_platform_device);
+	if (ret)
+		goto err;
+
+	/*
+	 * Codec platform device could be registered from elsewhere (board?),
+	 * but I do it here as it makes sense only if used with the card.
+	 */
+	cx20442_platform_device =
+		platform_device_register_simple("cx20442-codec", -1, NULL, 0);
+	return 0;
+err:
+	platform_device_put(ams_delta_audio_platform_device);
+	return ret;
+}
+module_init(ams_delta_module_init);
+
+static void __exit ams_delta_module_exit(void)
+{
+	if (tty_unregister_ldisc(N_V253) != 0)
+		dev_warn(&ams_delta_audio_platform_device->dev,
+			"failed to unregister V253 line discipline\n");
+
+	snd_soc_jack_free_gpios(&ams_delta_hook_switch,
+			ARRAY_SIZE(ams_delta_hook_switch_gpios),
+			ams_delta_hook_switch_gpios);
+
+	/* Keep modem power on */
+	ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
+
+	platform_device_unregister(cx20442_platform_device);
+	platform_device_unregister(ams_delta_audio_platform_device);
+}
+module_exit(ams_delta_module_exit);
+
+MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
+MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c
new file mode 100644
index 00000000..0ae34702
--- /dev/null
+++ b/sound/soc/omap/igep0020.c
@@ -0,0 +1,137 @@
+/*
+ * igep0020.c  --  SoC audio for IGEP v2
+ *
+ * Based on sound/soc/omap/overo.c by Steve Sakoman
+ *
+ * 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 <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+static int igep2_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops igep2_ops = {
+	.hw_params = igep2_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link igep2_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai_name = "omap-mcbsp-dai.1",
+	.codec_dai_name = "twl4030-hifi",
+	.platform_name = "omap-pcm-audio",
+	.codec_name = "twl4030-codec",
+	.ops = &igep2_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_card_igep2 = {
+	.name = "igep2",
+	.dai_link = &igep2_dai,
+	.num_links = 1,
+};
+
+static struct platform_device *igep2_snd_device;
+
+static int __init igep2_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_igep0020())
+		return -ENODEV;
+	printk(KERN_INFO "IGEP v2 SoC init\n");
+
+	igep2_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!igep2_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(igep2_snd_device, &snd_soc_card_igep2);
+
+	ret = platform_device_add(igep2_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(igep2_snd_device);
+
+	return ret;
+}
+module_init(igep2_soc_init);
+
+static void __exit igep2_soc_exit(void)
+{
+	platform_device_unregister(igep2_snd_device);
+}
+module_exit(igep2_soc_exit);
+
+MODULE_AUTHOR("Enric Balletbo i Serra <eballetbo@iseebcn.com>");
+MODULE_DESCRIPTION("ALSA SoC IGEP v2");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c
new file mode 100644
index 00000000..928f0370
--- /dev/null
+++ b/sound/soc/omap/mcpdm.c
@@ -0,0 +1,470 @@
+/*
+ * mcpdm.c  --	McPDM interface driver
+ *
+ * Author: Jorge Eduardo Candelaria <x0107209@ti.com>
+ * Copyright (C) 2009 - Texas Instruments, Inc.
+ *
+ * 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/module.h>
+#include <linux/init.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/wait.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <linux/err.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/irq.h>
+
+#include "mcpdm.h"
+
+static struct omap_mcpdm *mcpdm;
+
+static inline void omap_mcpdm_write(u16 reg, u32 val)
+{
+	__raw_writel(val, mcpdm->io_base + reg);
+}
+
+static inline int omap_mcpdm_read(u16 reg)
+{
+	return __raw_readl(mcpdm->io_base + reg);
+}
+
+static void omap_mcpdm_reg_dump(void)
+{
+	dev_dbg(mcpdm->dev, "***********************\n");
+	dev_dbg(mcpdm->dev, "IRQSTATUS_RAW:  0x%04x\n",
+			omap_mcpdm_read(MCPDM_IRQSTATUS_RAW));
+	dev_dbg(mcpdm->dev, "IRQSTATUS:	0x%04x\n",
+			omap_mcpdm_read(MCPDM_IRQSTATUS));
+	dev_dbg(mcpdm->dev, "IRQENABLE_SET:  0x%04x\n",
+			omap_mcpdm_read(MCPDM_IRQENABLE_SET));
+	dev_dbg(mcpdm->dev, "IRQENABLE_CLR:  0x%04x\n",
+			omap_mcpdm_read(MCPDM_IRQENABLE_CLR));
+	dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n",
+			omap_mcpdm_read(MCPDM_IRQWAKE_EN));
+	dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n",
+			omap_mcpdm_read(MCPDM_DMAENABLE_SET));
+	dev_dbg(mcpdm->dev, "DMAENABLE_CLR:  0x%04x\n",
+			omap_mcpdm_read(MCPDM_DMAENABLE_CLR));
+	dev_dbg(mcpdm->dev, "DMAWAKEEN:	0x%04x\n",
+			omap_mcpdm_read(MCPDM_DMAWAKEEN));
+	dev_dbg(mcpdm->dev, "CTRL:	0x%04x\n",
+			omap_mcpdm_read(MCPDM_CTRL));
+	dev_dbg(mcpdm->dev, "DN_DATA:  0x%04x\n",
+			omap_mcpdm_read(MCPDM_DN_DATA));
+	dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n",
+			omap_mcpdm_read(MCPDM_UP_DATA));
+	dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n",
+			omap_mcpdm_read(MCPDM_FIFO_CTRL_DN));
+	dev_dbg(mcpdm->dev, "FIFO_CTRL_UP:  0x%04x\n",
+			omap_mcpdm_read(MCPDM_FIFO_CTRL_UP));
+	dev_dbg(mcpdm->dev, "DN_OFFSET:	0x%04x\n",
+			omap_mcpdm_read(MCPDM_DN_OFFSET));
+	dev_dbg(mcpdm->dev, "***********************\n");
+}
+
+/*
+ * Takes the McPDM module in and out of reset state.
+ * Uplink and downlink can be reset individually.
+ */
+static void omap_mcpdm_reset_capture(int reset)
+{
+	int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+
+	if (reset)
+		ctrl |= SW_UP_RST;
+	else
+		ctrl &= ~SW_UP_RST;
+
+	omap_mcpdm_write(MCPDM_CTRL, ctrl);
+}
+
+static void omap_mcpdm_reset_playback(int reset)
+{
+	int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+
+	if (reset)
+		ctrl |= SW_DN_RST;
+	else
+		ctrl &= ~SW_DN_RST;
+
+	omap_mcpdm_write(MCPDM_CTRL, ctrl);
+}
+
+/*
+ * Enables the transfer through the PDM interface to/from the Phoenix
+ * codec by enabling the corresponding UP or DN channels.
+ */
+void omap_mcpdm_start(int stream)
+{
+	int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+
+	if (stream)
+		ctrl |= mcpdm->up_channels;
+	else
+		ctrl |= mcpdm->dn_channels;
+
+	omap_mcpdm_write(MCPDM_CTRL, ctrl);
+}
+
+/*
+ * Disables the transfer through the PDM interface to/from the Phoenix
+ * codec by disabling the corresponding UP or DN channels.
+ */
+void omap_mcpdm_stop(int stream)
+{
+	int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+
+	if (stream)
+		ctrl &= ~mcpdm->up_channels;
+	else
+		ctrl &= ~mcpdm->dn_channels;
+
+	omap_mcpdm_write(MCPDM_CTRL, ctrl);
+}
+
+/*
+ * Configures McPDM uplink for audio recording.
+ * This function should be called before omap_mcpdm_start.
+ */
+int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink)
+{
+	int irq_mask = 0;
+	int ctrl;
+
+	if (!uplink)
+		return -EINVAL;
+
+	mcpdm->uplink = uplink;
+
+	/* Enable irq request generation */
+	irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK;
+	omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask);
+
+	/* Configure uplink threshold */
+	if (uplink->threshold > UP_THRES_MAX)
+		uplink->threshold = UP_THRES_MAX;
+
+	omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold);
+
+	/* Configure DMA controller */
+	omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE);
+
+	/* Set pdm out format */
+	ctrl = omap_mcpdm_read(MCPDM_CTRL);
+	ctrl &= ~PDMOUTFORMAT;
+	ctrl |= uplink->format & PDMOUTFORMAT;
+
+	/* Uplink channels */
+	mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK);
+
+	omap_mcpdm_write(MCPDM_CTRL, ctrl);
+
+	return 0;
+}
+
+/*
+ * Configures McPDM downlink for audio playback.
+ * This function should be called before omap_mcpdm_start.
+ */
+int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink)
+{
+	int irq_mask = 0;
+	int ctrl;
+
+	if (!downlink)
+		return -EINVAL;
+
+	mcpdm->downlink = downlink;
+
+	/* Enable irq request generation */
+	irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK;
+	omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask);
+
+	/* Configure uplink threshold */
+	if (downlink->threshold > DN_THRES_MAX)
+		downlink->threshold = DN_THRES_MAX;
+
+	omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold);
+
+	/* Enable DMA request generation */
+	omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE);
+
+	/* Set pdm out format */
+	ctrl = omap_mcpdm_read(MCPDM_CTRL);
+	ctrl &= ~PDMOUTFORMAT;
+	ctrl |= downlink->format & PDMOUTFORMAT;
+
+	/* Downlink channels */
+	mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK);
+
+	omap_mcpdm_write(MCPDM_CTRL, ctrl);
+
+	return 0;
+}
+
+/*
+ * Cleans McPDM uplink configuration.
+ * This function should be called when the stream is closed.
+ */
+int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink)
+{
+	int irq_mask = 0;
+
+	if (!uplink)
+		return -EINVAL;
+
+	/* Disable irq request generation */
+	irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK;
+	omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask);
+
+	/* Disable DMA request generation */
+	omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE);
+
+	/* Clear Downlink channels */
+	mcpdm->up_channels = 0;
+
+	mcpdm->uplink = NULL;
+
+	return 0;
+}
+
+/*
+ * Cleans McPDM downlink configuration.
+ * This function should be called when the stream is closed.
+ */
+int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink)
+{
+	int irq_mask = 0;
+
+	if (!downlink)
+		return -EINVAL;
+
+	/* Disable irq request generation */
+	irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK;
+	omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask);
+
+	/* Disable DMA request generation */
+	omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE);
+
+	/* clear Downlink channels */
+	mcpdm->dn_channels = 0;
+
+	mcpdm->downlink = NULL;
+
+	return 0;
+}
+
+static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id)
+{
+	struct omap_mcpdm *mcpdm_irq = dev_id;
+	int irq_status;
+
+	irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS);
+
+	/* Acknowledge irq event */
+	omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status);
+
+	if (irq & MCPDM_DN_IRQ_FULL) {
+		dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status);
+		omap_mcpdm_reset_playback(1);
+		omap_mcpdm_playback_open(mcpdm_irq->downlink);
+		omap_mcpdm_reset_playback(0);
+	}
+
+	if (irq & MCPDM_DN_IRQ_EMPTY) {
+		dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status);
+		omap_mcpdm_reset_playback(1);
+		omap_mcpdm_playback_open(mcpdm_irq->downlink);
+		omap_mcpdm_reset_playback(0);
+	}
+
+	if (irq & MCPDM_DN_IRQ) {
+		dev_dbg(mcpdm_irq->dev, "DN write request\n");
+	}
+
+	if (irq & MCPDM_UP_IRQ_FULL) {
+		dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status);
+		omap_mcpdm_reset_capture(1);
+		omap_mcpdm_capture_open(mcpdm_irq->uplink);
+		omap_mcpdm_reset_capture(0);
+	}
+
+	if (irq & MCPDM_UP_IRQ_EMPTY) {
+		dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status);
+		omap_mcpdm_reset_capture(1);
+		omap_mcpdm_capture_open(mcpdm_irq->uplink);
+		omap_mcpdm_reset_capture(0);
+	}
+
+	if (irq & MCPDM_UP_IRQ) {
+		dev_dbg(mcpdm_irq->dev, "UP write request\n");
+	}
+
+	return IRQ_HANDLED;
+}
+
+int omap_mcpdm_request(void)
+{
+	int ret;
+
+	clk_enable(mcpdm->clk);
+
+	spin_lock(&mcpdm->lock);
+
+	if (!mcpdm->free) {
+		dev_err(mcpdm->dev, "McPDM interface is in use\n");
+		spin_unlock(&mcpdm->lock);
+		ret = -EBUSY;
+		goto err;
+	}
+	mcpdm->free = 0;
+
+	spin_unlock(&mcpdm->lock);
+
+	/* Disable lines while request is ongoing */
+	omap_mcpdm_write(MCPDM_CTRL, 0x00);
+
+	ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler,
+				0, "McPDM", (void *)mcpdm);
+	if (ret) {
+		dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n");
+		goto err;
+	}
+
+	return 0;
+
+err:
+	clk_disable(mcpdm->clk);
+	return ret;
+}
+
+void omap_mcpdm_free(void)
+{
+	spin_lock(&mcpdm->lock);
+	if (mcpdm->free) {
+		dev_err(mcpdm->dev, "McPDM interface is already free\n");
+		spin_unlock(&mcpdm->lock);
+		return;
+	}
+	mcpdm->free = 1;
+	spin_unlock(&mcpdm->lock);
+
+	clk_disable(mcpdm->clk);
+
+	free_irq(mcpdm->irq, (void *)mcpdm);
+}
+
+/* Enable/disable DC offset cancelation for the analog
+ * headset path (PDM channels 1 and 2).
+ */
+int omap_mcpdm_set_offset(int offset1, int offset2)
+{
+	int offset;
+
+	if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX))
+		return -EINVAL;
+
+	offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2);
+
+	/* offset cancellation for channel 1 */
+	if (offset1)
+		offset |= DN_OFST_RX1_EN;
+	else
+		offset &= ~DN_OFST_RX1_EN;
+
+	/* offset cancellation for channel 2 */
+	if (offset2)
+		offset |= DN_OFST_RX2_EN;
+	else
+		offset &= ~DN_OFST_RX2_EN;
+
+	omap_mcpdm_write(MCPDM_DN_OFFSET, offset);
+
+	return 0;
+}
+
+int __devinit omap_mcpdm_probe(struct platform_device *pdev)
+{
+	struct resource *res;
+	int ret = 0;
+
+	mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL);
+	if (!mcpdm) {
+		ret = -ENOMEM;
+		goto exit;
+	}
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (res == NULL) {
+		dev_err(&pdev->dev, "no resource\n");
+		goto err_resource;
+	}
+
+	spin_lock_init(&mcpdm->lock);
+	mcpdm->free = 1;
+	mcpdm->io_base = ioremap(res->start, resource_size(res));
+	if (!mcpdm->io_base) {
+		ret = -ENOMEM;
+		goto err_resource;
+	}
+
+	mcpdm->irq = platform_get_irq(pdev, 0);
+
+	mcpdm->clk = clk_get(&pdev->dev, "pdm_ck");
+	if (IS_ERR(mcpdm->clk)) {
+		ret = PTR_ERR(mcpdm->clk);
+		dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret);
+		goto err_clk;
+	}
+
+	mcpdm->dev = &pdev->dev;
+	platform_set_drvdata(pdev, mcpdm);
+
+	return 0;
+
+err_clk:
+	iounmap(mcpdm->io_base);
+err_resource:
+	kfree(mcpdm);
+exit:
+	return ret;
+}
+
+int __devexit omap_mcpdm_remove(struct platform_device *pdev)
+{
+	struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev);
+
+	platform_set_drvdata(pdev, NULL);
+
+	clk_put(mcpdm_ptr->clk);
+
+	iounmap(mcpdm_ptr->io_base);
+
+	mcpdm_ptr->clk = NULL;
+	mcpdm_ptr->free = 0;
+	mcpdm_ptr->dev = NULL;
+
+	kfree(mcpdm_ptr);
+
+	return 0;
+}
+
diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h
new file mode 100644
index 00000000..df3e16fb
--- /dev/null
+++ b/sound/soc/omap/mcpdm.h
@@ -0,0 +1,153 @@
+/*
+ * mcpdm.h -- Defines for McPDM driver
+ *
+ * Author: Jorge Eduardo Candelaria <x0107209@ti.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
+ *
+ */
+
+/* McPDM registers */
+
+#define MCPDM_REVISION         0x00
+#define MCPDM_SYSCONFIG                0x10
+#define MCPDM_IRQSTATUS_RAW    0x24
+#define MCPDM_IRQSTATUS                0x28
+#define MCPDM_IRQENABLE_SET    0x2C
+#define MCPDM_IRQENABLE_CLR    0x30
+#define MCPDM_IRQWAKE_EN       0x34
+#define MCPDM_DMAENABLE_SET    0x38
+#define MCPDM_DMAENABLE_CLR    0x3C
+#define MCPDM_DMAWAKEEN                0x40
+#define MCPDM_CTRL             0x44
+#define MCPDM_DN_DATA          0x48
+#define MCPDM_UP_DATA          0x4C
+#define MCPDM_FIFO_CTRL_DN     0x50
+#define MCPDM_FIFO_CTRL_UP     0x54
+#define MCPDM_DN_OFFSET                0x58
+
+/*
+ * MCPDM_IRQ bit fields
+ * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR
+ */
+
+#define MCPDM_DN_IRQ                   (1 << 0)
+#define MCPDM_DN_IRQ_EMPTY             (1 << 1)
+#define MCPDM_DN_IRQ_ALMST_EMPTY       (1 << 2)
+#define MCPDM_DN_IRQ_FULL              (1 << 3)
+
+#define MCPDM_UP_IRQ                   (1 << 8)
+#define MCPDM_UP_IRQ_EMPTY             (1 << 9)
+#define MCPDM_UP_IRQ_ALMST_FULL                (1 << 10)
+#define MCPDM_UP_IRQ_FULL              (1 << 11)
+
+#define MCPDM_DOWNLINK_IRQ_MASK                0x00F
+#define MCPDM_UPLINK_IRQ_MASK          0xF00
+
+/*
+ * MCPDM_DMAENABLE bit fields
+ */
+
+#define DMA_DN_ENABLE          0x1
+#define DMA_UP_ENABLE          0x2
+
+/*
+ * MCPDM_CTRL bit fields
+ */
+
+#define PDM_UP1_EN             0x0001
+#define PDM_UP2_EN             0x0002
+#define PDM_UP3_EN             0x0004
+#define PDM_DN1_EN             0x0008
+#define PDM_DN2_EN             0x0010
+#define PDM_DN3_EN             0x0020
+#define PDM_DN4_EN             0x0040
+#define PDM_DN5_EN             0x0080
+#define PDMOUTFORMAT           0x0100
+#define CMD_INT                        0x0200
+#define STATUS_INT             0x0400
+#define SW_UP_RST              0x0800
+#define SW_DN_RST              0x1000
+#define PDM_UP_MASK            0x007
+#define PDM_DN_MASK            0x0F8
+#define PDM_CMD_MASK           0x200
+#define PDM_STATUS_MASK                0x400
+
+
+#define PDMOUTFORMAT_LJUST     (0 << 8)
+#define PDMOUTFORMAT_RJUST     (1 << 8)
+
+/*
+ * MCPDM_FIFO_CTRL bit fields
+ */
+
+#define UP_THRES_MAX           0xF
+#define DN_THRES_MAX           0xF
+
+/*
+ * MCPDM_DN_OFFSET bit fields
+ */
+
+#define DN_OFST_RX1_EN         0x0001
+#define DN_OFST_RX2_EN         0x0100
+
+#define DN_OFST_RX1            1
+#define DN_OFST_RX2            9
+#define DN_OFST_MAX            0x1F
+
+#define MCPDM_UPLINK           1
+#define MCPDM_DOWNLINK         2
+
+struct omap_mcpdm_link {
+       int irq_mask;
+       int threshold;
+       int format;
+       int channels;
+};
+
+struct omap_mcpdm_platform_data {
+       unsigned long phys_base;
+       u16 irq;
+};
+
+struct omap_mcpdm {
+       struct device *dev;
+       unsigned long phys_base;
+       void __iomem *io_base;
+       u8 free;
+       int irq;
+
+       spinlock_t lock;
+       struct omap_mcpdm_platform_data *pdata;
+       struct clk *clk;
+       struct omap_mcpdm_link *downlink;
+       struct omap_mcpdm_link *uplink;
+       struct completion irq_completion;
+
+       int dn_channels;
+       int up_channels;
+};
+
+extern void omap_mcpdm_start(int stream);
+extern void omap_mcpdm_stop(int stream);
+extern int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink);
+extern int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink);
+extern int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink);
+extern int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink);
+extern int omap_mcpdm_request(void);
+extern void omap_mcpdm_free(void);
+extern int omap_mcpdm_set_offset(int offset1, int offset2);
+int __devinit omap_mcpdm_probe(struct platform_device *pdev);
+int __devexit omap_mcpdm_remove(struct platform_device *pdev);
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
new file mode 100644
index 00000000..83d213bf
--- /dev/null
+++ b/sound/soc/omap/n810.c
@@ -0,0 +1,407 @@
+/*
+ * n810.c  --  SoC audio for Nokia N810
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jhnikula@gmail.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/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+#include <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+#define N810_HEADSET_AMP_GPIO	10
+#define N810_SPEAKER_AMP_GPIO	101
+
+enum {
+	N810_JACK_DISABLED,
+	N810_JACK_HP,
+	N810_JACK_HS,
+	N810_JACK_MIC,
+};
+
+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 int n810_dmic_func;
+
+static void n810_ext_control(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int hp = 0, line1l = 0;
+
+	switch (n810_jack_func) {
+	case N810_JACK_HS:
+		line1l = 1;
+	case N810_JACK_HP:
+		hp = 1;
+		break;
+	case N810_JACK_MIC:
+		line1l = 1;
+		break;
+	}
+
+	if (n810_spk_func)
+		snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+
+	if (hp)
+		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+	if (line1l)
+		snd_soc_dapm_enable_pin(dapm, "LINE1L");
+	else
+		snd_soc_dapm_disable_pin(dapm, "LINE1L");
+
+	if (n810_dmic_func)
+		snd_soc_dapm_enable_pin(dapm, "DMic");
+	else
+		snd_soc_dapm_disable_pin(dapm, "DMic");
+
+	snd_soc_dapm_sync(dapm);
+}
+
+static int n810_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+
+	snd_pcm_hw_constraint_minmax(runtime,
+				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+
+	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_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int err;
+
+	/* Set codec DAI configuration */
+	err = snd_soc_dai_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 = snd_soc_dai_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 = snd_soc_dai_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_get_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = n810_dmic_func;
+
+	return 0;
+}
+
+static int n810_set_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (n810_dmic_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	n810_dmic_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))
+		gpio_set_value(N810_SPEAKER_AMP_GPIO, 1);
+	else
+		gpio_set_value(N810_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))
+		gpio_set_value(N810_HEADSET_AMP_GPIO, 1);
+	else
+		gpio_set_value(N810_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),
+	SND_SOC_DAPM_MIC("DMic", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Jack", NULL, "HPLOUT"},
+	{"Headphone Jack", NULL, "HPROUT"},
+
+	{"Ext Spk", NULL, "LLOUT"},
+	{"Ext Spk", NULL, "RLOUT"},
+
+	{"DMic Rate 64", NULL, "Mic Bias 2V"},
+	{"Mic Bias 2V", NULL, "DMic"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *jack_function[] = {"Off", "Headphone", "Headset", "Mic"};
+static const char *input_function[] = {"ADC", "Digital Mic"};
+static const struct soc_enum n810_enum[] = {
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_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),
+	SOC_ENUM_EXT("Input Select",  n810_enum[2],
+		     n810_get_input, n810_set_input),
+};
+
+static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int err;
+
+	/* Not connected */
+	snd_soc_dapm_nc_pin(dapm, "MONO_LOUT");
+	snd_soc_dapm_nc_pin(dapm, "HPLCOM");
+	snd_soc_dapm_nc_pin(dapm, "HPRCOM");
+	snd_soc_dapm_nc_pin(dapm, "MIC3L");
+	snd_soc_dapm_nc_pin(dapm, "MIC3R");
+	snd_soc_dapm_nc_pin(dapm, "LINE1R");
+	snd_soc_dapm_nc_pin(dapm, "LINE2L");
+	snd_soc_dapm_nc_pin(dapm, "LINE2R");
+
+	/* Add N810 specific controls */
+	err = snd_soc_add_controls(codec, aic33_n810_controls,
+				ARRAY_SIZE(aic33_n810_controls));
+	if (err < 0)
+		return err;
+
+	/* Add N810 specific widgets */
+	snd_soc_dapm_new_controls(dapm, aic33_dapm_widgets,
+				  ARRAY_SIZE(aic33_dapm_widgets));
+
+	/* Set up N810 specific audio path audio_map */
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(dapm);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link n810_dai = {
+	.name = "TLV320AIC33",
+	.stream_name = "AIC33",
+	.cpu_dai_name = "omap-mcbsp-dai.1",
+	.platform_name = "omap-pcm-audio",
+	.codec_name = "tlv320aic3x-codec.2-0018",
+	.codec_dai_name = "tlv320aic3x-hifi",
+	.init = n810_aic33_init,
+	.ops = &n810_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_n810 = {
+	.name = "N810",
+	.dai_link = &n810_dai,
+	.num_links = 1,
+};
+
+static struct platform_device *n810_snd_device;
+
+static int __init n810_soc_init(void)
+{
+	int err;
+	struct device *dev;
+
+	if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
+		return -ENODEV;
+
+	n810_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!n810_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(n810_snd_device, &snd_soc_n810);
+	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");
+		err = PTR_ERR(sys_clkout2_src);
+		goto err2;
+	}
+	sys_clkout2 = clk_get(dev, "sys_clkout2");
+	if (IS_ERR(sys_clkout2)) {
+		dev_err(dev, "Could not get sys_clkout2\n");
+		err = PTR_ERR(sys_clkout2);
+		goto err3;
+	}
+	/*
+	 * 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");
+		err = PTR_ERR(func96m_clk);
+		goto err4;
+	}
+	clk_set_parent(sys_clkout2_src, func96m_clk);
+	clk_set_rate(sys_clkout2, 12000000);
+
+	BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) ||
+	       (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0));
+
+	gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
+	gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
+
+	return 0;
+err4:
+	clk_put(sys_clkout2);
+err3:
+	clk_put(sys_clkout2_src);
+err2:
+	platform_device_del(n810_snd_device);
+err1:
+	platform_device_put(n810_snd_device);
+
+	return err;
+}
+
+static void __exit n810_soc_exit(void)
+{
+	gpio_free(N810_SPEAKER_AMP_GPIO);
+	gpio_free(N810_HEADSET_AMP_GPIO);
+	clk_put(sys_clkout2_src);
+	clk_put(sys_clkout2);
+	clk_put(func96m_clk);
+
+	platform_device_unregister(n810_snd_device);
+}
+
+module_init(n810_soc_init);
+module_exit(n810_soc_exit);
+
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.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 00000000..4b82290c
--- /dev/null
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -0,0 +1,791 @@
+/*
+ * omap-mcbsp.c  --  OMAP ALSA SoC DAI driver using McBSP port
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ *          Peter Ujfalusi <peter.ujfalusi@ti.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 <plat/dma.h>
+#include <plat/mcbsp.h>
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+#define OMAP_MCBSP_RATES	(SNDRV_PCM_RATE_8000_96000)
+
+#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \
+	xhandler_get, xhandler_put) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = omap_mcbsp_st_info_volsw, \
+	.get = xhandler_get, .put = xhandler_put, \
+	.private_value = (unsigned long) &(struct soc_mixer_control) \
+	{.min = xmin, .max = xmax} }
+
+struct omap_mcbsp_data {
+	unsigned int			bus_id;
+	struct omap_mcbsp_reg_cfg	regs;
+	unsigned int			fmt;
+	/*
+	 * Flags indicating is the bus already activated and configured by
+	 * another substream
+	 */
+	int				active;
+	int				configured;
+	unsigned int			in_freq;
+	int				clk_div;
+	int				wlen;
+};
+
+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];
+
+static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai);
+	struct omap_pcm_dma_data *dma_data;
+	int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id);
+	int words;
+
+	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+	/* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
+	if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
+		/*
+		 * Configure McBSP threshold based on either:
+		 * packet_size, when the sDMA is in packet mode, or
+		 * based on the period size.
+		 */
+		if (dma_data->packet_size)
+			words = dma_data->packet_size;
+		else
+			words = snd_pcm_lib_period_bytes(substream) /
+							(mcbsp_data->wlen / 8);
+	else
+		words = 1;
+
+	/* Configure McBSP internal buffer usage */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, words);
+	else
+		omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, words);
+}
+
+static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params,
+				    struct snd_pcm_hw_rule *rule)
+{
+	struct snd_interval *buffer_size = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_BUFFER_SIZE);
+	struct snd_interval *channels = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct omap_mcbsp_data *mcbsp_data = rule->private;
+	struct snd_interval frames;
+	int size;
+
+	snd_interval_any(&frames);
+	size = omap_mcbsp_get_fifo_size(mcbsp_data->bus_id);
+
+	frames.min = size / channels->min;
+	frames.integer = 1;
+	return snd_interval_refine(buffer_size, &frames);
+}
+
+static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *cpu_dai)
+{
+	struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai);
+	int bus_id = mcbsp_data->bus_id;
+	int err = 0;
+
+	if (!cpu_dai->active)
+		err = omap_mcbsp_request(bus_id);
+
+	/*
+	 * OMAP3 McBSP FIFO is word structured.
+	 * McBSP2 has 1024 + 256 = 1280 word long buffer,
+	 * McBSP1,3,4,5 has 128 word long buffer
+	 * This means that the size of the FIFO depends on the sample format.
+	 * For example on McBSP3:
+	 * 16bit samples: size is 128 * 2 = 256 bytes
+	 * 32bit samples: size is 128 * 4 = 512 bytes
+	 * It is simpler to place constraint for buffer and period based on
+	 * channels.
+	 * McBSP3 as example again (16 or 32 bit samples):
+	 * 1 channel (mono): size is 128 frames (128 words)
+	 * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
+	 * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
+	 */
+	if (cpu_is_omap34xx() || cpu_is_omap44xx()) {
+		/*
+		* Rule for the buffer size. We should not allow
+		* smaller buffer than the FIFO size to avoid underruns
+		*/
+		snd_pcm_hw_rule_add(substream->runtime, 0,
+				    SNDRV_PCM_HW_PARAM_CHANNELS,
+				    omap_mcbsp_hwrule_min_buffersize,
+				    mcbsp_data,
+				    SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1);
+
+		/* Make sure, that the period size is always even */
+		snd_pcm_hw_constraint_step(substream->runtime, 0,
+					   SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2);
+	}
+
+	return err;
+}
+
+static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *cpu_dai)
+{
+	struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai);
+
+	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_dai *cpu_dai)
+{
+	struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai);
+	int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		mcbsp_data->active++;
+		omap_mcbsp_start(mcbsp_data->bus_id, play, !play);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		omap_mcbsp_stop(mcbsp_data->bus_id, play, !play);
+		mcbsp_data->active--;
+		break;
+	default:
+		err = -EINVAL;
+	}
+
+	return err;
+}
+
+static snd_pcm_sframes_t omap_mcbsp_dai_delay(
+			struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai);
+	u16 fifo_use;
+	snd_pcm_sframes_t delay;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		fifo_use = omap_mcbsp_get_tx_delay(mcbsp_data->bus_id);
+	else
+		fifo_use = omap_mcbsp_get_rx_delay(mcbsp_data->bus_id);
+
+	/*
+	 * Divide the used locations with the channel count to get the
+	 * FIFO usage in samples (don't care about partial samples in the
+	 * buffer).
+	 */
+	delay = fifo_use / substream->runtime->channels;
+
+	return delay;
+}
+
+static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_dai *cpu_dai)
+{
+	struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai);
+	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+	struct omap_pcm_dma_data *dma_data;
+	int dma, bus_id = mcbsp_data->bus_id;
+	int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT;
+	int pkt_size = 0;
+	unsigned long port;
+	unsigned int format, div, framesize, master;
+
+	dma_data = &omap_mcbsp_dai_dma_params[cpu_dai->id][substream->stream];
+
+	dma = omap_mcbsp_dma_ch_params(bus_id, substream->stream);
+	port = omap_mcbsp_dma_reg_params(bus_id, substream->stream);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		dma_data->data_type = OMAP_DMA_DATA_TYPE_S16;
+		wlen = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		dma_data->data_type = OMAP_DMA_DATA_TYPE_S32;
+		wlen = 32;
+		break;
+	default:
+		return -EINVAL;
+	}
+	if (cpu_is_omap34xx()) {
+		dma_data->set_threshold = omap_mcbsp_set_threshold;
+		/* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
+		if (omap_mcbsp_get_dma_op_mode(bus_id) ==
+						MCBSP_DMA_MODE_THRESHOLD) {
+			int period_words, max_thrsh;
+
+			period_words = params_period_bytes(params) / (wlen / 8);
+			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+				max_thrsh = omap_mcbsp_get_max_tx_threshold(
+							    mcbsp_data->bus_id);
+			else
+				max_thrsh = omap_mcbsp_get_max_rx_threshold(
+							    mcbsp_data->bus_id);
+			/*
+			 * If the period contains less or equal number of words,
+			 * we are using the original threshold mode setup:
+			 * McBSP threshold = sDMA frame size = period_size
+			 * Otherwise we switch to sDMA packet mode:
+			 * McBSP threshold = sDMA packet size
+			 * sDMA frame size = period size
+			 */
+			if (period_words > max_thrsh) {
+				int divider = 0;
+
+				/*
+				 * Look for the biggest threshold value, which
+				 * divides the period size evenly.
+				 */
+				divider = period_words / max_thrsh;
+				if (period_words % max_thrsh)
+					divider++;
+				while (period_words % divider &&
+					divider < period_words)
+					divider++;
+				if (divider == period_words)
+					return -EINVAL;
+
+				pkt_size = period_words / divider;
+				sync_mode = OMAP_DMA_SYNC_PACKET;
+			} else {
+				sync_mode = OMAP_DMA_SYNC_FRAME;
+			}
+		}
+	}
+
+	dma_data->name = substream->stream ? "Audio Capture" : "Audio Playback";
+	dma_data->dma_req = dma;
+	dma_data->port_addr = port;
+	dma_data->sync_mode = sync_mode;
+	dma_data->packet_size = pkt_size;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
+
+	if (mcbsp_data->configured) {
+		/* McBSP already configured by another stream */
+		return 0;
+	}
+
+	format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+	wpf = channels = params_channels(params);
+	if (channels == 2 && (format == SND_SOC_DAIFMT_I2S ||
+			      format == SND_SOC_DAIFMT_LEFT_J)) {
+		/* Use dual-phase frames */
+		regs->rcr2	|= RPHASE;
+		regs->xcr2	|= XPHASE;
+		/* Set 1 word per (McBSP) frame for phase1 and phase2 */
+		wpf--;
+		regs->rcr2	|= RFRLEN2(wpf - 1);
+		regs->xcr2	|= XFRLEN2(wpf - 1);
+	}
+
+	regs->rcr1	|= RFRLEN1(wpf - 1);
+	regs->xcr1	|= XFRLEN1(wpf - 1);
+
+	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);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		/* Set word lengths */
+		regs->rcr2	|= RWDLEN2(OMAP_MCBSP_WORD_32);
+		regs->rcr1	|= RWDLEN1(OMAP_MCBSP_WORD_32);
+		regs->xcr2	|= XWDLEN2(OMAP_MCBSP_WORD_32);
+		regs->xcr1	|= XWDLEN1(OMAP_MCBSP_WORD_32);
+		break;
+	default:
+		/* Unsupported PCM format */
+		return -EINVAL;
+	}
+
+	/* In McBSP master modes, FRAME (i.e. sample rate) is generated
+	 * by _counting_ BCLKs. Calculate frame size in BCLKs */
+	master = mcbsp_data->fmt & SND_SOC_DAIFMT_MASTER_MASK;
+	if (master ==	SND_SOC_DAIFMT_CBS_CFS) {
+		div = mcbsp_data->clk_div ? mcbsp_data->clk_div : 1;
+		framesize = (mcbsp_data->in_freq / div) / params_rate(params);
+
+		if (framesize < wlen * channels) {
+			printk(KERN_ERR "%s: not enough bandwidth for desired rate and "
+					"channels\n", __func__);
+			return -EINVAL;
+		}
+	} else
+		framesize = wlen * channels;
+
+	/* Set FS period and length in terms of bit clock periods */
+	switch (format) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_LEFT_J:
+		regs->srgr2	|= FPER(framesize - 1);
+		regs->srgr1	|= FWID((framesize >> 1) - 1);
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+	case SND_SOC_DAIFMT_DSP_B:
+		regs->srgr2	|= FPER(framesize - 1);
+		regs->srgr1	|= FWID(0);
+		break;
+	}
+
+	omap_mcbsp_config(bus_id, &mcbsp_data->regs);
+	mcbsp_data->wlen = wlen;
+	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_dai *cpu_dai,
+				      unsigned int fmt)
+{
+	struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai);
+	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+	unsigned int temp_fmt = fmt;
+
+	if (mcbsp_data->configured)
+		return 0;
+
+	mcbsp_data->fmt = fmt;
+	memset(regs, 0, sizeof(*regs));
+	/* Generic McBSP register settings */
+	regs->spcr2	|= XINTM(3) | FREE;
+	regs->spcr1	|= RINTM(3);
+	/* RFIG and XFIG are not defined in 34xx */
+	if (!cpu_is_omap34xx() && !cpu_is_omap44xx()) {
+		regs->rcr2	|= RFIG;
+		regs->xcr2	|= XFIG;
+	}
+	if (cpu_is_omap2430() || cpu_is_omap34xx() || cpu_is_omap44xx()) {
+		regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE;
+		regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE;
+	}
+
+	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;
+	case SND_SOC_DAIFMT_LEFT_J:
+		/* 0-bit data delay */
+		regs->rcr2	|= RDATDLY(0);
+		regs->xcr2	|= XDATDLY(0);
+		regs->spcr1	|= RJUST(2);
+		/* Invert FS polarity configuration */
+		temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		/* 1-bit data delay */
+		regs->rcr2      |= RDATDLY(1);
+		regs->xcr2      |= XDATDLY(1);
+		/* Invert FS polarity configuration */
+		temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		/* 0-bit data delay */
+		regs->rcr2      |= RDATDLY(0);
+		regs->xcr2      |= XDATDLY(0);
+		/* Invert FS polarity configuration */
+		temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
+		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 (temp_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_dai *cpu_dai,
+				     int div_id, int div)
+{
+	struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai);
+	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+
+	if (div_id != OMAP_MCBSP_CLKGDV)
+		return -ENODEV;
+
+	mcbsp_data->clk_div = div;
+	regs->srgr1	|= CLKGDV(div - 1);
+
+	return 0;
+}
+
+static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+					 int clk_id, unsigned int freq,
+					 int dir)
+{
+	struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai);
+	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+	int err = 0;
+
+	if (mcbsp_data->active)
+		if (freq == mcbsp_data->in_freq)
+			return 0;
+		else
+			return -EBUSY;
+
+	/* The McBSP signal muxing functions are only available on McBSP1 */
+	if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR ||
+	    clk_id == OMAP_MCBSP_CLKR_SRC_CLKX ||
+	    clk_id == OMAP_MCBSP_FSR_SRC_FSR ||
+	    clk_id == OMAP_MCBSP_FSR_SRC_FSX)
+		if (cpu_class_is_omap1() || mcbsp_data->bus_id != 0)
+			return -EINVAL;
+
+	mcbsp_data->in_freq = freq;
+
+	switch (clk_id) {
+	case OMAP_MCBSP_SYSCLK_CLK:
+		regs->srgr2	|= CLKSM;
+		break;
+	case OMAP_MCBSP_SYSCLK_CLKS_FCLK:
+		if (cpu_class_is_omap1()) {
+			err = -EINVAL;
+			break;
+		}
+		err = omap2_mcbsp_set_clks_src(mcbsp_data->bus_id,
+					       MCBSP_CLKS_PRCM_SRC);
+		break;
+	case OMAP_MCBSP_SYSCLK_CLKS_EXT:
+		if (cpu_class_is_omap1()) {
+			err = 0;
+			break;
+		}
+		err = omap2_mcbsp_set_clks_src(mcbsp_data->bus_id,
+					       MCBSP_CLKS_PAD_SRC);
+		break;
+
+	case OMAP_MCBSP_SYSCLK_CLKX_EXT:
+		regs->srgr2	|= CLKSM;
+	case OMAP_MCBSP_SYSCLK_CLKR_EXT:
+		regs->pcr0	|= SCLKME;
+		break;
+
+
+	case OMAP_MCBSP_CLKR_SRC_CLKR:
+		if (cpu_class_is_omap1())
+			break;
+		omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKR);
+		break;
+	case OMAP_MCBSP_CLKR_SRC_CLKX:
+		if (cpu_class_is_omap1())
+			break;
+		omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKX);
+		break;
+	case OMAP_MCBSP_FSR_SRC_FSR:
+		if (cpu_class_is_omap1())
+			break;
+		omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSR);
+		break;
+	case OMAP_MCBSP_FSR_SRC_FSX:
+		if (cpu_class_is_omap1())
+			break;
+		omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSX);
+		break;
+	default:
+		err = -ENODEV;
+	}
+
+	return err;
+}
+
+static struct snd_soc_dai_ops mcbsp_dai_ops = {
+	.startup	= omap_mcbsp_dai_startup,
+	.shutdown	= omap_mcbsp_dai_shutdown,
+	.trigger	= omap_mcbsp_dai_trigger,
+	.delay		= omap_mcbsp_dai_delay,
+	.hw_params	= omap_mcbsp_dai_hw_params,
+	.set_fmt	= omap_mcbsp_dai_set_dai_fmt,
+	.set_clkdiv	= omap_mcbsp_dai_set_clkdiv,
+	.set_sysclk	= omap_mcbsp_dai_set_dai_sysclk,
+};
+
+static int mcbsp_dai_probe(struct snd_soc_dai *dai)
+{
+	mcbsp_data[dai->id].bus_id = dai->id;
+	snd_soc_dai_set_drvdata(dai, &mcbsp_data[dai->id].bus_id);
+	return 0;
+}
+
+static struct snd_soc_dai_driver omap_mcbsp_dai =
+{
+	.probe = mcbsp_dai_probe,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 16,
+		.rates = OMAP_MCBSP_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 16,
+		.rates = OMAP_MCBSP_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+	},
+	.ops = &mcbsp_dai_ops,
+};
+
+static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_info *uinfo)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int max = mc->max;
+	int min = mc->min;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = min;
+	uinfo->value.integer.max = max;
+	return 0;
+}
+
+#define OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(id, channel)			\
+static int								\
+omap_mcbsp##id##_set_st_ch##channel##_volume(struct snd_kcontrol *kc,	\
+					struct snd_ctl_elem_value *uc)	\
+{									\
+	struct soc_mixer_control *mc =					\
+		(struct soc_mixer_control *)kc->private_value;		\
+	int max = mc->max;						\
+	int min = mc->min;						\
+	int val = uc->value.integer.value[0];				\
+									\
+	if (val < min || val > max)					\
+		return -EINVAL;						\
+									\
+	/* OMAP McBSP implementation uses index values 0..4 */		\
+	return omap_st_set_chgain((id)-1, channel, val);		\
+}
+
+#define OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(id, channel)			\
+static int								\
+omap_mcbsp##id##_get_st_ch##channel##_volume(struct snd_kcontrol *kc,	\
+					struct snd_ctl_elem_value *uc)	\
+{									\
+	s16 chgain;							\
+									\
+	if (omap_st_get_chgain((id)-1, channel, &chgain))		\
+		return -EAGAIN;						\
+									\
+	uc->value.integer.value[0] = chgain;				\
+	return 0;							\
+}
+
+OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 0)
+OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 1)
+OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 0)
+OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 1)
+OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 0)
+OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 1)
+OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 0)
+OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 1)
+
+static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	u8 value = ucontrol->value.integer.value[0];
+
+	if (value == omap_st_is_enabled(mc->reg))
+		return 0;
+
+	if (value)
+		omap_st_enable(mc->reg);
+	else
+		omap_st_disable(mc->reg);
+
+	return 1;
+}
+
+static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	ucontrol->value.integer.value[0] = omap_st_is_enabled(mc->reg);
+	return 0;
+}
+
+static const struct snd_kcontrol_new omap_mcbsp2_st_controls[] = {
+	SOC_SINGLE_EXT("McBSP2 Sidetone Switch", 1, 0, 1, 0,
+			omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode),
+	OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 0 Volume",
+				      -32768, 32767,
+				      omap_mcbsp2_get_st_ch0_volume,
+				      omap_mcbsp2_set_st_ch0_volume),
+	OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 1 Volume",
+				      -32768, 32767,
+				      omap_mcbsp2_get_st_ch1_volume,
+				      omap_mcbsp2_set_st_ch1_volume),
+};
+
+static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = {
+	SOC_SINGLE_EXT("McBSP3 Sidetone Switch", 2, 0, 1, 0,
+			omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode),
+	OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 0 Volume",
+				      -32768, 32767,
+				      omap_mcbsp3_get_st_ch0_volume,
+				      omap_mcbsp3_set_st_ch0_volume),
+	OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 1 Volume",
+				      -32768, 32767,
+				      omap_mcbsp3_get_st_ch1_volume,
+				      omap_mcbsp3_set_st_ch1_volume),
+};
+
+int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id)
+{
+	if (!cpu_is_omap34xx())
+		return -ENODEV;
+
+	switch (mcbsp_id) {
+	case 1: /* McBSP 2 */
+		return snd_soc_add_controls(codec, omap_mcbsp2_st_controls,
+					ARRAY_SIZE(omap_mcbsp2_st_controls));
+	case 2: /* McBSP 3 */
+		return snd_soc_add_controls(codec, omap_mcbsp3_st_controls,
+					ARRAY_SIZE(omap_mcbsp3_st_controls));
+	default:
+		break;
+	}
+
+	return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls);
+
+static __devinit int asoc_mcbsp_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_dai(&pdev->dev, &omap_mcbsp_dai);
+}
+
+static int __devexit asoc_mcbsp_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_dai(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver asoc_mcbsp_driver = {
+	.driver = {
+			.name = "omap-mcbsp-dai",
+			.owner = THIS_MODULE,
+	},
+
+	.probe = asoc_mcbsp_probe,
+	.remove = __devexit_p(asoc_mcbsp_remove),
+};
+
+static int __init snd_omap_mcbsp_init(void)
+{
+	return platform_driver_register(&asoc_mcbsp_driver);
+}
+module_init(snd_omap_mcbsp_init);
+
+static void __exit snd_omap_mcbsp_exit(void)
+{
+	platform_driver_unregister(&asoc_mcbsp_driver);
+}
+module_exit(snd_omap_mcbsp_exit);
+
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.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 00000000..9a7dedd6
--- /dev/null
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -0,0 +1,64 @@
+/*
+ * omap-mcbsp.h
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ *          Peter Ujfalusi <peter.ujfalusi@ti.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 */
+	OMAP_MCBSP_CLKR_SRC_CLKR,	/* CLKR from CLKR pin */
+	OMAP_MCBSP_CLKR_SRC_CLKX,	/* CLKR from CLKX pin */
+	OMAP_MCBSP_FSR_SRC_FSR,		/* FSR from FSR pin */
+	OMAP_MCBSP_FSR_SRC_FSX,		/* FSR from FSX pin */
+};
+
+/* McBSP dividers */
+enum omap_mcbsp_div {
+	OMAP_MCBSP_CLKGDV,		/* Sample rate generator divider */
+};
+
+#if defined(CONFIG_SOC_OMAP2420)
+#define NUM_LINKS	2
+#endif
+#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
+#undef  NUM_LINKS
+#define NUM_LINKS	3
+#endif
+#if defined(CONFIG_ARCH_OMAP4)
+#undef  NUM_LINKS
+#define NUM_LINKS	4
+#endif
+#if defined(CONFIG_ARCH_OMAP3) || defined(CONFIG_SOC_OMAP2430)
+#undef  NUM_LINKS
+#define NUM_LINKS	5
+#endif
+
+int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id);
+
+#endif
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
new file mode 100644
index 00000000..bed09c27
--- /dev/null
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -0,0 +1,272 @@
+/*
+ * omap-mcpdm.c  --  OMAP ALSA SoC DAI driver using McPDM port
+ *
+ * Copyright (C) 2009 Texas Instruments
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.com>
+ * Contact: Jorge Eduardo Candelaria <x0107209@ti.com>
+ *          Margarita Olaya <magi.olaya@ti.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 <plat/dma.h>
+#include <plat/mcbsp.h>
+#include "mcpdm.h"
+#include "omap-pcm.h"
+
+struct omap_mcpdm_data {
+	struct omap_mcpdm_link *links;
+	int active;
+};
+
+static struct omap_mcpdm_link omap_mcpdm_links[] = {
+	/* downlink */
+	{
+		.irq_mask = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL,
+		.threshold = 1,
+		.format = PDMOUTFORMAT_LJUST,
+	},
+	/* uplink */
+	{
+		.irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL,
+		.threshold = 1,
+		.format = PDMOUTFORMAT_LJUST,
+	},
+};
+
+static struct omap_mcpdm_data mcpdm_data = {
+	.links = omap_mcpdm_links,
+	.active = 0,
+};
+
+/*
+ * Stream DMA parameters
+ */
+static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = {
+	{
+		.name = "Audio playback",
+		.dma_req = OMAP44XX_DMA_MCPDM_DL,
+		.data_type = OMAP_DMA_DATA_TYPE_S32,
+		.sync_mode = OMAP_DMA_SYNC_PACKET,
+		.packet_size = 16,
+		.port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA,
+	},
+	{
+		.name = "Audio capture",
+		.dma_req = OMAP44XX_DMA_MCPDM_UP,
+		.data_type = OMAP_DMA_DATA_TYPE_S32,
+		.sync_mode = OMAP_DMA_SYNC_PACKET,
+		.packet_size = 16,
+		.port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_UP_DATA,
+	},
+};
+
+static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	int err = 0;
+
+	if (!dai->active)
+		err = omap_mcpdm_request();
+
+	return err;
+}
+
+static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *dai)
+{
+	if (!dai->active)
+		omap_mcpdm_free();
+}
+
+static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+				  struct snd_soc_dai *dai)
+{
+	struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai);
+	int stream = substream->stream;
+	int err = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (!mcpdm_priv->active++)
+			omap_mcpdm_start(stream);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (!--mcpdm_priv->active)
+			omap_mcpdm_stop(stream);
+		break;
+	default:
+		err = -EINVAL;
+	}
+
+	return err;
+}
+
+static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_dai *dai)
+{
+	struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai);
+	struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links;
+	int stream = substream->stream;
+	int channels, err, link_mask = 0;
+
+	snd_soc_dai_set_dma_data(dai, substream,
+				 &omap_mcpdm_dai_dma_params[stream]);
+
+	channels = params_channels(params);
+	switch (channels) {
+	case 4:
+		if (stream == SNDRV_PCM_STREAM_CAPTURE)
+			/* up to 2 channels for capture */
+			return -EINVAL;
+		link_mask |= 1 << 3;
+	case 3:
+		if (stream == SNDRV_PCM_STREAM_CAPTURE)
+			/* up to 2 channels for capture */
+			return -EINVAL;
+		link_mask |= 1 << 2;
+	case 2:
+		link_mask |= 1 << 1;
+	case 1:
+		link_mask |= 1 << 0;
+		break;
+	default:
+		/* unsupported number of channels */
+		return -EINVAL;
+	}
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		mcpdm_links[stream].channels = link_mask << 3;
+		err = omap_mcpdm_playback_open(&mcpdm_links[stream]);
+	} else {
+		mcpdm_links[stream].channels = link_mask << 0;
+		err = omap_mcpdm_capture_open(&mcpdm_links[stream]);
+	}
+
+	return err;
+}
+
+static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct omap_mcpdm_data *mcpdm_priv = snd_soc_dai_get_drvdata(dai);
+	struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links;
+	int stream = substream->stream;
+	int err;
+
+	if (substream->stream ==  SNDRV_PCM_STREAM_PLAYBACK)
+		err = omap_mcpdm_playback_close(&mcpdm_links[stream]);
+	else
+		err = omap_mcpdm_capture_close(&mcpdm_links[stream]);
+
+	return err;
+}
+
+static struct snd_soc_dai_ops omap_mcpdm_dai_ops = {
+	.startup	= omap_mcpdm_dai_startup,
+	.shutdown	= omap_mcpdm_dai_shutdown,
+	.trigger	= omap_mcpdm_dai_trigger,
+	.hw_params	= omap_mcpdm_dai_hw_params,
+	.hw_free	= omap_mcpdm_dai_hw_free,
+};
+
+#define OMAP_MCPDM_RATES	(SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+#define OMAP_MCPDM_FORMATS	(SNDRV_PCM_FMTBIT_S32_LE)
+
+static int omap_mcpdm_dai_probe(struct snd_soc_dai *dai)
+{
+	snd_soc_dai_set_drvdata(dai, &mcpdm_data);
+	return 0;
+}
+
+static struct snd_soc_dai_driver omap_mcpdm_dai = {
+	.probe = omap_mcpdm_dai_probe,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 4,
+		.rates = OMAP_MCPDM_RATES,
+		.formats = OMAP_MCPDM_FORMATS,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = OMAP_MCPDM_RATES,
+		.formats = OMAP_MCPDM_FORMATS,
+	},
+	.ops = &omap_mcpdm_dai_ops,
+};
+
+static __devinit int asoc_mcpdm_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	ret = omap_mcpdm_probe(pdev);
+	if (ret < 0)
+		return ret;
+	ret = snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai);
+	if (ret < 0)
+		omap_mcpdm_remove(pdev);
+	return ret;
+}
+
+static int __devexit asoc_mcpdm_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_dai(&pdev->dev);
+	omap_mcpdm_remove(pdev);
+	return 0;
+}
+
+static struct platform_driver asoc_mcpdm_driver = {
+	.driver = {
+			.name = "omap-mcpdm-dai",
+			.owner = THIS_MODULE,
+	},
+
+	.probe = asoc_mcpdm_probe,
+	.remove = __devexit_p(asoc_mcpdm_remove),
+};
+
+static int __init snd_omap_mcpdm_init(void)
+{
+	return platform_driver_register(&asoc_mcpdm_driver);
+}
+module_init(snd_omap_mcpdm_init);
+
+static void __exit snd_omap_mcpdm_exit(void)
+{
+	platform_driver_unregister(&asoc_mcpdm_driver);
+}
+module_exit(snd_omap_mcpdm_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("OMAP PDM SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
new file mode 100644
index 00000000..e6a6b991
--- /dev/null
+++ b/sound/soc/omap/omap-pcm.c
@@ -0,0 +1,439 @@
+/*
+ * omap-pcm.c  --  ALSA PCM interface for the OMAP SoC
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ *          Peter Ujfalusi <peter.ujfalusi@ti.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 <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <plat/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 |
+				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
+	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
+				  SNDRV_PCM_FMTBIT_S32_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 fully support DMA progress counter
+		 * and there is no software emulation implemented yet,
+		 * so have to maintain our own progress counters
+		 * that can be used by omap_pcm_pointer() instead.
+		 */
+		spin_lock_irqsave(&prtd->lock, flags);
+		if ((stat == OMAP_DMA_LAST_IRQ) &&
+				(prtd->period_index == runtime->periods - 1)) {
+			/* we are in sync, do nothing */
+			spin_unlock_irqrestore(&prtd->lock, flags);
+			return;
+		}
+		if (prtd->period_index >= 0) {
+			if (stat & OMAP_DMA_BLOCK_IRQ) {
+				/* end of buffer reached, loop back */
+				prtd->period_index = 0;
+			} else if (stat & OMAP_DMA_LAST_IRQ) {
+				/* update the counter for the last period */
+				prtd->period_index = runtime->periods - 1;
+			} else if (++prtd->period_index >= runtime->periods) {
+				/* end of buffer missed? loop back */
+				prtd->period_index = 0;
+			}
+		}
+		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;
+
+	int err = 0;
+
+	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+	/* return if this is a bufferless transfer e.g.
+	 * codec <--> BT codec or GSM modem -- lg FIXME */
+	if (!dma_data)
+		return 0;
+
+	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 (!err) {
+		/*
+		 * 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;
+
+	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;
+	int bytes;
+
+	/* return if this is a bufferless transfer e.g.
+	 * codec <--> BT codec or GSM modem -- lg FIXME */
+	if (!prtd->dma_data)
+		return 0;
+
+	memset(&dma_params, 0, sizeof(dma_params));
+	dma_params.data_type			= dma_data->data_type;
+	dma_params.trigger			= dma_data->dma_req;
+	dma_params.sync_mode			= dma_data->sync_mode;
+	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;
+		dma_params.dst_port		= OMAP_DMA_PORT_MPUI;
+		dma_params.dst_fi		= dma_data->packet_size;
+	} 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;
+		dma_params.src_port		= OMAP_DMA_PORT_MPUI;
+		dma_params.src_fi		= dma_data->packet_size;
+	}
+	/*
+	 * 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
+	 */
+	bytes = snd_pcm_lib_period_bytes(substream);
+	dma_params.elem_count	= bytes >> dma_data->data_type;
+	dma_params.frame_count	= runtime->periods;
+	omap_set_dma_params(prtd->dma_ch, &dma_params);
+
+	if ((cpu_is_omap1510()))
+		omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ |
+			      OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ);
+	else if (!substream->runtime->no_period_wakeup)
+		omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
+
+	if (!(cpu_class_is_omap1())) {
+		omap_set_dma_src_burst_mode(prtd->dma_ch,
+						OMAP_DMA_DATA_BURST_16);
+		omap_set_dma_dest_burst_mode(prtd->dma_ch,
+						OMAP_DMA_DATA_BURST_16);
+	}
+
+	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;
+	struct omap_pcm_dma_data *dma_data = prtd->dma_data;
+	unsigned long flags;
+	int ret = 0;
+
+	spin_lock_irqsave(&prtd->lock, flags);
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		prtd->period_index = 0;
+		/* Configure McBSP internal buffer usage */
+		if (dma_data->set_threshold)
+			dma_data->set_threshold(substream);
+
+		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_irqrestore(&prtd->lock, flags);
+
+	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 (cpu_is_omap1510()) {
+		offset = prtd->period_index * runtime->period_size;
+	} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		ptr = omap_get_dma_dst_pos(prtd->dma_ch);
+		offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+	} else {
+		ptr = omap_get_dma_src_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);
+}
+
+static 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(64);
+
+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;
+	}
+}
+
+static int omap_pcm_new(struct snd_card *card, struct snd_soc_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_BIT_MASK(64);
+
+	if (dai->driver->playback.channels_min) {
+		ret = omap_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto out;
+	}
+
+	if (dai->driver->capture.channels_min) {
+		ret = omap_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto out;
+	}
+
+out:
+	return ret;
+}
+
+static struct snd_soc_platform_driver omap_soc_platform = {
+	.ops		= &omap_pcm_ops,
+	.pcm_new	= omap_pcm_new,
+	.pcm_free	= omap_pcm_free_dma_buffers,
+};
+
+static __devinit int omap_pcm_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_platform(&pdev->dev,
+			&omap_soc_platform);
+}
+
+static int __devexit omap_pcm_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver omap_pcm_driver = {
+	.driver = {
+			.name = "omap-pcm-audio",
+			.owner = THIS_MODULE,
+	},
+
+	.probe = omap_pcm_probe,
+	.remove = __devexit_p(omap_pcm_remove),
+};
+
+static int __init snd_omap_pcm_init(void)
+{
+	return platform_driver_register(&omap_pcm_driver);
+}
+module_init(snd_omap_pcm_init);
+
+static void __exit snd_omap_pcm_exit(void)
+{
+	platform_driver_unregister(&omap_pcm_driver);
+}
+module_exit(snd_omap_pcm_exit);
+
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.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 00000000..a0ed1dbb
--- /dev/null
+++ b/sound/soc/omap/omap-pcm.h
@@ -0,0 +1,38 @@
+/*
+ * omap-pcm.h
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ *          Peter Ujfalusi <peter.ujfalusi@ti.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 */
+	void (*set_threshold)(struct snd_pcm_substream *substream);
+	int		data_type;	/* data type 8,16,32 */
+	int		sync_mode;	/* DMA sync mode */
+	int		packet_size;	/* packet size only in PACKET mode */
+};
+
+#endif
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
new file mode 100644
index 00000000..40db813c
--- /dev/null
+++ b/sound/soc/omap/omap3beagle.c
@@ -0,0 +1,149 @@
+/*
+ * omap3beagle.c  --  SoC audio for OMAP3 Beagle
+ *
+ * Author: Steve Sakoman <steve@sakoman.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 <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+static int omap3beagle_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	unsigned int fmt;
+	int ret;
+
+	switch (params_channels(params)) {
+	case 2: /* Stereo I2S mode */
+		fmt =	SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBM_CFM;
+		break;
+	case 4: /* Four channel TDM mode */
+		fmt =	SND_SOC_DAIFMT_DSP_A |
+			SND_SOC_DAIFMT_IB_NF |
+			SND_SOC_DAIFMT_CBM_CFM;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops omap3beagle_ops = {
+	.hw_params = omap3beagle_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3beagle_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai_name = "omap-mcbsp-dai.1",
+	.platform_name = "omap-pcm-audio",
+	.codec_dai_name = "twl4030-hifi",
+	.codec_name = "twl4030-codec",
+	.ops = &omap3beagle_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap3beagle = {
+	.name = "omap3beagle",
+	.owner = THIS_MODULE,
+	.dai_link = &omap3beagle_dai,
+	.num_links = 1,
+};
+
+static struct platform_device *omap3beagle_snd_device;
+
+static int __init omap3beagle_soc_init(void)
+{
+	int ret;
+
+	if (!(machine_is_omap3_beagle() || machine_is_devkit8000()))
+		return -ENODEV;
+	pr_info("OMAP3 Beagle/Devkit8000 SoC init\n");
+
+	omap3beagle_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!omap3beagle_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(omap3beagle_snd_device, &snd_soc_omap3beagle);
+
+	ret = platform_device_add(omap3beagle_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(omap3beagle_snd_device);
+
+	return ret;
+}
+
+static void __exit omap3beagle_soc_exit(void)
+{
+	platform_device_unregister(omap3beagle_snd_device);
+}
+
+module_init(omap3beagle_soc_init);
+module_exit(omap3beagle_soc_exit);
+
+MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 Beagle");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
new file mode 100644
index 00000000..0daa0446
--- /dev/null
+++ b/sound/soc/omap/omap3evm.c
@@ -0,0 +1,135 @@
+/*
+ * omap3evm.c  -- ALSA SoC support for OMAP3 EVM
+ *
+ * Author: Anuj Aggarwal <anuj.aggarwal@ti.com>
+ *
+ * Based on sound/soc/omap/beagle.c by Steve Sakoman
+ *
+ * Copyright (C) 2008 Texas Instruments, Incorporated
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind,
+ * whether express or implied; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+static int omap3evm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "Can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "Can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "Can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops omap3evm_ops = {
+	.hw_params = omap3evm_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3evm_dai = {
+	.name 		= "TWL4030",
+	.stream_name 	= "TWL4030",
+	.cpu_dai_name = "omap-mcbsp-dai.1",
+	.codec_dai_name = "twl4030-hifi",
+	.platform_name = "omap-pcm-audio",
+	.codec_name = "twl4030-codec",
+	.ops 		= &omap3evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap3evm = {
+	.name = "omap3evm",
+	.dai_link = &omap3evm_dai,
+	.num_links = 1,
+};
+
+static struct platform_device *omap3evm_snd_device;
+
+static int __init omap3evm_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3evm())
+		return -ENODEV;
+	pr_info("OMAP3 EVM SoC init\n");
+
+	omap3evm_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!omap3evm_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(omap3evm_snd_device, &snd_soc_omap3evm);
+	ret = platform_device_add(omap3evm_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(omap3evm_snd_device);
+
+	return ret;
+}
+
+static void __exit omap3evm_soc_exit(void)
+{
+	platform_device_unregister(omap3evm_snd_device);
+}
+
+module_init(omap3evm_soc_init);
+module_exit(omap3evm_soc_exit);
+
+MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
new file mode 100644
index 00000000..8047c521
--- /dev/null
+++ b/sound/soc/omap/omap3pandora.c
@@ -0,0 +1,339 @@
+/*
+ * omap3pandora.c  --  SoC audio for Pandora Handheld Console
+ *
+ * Author: Gražvydas Ignotas <notasas@gmail.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 <linux/gpio.h>
+#include <linux/delay.h>
+#include <linux/regulator/consumer.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+#define OMAP3_PANDORA_DAC_POWER_GPIO	118
+#define OMAP3_PANDORA_AMP_POWER_GPIO	14
+
+#define PREFIX "ASoC omap3pandora: "
+
+static struct regulator *omap3pandora_dac_reg;
+
+static int omap3pandora_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		  SND_SOC_DAIFMT_CBS_CFS;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set codec system clock\n");
+		return ret;
+	}
+
+	/* Set McBSP clock to external */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT,
+				     256 * params_rate(params),
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set cpu system clock\n");
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set SRG clock divider\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
+{
+	/*
+	 * The PCM1773 DAC datasheet requires 1ms delay between switching
+	 * VCC power on/off and /PD pin high/low
+	 */
+	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		regulator_enable(omap3pandora_dac_reg);
+		mdelay(1);
+		gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1);
+	} else {
+		gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
+		mdelay(1);
+		regulator_disable(omap3pandora_dac_reg);
+	}
+
+	return 0;
+}
+
+static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1);
+	else
+		gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
+
+	return 0;
+}
+
+/*
+ * Audio paths on Pandora board:
+ *
+ *  |O| ---> PCM DAC +-> AMP -> Headphone Jack
+ *  |M|         A    +--------> Line Out
+ *  |A| <~~clk~~+
+ *  |P| <--- TWL4030 <--------- Line In and MICs
+ */
+static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC_E("PCM DAC", "HiFi Playback", SND_SOC_NOPM,
+			   0, 0, omap3pandora_dac_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
+			   0, 0, NULL, 0, omap3pandora_hp_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+};
+
+static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Mic (internal)", NULL),
+	SND_SOC_DAPM_MIC("Mic (external)", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+	{"PCM DAC", NULL, "APLL Enable"},
+	{"Headphone Amplifier", NULL, "PCM DAC"},
+	{"Line Out", NULL, "PCM DAC"},
+	{"Headphone Jack", NULL, "Headphone Amplifier"},
+};
+
+static const struct snd_soc_dapm_route omap3pandora_in_map[] = {
+	{"AUXL", NULL, "Line In"},
+	{"AUXR", NULL, "Line In"},
+
+	{"MAINMIC", NULL, "Mic Bias 1"},
+	{"Mic Bias 1", NULL, "Mic (internal)"},
+
+	{"SUBMIC", NULL, "Mic Bias 2"},
+	{"Mic Bias 2", NULL, "Mic (external)"},
+};
+
+static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	/* All TWL4030 output pins are floating */
+	snd_soc_dapm_nc_pin(dapm, "EARPIECE");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
+	snd_soc_dapm_nc_pin(dapm, "HSOL");
+	snd_soc_dapm_nc_pin(dapm, "HSOR");
+	snd_soc_dapm_nc_pin(dapm, "CARKITL");
+	snd_soc_dapm_nc_pin(dapm, "CARKITR");
+	snd_soc_dapm_nc_pin(dapm, "HFL");
+	snd_soc_dapm_nc_pin(dapm, "HFR");
+	snd_soc_dapm_nc_pin(dapm, "VIBRA");
+
+	ret = snd_soc_dapm_new_controls(dapm, omap3pandora_out_dapm_widgets,
+				ARRAY_SIZE(omap3pandora_out_dapm_widgets));
+	if (ret < 0)
+		return ret;
+
+	snd_soc_dapm_add_routes(dapm, omap3pandora_out_map,
+		ARRAY_SIZE(omap3pandora_out_map));
+
+	return snd_soc_dapm_sync(dapm);
+}
+
+static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	/* Not comnnected */
+	snd_soc_dapm_nc_pin(dapm, "HSMIC");
+	snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
+
+	ret = snd_soc_dapm_new_controls(dapm, omap3pandora_in_dapm_widgets,
+				ARRAY_SIZE(omap3pandora_in_dapm_widgets));
+	if (ret < 0)
+		return ret;
+
+	snd_soc_dapm_add_routes(dapm, omap3pandora_in_map,
+		ARRAY_SIZE(omap3pandora_in_map));
+
+	return snd_soc_dapm_sync(dapm);
+}
+
+static struct snd_soc_ops omap3pandora_ops = {
+	.hw_params = omap3pandora_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3pandora_dai[] = {
+	{
+		.name = "PCM1773",
+		.stream_name = "HiFi Out",
+		.cpu_dai_name = "omap-mcbsp-dai.1",
+		.codec_dai_name = "twl4030-hifi",
+		.platform_name = "omap-pcm-audio",
+		.codec_name = "twl4030-codec",
+		.ops = &omap3pandora_ops,
+		.init = omap3pandora_out_init,
+	}, {
+		.name = "TWL4030",
+		.stream_name = "Line/Mic In",
+		.cpu_dai_name = "omap-mcbsp-dai.3",
+		.codec_dai_name = "twl4030-hifi",
+		.platform_name = "omap-pcm-audio",
+		.codec_name = "twl4030-codec",
+		.ops = &omap3pandora_ops,
+		.init = omap3pandora_in_init,
+	}
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_omap3pandora = {
+	.name = "omap3pandora",
+	.dai_link = omap3pandora_dai,
+	.num_links = ARRAY_SIZE(omap3pandora_dai),
+};
+
+static struct platform_device *omap3pandora_snd_device;
+
+static int __init omap3pandora_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3_pandora())
+		return -ENODEV;
+
+	pr_info("OMAP3 Pandora SoC init\n");
+
+	ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power");
+	if (ret) {
+		pr_err(PREFIX "Failed to get DAC power GPIO\n");
+		return ret;
+	}
+
+	ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
+	if (ret) {
+		pr_err(PREFIX "Failed to set DAC power GPIO direction\n");
+		goto fail0;
+	}
+
+	ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power");
+	if (ret) {
+		pr_err(PREFIX "Failed to get amp power GPIO\n");
+		goto fail0;
+	}
+
+	ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
+	if (ret) {
+		pr_err(PREFIX "Failed to set amp power GPIO direction\n");
+		goto fail1;
+	}
+
+	omap3pandora_snd_device = platform_device_alloc("soc-audio", -1);
+	if (omap3pandora_snd_device == NULL) {
+		pr_err(PREFIX "Platform device allocation failed\n");
+		ret = -ENOMEM;
+		goto fail1;
+	}
+
+	platform_set_drvdata(omap3pandora_snd_device, &snd_soc_card_omap3pandora);
+
+	ret = platform_device_add(omap3pandora_snd_device);
+	if (ret) {
+		pr_err(PREFIX "Unable to add platform device\n");
+		goto fail2;
+	}
+
+	omap3pandora_dac_reg = regulator_get(&omap3pandora_snd_device->dev, "vcc");
+	if (IS_ERR(omap3pandora_dac_reg)) {
+		pr_err(PREFIX "Failed to get DAC regulator from %s: %ld\n",
+			dev_name(&omap3pandora_snd_device->dev),
+			PTR_ERR(omap3pandora_dac_reg));
+		ret = PTR_ERR(omap3pandora_dac_reg);
+		goto fail3;
+	}
+
+	return 0;
+
+fail3:
+	platform_device_del(omap3pandora_snd_device);
+fail2:
+	platform_device_put(omap3pandora_snd_device);
+fail1:
+	gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
+fail0:
+	gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
+	return ret;
+}
+module_init(omap3pandora_soc_init);
+
+static void __exit omap3pandora_soc_exit(void)
+{
+	regulator_put(omap3pandora_dac_reg);
+	platform_device_unregister(omap3pandora_snd_device);
+	gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
+	gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
+}
+module_exit(omap3pandora_soc_exit);
+
+MODULE_AUTHOR("Grazvydas Ignotas <notasas@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
new file mode 100644
index 00000000..7e75e775
--- /dev/null
+++ b/sound/soc/omap/osk5912.c
@@ -0,0 +1,223 @@
+/*
+ * osk5912.c  --  SoC audio for OSK 5912
+ *
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * Contact: Arun KS  <arunks@mistralsolutions.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 <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+#include <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic23.h"
+
+#define CODEC_CLOCK 	12000000
+
+static struct clk *tlv320aic23_mclk;
+
+static int osk_startup(struct snd_pcm_substream *substream)
+{
+	return clk_enable(tlv320aic23_mclk);
+}
+
+static void osk_shutdown(struct snd_pcm_substream *substream)
+{
+	clk_disable(tlv320aic23_mclk);
+}
+
+static int osk_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int err;
+
+	/* Set codec DAI configuration */
+	err = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_DSP_B |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return err;
+	}
+
+	/* Set cpu DAI configuration */
+	err = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_DSP_B |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return err;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	err =
+	    snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
+
+	if (err < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return err;
+	}
+
+	return err;
+}
+
+static struct snd_soc_ops osk_ops = {
+	.startup = osk_startup,
+	.hw_params = osk_hw_params,
+	.shutdown = osk_shutdown,
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Jack", NULL, "LHPOUT"},
+	{"Headphone Jack", NULL, "RHPOUT"},
+
+	{"LLINEIN", NULL, "Line In"},
+	{"RLINEIN", NULL, "Line In"},
+
+	{"MICIN", NULL, "Mic Jack"},
+};
+
+static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	/* Add osk5912 specific widgets */
+	snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* Set up osk5912 specific audio path audio_map */
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	snd_soc_dapm_enable_pin(dapm, "Line In");
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+
+	snd_soc_dapm_sync(dapm);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link osk_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai_name = "omap-mcbsp-dai.0",
+	.codec_dai_name = "tlv320aic23-hifi",
+	.platform_name = "omap-pcm-audio",
+	.codec_name = "tlv320aic23-codec",
+	.init = osk_tlv320aic23_init,
+	.ops = &osk_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_card_osk = {
+	.name = "OSK5912",
+	.dai_link = &osk_dai,
+	.num_links = 1,
+};
+
+static struct platform_device *osk_snd_device;
+
+static int __init osk_soc_init(void)
+{
+	int err;
+	u32 curRate;
+	struct device *dev;
+
+	if (!(machine_is_omap_osk()))
+		return -ENODEV;
+
+	osk_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!osk_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(osk_snd_device, &snd_soc_card_osk);
+	err = platform_device_add(osk_snd_device);
+	if (err)
+		goto err1;
+
+	dev = &osk_snd_device->dev;
+
+	tlv320aic23_mclk = clk_get(dev, "mclk");
+	if (IS_ERR(tlv320aic23_mclk)) {
+		printk(KERN_ERR "Could not get mclk clock\n");
+		err = PTR_ERR(tlv320aic23_mclk);
+		goto err2;
+	}
+
+	/*
+	 * Configure 12 MHz output on MCLK.
+	 */
+	curRate = (uint) clk_get_rate(tlv320aic23_mclk);
+	if (curRate != CODEC_CLOCK) {
+		if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
+			printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
+			err = -ECANCELED;
+			goto err3;
+		}
+	}
+
+	printk(KERN_INFO "MCLK = %d [%d]\n",
+	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
+
+	return 0;
+
+err3:
+	clk_put(tlv320aic23_mclk);
+err2:
+	platform_device_del(osk_snd_device);
+err1:
+	platform_device_put(osk_snd_device);
+
+	return err;
+
+}
+
+static void __exit osk_soc_exit(void)
+{
+	clk_put(tlv320aic23_mclk);
+	platform_device_unregister(osk_snd_device);
+}
+
+module_init(osk_soc_init);
+module_exit(osk_soc_exit);
+
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_DESCRIPTION("ALSA SoC OSK 5912");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
new file mode 100644
index 00000000..bbcf380b
--- /dev/null
+++ b/sound/soc/omap/overo.c
@@ -0,0 +1,139 @@
+/*
+ * overo.c  --  SoC audio for Gumstix Overo
+ *
+ * Author: Steve Sakoman <steve@sakoman.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 <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+static int overo_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops overo_ops = {
+	.hw_params = overo_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link overo_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai_name = "omap-mcbsp-dai.1",
+	.codec_dai_name = "twl4030-hifi",
+	.platform_name = "omap-pcm-audio",
+	.codec_name = "twl4030-codec",
+	.ops = &overo_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_card_overo = {
+	.name = "overo",
+	.dai_link = &overo_dai,
+	.num_links = 1,
+};
+
+static struct platform_device *overo_snd_device;
+
+static int __init overo_soc_init(void)
+{
+	int ret;
+
+	if (!(machine_is_overo() || machine_is_cm_t35())) {
+		pr_debug("Incomatible machine!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "overo SoC init\n");
+
+	overo_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!overo_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(overo_snd_device, &snd_soc_card_overo);
+
+	ret = platform_device_add(overo_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(overo_snd_device);
+
+	return ret;
+}
+module_init(overo_soc_init);
+
+static void __exit overo_soc_exit(void)
+{
+	platform_device_unregister(overo_snd_device);
+}
+module_exit(overo_soc_exit);
+
+MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>");
+MODULE_DESCRIPTION("ALSA SoC overo");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
new file mode 100644
index 00000000..0aae998b
--- /dev/null
+++ b/sound/soc/omap/rx51.c
@@ -0,0 +1,468 @@
+/*
+ * rx51.c  --  SoC audio for Nokia RX-51
+ *
+ * Copyright (C) 2008 - 2009 Nokia Corporation
+ *
+ * Contact: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *          Eduardo Valentin <eduardo.valentin@nokia.com>
+ *          Jarkko Nikula <jhnikula@gmail.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/delay.h>
+#include <linux/gpio.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <plat/mcbsp.h>
+#include "../codecs/tpa6130a2.h"
+
+#include <asm/mach-types.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+#define RX51_TVOUT_SEL_GPIO		40
+#define RX51_JACK_DETECT_GPIO		177
+#define RX51_ECI_SW_GPIO		182
+/*
+ * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This
+ * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c
+ */
+#define RX51_SPEAKER_AMP_TWL_GPIO	(192 + 7)
+
+enum {
+	RX51_JACK_DISABLED,
+	RX51_JACK_TVOUT,		/* tv-out with stereo output */
+	RX51_JACK_HP,			/* headphone: stereo output, no mic */
+	RX51_JACK_HS,			/* headset: stereo output with mic */
+};
+
+static int rx51_spk_func;
+static int rx51_dmic_func;
+static int rx51_jack_func;
+
+static void rx51_ext_control(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int hp = 0, hs = 0, tvout = 0;
+
+	switch (rx51_jack_func) {
+	case RX51_JACK_TVOUT:
+		tvout = 1;
+		hp = 1;
+		break;
+	case RX51_JACK_HS:
+		hs = 1;
+	case RX51_JACK_HP:
+		hp = 1;
+		break;
+	}
+
+	if (rx51_spk_func)
+		snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+	if (rx51_dmic_func)
+		snd_soc_dapm_enable_pin(dapm, "DMic");
+	else
+		snd_soc_dapm_disable_pin(dapm, "DMic");
+	if (hp)
+		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+	if (hs)
+		snd_soc_dapm_enable_pin(dapm, "HS Mic");
+	else
+		snd_soc_dapm_disable_pin(dapm, "HS Mic");
+
+	gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout);
+
+	snd_soc_dapm_sync(dapm);
+}
+
+static int rx51_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+
+	snd_pcm_hw_constraint_minmax(runtime,
+				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+	rx51_ext_control(codec);
+
+	return 0;
+}
+
+static int rx51_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int err;
+
+	/* Set codec DAI configuration */
+	err = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_DSP_A |
+				  SND_SOC_DAIFMT_IB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0)
+		return err;
+
+	/* Set cpu DAI configuration */
+	err = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_DSP_A |
+				  SND_SOC_DAIFMT_IB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0)
+		return err;
+
+	/* Set the codec system clock for DAC and ADC */
+	return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000,
+				      SND_SOC_CLOCK_IN);
+}
+
+static struct snd_soc_ops rx51_ops = {
+	.startup = rx51_startup,
+	.hw_params = rx51_hw_params,
+};
+
+static int rx51_get_spk(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = rx51_spk_func;
+
+	return 0;
+}
+
+static int rx51_set_spk(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (rx51_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	rx51_spk_func = ucontrol->value.integer.value[0];
+	rx51_ext_control(codec);
+
+	return 1;
+}
+
+static int rx51_spk_event(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		gpio_set_value_cansleep(RX51_SPEAKER_AMP_TWL_GPIO, 1);
+	else
+		gpio_set_value_cansleep(RX51_SPEAKER_AMP_TWL_GPIO, 0);
+
+	return 0;
+}
+
+static int rx51_hp_event(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *k, int event)
+{
+	struct snd_soc_codec *codec = w->dapm->codec;
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		tpa6130a2_stereo_enable(codec, 1);
+	else
+		tpa6130a2_stereo_enable(codec, 0);
+
+	return 0;
+}
+
+static int rx51_get_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = rx51_dmic_func;
+
+	return 0;
+}
+
+static int rx51_set_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (rx51_dmic_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	rx51_dmic_func = ucontrol->value.integer.value[0];
+	rx51_ext_control(codec);
+
+	return 1;
+}
+
+static int rx51_get_jack(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = rx51_jack_func;
+
+	return 0;
+}
+
+static int rx51_set_jack(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (rx51_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	rx51_jack_func = ucontrol->value.integer.value[0];
+	rx51_ext_control(codec);
+
+	return 1;
+}
+
+static struct snd_soc_jack rx51_av_jack;
+
+static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = {
+	{
+		.gpio = RX51_JACK_DETECT_GPIO,
+		.name = "avdet-gpio",
+		.report = SND_JACK_HEADSET,
+		.invert = 1,
+		.debounce_time = 200,
+	},
+};
+
+static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event),
+	SND_SOC_DAPM_MIC("DMic", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event),
+	SND_SOC_DAPM_MIC("HS Mic", NULL),
+	SND_SOC_DAPM_LINE("FM Transmitter", NULL),
+};
+
+static const struct snd_soc_dapm_widget aic34_dapm_widgetsb[] = {
+	SND_SOC_DAPM_SPK("Earphone", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Ext Spk", NULL, "HPLOUT"},
+	{"Ext Spk", NULL, "HPROUT"},
+	{"Headphone Jack", NULL, "LLOUT"},
+	{"Headphone Jack", NULL, "RLOUT"},
+	{"FM Transmitter", NULL, "LLOUT"},
+	{"FM Transmitter", NULL, "RLOUT"},
+
+	{"DMic Rate 64", NULL, "Mic Bias 2V"},
+	{"Mic Bias 2V", NULL, "DMic"},
+};
+
+static const struct snd_soc_dapm_route audio_mapb[] = {
+	{"b LINE2R", NULL, "MONO_LOUT"},
+	{"Earphone", NULL, "b HPLOUT"},
+
+	{"LINE1L", NULL, "b Mic Bias 2.5V"},
+	{"b Mic Bias 2.5V", NULL, "HS Mic"}
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *input_function[] = {"ADC", "Digital Mic"};
+static const char *jack_function[] = {"Off", "TV-OUT", "Headphone", "Headset"};
+
+static const struct soc_enum rx51_enum[] = {
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
+};
+
+static const struct snd_kcontrol_new aic34_rx51_controls[] = {
+	SOC_ENUM_EXT("Speaker Function", rx51_enum[0],
+		     rx51_get_spk, rx51_set_spk),
+	SOC_ENUM_EXT("Input Select",  rx51_enum[1],
+		     rx51_get_input, rx51_set_input),
+	SOC_ENUM_EXT("Jack Function", rx51_enum[2],
+		     rx51_get_jack, rx51_set_jack),
+	SOC_DAPM_PIN_SWITCH("FM Transmitter"),
+};
+
+static const struct snd_kcontrol_new aic34_rx51_controlsb[] = {
+	SOC_DAPM_PIN_SWITCH("Earphone"),
+};
+
+static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int err;
+
+	/* Set up NC codec pins */
+	snd_soc_dapm_nc_pin(dapm, "MIC3L");
+	snd_soc_dapm_nc_pin(dapm, "MIC3R");
+	snd_soc_dapm_nc_pin(dapm, "LINE1R");
+
+	/* Add RX-51 specific controls */
+	err = snd_soc_add_controls(codec, aic34_rx51_controls,
+				   ARRAY_SIZE(aic34_rx51_controls));
+	if (err < 0)
+		return err;
+
+	/* Add RX-51 specific widgets */
+	snd_soc_dapm_new_controls(dapm, aic34_dapm_widgets,
+				  ARRAY_SIZE(aic34_dapm_widgets));
+
+	/* Set up RX-51 specific audio path audio_map */
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+
+	err = tpa6130a2_add_controls(codec);
+	if (err < 0)
+		return err;
+	snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42);
+
+	err = omap_mcbsp_st_add_controls(codec, 1);
+	if (err < 0)
+		return err;
+
+	snd_soc_dapm_sync(dapm);
+
+	/* AV jack detection */
+	err = snd_soc_jack_new(codec, "AV Jack",
+			       SND_JACK_HEADSET | SND_JACK_VIDEOOUT,
+			       &rx51_av_jack);
+	if (err)
+		return err;
+	err = snd_soc_jack_add_gpios(&rx51_av_jack,
+				     ARRAY_SIZE(rx51_av_jack_gpios),
+				     rx51_av_jack_gpios);
+
+	return err;
+}
+
+static int rx51_aic34b_init(struct snd_soc_dapm_context *dapm)
+{
+	int err;
+
+	err = snd_soc_add_controls(dapm->codec, aic34_rx51_controlsb,
+				   ARRAY_SIZE(aic34_rx51_controlsb));
+	if (err < 0)
+		return err;
+
+	err = snd_soc_dapm_new_controls(dapm, aic34_dapm_widgetsb,
+					ARRAY_SIZE(aic34_dapm_widgetsb));
+	if (err < 0)
+		return 0;
+
+	return snd_soc_dapm_add_routes(dapm, audio_mapb,
+				       ARRAY_SIZE(audio_mapb));
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link rx51_dai[] = {
+	{
+		.name = "TLV320AIC34",
+		.stream_name = "AIC34",
+		.cpu_dai_name = "omap-mcbsp-dai.1",
+		.codec_dai_name = "tlv320aic3x-hifi",
+		.platform_name = "omap-pcm-audio",
+		.codec_name = "tlv320aic3x-codec.2-0018",
+		.init = rx51_aic34_init,
+		.ops = &rx51_ops,
+	},
+};
+
+struct snd_soc_aux_dev rx51_aux_dev[] = {
+	{
+		.name = "TLV320AIC34b",
+		.codec_name = "tlv320aic3x-codec.2-0019",
+		.init = rx51_aic34b_init,
+	},
+};
+
+static struct snd_soc_codec_conf rx51_codec_conf[] = {
+	{
+		.dev_name = "tlv320aic3x-codec.2-0019",
+		.name_prefix = "b",
+	},
+};
+
+/* Audio card */
+static struct snd_soc_card rx51_sound_card = {
+	.name = "RX-51",
+	.dai_link = rx51_dai,
+	.num_links = ARRAY_SIZE(rx51_dai),
+	.aux_dev = rx51_aux_dev,
+	.num_aux_devs = ARRAY_SIZE(rx51_aux_dev),
+	.codec_conf = rx51_codec_conf,
+	.num_configs = ARRAY_SIZE(rx51_codec_conf),
+};
+
+static struct platform_device *rx51_snd_device;
+
+static int __init rx51_soc_init(void)
+{
+	int err;
+
+	if (!machine_is_nokia_rx51())
+		return -ENODEV;
+
+	err = gpio_request_one(RX51_TVOUT_SEL_GPIO,
+			       GPIOF_DIR_OUT | GPIOF_INIT_LOW, "tvout_sel");
+	if (err)
+		goto err_gpio_tvout_sel;
+	err = gpio_request_one(RX51_ECI_SW_GPIO,
+			       GPIOF_DIR_OUT | GPIOF_INIT_HIGH, "eci_sw");
+	if (err)
+		goto err_gpio_eci_sw;
+
+	rx51_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!rx51_snd_device) {
+		err = -ENOMEM;
+		goto err1;
+	}
+
+	platform_set_drvdata(rx51_snd_device, &rx51_sound_card);
+
+	err = platform_device_add(rx51_snd_device);
+	if (err)
+		goto err2;
+
+	return 0;
+err2:
+	platform_device_put(rx51_snd_device);
+err1:
+	gpio_free(RX51_ECI_SW_GPIO);
+err_gpio_eci_sw:
+	gpio_free(RX51_TVOUT_SEL_GPIO);
+err_gpio_tvout_sel:
+
+	return err;
+}
+
+static void __exit rx51_soc_exit(void)
+{
+	snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios),
+				rx51_av_jack_gpios);
+
+	platform_device_unregister(rx51_snd_device);
+	gpio_free(RX51_ECI_SW_GPIO);
+	gpio_free(RX51_TVOUT_SEL_GPIO);
+}
+
+module_init(rx51_soc_init);
+module_exit(rx51_soc_exit);
+
+MODULE_AUTHOR("Nokia Corporation");
+MODULE_DESCRIPTION("ALSA SoC Nokia RX-51");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
new file mode 100644
index 00000000..3f72d17d
--- /dev/null
+++ b/sound/soc/omap/sdp3430.c
@@ -0,0 +1,345 @@
+/*
+ * sdp3430.c  --  SoC audio for TI OMAP3430 SDP
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.com>
+ *
+ * Based on:
+ * Author: Steve Sakoman <steve@sakoman.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 <linux/i2c/twl.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <plat/mcbsp.h>
+
+/* Register descriptions for twl4030 codec part */
+#include <linux/mfd/twl4030-codec.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+/* TWL4030 PMBR1 Register */
+#define TWL4030_INTBR_PMBR1		0x0D
+/* TWL4030 PMBR1 Register GPIO6 mux bit */
+#define TWL4030_GPIO6_PWM0_MUTE(value)	(value << 2)
+
+static struct snd_soc_card snd_soc_sdp3430;
+
+static int sdp3430_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops sdp3430_ops = {
+	.hw_params = sdp3430_hw_params,
+};
+
+static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				SND_SOC_DAIFMT_DSP_A |
+				SND_SOC_DAIFMT_IB_NF |
+				SND_SOC_DAIFMT_CBM_CFM);
+	if (ret) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				SND_SOC_DAIFMT_DSP_A |
+				SND_SOC_DAIFMT_IB_NF |
+				SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops sdp3430_voice_ops = {
+	.hw_params = sdp3430_hw_voice_params,
+};
+
+/* Headset jack */
+static struct snd_soc_jack hs_jack;
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{
+		.pin = "Headset Mic",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headset Stereophone",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+/* Headset jack detection gpios */
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+	{
+		.gpio = (OMAP_MAX_GPIO_LINES + 2),
+		.name = "hsdet-gpio",
+		.report = SND_JACK_HEADSET,
+		.debounce_time = 200,
+	},
+};
+
+/* SDP3430 machine DAPM */
+static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Ext Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* External Mics: MAINMIC, SUBMIC with bias*/
+	{"MAINMIC", NULL, "Mic Bias 1"},
+	{"SUBMIC", NULL, "Mic Bias 2"},
+	{"Mic Bias 1", NULL, "Ext Mic"},
+	{"Mic Bias 2", NULL, "Ext Mic"},
+
+	/* External Speakers: HFL, HFR */
+	{"Ext Spk", NULL, "HFL"},
+	{"Ext Spk", NULL, "HFR"},
+
+	/* Headset Mic: HSMIC with bias */
+	{"HSMIC", NULL, "Headset Mic Bias"},
+	{"Headset Mic Bias", NULL, "Headset Mic"},
+
+	/* Headset Stereophone (Headphone): HSOL, HSOR */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+};
+
+static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	/* Add SDP3430 specific widgets */
+	ret = snd_soc_dapm_new_controls(dapm, sdp3430_twl4030_dapm_widgets,
+				ARRAY_SIZE(sdp3430_twl4030_dapm_widgets));
+	if (ret)
+		return ret;
+
+	/* Set up SDP3430 specific audio path audio_map */
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+
+	/* SDP3430 connected pins */
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic");
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+	snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
+
+	/* TWL4030 not connected pins */
+	snd_soc_dapm_nc_pin(dapm, "AUXL");
+	snd_soc_dapm_nc_pin(dapm, "AUXR");
+	snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
+
+	snd_soc_dapm_nc_pin(dapm, "OUTL");
+	snd_soc_dapm_nc_pin(dapm, "OUTR");
+	snd_soc_dapm_nc_pin(dapm, "EARPIECE");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
+	snd_soc_dapm_nc_pin(dapm, "CARKITL");
+	snd_soc_dapm_nc_pin(dapm, "CARKITR");
+
+	ret = snd_soc_dapm_sync(dapm);
+	if (ret)
+		return ret;
+
+	/* Headset jack detection */
+	ret = snd_soc_jack_new(codec, "Headset Jack",
+				SND_JACK_HEADSET, &hs_jack);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+				hs_jack_pins);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+				hs_jack_gpios);
+
+	return ret;
+}
+
+static int sdp3430_twl4030_voice_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	unsigned short reg;
+
+	/* Enable voice interface */
+	reg = codec->driver->read(codec, TWL4030_REG_VOICE_IF);
+	reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
+	codec->driver->write(codec, TWL4030_REG_VOICE_IF, reg);
+
+	return 0;
+}
+
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sdp3430_dai[] = {
+	{
+		.name = "TWL4030 I2S",
+		.stream_name = "TWL4030 Audio",
+		.cpu_dai_name = "omap-mcbsp-dai.1",
+		.codec_dai_name = "twl4030-hifi",
+		.platform_name = "omap-pcm-audio",
+		.codec_name = "twl4030-codec",
+		.init = sdp3430_twl4030_init,
+		.ops = &sdp3430_ops,
+	},
+	{
+		.name = "TWL4030 PCM",
+		.stream_name = "TWL4030 Voice",
+		.cpu_dai_name = "omap-mcbsp-dai.2",
+		.codec_dai_name = "twl4030-voice",
+		.platform_name = "omap-pcm-audio",
+		.codec_name = "twl4030-codec",
+		.init = sdp3430_twl4030_voice_init,
+		.ops = &sdp3430_voice_ops,
+	},
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_sdp3430 = {
+	.name = "SDP3430",
+	.dai_link = sdp3430_dai,
+	.num_links = ARRAY_SIZE(sdp3430_dai),
+};
+
+static struct platform_device *sdp3430_snd_device;
+
+static int __init sdp3430_soc_init(void)
+{
+	int ret;
+	u8 pin_mux;
+
+	if (!machine_is_omap_3430sdp())
+		return -ENODEV;
+	printk(KERN_INFO "SDP3430 SoC init\n");
+
+	sdp3430_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!sdp3430_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(sdp3430_snd_device, &snd_soc_sdp3430);
+
+	/* Set TWL4030 GPIO6 as EXTMUTE signal */
+	twl_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux,
+						TWL4030_INTBR_PMBR1);
+	pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03);
+	pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02);
+	twl_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux,
+						TWL4030_INTBR_PMBR1);
+
+	ret = platform_device_add(sdp3430_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(sdp3430_snd_device);
+
+	return ret;
+}
+module_init(sdp3430_soc_init);
+
+static void __exit sdp3430_soc_exit(void)
+{
+	snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+				hs_jack_gpios);
+
+	platform_device_unregister(sdp3430_snd_device);
+}
+module_exit(sdp3430_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC SDP3430");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c
new file mode 100644
index 00000000..189e0390
--- /dev/null
+++ b/sound/soc/omap/sdp4430.c
@@ -0,0 +1,261 @@
+/*
+ * sdp4430.c  --  SoC audio for TI OMAP4430 SDP
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.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/jack.h>
+
+#include <asm/mach-types.h>
+#include <plat/hardware.h>
+#include <plat/mux.h>
+
+#include "mcpdm.h"
+#include "omap-pcm.h"
+#include "../codecs/twl6040.h"
+
+static int twl6040_power_mode;
+
+static int sdp4430_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int clk_id, freq;
+	int ret;
+
+	if (twl6040_power_mode) {
+		clk_id = TWL6040_SYSCLK_SEL_HPPLL;
+		freq = 38400000;
+	} else {
+		clk_id = TWL6040_SYSCLK_SEL_LPPLL;
+		freq = 32768;
+	}
+
+	/* set the codec mclk */
+	ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
+				SND_SOC_CLOCK_IN);
+	if (ret) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+	return ret;
+}
+
+static struct snd_soc_ops sdp4430_ops = {
+	.hw_params = sdp4430_hw_params,
+};
+
+/* Headset jack */
+static struct snd_soc_jack hs_jack;
+
+/*Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{
+		.pin = "Headset Mic",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headset Stereophone",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = twl6040_power_mode;
+	return 0;
+}
+
+static int sdp4430_set_power_mode(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	if (twl6040_power_mode == ucontrol->value.integer.value[0])
+		return 0;
+
+	twl6040_power_mode = ucontrol->value.integer.value[0];
+
+	return 1;
+}
+
+static const char *power_texts[] = {"Low-Power", "High-Performance"};
+
+static const struct soc_enum sdp4430_enum[] = {
+	SOC_ENUM_SINGLE_EXT(2, power_texts),
+};
+
+static const struct snd_kcontrol_new sdp4430_controls[] = {
+	SOC_ENUM_EXT("TWL6040 Power Mode", sdp4430_enum[0],
+		sdp4430_get_power_mode, sdp4430_set_power_mode),
+};
+
+/* SDP4430 machine DAPM */
+static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Ext Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_SPK("Earphone Spk", NULL),
+	SND_SOC_DAPM_INPUT("Aux/FM Stereo In"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* External Mics: MAINMIC, SUBMIC with bias*/
+	{"MAINMIC", NULL, "Main Mic Bias"},
+	{"SUBMIC", NULL, "Main Mic Bias"},
+	{"Main Mic Bias", NULL, "Ext Mic"},
+
+	/* External Speakers: HFL, HFR */
+	{"Ext Spk", NULL, "HFL"},
+	{"Ext Spk", NULL, "HFR"},
+
+	/* Headset Mic: HSMIC with bias */
+	{"HSMIC", NULL, "Headset Mic Bias"},
+	{"Headset Mic Bias", NULL, "Headset Mic"},
+
+	/* Headset Stereophone (Headphone): HSOL, HSOR */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+
+	/* Earphone speaker */
+	{"Earphone Spk", NULL, "EP"},
+
+	/* Aux/FM Stereo In: AFML, AFMR */
+	{"AFML", NULL, "Aux/FM Stereo In"},
+	{"AFMR", NULL, "Aux/FM Stereo In"},
+};
+
+static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	/* Add SDP4430 specific controls */
+	ret = snd_soc_add_controls(codec, sdp4430_controls,
+				ARRAY_SIZE(sdp4430_controls));
+	if (ret)
+		return ret;
+
+	/* Add SDP4430 specific widgets */
+	ret = snd_soc_dapm_new_controls(dapm, sdp4430_twl6040_dapm_widgets,
+				ARRAY_SIZE(sdp4430_twl6040_dapm_widgets));
+	if (ret)
+		return ret;
+
+	/* Set up SDP4430 specific audio path audio_map */
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+
+	/* SDP4430 connected pins */
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic");
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	snd_soc_dapm_enable_pin(dapm, "AFML");
+	snd_soc_dapm_enable_pin(dapm, "AFMR");
+	snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+	snd_soc_dapm_enable_pin(dapm, "Headset Stereophone");
+
+	ret = snd_soc_dapm_sync(dapm);
+	if (ret)
+		return ret;
+
+	/* Headset jack detection */
+	ret = snd_soc_jack_new(codec, "Headset Jack",
+				SND_JACK_HEADSET, &hs_jack);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+				hs_jack_pins);
+
+	if (machine_is_omap_4430sdp())
+		twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
+	else
+		snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET);
+
+	return ret;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sdp4430_dai = {
+	.name = "TWL6040",
+	.stream_name = "TWL6040",
+	.cpu_dai_name ="omap-mcpdm-dai",
+	.codec_dai_name = "twl6040-hifi",
+	.platform_name = "omap-pcm-audio",
+	.codec_name = "twl6040-codec",
+	.init = sdp4430_twl6040_init,
+	.ops = &sdp4430_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_sdp4430 = {
+	.name = "SDP4430",
+	.dai_link = &sdp4430_dai,
+	.num_links = 1,
+};
+
+static struct platform_device *sdp4430_snd_device;
+
+static int __init sdp4430_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap_4430sdp())
+		return -ENODEV;
+	printk(KERN_INFO "SDP4430 SoC init\n");
+
+	sdp4430_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!sdp4430_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430);
+
+	ret = platform_device_add(sdp4430_snd_device);
+	if (ret)
+		goto err;
+
+	/* Codec starts in HP mode */
+	twl6040_power_mode = 1;
+
+	return 0;
+
+err:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(sdp4430_snd_device);
+	return ret;
+}
+module_init(sdp4430_soc_init);
+
+static void __exit sdp4430_soc_exit(void)
+{
+	platform_device_unregister(sdp4430_snd_device);
+}
+module_exit(sdp4430_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC SDP4430");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c
new file mode 100644
index 00000000..01709940
--- /dev/null
+++ b/sound/soc/omap/zoom2.c
@@ -0,0 +1,291 @@
+/*
+ * zoom2.c  --  SoC audio for Zoom2
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.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 <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/board-zoom.h>
+#include <plat/mcbsp.h>
+
+/* Register descriptions for twl4030 codec part */
+#include <linux/mfd/twl4030-codec.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+#define ZOOM2_HEADSET_MUX_GPIO		(OMAP_MAX_GPIO_LINES + 15)
+
+static int zoom2_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops zoom2_ops = {
+	.hw_params = zoom2_hw_params,
+};
+
+static int zoom2_hw_voice_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				SND_SOC_DAIFMT_DSP_A |
+				SND_SOC_DAIFMT_IB_NF |
+				SND_SOC_DAIFMT_CBM_CFM);
+	if (ret) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				SND_SOC_DAIFMT_DSP_A |
+				SND_SOC_DAIFMT_IB_NF |
+				SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops zoom2_voice_ops = {
+	.hw_params = zoom2_hw_voice_params,
+};
+
+/* Zoom2 machine DAPM */
+static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Ext Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Aux In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* External Mics: MAINMIC, SUBMIC with bias*/
+	{"MAINMIC", NULL, "Mic Bias 1"},
+	{"SUBMIC", NULL, "Mic Bias 2"},
+	{"Mic Bias 1", NULL, "Ext Mic"},
+	{"Mic Bias 2", NULL, "Ext Mic"},
+
+	/* External Speakers: HFL, HFR */
+	{"Ext Spk", NULL, "HFL"},
+	{"Ext Spk", NULL, "HFR"},
+
+	/* Headset Stereophone:  HSOL, HSOR */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+
+	/* Headset Mic: HSMIC with bias */
+	{"HSMIC", NULL, "Headset Mic Bias"},
+	{"Headset Mic Bias", NULL, "Headset Mic"},
+
+	/* Aux In: AUXL, AUXR */
+	{"Aux In", NULL, "AUXL"},
+	{"Aux In", NULL, "AUXR"},
+};
+
+static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	/* Add Zoom2 specific widgets */
+	ret = snd_soc_dapm_new_controls(dapm, zoom2_twl4030_dapm_widgets,
+				ARRAY_SIZE(zoom2_twl4030_dapm_widgets));
+	if (ret)
+		return ret;
+
+	/* Set up Zoom2 specific audio path audio_map */
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+
+	/* Zoom2 connected pins */
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic");
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+	snd_soc_dapm_enable_pin(dapm, "Headset Stereophone");
+	snd_soc_dapm_enable_pin(dapm, "Aux In");
+
+	/* TWL4030 not connected pins */
+	snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
+	snd_soc_dapm_nc_pin(dapm, "EARPIECE");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
+	snd_soc_dapm_nc_pin(dapm, "CARKITL");
+	snd_soc_dapm_nc_pin(dapm, "CARKITR");
+
+	ret = snd_soc_dapm_sync(dapm);
+
+	return ret;
+}
+
+static int zoom2_twl4030_voice_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	unsigned short reg;
+
+	/* Enable voice interface */
+	reg = codec->driver->read(codec, TWL4030_REG_VOICE_IF);
+	reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
+	codec->driver->write(codec, TWL4030_REG_VOICE_IF, reg);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link zoom2_dai[] = {
+	{
+		.name = "TWL4030 I2S",
+		.stream_name = "TWL4030 Audio",
+		.cpu_dai_name = "omap-mcbsp-dai.1",
+		.codec_dai_name = "twl4030-hifi",
+		.platform_name = "omap-pcm-audio",
+		.codec_name = "twl4030-codec",
+		.init = zoom2_twl4030_init,
+		.ops = &zoom2_ops,
+	},
+	{
+		.name = "TWL4030 PCM",
+		.stream_name = "TWL4030 Voice",
+		.cpu_dai_name = "omap-mcbsp-dai.2",
+		.codec_dai_name = "twl4030-voice",
+		.platform_name = "omap-pcm-audio",
+		.codec_name = "twl4030-codec",
+		.init = zoom2_twl4030_voice_init,
+		.ops = &zoom2_voice_ops,
+	},
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_zoom2 = {
+	.name = "Zoom2",
+	.dai_link = zoom2_dai,
+	.num_links = ARRAY_SIZE(zoom2_dai),
+};
+
+static struct platform_device *zoom2_snd_device;
+
+static int __init zoom2_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap_zoom2())
+		return -ENODEV;
+	printk(KERN_INFO "Zoom2 SoC init\n");
+
+	zoom2_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!zoom2_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(zoom2_snd_device, &snd_soc_zoom2);
+	ret = platform_device_add(zoom2_snd_device);
+	if (ret)
+		goto err1;
+
+	BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0);
+	gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0);
+
+	BUG_ON(gpio_request(ZOOM2_HEADSET_EXTMUTE_GPIO, "ext_mute") < 0);
+	gpio_direction_output(ZOOM2_HEADSET_EXTMUTE_GPIO, 0);
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(zoom2_snd_device);
+
+	return ret;
+}
+module_init(zoom2_soc_init);
+
+static void __exit zoom2_soc_exit(void)
+{
+	gpio_free(ZOOM2_HEADSET_MUX_GPIO);
+	gpio_free(ZOOM2_HEADSET_EXTMUTE_GPIO);
+
+	platform_device_unregister(zoom2_snd_device);
+}
+module_exit(zoom2_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC Zoom2");
+MODULE_LICENSE("GPL");
+