Initial import

30 files changed, 7114 insertions, 0 deletions

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
new file mode 100644
index 000000000..39cea8084
--- /dev/null
+++ b/sound/soc/pxa/Kconfig
@@ -0,0 +1,219 @@
+config SND_PXA2XX_SOC
+	tristate "SoC Audio for the Intel PXA2xx chip"
+	depends on ARCH_PXA
+	select SND_ARM
+	select SND_PXA2XX_LIB
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the PXA2xx AC97, I2S or SSP interface. You will also need
+	  to select the audio interfaces to support below.
+
+config SND_MMP_SOC
+	bool "Soc Audio for Marvell MMP chips"
+	depends on ARCH_MMP
+	select MMP_SRAM
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	select SND_ARM
+	help
+	  Say Y if you want to add support for codecs attached to
+	  the MMP SSPA interface.
+
+config SND_PXA2XX_AC97
+	tristate
+	select SND_AC97_CODEC
+
+config SND_PXA2XX_SOC_AC97
+	tristate
+	select AC97_BUS
+	select SND_ARM
+	select SND_PXA2XX_LIB_AC97
+	select SND_SOC_AC97_BUS
+
+config SND_PXA2XX_SOC_I2S
+	tristate
+
+config SND_PXA_SOC_SSP
+	tristate
+	select PXA_SSP
+
+config SND_MMP_SOC_SSPA
+	tristate
+
+config SND_PXA2XX_SOC_CORGI
+	tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
+	depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx && I2C
+	select SND_PXA2XX_SOC_I2S
+	select SND_SOC_WM8731
+	help
+	  Say Y if you want to add support for SoC audio on Sharp
+	  Zaurus SL-C7x0 models (Corgi, Shepherd, Husky).
+
+config SND_PXA2XX_SOC_SPITZ
+	tristate "SoC Audio support for Sharp Zaurus SL-Cxx00"
+	depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 && I2C
+	select SND_PXA2XX_SOC_I2S
+	select SND_SOC_WM8750
+	help
+	  Say Y if you want to add support for SoC audio on Sharp
+	  Zaurus SL-Cxx00 models (Spitz, Borzoi and Akita).
+
+config SND_PXA2XX_SOC_Z2
+	tristate "SoC Audio support for Zipit Z2"
+	depends on SND_PXA2XX_SOC && MACH_ZIPIT2 && I2C
+	select SND_PXA2XX_SOC_I2S
+	select SND_SOC_WM8750
+	help
+	  Say Y if you want to add support for SoC audio on Zipit Z2.
+
+config SND_PXA2XX_SOC_POODLE
+	tristate "SoC Audio support for Poodle"
+	depends on SND_PXA2XX_SOC && MACH_POODLE && I2C
+	select SND_PXA2XX_SOC_I2S
+	select SND_SOC_WM8731
+	help
+	  Say Y if you want to add support for SoC audio on Sharp
+	  Zaurus SL-5600 model (Poodle).
+
+config SND_PXA2XX_SOC_TOSA
+	tristate "SoC AC97 Audio support for Tosa"
+	depends on SND_PXA2XX_SOC && MACH_TOSA
+	depends on MFD_TC6393XB
+	select SND_PXA2XX_SOC_AC97
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for SoC audio on Sharp
+	  Zaurus SL-C6000x models (Tosa).
+
+config SND_PXA2XX_SOC_E740
+	tristate "SoC AC97 Audio support for e740"
+	depends on SND_PXA2XX_SOC && MACH_E740
+	select SND_SOC_WM9705
+	select SND_PXA2XX_SOC_AC97
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  toshiba e740 PDA
+
+config SND_PXA2XX_SOC_E750
+	tristate "SoC AC97 Audio support for e750"
+	depends on SND_PXA2XX_SOC && MACH_E750
+	select SND_SOC_WM9705
+	select SND_PXA2XX_SOC_AC97
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  toshiba e750 PDA
+
+config SND_PXA2XX_SOC_E800
+	tristate "SoC AC97 Audio support for e800"
+	depends on SND_PXA2XX_SOC && MACH_E800
+	select SND_SOC_WM9712
+	select SND_PXA2XX_SOC_AC97
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Toshiba e800 PDA
+
+config SND_PXA2XX_SOC_EM_X270
+	tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
+	depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \
+			MACH_CM_X300)
+	select SND_PXA2XX_SOC_AC97
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for SoC audio on
+	  CompuLab EM-x270, eXeda and CM-X300 machines.
+
+config SND_PXA2XX_SOC_PALM27X
+	bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive"
+	depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \
+			MACH_PALMT5 || MACH_PALMTE2)
+	select SND_PXA2XX_SOC_AC97
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for SoC audio on
+	  Palm T|X, T5, E2 or LifeDrive handheld computer.
+
+config SND_PXA910_SOC
+	tristate "SoC Audio for Marvell PXA910 chip"
+	depends on ARCH_MMP && SND
+	select SND_PCM
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell PXA910 reference platform.
+
+config SND_SOC_TTC_DKB
+	tristate "SoC Audio support for TTC DKB"
+	depends on SND_PXA910_SOC && MACH_TTC_DKB && I2C=y
+	select PXA_SSP
+	select SND_PXA_SOC_SSP
+	select SND_MMP_SOC
+	select MFD_88PM860X
+	select SND_SOC_88PM860X
+	help
+	  Say Y if you want to add support for SoC audio on TTC DKB
+
+
+config SND_SOC_ZYLONITE
+	tristate "SoC Audio support for Marvell Zylonite"
+	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
+	select SND_PXA2XX_SOC_AC97
+	select SND_PXA_SOC_SSP
+	select SND_SOC_WM9713
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Zylonite reference platform.
+
+config SND_SOC_RAUMFELD
+	tristate "SoC Audio support Raumfeld audio adapter"
+	depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR)
+	depends on I2C && SPI_MASTER
+	select SND_PXA_SOC_SSP
+	select SND_SOC_CS4270
+	select SND_SOC_AK4104
+	help
+	  Say Y if you want to add support for SoC audio on Raumfeld devices
+
+config SND_PXA2XX_SOC_HX4700
+	tristate "SoC Audio support for HP iPAQ hx4700"
+	depends on SND_PXA2XX_SOC && MACH_H4700 && I2C
+	select SND_PXA2XX_SOC_I2S
+	select SND_SOC_AK4641
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  HP iPAQ hx4700.
+
+config SND_PXA2XX_SOC_MAGICIAN
+	tristate "SoC Audio support for HTC Magician"
+	depends on SND_PXA2XX_SOC && MACH_MAGICIAN && I2C
+	select SND_PXA2XX_SOC_I2S
+	select SND_PXA_SOC_SSP
+	select SND_SOC_UDA1380
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  HTC Magician.
+
+config SND_PXA2XX_SOC_MIOA701
+        tristate "SoC Audio support for MIO A701"
+        depends on SND_PXA2XX_SOC && MACH_MIOA701
+        select SND_PXA2XX_SOC_AC97
+        select SND_SOC_WM9713
+        help
+          Say Y if you want to add support for SoC audio on the
+          MIO A701.
+
+config SND_PXA2XX_SOC_IMOTE2
+       tristate "SoC Audio support for IMote 2"
+       depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C
+       select SND_PXA2XX_SOC_I2S
+       select SND_SOC_WM8940
+       help
+         Say Y if you want to add support for SoC audio on the
+	 IMote 2.
+
+config SND_MMP_SOC_BROWNSTONE
+	tristate "SoC Audio support for Marvell Brownstone"
+	depends on SND_MMP_SOC && MACH_BROWNSTONE
+	select SND_MMP_SOC_SSPA
+	select MFD_WM8994
+	select SND_SOC_WM8994
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Brownstone reference platform.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
new file mode 100644
index 000000000..2cff67b61
--- /dev/null
+++ b/sound/soc/pxa/Makefile
@@ -0,0 +1,53 @@
+# PXA Platform Support
+snd-soc-pxa2xx-objs := pxa2xx-pcm.o
+snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
+snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
+snd-soc-pxa-ssp-objs := pxa-ssp.o
+snd-soc-mmp-objs := mmp-pcm.o
+snd-soc-mmp-sspa-objs := mmp-sspa.o
+
+obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
+obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
+obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
+obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
+obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o
+obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o
+
+# PXA Machine Support
+snd-soc-corgi-objs := corgi.o
+snd-soc-poodle-objs := poodle.o
+snd-soc-tosa-objs := tosa.o
+snd-soc-e740-objs := e740_wm9705.o
+snd-soc-e750-objs := e750_wm9705.o
+snd-soc-e800-objs := e800_wm9712.o
+snd-soc-spitz-objs := spitz.o
+snd-soc-em-x270-objs := em-x270.o
+snd-soc-palm27x-objs := palm27x.o
+snd-soc-zylonite-objs := zylonite.o
+snd-soc-hx4700-objs := hx4700.o
+snd-soc-magician-objs := magician.o
+snd-soc-mioa701-objs := mioa701_wm9713.o
+snd-soc-z2-objs := z2.o
+snd-soc-imote2-objs := imote2.o
+snd-soc-raumfeld-objs := raumfeld.o
+snd-soc-brownstone-objs := brownstone.o
+snd-soc-ttc-dkb-objs := ttc-dkb.o
+
+obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
+obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
+obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
+obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
+obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
+obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o
+obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
+obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
+obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o
+obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
+obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
+obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
+obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o
+obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
new file mode 100644
index 000000000..79936e3e8
--- /dev/null
+++ b/sound/soc/pxa/brownstone.c
@@ -0,0 +1,168 @@
+/*
+ * linux/sound/soc/pxa/brownstone.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "../codecs/wm8994.h"
+#include "mmp-sspa.h"
+
+static const struct snd_kcontrol_new brownstone_dapm_control[] = {
+	SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Main Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route brownstone_audio_map[] = {
+	{"Ext Spk", NULL, "SPKOUTLP"},
+	{"Ext Spk", NULL, "SPKOUTLN"},
+	{"Ext Spk", NULL, "SPKOUTRP"},
+	{"Ext Spk", NULL, "SPKOUTRN"},
+
+	{"Headset Stereophone", NULL, "HPOUT1L"},
+	{"Headset Stereophone", NULL, "HPOUT1R"},
+
+	{"IN1RN", NULL, "Headset Mic"},
+
+	{"DMIC1DAT", NULL, "MICBIAS1"},
+	{"MICBIAS1", NULL, "Main Mic"},
+};
+
+static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	/* set endpoints to not connected */
+	snd_soc_dapm_nc_pin(dapm, "HPOUT2P");
+	snd_soc_dapm_nc_pin(dapm, "HPOUT2N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
+	snd_soc_dapm_nc_pin(dapm, "IN1LN");
+	snd_soc_dapm_nc_pin(dapm, "IN1LP");
+	snd_soc_dapm_nc_pin(dapm, "IN1RP");
+	snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
+	snd_soc_dapm_nc_pin(dapm, "IN2RN");
+	snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
+	snd_soc_dapm_nc_pin(dapm, "IN2LN");
+
+	return 0;
+}
+
+static int brownstone_wm8994_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 freq_out, sspa_mclk, sysclk;
+	int sspa_div;
+
+	if (params_rate(params) > 11025) {
+		freq_out  = params_rate(params) * 512;
+		sysclk    = params_rate(params) * 256;
+		sspa_mclk = params_rate(params) * 64;
+	} else {
+		freq_out  = params_rate(params) * 1024;
+		sysclk    = params_rate(params) * 512;
+		sspa_mclk = params_rate(params) * 64;
+	}
+	sspa_div = freq_out;
+	do_div(sspa_div, sspa_mclk);
+
+	snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0);
+	snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk);
+	snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk);
+
+	/* set wm8994 sysclk */
+	snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0);
+
+	return 0;
+}
+
+/* machine stream operations */
+static struct snd_soc_ops brownstone_ops = {
+	.hw_params = brownstone_wm8994_hw_params,
+};
+
+static struct snd_soc_dai_link brownstone_wm8994_dai[] = {
+{
+	.name		= "WM8994",
+	.stream_name	= "WM8994 HiFi",
+	.cpu_dai_name	= "mmp-sspa-dai.0",
+	.codec_dai_name	= "wm8994-aif1",
+	.platform_name	= "mmp-pcm-audio",
+	.codec_name	= "wm8994-codec",
+	.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+				SND_SOC_DAIFMT_CBS_CFS,
+	.ops		= &brownstone_ops,
+	.init		= brownstone_wm8994_init,
+},
+};
+
+/* audio machine driver */
+static struct snd_soc_card brownstone = {
+	.name         = "brownstone",
+	.owner        = THIS_MODULE,
+	.dai_link     = brownstone_wm8994_dai,
+	.num_links    = ARRAY_SIZE(brownstone_wm8994_dai),
+
+	.controls = brownstone_dapm_control,
+	.num_controls = ARRAY_SIZE(brownstone_dapm_control),
+	.dapm_widgets = brownstone_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets),
+	.dapm_routes = brownstone_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(brownstone_audio_map),
+};
+
+static int brownstone_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	brownstone.dev = &pdev->dev;
+	ret = snd_soc_register_card(&brownstone);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+				ret);
+	return ret;
+}
+
+static int brownstone_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&brownstone);
+	return 0;
+}
+
+static struct platform_driver mmp_driver = {
+	.driver		= {
+		.name	= "brownstone-audio",
+		.pm     = &snd_soc_pm_ops,
+	},
+	.probe		= brownstone_probe,
+	.remove		= brownstone_remove,
+};
+
+module_platform_driver(mmp_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC Brownstone");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
new file mode 100644
index 000000000..3580d10c9
--- /dev/null
+++ b/sound/soc/pxa/corgi.c
@@ -0,0 +1,328 @@
+/*
+ * corgi.c  --  SoC audio for Corgi
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/i2c.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/corgi.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm8731.h"
+#include "pxa2xx-i2s.h"
+
+#define CORGI_HP        0
+#define CORGI_MIC       1
+#define CORGI_LINE      2
+#define CORGI_HEADSET   3
+#define CORGI_HP_OFF    4
+#define CORGI_SPK_ON    0
+#define CORGI_SPK_OFF   1
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define CORGI_AUDIO_CLOCK 12288000
+
+static int corgi_jack_func;
+static int corgi_spk_func;
+
+static void corgi_ext_control(struct snd_soc_dapm_context *dapm)
+{
+	snd_soc_dapm_mutex_lock(dapm);
+
+	/* set up jack connection */
+	switch (corgi_jack_func) {
+	case CORGI_HP:
+		/* set = unmute headphone */
+		gpio_set_value(CORGI_GPIO_MUTE_L, 1);
+		gpio_set_value(CORGI_GPIO_MUTE_R, 1);
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+		break;
+	case CORGI_MIC:
+		/* reset = mute headphone */
+		gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+		gpio_set_value(CORGI_GPIO_MUTE_R, 0);
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+		break;
+	case CORGI_LINE:
+		gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+		gpio_set_value(CORGI_GPIO_MUTE_R, 0);
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+		break;
+	case CORGI_HEADSET:
+		gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+		gpio_set_value(CORGI_GPIO_MUTE_R, 1);
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
+		break;
+	}
+
+	if (corgi_spk_func == CORGI_SPK_ON)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
+
+	/* signal a DAPM event */
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static int corgi_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* check the jack status at stream startup */
+	corgi_ext_control(&rtd->card->dapm);
+
+	return 0;
+}
+
+/* we need to unmute the HP at shutdown as the mute burns power on corgi */
+static void corgi_shutdown(struct snd_pcm_substream *substream)
+{
+	/* set = unmute headphone */
+	gpio_set_value(CORGI_GPIO_MUTE_L, 1);
+	gpio_set_value(CORGI_GPIO_MUTE_R, 1);
+}
+
+static int corgi_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 clk = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk = 11289600;
+		break;
+	}
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as input (unused) */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops corgi_ops = {
+	.startup = corgi_startup,
+	.hw_params = corgi_hw_params,
+	.shutdown = corgi_shutdown,
+};
+
+static int corgi_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = corgi_jack_func;
+	return 0;
+}
+
+static int corgi_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+	if (corgi_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	corgi_jack_func = ucontrol->value.integer.value[0];
+	corgi_ext_control(&card->dapm);
+	return 1;
+}
+
+static int corgi_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = corgi_spk_func;
+	return 0;
+}
+
+static int corgi_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card =  snd_kcontrol_chip(kcontrol);
+
+	if (corgi_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	corgi_spk_func = ucontrol->value.integer.value[0];
+	corgi_ext_control(&card->dapm);
+	return 1;
+}
+
+static int corgi_amp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+static int corgi_mic_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(CORGI_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+/* corgi machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", NULL),
+SND_SOC_DAPM_MIC("Mic Jack", corgi_mic_event),
+SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event),
+SND_SOC_DAPM_LINE("Line Jack", NULL),
+SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Corgi machine audio map (connections to the codec pins) */
+static const struct snd_soc_dapm_route corgi_audio_map[] = {
+
+	/* headset Jack  - in = micin, out = LHPOUT*/
+	{"Headset Jack", NULL, "LHPOUT"},
+
+	/* headphone connected to LHPOUT1, RHPOUT1 */
+	{"Headphone Jack", NULL, "LHPOUT"},
+	{"Headphone Jack", NULL, "RHPOUT"},
+
+	/* speaker connected to LOUT, ROUT */
+	{"Ext Spk", NULL, "ROUT"},
+	{"Ext Spk", NULL, "LOUT"},
+
+	/* mic is connected to MICIN (via right channel of headphone jack) */
+	{"MICIN", NULL, "Mic Jack"},
+
+	/* Same as the above but no mic bias for line signals */
+	{"MICIN", NULL, "Line Jack"},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+	"Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum corgi_enum[] = {
+	SOC_ENUM_SINGLE_EXT(5, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
+	SOC_ENUM_EXT("Jack Function", corgi_enum[0], corgi_get_jack,
+		corgi_set_jack),
+	SOC_ENUM_EXT("Speaker Function", corgi_enum[1], corgi_get_spk,
+		corgi_set_spk),
+};
+
+/* corgi digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link corgi_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731",
+	.cpu_dai_name = "pxa2xx-i2s",
+	.codec_dai_name = "wm8731-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm8731.0-001b",
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBS_CFS,
+	.ops = &corgi_ops,
+};
+
+/* corgi audio machine driver */
+static struct snd_soc_card corgi = {
+	.name = "Corgi",
+	.owner = THIS_MODULE,
+	.dai_link = &corgi_dai,
+	.num_links = 1,
+
+	.controls = wm8731_corgi_controls,
+	.num_controls = ARRAY_SIZE(wm8731_corgi_controls),
+	.dapm_widgets = wm8731_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets),
+	.dapm_routes = corgi_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(corgi_audio_map),
+	.fully_routed = true,
+};
+
+static int corgi_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &corgi;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+	return ret;
+}
+
+static int corgi_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static struct platform_driver corgi_driver = {
+	.driver		= {
+		.name	= "corgi-audio",
+		.pm     = &snd_soc_pm_ops,
+	},
+	.probe		= corgi_probe,
+	.remove		= corgi_remove,
+};
+
+module_platform_driver(corgi_driver);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Corgi");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:corgi-audio");
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
new file mode 100644
index 000000000..d72e124a3
--- /dev/null
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -0,0 +1,174 @@
+/*
+ * e740-wm9705.c  --  SoC audio for e740
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
+ *
+ *  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 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm9705.h"
+#include "pxa2xx-ac97.h"
+
+
+#define E740_AUDIO_OUT 1
+#define E740_AUDIO_IN  2
+
+static int e740_audio_power;
+
+static void e740_sync_audio_power(int status)
+{
+	gpio_set_value(GPIO_E740_WM9705_nAVDD2, !status);
+	gpio_set_value(GPIO_E740_AMP_ON, (status & E740_AUDIO_OUT) ? 1 : 0);
+	gpio_set_value(GPIO_E740_MIC_ON, (status & E740_AUDIO_IN) ? 1 : 0);
+}
+
+static int e740_mic_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		e740_audio_power |= E740_AUDIO_IN;
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		e740_audio_power &= ~E740_AUDIO_IN;
+
+	e740_sync_audio_power(e740_audio_power);
+
+	return 0;
+}
+
+static int e740_output_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		e740_audio_power |= E740_AUDIO_OUT;
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		e740_audio_power &= ~E740_AUDIO_OUT;
+
+	e740_sync_audio_power(e740_audio_power);
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget e740_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+	SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e740_output_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Output Amp", NULL, "LOUT"},
+	{"Output Amp", NULL, "ROUT"},
+	{"Output Amp", NULL, "MONOOUT"},
+
+	{"Speaker", NULL, "Output Amp"},
+	{"Headphone Jack", NULL, "Output Amp"},
+
+	{"MIC1", NULL, "Mic Amp"},
+	{"Mic Amp", NULL, "Mic (Internal)"},
+};
+
+static struct snd_soc_dai_link e740_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai_name = "pxa2xx-ac97",
+		.codec_dai_name = "wm9705-hifi",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9705-codec",
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
+		.codec_dai_name = "wm9705-aux",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9705-codec",
+	},
+};
+
+static struct snd_soc_card e740 = {
+	.name = "Toshiba e740",
+	.owner = THIS_MODULE,
+	.dai_link = e740_dai,
+	.num_links = ARRAY_SIZE(e740_dai),
+
+	.dapm_widgets = e740_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(e740_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+	.fully_routed = true,
+};
+
+static struct gpio e740_audio_gpios[] = {
+	{ GPIO_E740_MIC_ON, GPIOF_OUT_INIT_LOW, "Mic amp" },
+	{ GPIO_E740_AMP_ON, GPIOF_OUT_INIT_LOW, "Output amp" },
+	{ GPIO_E740_WM9705_nAVDD2, GPIOF_OUT_INIT_HIGH, "Audio power" },
+};
+
+static int e740_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &e740;
+	int ret;
+
+	ret = gpio_request_array(e740_audio_gpios,
+				 ARRAY_SIZE(e740_audio_gpios));
+	if (ret)
+		return ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+		gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios));
+	}
+	return ret;
+}
+
+static int e740_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios));
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static struct platform_driver e740_driver = {
+	.driver		= {
+		.name	= "e740-audio",
+		.pm     = &snd_soc_pm_ops,
+	},
+	.probe		= e740_probe,
+	.remove		= e740_remove,
+};
+
+module_platform_driver(e740_driver);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e740");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:e740-audio");
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
new file mode 100644
index 000000000..48f2d7c2e
--- /dev/null
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -0,0 +1,156 @@
+/*
+ * e750-wm9705.c  --  SoC audio for e750
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
+ *
+ *  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 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm9705.h"
+#include "pxa2xx-ac97.h"
+
+static int e750_spk_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		gpio_set_value(GPIO_E750_SPK_AMP_OFF, 0);
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		gpio_set_value(GPIO_E750_SPK_AMP_OFF, 1);
+
+	return 0;
+}
+
+static int e750_hp_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		gpio_set_value(GPIO_E750_HP_AMP_OFF, 0);
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		gpio_set_value(GPIO_E750_HP_AMP_OFF, 1);
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget e750_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+	SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e750_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e750_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Amp", NULL, "HPOUTL"},
+	{"Headphone Amp", NULL, "HPOUTR"},
+	{"Headphone Jack", NULL, "Headphone Amp"},
+
+	{"Speaker Amp", NULL, "MONOOUT"},
+	{"Speaker", NULL, "Speaker Amp"},
+
+	{"MIC1", NULL, "Mic (Internal)"},
+};
+
+static struct snd_soc_dai_link e750_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai_name = "pxa2xx-ac97",
+		.codec_dai_name = "wm9705-hifi",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9705-codec",
+		/* use ops to check startup state */
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
+		.codec_dai_name ="wm9705-aux",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9705-codec",
+	},
+};
+
+static struct snd_soc_card e750 = {
+	.name = "Toshiba e750",
+	.owner = THIS_MODULE,
+	.dai_link = e750_dai,
+	.num_links = ARRAY_SIZE(e750_dai),
+
+	.dapm_widgets = e750_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(e750_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+	.fully_routed = true,
+};
+
+static struct gpio e750_audio_gpios[] = {
+	{ GPIO_E750_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Headphone amp" },
+	{ GPIO_E750_SPK_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Speaker amp" },
+};
+
+static int e750_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &e750;
+	int ret;
+
+	ret = gpio_request_array(e750_audio_gpios,
+				 ARRAY_SIZE(e750_audio_gpios));
+	if (ret)
+		return ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+		gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios));
+	}
+	return ret;
+}
+
+static int e750_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios));
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static struct platform_driver e750_driver = {
+	.driver		= {
+		.name	= "e750-audio",
+		.pm     = &snd_soc_pm_ops,
+	},
+	.probe		= e750_probe,
+	.remove		= e750_remove,
+};
+
+module_platform_driver(e750_driver);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e750");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:e750-audio");
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
new file mode 100644
index 000000000..45d4bd46f
--- /dev/null
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -0,0 +1,155 @@
+/*
+ * e800-wm9712.c  --  SoC audio for e800
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
+ *
+ *  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 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-ac97.h"
+
+static int e800_spk_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		gpio_set_value(GPIO_E800_SPK_AMP_ON, 1);
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		gpio_set_value(GPIO_E800_SPK_AMP_ON, 0);
+
+	return 0;
+}
+
+static int e800_hp_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		gpio_set_value(GPIO_E800_HP_AMP_OFF, 0);
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		gpio_set_value(GPIO_E800_HP_AMP_OFF, 1);
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget e800_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic (Internal1)", NULL),
+	SND_SOC_DAPM_MIC("Mic (Internal2)", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Jack", NULL, "HPOUTL"},
+	{"Headphone Jack", NULL, "HPOUTR"},
+	{"Headphone Jack", NULL, "Headphone Amp"},
+
+	{"Speaker Amp", NULL, "MONOOUT"},
+	{"Speaker", NULL, "Speaker Amp"},
+
+	{"MIC1", NULL, "Mic (Internal1)"},
+	{"MIC2", NULL, "Mic (Internal2)"},
+};
+
+static struct snd_soc_dai_link e800_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai_name = "pxa2xx-ac97",
+		.codec_dai_name = "wm9712-hifi",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9712-codec",
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
+		.codec_dai_name ="wm9712-aux",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9712-codec",
+	},
+};
+
+static struct snd_soc_card e800 = {
+	.name = "Toshiba e800",
+	.owner = THIS_MODULE,
+	.dai_link = e800_dai,
+	.num_links = ARRAY_SIZE(e800_dai),
+
+	.dapm_widgets = e800_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(e800_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static struct gpio e800_audio_gpios[] = {
+	{ GPIO_E800_SPK_AMP_ON, GPIOF_OUT_INIT_HIGH, "Headphone amp" },
+	{ GPIO_E800_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Speaker amp" },
+};
+
+static int e800_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &e800;
+	int ret;
+
+	ret = gpio_request_array(e800_audio_gpios,
+				 ARRAY_SIZE(e800_audio_gpios));
+	if (ret)
+		return ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+		gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios));
+	}
+	return ret;
+}
+
+static int e800_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios));
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static struct platform_driver e800_driver = {
+	.driver		= {
+		.name	= "e800-audio",
+		.pm     = &snd_soc_pm_ops,
+	},
+	.probe		= e800_probe,
+	.remove		= e800_remove,
+};
+
+module_platform_driver(e800_driver);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e800");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:e800-audio");
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
new file mode 100644
index 000000000..64743a05a
--- /dev/null
+++ b/sound/soc/pxa/em-x270.c
@@ -0,0 +1,96 @@
+/*
+ * SoC audio driver for EM-X270, eXeda and CM-X300
+ *
+ * Copyright 2007, 2009 CompuLab, Ltd.
+ *
+ * Author: Mike Rapoport <mike@compulab.co.il>
+ *
+ * Copied from tosa.c:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-ac97.h"
+
+static struct snd_soc_dai_link em_x270_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai_name = "pxa2xx-ac97",
+		.codec_dai_name = "wm9712-hifi",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9712-codec",
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
+		.codec_dai_name ="wm9712-aux",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9712-codec",
+	},
+};
+
+static struct snd_soc_card em_x270 = {
+	.name = "EM-X270",
+	.owner = THIS_MODULE,
+	.dai_link = em_x270_dai,
+	.num_links = ARRAY_SIZE(em_x270_dai),
+};
+
+static struct platform_device *em_x270_snd_device;
+
+static int __init em_x270_init(void)
+{
+	int ret;
+
+	if (!(machine_is_em_x270() || machine_is_exeda()
+	      || machine_is_cm_x300()))
+		return -ENODEV;
+
+	em_x270_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!em_x270_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(em_x270_snd_device, &em_x270);
+	ret = platform_device_add(em_x270_snd_device);
+
+	if (ret)
+		platform_device_put(em_x270_snd_device);
+
+	return ret;
+}
+
+static void __exit em_x270_exit(void)
+{
+	platform_device_unregister(em_x270_snd_device);
+}
+
+module_init(em_x270_init);
+module_exit(em_x270_exit);
+
+/* Module information */
+MODULE_AUTHOR("Mike Rapoport");
+MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
new file mode 100644
index 000000000..9f8be7cd5
--- /dev/null
+++ b/sound/soc/pxa/hx4700.c
@@ -0,0 +1,229 @@
+/*
+ * SoC audio for HP iPAQ hx4700
+ *
+ * Copyright (c) 2009 Philipp Zabel
+ *
+ *  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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/hx4700.h>
+#include <asm/mach-types.h>
+#include "pxa2xx-i2s.h"
+
+#include "../codecs/ak4641.h"
+
+static struct snd_soc_jack hs_jack;
+
+/* Headphones jack detection DAPM pin */
+static struct snd_soc_jack_pin hs_jack_pin[] = {
+	{
+		.pin	= "Headphone Jack",
+		.mask	= SND_JACK_HEADPHONE,
+	},
+	{
+		.pin	= "Speaker",
+		/* disable speaker when hp jack is inserted */
+		.mask   = SND_JACK_HEADPHONE,
+		.invert	= 1,
+	},
+};
+
+/* Headphones jack detection GPIO */
+static struct snd_soc_jack_gpio hs_jack_gpio = {
+	.gpio		= GPIO75_HX4700_EARPHONE_nDET,
+	.invert		= true,
+	.name		= "hp-gpio",
+	.report		= SND_JACK_HEADPHONE,
+	.debounce_time	= 200,
+};
+
+/*
+ * iPAQ hx4700 uses I2S for capture and playback.
+ */
+static int hx4700_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 = 0;
+
+	/* set the I2S system clock as output */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+			SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	/* inform codec driver about clock freq *
+	 * (PXA I2S always uses divider 256)    */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 256 * params_rate(params),
+			SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops hx4700_ops = {
+	.hw_params = hx4700_hw_params,
+};
+
+static int hx4700_spk_power(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(GPIO107_HX4700_SPK_nSD, !!SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+static int hx4700_hp_power(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(GPIO92_HX4700_HP_DRIVER, !!SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+/* hx4700 machine dapm widgets */
+static const struct snd_soc_dapm_widget hx4700_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", hx4700_hp_power),
+	SND_SOC_DAPM_SPK("Speaker", hx4700_spk_power),
+	SND_SOC_DAPM_MIC("Built-in Microphone", NULL),
+};
+
+/* hx4700 machine audio_map */
+static const struct snd_soc_dapm_route hx4700_audio_map[] = {
+
+	/* Headphone connected to LOUT, ROUT */
+	{"Headphone Jack", NULL, "LOUT"},
+	{"Headphone Jack", NULL, "ROUT"},
+
+	/* Speaker connected to MOUT2 */
+	{"Speaker", NULL, "MOUT2"},
+
+	/* Microphone connected to MICIN */
+	{"MICIN", NULL, "Built-in Microphone"},
+	{"AIN", NULL, "MICOUT"},
+};
+
+/*
+ * Logic for a ak4641 as connected on a HP iPAQ hx4700
+ */
+static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd)
+{
+	int err;
+
+	/* Jack detection API stuff */
+	err = snd_soc_card_jack_new(rtd->card, "Headphone Jack",
+				    SND_JACK_HEADPHONE, &hs_jack, hs_jack_pin,
+				    ARRAY_SIZE(hs_jack_pin));
+	if (err)
+		return err;
+
+	err = snd_soc_jack_add_gpios(&hs_jack, 1, &hs_jack_gpio);
+
+	return err;
+}
+
+static int hx4700_card_remove(struct snd_soc_card *card)
+{
+	snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio);
+
+	return 0;
+}
+
+/* hx4700 digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link hx4700_dai = {
+	.name = "ak4641",
+	.stream_name = "AK4641",
+	.cpu_dai_name = "pxa2xx-i2s",
+	.codec_dai_name = "ak4641-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "ak4641.0-0012",
+	.init = hx4700_ak4641_init,
+	.dai_fmt = SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBS_CFS,
+	.ops = &hx4700_ops,
+};
+
+/* hx4700 audio machine driver */
+static struct snd_soc_card snd_soc_card_hx4700 = {
+	.name			= "iPAQ hx4700",
+	.owner			= THIS_MODULE,
+	.remove			= hx4700_card_remove,
+	.dai_link		= &hx4700_dai,
+	.num_links		= 1,
+	.dapm_widgets		= hx4700_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(hx4700_dapm_widgets),
+	.dapm_routes		= hx4700_audio_map,
+	.num_dapm_routes	= ARRAY_SIZE(hx4700_audio_map),
+	.fully_routed		= true,
+};
+
+static struct gpio hx4700_audio_gpios[] = {
+	{ GPIO107_HX4700_SPK_nSD, GPIOF_OUT_INIT_HIGH, "SPK_POWER" },
+	{ GPIO92_HX4700_HP_DRIVER, GPIOF_OUT_INIT_LOW, "EP_POWER" },
+};
+
+static int hx4700_audio_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	if (!machine_is_h4700())
+		return -ENODEV;
+
+	ret = gpio_request_array(hx4700_audio_gpios,
+				ARRAY_SIZE(hx4700_audio_gpios));
+	if (ret)
+		return ret;
+
+	snd_soc_card_hx4700.dev = &pdev->dev;
+	ret = snd_soc_register_card(&snd_soc_card_hx4700);
+	if (ret)
+		gpio_free_array(hx4700_audio_gpios,
+				ARRAY_SIZE(hx4700_audio_gpios));
+
+	return ret;
+}
+
+static int hx4700_audio_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&snd_soc_card_hx4700);
+
+	gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0);
+	gpio_set_value(GPIO107_HX4700_SPK_nSD, 0);
+
+	gpio_free_array(hx4700_audio_gpios, ARRAY_SIZE(hx4700_audio_gpios));
+	return 0;
+}
+
+static struct platform_driver hx4700_audio_driver = {
+	.driver	= {
+		.name = "hx4700-audio",
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe	= hx4700_audio_probe,
+	.remove	= hx4700_audio_remove,
+};
+
+module_platform_driver(hx4700_audio_driver);
+
+MODULE_AUTHOR("Philipp Zabel");
+MODULE_DESCRIPTION("ALSA SoC iPAQ hx4700");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:hx4700-audio");
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
new file mode 100644
index 000000000..29fabbfd2
--- /dev/null
+++ b/sound/soc/pxa/imote2.c
@@ -0,0 +1,104 @@
+
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm8940.h"
+#include "pxa2xx-i2s.h"
+
+static int imote2_asoc_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 clk = 0;
+	int ret;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk = 11289600;
+		break;
+	}
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as input (unused) */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, clk,
+		SND_SOC_CLOCK_OUT);
+
+	return ret;
+}
+
+static struct snd_soc_ops imote2_asoc_ops = {
+	.hw_params = imote2_asoc_hw_params,
+};
+
+static struct snd_soc_dai_link imote2_dai = {
+	.name = "WM8940",
+	.stream_name = "WM8940",
+	.cpu_dai_name = "pxa2xx-i2s",
+	.codec_dai_name = "wm8940-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm8940-codec.0-0034",
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBS_CFS,
+	.ops = &imote2_asoc_ops,
+};
+
+static struct snd_soc_card imote2 = {
+	.name = "Imote2",
+	.owner = THIS_MODULE,
+	.dai_link = &imote2_dai,
+	.num_links = 1,
+};
+
+static int imote2_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &imote2;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+	return ret;
+}
+
+static int imote2_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static struct platform_driver imote2_driver = {
+	.driver		= {
+		.name	= "imote2-audio",
+		.pm     = &snd_soc_pm_ops,
+	},
+	.probe		= imote2_probe,
+	.remove		= imote2_remove,
+};
+
+module_platform_driver(imote2_driver);
+
+MODULE_AUTHOR("Jonathan Cameron");
+MODULE_DESCRIPTION("ALSA SoC Imote 2");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imote2-audio");
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
new file mode 100644
index 000000000..241d0be42
--- /dev/null
+++ b/sound/soc/pxa/magician.c
@@ -0,0 +1,534 @@
+/*
+ * SoC audio for HTC Magician
+ *
+ * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
+ *
+ * based on spitz.c,
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/uda1380.h>
+
+#include <mach/magician.h>
+#include <asm/mach-types.h>
+#include "../codecs/uda1380.h"
+#include "pxa2xx-i2s.h"
+#include "pxa-ssp.h"
+
+#define MAGICIAN_MIC       0
+#define MAGICIAN_MIC_EXT   1
+
+static int magician_hp_switch;
+static int magician_spk_switch = 1;
+static int magician_in_sel = MAGICIAN_MIC;
+
+static void magician_ext_control(struct snd_soc_dapm_context *dapm)
+{
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	if (magician_spk_switch)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+	if (magician_hp_switch)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+
+	switch (magician_in_sel) {
+	case MAGICIAN_MIC:
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Mic");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Call Mic");
+		break;
+	case MAGICIAN_MIC_EXT:
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Call Mic");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Mic");
+		break;
+	}
+
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static int magician_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* check the jack status at stream startup */
+	magician_ext_control(&rtd->card->dapm);
+
+	return 0;
+}
+
+/*
+ * Magician uses SSP port for playback.
+ */
+static int magician_playback_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 acps, acds, width;
+	unsigned int div4 = PXA_SSP_CLK_SCDB_4;
+	int ret = 0;
+
+	width = snd_pcm_format_physical_width(params_format(params));
+
+	/*
+	 * rate = SSPSCLK / (2 * width(16 or 32))
+	 * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
+	 */
+	switch (params_rate(params)) {
+	case 8000:
+		/* off by a factor of 2: bug in the PXA27x audio clock? */
+		acps = 32842000;
+		switch (width) {
+		case 16:
+			/* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_16;
+			break;
+		default: /* 32 */
+			/* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_8;
+		}
+		break;
+	case 11025:
+		acps = 5622000;
+		switch (width) {
+		case 16:
+			/* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_4;
+			break;
+		default: /* 32 */
+			/* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_2;
+		}
+		break;
+	case 22050:
+		acps = 5622000;
+		switch (width) {
+		case 16:
+			/* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_2;
+			break;
+		default: /* 32 */
+			/* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_1;
+		}
+		break;
+	case 44100:
+		acps = 5622000;
+		switch (width) {
+		case 16:
+			/* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_2;
+			break;
+		default: /* 32 */
+			/* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_1;
+		}
+		break;
+	case 48000:
+		acps = 12235000;
+		switch (width) {
+		case 16:
+			/* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_2;
+			break;
+		default: /* 32 */
+			/* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_1;
+		}
+		break;
+	case 96000:
+	default:
+		acps = 12235000;
+		switch (width) {
+		case 16:
+			/* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_1;
+			break;
+		default: /* 32 */
+			/* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_2;
+			div4 = PXA_SSP_CLK_SCDB_1;
+			break;
+		}
+		break;
+	}
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+			SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width);
+	if (ret < 0)
+		return ret;
+
+	/* set audio clock as clock source */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
+			SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	/* set the SSP audio system clock ACDS divider */
+	ret = snd_soc_dai_set_clkdiv(cpu_dai,
+			PXA_SSP_AUDIO_DIV_ACDS, acds);
+	if (ret < 0)
+		return ret;
+
+	/* set the SSP audio system clock SCDB divider4 */
+	ret = snd_soc_dai_set_clkdiv(cpu_dai,
+			PXA_SSP_AUDIO_DIV_SCDB, div4);
+	if (ret < 0)
+		return ret;
+
+	/* set SSP audio pll clock */
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+/*
+ * Magician uses I2S for capture.
+ */
+static int magician_capture_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 = 0;
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as output */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+			SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops magician_capture_ops = {
+	.startup = magician_startup,
+	.hw_params = magician_capture_hw_params,
+};
+
+static struct snd_soc_ops magician_playback_ops = {
+	.startup = magician_startup,
+	.hw_params = magician_playback_hw_params,
+};
+
+static int magician_get_hp(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = magician_hp_switch;
+	return 0;
+}
+
+static int magician_set_hp(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+	if (magician_hp_switch == ucontrol->value.integer.value[0])
+		return 0;
+
+	magician_hp_switch = ucontrol->value.integer.value[0];
+	magician_ext_control(&card->dapm);
+	return 1;
+}
+
+static int magician_get_spk(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = magician_spk_switch;
+	return 0;
+}
+
+static int magician_set_spk(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+	if (magician_spk_switch == ucontrol->value.integer.value[0])
+		return 0;
+
+	magician_spk_switch = ucontrol->value.integer.value[0];
+	magician_ext_control(&card->dapm);
+	return 1;
+}
+
+static int magician_get_input(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = magician_in_sel;
+	return 0;
+}
+
+static int magician_set_input(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	if (magician_in_sel == ucontrol->value.integer.value[0])
+		return 0;
+
+	magician_in_sel = ucontrol->value.integer.value[0];
+
+	switch (magician_in_sel) {
+	case MAGICIAN_MIC:
+		gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1);
+		break;
+	case MAGICIAN_MIC_EXT:
+		gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0);
+	}
+
+	return 1;
+}
+
+static int magician_spk_power(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+static int magician_hp_power(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+static int magician_mic_bias(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+/* magician machine dapm widgets */
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
+	SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
+	SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
+	SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
+};
+
+/* magician machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* Headphone connected to VOUTL, VOUTR */
+	{"Headphone Jack", NULL, "VOUTL"},
+	{"Headphone Jack", NULL, "VOUTR"},
+
+	/* Speaker connected to VOUTL, VOUTR */
+	{"Speaker", NULL, "VOUTL"},
+	{"Speaker", NULL, "VOUTR"},
+
+	/* Mics are connected to VINM */
+	{"VINM", NULL, "Headset Mic"},
+	{"VINM", NULL, "Call Mic"},
+};
+
+static const char *input_select[] = {"Call Mic", "Headset Mic"};
+static const struct soc_enum magician_in_sel_enum =
+	SOC_ENUM_SINGLE_EXT(2, input_select);
+
+static const struct snd_kcontrol_new uda1380_magician_controls[] = {
+	SOC_SINGLE_BOOL_EXT("Headphone Switch",
+			(unsigned long)&magician_hp_switch,
+			magician_get_hp, magician_set_hp),
+	SOC_SINGLE_BOOL_EXT("Speaker Switch",
+			(unsigned long)&magician_spk_switch,
+			magician_get_spk, magician_set_spk),
+	SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
+			magician_get_input, magician_set_input),
+};
+
+/* magician digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link magician_dai[] = {
+{
+	.name = "uda1380",
+	.stream_name = "UDA1380 Playback",
+	.cpu_dai_name = "pxa-ssp-dai.0",
+	.codec_dai_name = "uda1380-hifi-playback",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "uda1380-codec.0-0018",
+	.ops = &magician_playback_ops,
+},
+{
+	.name = "uda1380",
+	.stream_name = "UDA1380 Capture",
+	.cpu_dai_name = "pxa2xx-i2s",
+	.codec_dai_name = "uda1380-hifi-capture",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "uda1380-codec.0-0018",
+	.ops = &magician_capture_ops,
+}
+};
+
+/* magician audio machine driver */
+static struct snd_soc_card snd_soc_card_magician = {
+	.name = "Magician",
+	.owner = THIS_MODULE,
+	.dai_link = magician_dai,
+	.num_links = ARRAY_SIZE(magician_dai),
+
+	.controls = uda1380_magician_controls,
+	.num_controls = ARRAY_SIZE(uda1380_magician_controls),
+	.dapm_widgets = uda1380_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+	.fully_routed = true,
+};
+
+static struct platform_device *magician_snd_device;
+
+/*
+ * FIXME: move into magician board file once merged into the pxa tree
+ */
+static struct uda1380_platform_data uda1380_info = {
+	.gpio_power = EGPIO_MAGICIAN_CODEC_POWER,
+	.gpio_reset = EGPIO_MAGICIAN_CODEC_RESET,
+	.dac_clk    = UDA1380_DAC_CLK_WSPLL,
+};
+
+static struct i2c_board_info i2c_board_info[] = {
+	{
+		I2C_BOARD_INFO("uda1380", 0x18),
+		.platform_data = &uda1380_info,
+	},
+};
+
+static int __init magician_init(void)
+{
+	int ret;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+
+	if (!machine_is_magician())
+		return -ENODEV;
+
+	adapter = i2c_get_adapter(0);
+	if (!adapter)
+		return -ENODEV;
+	client = i2c_new_device(adapter, i2c_board_info);
+	i2c_put_adapter(adapter);
+	if (!client)
+		return -ENODEV;
+
+	ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
+	if (ret)
+		goto err_request_spk;
+	ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER");
+	if (ret)
+		goto err_request_ep;
+	ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER");
+	if (ret)
+		goto err_request_mic;
+	ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0");
+	if (ret)
+		goto err_request_in_sel0;
+	ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1");
+	if (ret)
+		goto err_request_in_sel1;
+
+	gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
+
+	magician_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!magician_snd_device) {
+		ret = -ENOMEM;
+		goto err_pdev;
+	}
+
+	platform_set_drvdata(magician_snd_device, &snd_soc_card_magician);
+	ret = platform_device_add(magician_snd_device);
+	if (ret) {
+		platform_device_put(magician_snd_device);
+		goto err_pdev;
+	}
+
+	return 0;
+
+err_pdev:
+	gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+err_request_in_sel1:
+	gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+err_request_in_sel0:
+	gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+err_request_mic:
+	gpio_free(EGPIO_MAGICIAN_EP_POWER);
+err_request_ep:
+	gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+err_request_spk:
+	return ret;
+}
+
+static void __exit magician_exit(void)
+{
+	platform_device_unregister(magician_snd_device);
+
+	gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
+	gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
+	gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
+
+	gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+	gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+	gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+	gpio_free(EGPIO_MAGICIAN_EP_POWER);
+	gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+}
+
+module_init(magician_init);
+module_exit(magician_exit);
+
+MODULE_AUTHOR("Philipp Zabel");
+MODULE_DESCRIPTION("ALSA SoC Magician");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
new file mode 100644
index 000000000..a9615a574
--- /dev/null
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -0,0 +1,214 @@
+/*
+ * Handles the Mitac mioa701 SoC system
+ *
+ * Copyright (C) 2008 Robert Jarzmik
+ *
+ * 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 in version 2 of the License.
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ * This is a little schema of the sound interconnections :
+ *
+ *    Sagem X200                 Wolfson WM9713
+ *    +--------+             +-------------------+      Rear Speaker
+ *    |        |             |                   |           /-+
+ *    |        +--->----->---+MONOIN         SPKL+--->----+-+  |
+ *    |  GSM   |             |                   |        | |  |
+ *    |        +--->----->---+PCBEEP         SPKR+--->----+-+  |
+ *    |  CHIP  |             |                   |           \-+
+ *    |        +---<-----<---+MONO               |
+ *    |        |             |                   |      Front Speaker
+ *    +--------+             |                   |           /-+
+ *                           |                HPL+--->----+-+  |
+ *                           |                   |        | |  |
+ *                           |               OUT3+--->----+-+  |
+ *                           |                   |           \-+
+ *                           |                   |
+ *                           |                   |     Front Micro
+ *                           |                   |         +
+ *                           |               MIC1+-----<--+o+
+ *                           |                   |         +
+ *                           +-------------------+        ---
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/ac97_codec.h>
+
+#include "pxa2xx-ac97.h"
+#include "../codecs/wm9713.h"
+
+#define AC97_GPIO_PULL		0x58
+
+/* Use GPIO8 for rear speaker amplifier */
+static int rear_amp_power(struct snd_soc_codec *codec, int power)
+{
+	unsigned short reg;
+
+	if (power) {
+		reg = snd_soc_read(codec, AC97_GPIO_CFG);
+		snd_soc_write(codec, AC97_GPIO_CFG, reg | 0x0100);
+		reg = snd_soc_read(codec, AC97_GPIO_PULL);
+		snd_soc_write(codec, AC97_GPIO_PULL, reg | (1<<15));
+	} else {
+		reg = snd_soc_read(codec, AC97_GPIO_CFG);
+		snd_soc_write(codec, AC97_GPIO_CFG, reg & ~0x0100);
+		reg = snd_soc_read(codec, AC97_GPIO_PULL);
+		snd_soc_write(codec, AC97_GPIO_PULL, reg & ~(1<<15));
+	}
+
+	return 0;
+}
+
+static int rear_amp_event(struct snd_soc_dapm_widget *widget,
+			  struct snd_kcontrol *kctl, int event)
+{
+	struct snd_soc_codec *codec = widget->dapm->card->rtd[0].codec;
+
+	return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event));
+}
+
+/* mioa701 machine dapm widgets */
+static const struct snd_soc_dapm_widget mioa701_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Front Speaker", NULL),
+	SND_SOC_DAPM_SPK("Rear Speaker", rear_amp_event),
+	SND_SOC_DAPM_MIC("Headset", NULL),
+	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+	SND_SOC_DAPM_LINE("GSM Line In", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Front Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Call Mic */
+	{"Mic Bias", NULL, "Front Mic"},
+	{"MIC1", NULL, "Mic Bias"},
+
+	/* Headset Mic */
+	{"LINEL", NULL, "Headset Mic"},
+	{"LINER", NULL, "Headset Mic"},
+
+	/* GSM Module */
+	{"MONOIN", NULL, "GSM Line Out"},
+	{"PCBEEP", NULL, "GSM Line Out"},
+	{"GSM Line In", NULL, "MONO"},
+
+	/* headphone connected to HPL, HPR */
+	{"Headset", NULL, "HPL"},
+	{"Headset", NULL, "HPR"},
+
+	/* front speaker connected to HPL, OUT3 */
+	{"Front Speaker", NULL, "HPL"},
+	{"Front Speaker", NULL, "OUT3"},
+
+	/* rear speaker connected to SPKL, SPKR */
+	{"Rear Speaker", NULL, "SPKL"},
+	{"Rear Speaker", NULL, "SPKR"},
+};
+
+static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+
+	/* Prepare GPIO8 for rear speaker amplifier */
+	snd_soc_update_bits(codec, AC97_GPIO_CFG, 0x100, 0x100);
+
+	/* Prepare MIC input */
+	snd_soc_update_bits(codec, AC97_3D_CONTROL, 0xc000, 0xc000);
+
+	return 0;
+}
+
+static struct snd_soc_ops mioa701_ops;
+
+static struct snd_soc_dai_link mioa701_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai_name = "pxa2xx-ac97",
+		.codec_dai_name = "wm9713-hifi",
+		.codec_name = "wm9713-codec",
+		.init = mioa701_wm9713_init,
+		.platform_name = "pxa-pcm-audio",
+		.ops = &mioa701_ops,
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
+		.codec_dai_name ="wm9713-aux",
+		.codec_name = "wm9713-codec",
+		.platform_name = "pxa-pcm-audio",
+		.ops = &mioa701_ops,
+	},
+};
+
+static struct snd_soc_card mioa701 = {
+	.name = "MioA701",
+	.owner = THIS_MODULE,
+	.dai_link = mioa701_dai,
+	.num_links = ARRAY_SIZE(mioa701_dai),
+
+	.dapm_widgets = mioa701_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(mioa701_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static int mioa701_wm9713_probe(struct platform_device *pdev)
+{
+	int rc;
+
+	if (!machine_is_mioa701())
+		return -ENODEV;
+
+	mioa701.dev = &pdev->dev;
+	rc =  snd_soc_register_card(&mioa701);
+	if (!rc)
+		dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
+			 "lead to overheating and possible destruction of your device."
+			 " Do not use without a good knowledge of mio's board design!\n");
+	return rc;
+}
+
+static int mioa701_wm9713_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static struct platform_driver mioa701_wm9713_driver = {
+	.probe		= mioa701_wm9713_probe,
+	.remove		= mioa701_wm9713_remove,
+	.driver		= {
+		.name		= "mioa701-wm9713",
+		.pm     = &snd_soc_pm_ops,
+	},
+};
+
+module_platform_driver(mioa701_wm9713_driver);
+
+/* Module information */
+MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)");
+MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
new file mode 100644
index 000000000..1eb45dcfb
--- /dev/null
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -0,0 +1,257 @@
+/*
+ * linux/sound/soc/pxa/mmp-pcm.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ */
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
+#include <linux/platform_data/dma-mmp_tdma.h>
+#include <linux/platform_data/mmp_audio.h>
+
+#include <sound/pxa2xx-lib.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+struct mmp_dma_data {
+	int ssp_id;
+	struct resource *dma_res;
+};
+
+#define MMP_PCM_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)
+
+static struct snd_pcm_hardware mmp_pcm_hardware[] = {
+	{
+		.info			= MMP_PCM_INFO,
+		.period_bytes_min	= 1024,
+		.period_bytes_max	= 2048,
+		.periods_min		= 2,
+		.periods_max		= 32,
+		.buffer_bytes_max	= 4096,
+		.fifo_size		= 32,
+	},
+	{
+		.info			= MMP_PCM_INFO,
+		.period_bytes_min	= 1024,
+		.period_bytes_max	= 2048,
+		.periods_min		= 2,
+		.periods_max		= 32,
+		.buffer_bytes_max	= 4096,
+		.fifo_size		= 32,
+	},
+};
+
+static int mmp_pcm_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params)
+{
+	struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+	struct dma_slave_config slave_config;
+	int ret;
+
+	ret =
+	    snd_dmaengine_pcm_prepare_slave_config(substream, params,
+						   &slave_config);
+	if (ret)
+		return ret;
+
+	ret = dmaengine_slave_config(chan, &slave_config);
+	if (ret)
+		return ret;
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+	return 0;
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+	struct mmp_dma_data *dma_data = param;
+	bool found = false;
+	char *devname;
+
+	devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name,
+		dma_data->ssp_id);
+	if ((strcmp(dev_name(chan->device->dev), devname) == 0) &&
+		(chan->chan_id == dma_data->dma_res->start)) {
+		found = true;
+	}
+
+	kfree(devname);
+	return found;
+}
+
+static int mmp_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct platform_device *pdev = to_platform_device(rtd->platform->dev);
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct mmp_dma_data dma_data;
+	struct resource *r;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream);
+	if (!r)
+		return -EBUSY;
+
+	snd_soc_set_runtime_hwparams(substream,
+				&mmp_pcm_hardware[substream->stream]);
+
+	dma_data.dma_res = r;
+	dma_data.ssp_id = cpu_dai->id;
+
+	return snd_dmaengine_pcm_open_request_chan(substream, filter,
+		    &dma_data);
+}
+
+static int mmp_pcm_mmap(struct snd_pcm_substream *substream,
+			 struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned long off = vma->vm_pgoff;
+
+	vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+	return remap_pfn_range(vma, vma->vm_start,
+		__phys_to_pfn(runtime->dma_addr) + off,
+		vma->vm_end - vma->vm_start, vma->vm_page_prot);
+}
+
+static struct snd_pcm_ops mmp_pcm_ops = {
+	.open		= mmp_pcm_open,
+	.close		= snd_dmaengine_pcm_close_release_chan,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= mmp_pcm_hw_params,
+	.trigger	= snd_dmaengine_pcm_trigger,
+	.pointer	= snd_dmaengine_pcm_pointer,
+	.mmap		= mmp_pcm_mmap,
+};
+
+static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+	int stream;
+	struct gen_pool *gpool;
+
+	gpool = sram_get_gpool("asram");
+	if (!gpool)
+		return;
+
+	for (stream = 0; stream < 2; stream++) {
+		size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+
+		substream = pcm->streams[stream].substream;
+		if (!substream)
+			continue;
+
+		buf = &substream->dma_buffer;
+		if (!buf->area)
+			continue;
+		gen_pool_free(gpool, (unsigned long)buf->area, size);
+		buf->area = NULL;
+	}
+
+	return;
+}
+
+static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream,
+								int stream)
+{
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+	struct gen_pool *gpool;
+
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = substream->pcm->card->dev;
+	buf->private_data = NULL;
+
+	gpool = sram_get_gpool("asram");
+	if (!gpool)
+		return -ENOMEM;
+
+	buf->area = gen_pool_dma_alloc(gpool, size, &buf->addr);
+	if (!buf->area)
+		return -ENOMEM;
+	buf->bytes = size;
+	return 0;
+}
+
+static int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_pcm *pcm = rtd->pcm;
+	int ret = 0, stream;
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+
+		ret = mmp_pcm_preallocate_dma_buffer(substream,	stream);
+		if (ret)
+			goto err;
+	}
+
+	return 0;
+
+err:
+	mmp_pcm_free_dma_buffers(pcm);
+	return ret;
+}
+
+static struct snd_soc_platform_driver mmp_soc_platform = {
+	.ops		= &mmp_pcm_ops,
+	.pcm_new	= mmp_pcm_new,
+	.pcm_free	= mmp_pcm_free_dma_buffers,
+};
+
+static int mmp_pcm_probe(struct platform_device *pdev)
+{
+	struct mmp_audio_platdata *pdata = pdev->dev.platform_data;
+
+	if (pdata) {
+		mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max =
+						pdata->buffer_max_playback;
+		mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max =
+						pdata->period_max_playback;
+		mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max =
+						pdata->buffer_max_capture;
+		mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max =
+						pdata->period_max_capture;
+	}
+	return snd_soc_register_platform(&pdev->dev, &mmp_soc_platform);
+}
+
+static int mmp_pcm_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver mmp_pcm_driver = {
+	.driver = {
+		.name = "mmp-pcm-audio",
+	},
+
+	.probe = mmp_pcm_probe,
+	.remove = mmp_pcm_remove,
+};
+
+module_platform_driver(mmp_pcm_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP Soc Audio DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
new file mode 100644
index 000000000..eca60c297
--- /dev/null
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -0,0 +1,484 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.c
+ * Base on pxa2xx-ssp.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ */
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/slab.h>
+#include <linux/pxa2xx_ssp.h>
+#include <linux/io.h>
+#include <linux/dmaengine.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
+#include "mmp-sspa.h"
+
+/*
+ * SSPA audio private data
+ */
+struct sspa_priv {
+	struct ssp_device *sspa;
+	struct snd_dmaengine_dai_dma_data *dma_params;
+	struct clk *audio_clk;
+	struct clk *sysclk;
+	int dai_fmt;
+	int running_cnt;
+};
+
+static void mmp_sspa_write_reg(struct ssp_device *sspa, u32 reg, u32 val)
+{
+	__raw_writel(val, sspa->mmio_base + reg);
+}
+
+static u32 mmp_sspa_read_reg(struct ssp_device *sspa, u32 reg)
+{
+	return __raw_readl(sspa->mmio_base + reg);
+}
+
+static void mmp_sspa_tx_enable(struct ssp_device *sspa)
+{
+	unsigned int sspa_sp;
+
+	sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+	sspa_sp |= SSPA_SP_S_EN;
+	sspa_sp |= SSPA_SP_WEN;
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_tx_disable(struct ssp_device *sspa)
+{
+	unsigned int sspa_sp;
+
+	sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+	sspa_sp &= ~SSPA_SP_S_EN;
+	sspa_sp |= SSPA_SP_WEN;
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_enable(struct ssp_device *sspa)
+{
+	unsigned int sspa_sp;
+
+	sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+	sspa_sp |= SSPA_SP_S_EN;
+	sspa_sp |= SSPA_SP_WEN;
+	mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_disable(struct ssp_device *sspa)
+{
+	unsigned int sspa_sp;
+
+	sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+	sspa_sp &= ~SSPA_SP_S_EN;
+	sspa_sp |= SSPA_SP_WEN;
+	mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static int mmp_sspa_startup(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+	clk_enable(priv->sysclk);
+	clk_enable(priv->sspa->clk);
+
+	return 0;
+}
+
+static void mmp_sspa_shutdown(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable(priv->sspa->clk);
+	clk_disable(priv->sysclk);
+
+	return;
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int mmp_sspa_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+				    int clk_id, unsigned int freq, int dir)
+{
+	struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret = 0;
+
+	switch (clk_id) {
+	case MMP_SSPA_CLK_AUDIO:
+		ret = clk_set_rate(priv->audio_clk, freq);
+		if (ret)
+			return ret;
+		break;
+	case MMP_SSPA_CLK_PLL:
+	case MMP_SSPA_CLK_VCXO:
+		/* not support yet */
+		return -EINVAL;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int mmp_sspa_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
+				 int source, unsigned int freq_in,
+				 unsigned int freq_out)
+{
+	struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret = 0;
+
+	switch (pll_id) {
+	case MMP_SYSCLK:
+		ret = clk_set_rate(priv->sysclk, freq_out);
+		if (ret)
+			return ret;
+		break;
+	case MMP_SSPA_CLK:
+		ret = clk_set_rate(priv->sspa->clk, freq_out);
+		if (ret)
+			return ret;
+		break;
+	default:
+		return -ENODEV;
+	}
+
+	return 0;
+}
+
+/*
+ * Set up the sspa dai format. The sspa port must be inactive
+ * before calling this function as the physical
+ * interface format is changed.
+ */
+static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+				 unsigned int fmt)
+{
+	struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *sspa = sspa_priv->sspa;
+	u32 sspa_sp, sspa_ctrl;
+
+	/* check if we need to change anything at all */
+	if (sspa_priv->dai_fmt == fmt)
+		return 0;
+
+	/* we can only change the settings if the port is not in use */
+	if ((mmp_sspa_read_reg(sspa, SSPA_TXSP) & SSPA_SP_S_EN) ||
+	    (mmp_sspa_read_reg(sspa, SSPA_RXSP) & SSPA_SP_S_EN)) {
+		dev_err(&sspa->pdev->dev,
+			"can't change hardware dai format: stream is in use\n");
+		return -EINVAL;
+	}
+
+	/* reset port settings */
+	sspa_sp   = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH;
+	sspa_ctrl = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		sspa_sp |= SSPA_SP_MSL;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		sspa_sp |= SSPA_SP_FSP;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		sspa_sp |= SSPA_TXSP_FPER(63);
+		sspa_sp |= SSPA_SP_FWID(31);
+		sspa_ctrl |= SSPA_CTL_XDATDLY(1);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+	mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+	sspa_sp &= ~(SSPA_SP_S_RST | SSPA_SP_FFLUSH);
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+	mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+	/*
+	 * FIXME: hw issue, for the tx serial port,
+	 * can not config the master/slave mode;
+	 * so must clean this bit.
+	 * The master/slave mode has been set in the
+	 * rx port.
+	 */
+	sspa_sp &= ~SSPA_SP_MSL;
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+
+	mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+	mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+
+	/* Since we are configuring the timings for the format by hand
+	 * we have to defer some things until hw_params() where we
+	 * know parameters like the sample size.
+	 */
+	sspa_priv->dai_fmt = fmt;
+	return 0;
+}
+
+/*
+ * Set the SSPA audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int mmp_sspa_hw_params(struct snd_pcm_substream *substream,
+			       struct snd_pcm_hw_params *params,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+	struct ssp_device *sspa = sspa_priv->sspa;
+	struct snd_dmaengine_dai_dma_data *dma_params;
+	u32 sspa_ctrl;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_TXCTL);
+	else
+		sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_RXCTL);
+
+	sspa_ctrl &= ~SSPA_CTL_XFRLEN1_MASK;
+	sspa_ctrl |= SSPA_CTL_XFRLEN1(params_channels(params) - 1);
+	sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK;
+	sspa_ctrl |= SSPA_CTL_XWDLEN1(SSPA_CTL_32_BITS);
+	sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_8_BITS);
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_16_BITS);
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_20_BITS);
+		break;
+	case SNDRV_PCM_FORMAT_S24_3LE:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_24_BITS);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_32_BITS);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+		mmp_sspa_write_reg(sspa, SSPA_TXFIFO_LL, 0x1);
+	} else {
+		mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+		mmp_sspa_write_reg(sspa, SSPA_RXFIFO_UL, 0x0);
+	}
+
+	dma_params = &sspa_priv->dma_params[substream->stream];
+	dma_params->addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+				(sspa->phys_base + SSPA_TXD) :
+				(sspa->phys_base + SSPA_RXD);
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params);
+	return 0;
+}
+
+static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd,
+			     struct snd_soc_dai *dai)
+{
+	struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+	struct ssp_device *sspa = sspa_priv->sspa;
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		/*
+		 * whatever playback or capture, must enable rx.
+		 * this is a hw issue, so need check if rx has been
+		 * enabled or not; if has been enabled by another
+		 * stream, do not enable again.
+		 */
+		if (!sspa_priv->running_cnt)
+			mmp_sspa_rx_enable(sspa);
+
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			mmp_sspa_tx_enable(sspa);
+
+		sspa_priv->running_cnt++;
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		sspa_priv->running_cnt--;
+
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			mmp_sspa_tx_disable(sspa);
+
+		/* have no capture stream, disable rx port */
+		if (!sspa_priv->running_cnt)
+			mmp_sspa_rx_disable(sspa);
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static int mmp_sspa_probe(struct snd_soc_dai *dai)
+{
+	struct sspa_priv *priv = dev_get_drvdata(dai->dev);
+
+	snd_soc_dai_set_drvdata(dai, priv);
+	return 0;
+
+}
+
+#define MMP_SSPA_RATES SNDRV_PCM_RATE_8000_192000
+#define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+		SNDRV_PCM_FMTBIT_S16_LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | \
+		SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops mmp_sspa_dai_ops = {
+	.startup	= mmp_sspa_startup,
+	.shutdown	= mmp_sspa_shutdown,
+	.trigger	= mmp_sspa_trigger,
+	.hw_params	= mmp_sspa_hw_params,
+	.set_sysclk	= mmp_sspa_set_dai_sysclk,
+	.set_pll	= mmp_sspa_set_dai_pll,
+	.set_fmt	= mmp_sspa_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver mmp_sspa_dai = {
+	.probe = mmp_sspa_probe,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 128,
+		.rates = MMP_SSPA_RATES,
+		.formats = MMP_SSPA_FORMATS,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = MMP_SSPA_RATES,
+		.formats = MMP_SSPA_FORMATS,
+	},
+	.ops = &mmp_sspa_dai_ops,
+};
+
+static const struct snd_soc_component_driver mmp_sspa_component = {
+	.name		= "mmp-sspa",
+};
+
+static int asoc_mmp_sspa_probe(struct platform_device *pdev)
+{
+	struct sspa_priv *priv;
+	struct resource *res;
+
+	priv = devm_kzalloc(&pdev->dev,
+				sizeof(struct sspa_priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->sspa = devm_kzalloc(&pdev->dev,
+				sizeof(struct ssp_device), GFP_KERNEL);
+	if (priv->sspa == NULL)
+		return -ENOMEM;
+
+	priv->dma_params = devm_kzalloc(&pdev->dev,
+			2 * sizeof(struct snd_dmaengine_dai_dma_data),
+			GFP_KERNEL);
+	if (priv->dma_params == NULL)
+		return -ENOMEM;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	priv->sspa->mmio_base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(priv->sspa->mmio_base))
+		return PTR_ERR(priv->sspa->mmio_base);
+
+	priv->sspa->clk = devm_clk_get(&pdev->dev, NULL);
+	if (IS_ERR(priv->sspa->clk))
+		return PTR_ERR(priv->sspa->clk);
+
+	priv->audio_clk = clk_get(NULL, "mmp-audio");
+	if (IS_ERR(priv->audio_clk))
+		return PTR_ERR(priv->audio_clk);
+
+	priv->sysclk = clk_get(NULL, "mmp-sysclk");
+	if (IS_ERR(priv->sysclk)) {
+		clk_put(priv->audio_clk);
+		return PTR_ERR(priv->sysclk);
+	}
+	clk_enable(priv->audio_clk);
+	priv->dai_fmt = (unsigned int) -1;
+	platform_set_drvdata(pdev, priv);
+
+	return devm_snd_soc_register_component(&pdev->dev, &mmp_sspa_component,
+					       &mmp_sspa_dai, 1);
+}
+
+static int asoc_mmp_sspa_remove(struct platform_device *pdev)
+{
+	struct sspa_priv *priv = platform_get_drvdata(pdev);
+
+	clk_disable(priv->audio_clk);
+	clk_put(priv->audio_clk);
+	clk_put(priv->sysclk);
+	return 0;
+}
+
+static struct platform_driver asoc_mmp_sspa_driver = {
+	.driver = {
+		.name = "mmp-sspa-dai",
+	},
+	.probe = asoc_mmp_sspa_probe,
+	.remove = asoc_mmp_sspa_remove,
+};
+
+module_platform_driver(asoc_mmp_sspa_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP SSPA SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mmp-sspa.h b/sound/soc/pxa/mmp-sspa.h
new file mode 100644
index 000000000..ea365cb9e
--- /dev/null
+++ b/sound/soc/pxa/mmp-sspa.h
@@ -0,0 +1,92 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.h
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ */
+#ifndef _MMP_SSPA_H
+#define _MMP_SSPA_H
+
+/*
+ * SSPA Registers
+ */
+#define SSPA_RXD		(0x00)
+#define SSPA_RXID		(0x04)
+#define SSPA_RXCTL		(0x08)
+#define SSPA_RXSP		(0x0c)
+#define SSPA_RXFIFO_UL		(0x10)
+#define SSPA_RXINT_MASK		(0x14)
+#define SSPA_RXC		(0x18)
+#define SSPA_RXFIFO_NOFS	(0x1c)
+#define SSPA_RXFIFO_SIZE	(0x20)
+
+#define SSPA_TXD		(0x80)
+#define SSPA_TXID		(0x84)
+#define SSPA_TXCTL		(0x88)
+#define SSPA_TXSP		(0x8c)
+#define SSPA_TXFIFO_LL		(0x90)
+#define SSPA_TXINT_MASK		(0x94)
+#define SSPA_TXC		(0x98)
+#define SSPA_TXFIFO_NOFS	(0x9c)
+#define SSPA_TXFIFO_SIZE	(0xa0)
+
+/* SSPA Control Register */
+#define	SSPA_CTL_XPH		(1 << 31)	/* Read Phase */
+#define	SSPA_CTL_XFIG		(1 << 15)	/* Transmit Zeros when FIFO Empty */
+#define	SSPA_CTL_JST		(1 << 3)	/* Audio Sample Justification */
+#define	SSPA_CTL_XFRLEN2_MASK	(7 << 24)
+#define	SSPA_CTL_XFRLEN2(x)	((x) << 24)	/* Transmit Frame Length in Phase 2 */
+#define	SSPA_CTL_XWDLEN2_MASK	(7 << 21)
+#define	SSPA_CTL_XWDLEN2(x)	((x) << 21)	/* Transmit Word Length in Phase 2 */
+#define	SSPA_CTL_XDATDLY(x)	((x) << 19)	/* Tansmit Data Delay */
+#define	SSPA_CTL_XSSZ2_MASK	(7 << 16)
+#define	SSPA_CTL_XSSZ2(x)	((x) << 16)	/* Transmit Sample Audio Size */
+#define	SSPA_CTL_XFRLEN1_MASK	(7 << 8)
+#define	SSPA_CTL_XFRLEN1(x)	((x) << 8)	/* Transmit Frame Length in Phase 1 */
+#define	SSPA_CTL_XWDLEN1_MASK	(7 << 5)
+#define	SSPA_CTL_XWDLEN1(x)	((x) << 5)	/* Transmit Word Length in Phase 1 */
+#define	SSPA_CTL_XSSZ1_MASK	(7 << 0)
+#define	SSPA_CTL_XSSZ1(x)	((x) << 0)	/* XSSZ1 */
+
+#define SSPA_CTL_8_BITS		(0x0)		/* Sample Size */
+#define SSPA_CTL_12_BITS	(0x1)
+#define SSPA_CTL_16_BITS	(0x2)
+#define SSPA_CTL_20_BITS	(0x3)
+#define SSPA_CTL_24_BITS	(0x4)
+#define SSPA_CTL_32_BITS	(0x5)
+
+/* SSPA Serial Port Register */
+#define	SSPA_SP_WEN		(1 << 31)	/* Write Configuration Enable */
+#define	SSPA_SP_MSL		(1 << 18)	/* Master Slave Configuration */
+#define	SSPA_SP_CLKP		(1 << 17)	/* CLKP Polarity Clock Edge Select */
+#define	SSPA_SP_FSP		(1 << 16)	/* FSP Polarity Clock Edge Select */
+#define	SSPA_SP_FFLUSH		(1 << 2)	/* FIFO Flush */
+#define	SSPA_SP_S_RST		(1 << 1)	/* Active High Reset Signal */
+#define	SSPA_SP_S_EN		(1 << 0)	/* Serial Clock Domain Enable */
+#define	SSPA_SP_FWID(x)		((x) << 20)	/* Frame-Sync Width */
+#define	SSPA_TXSP_FPER(x)	((x) << 4)	/* Frame-Sync Active */
+
+/* sspa clock sources */
+#define MMP_SSPA_CLK_PLL	0
+#define MMP_SSPA_CLK_VCXO	1
+#define MMP_SSPA_CLK_AUDIO	3
+
+/* sspa pll id */
+#define MMP_SYSCLK		0
+#define MMP_SSPA_CLK		1
+
+#endif /* _MMP_SSPA_H */
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
new file mode 100644
index 000000000..c20bbc042
--- /dev/null
+++ b/sound/soc/pxa/palm27x.c
@@ -0,0 +1,170 @@
+/*
+ * linux/sound/soc/pxa/palm27x.c
+ *
+ * SoC Audio driver for Palm T|X, T5 and LifeDrive
+ *
+ * based on tosa.c
+ *
+ * Copyright (C) 2008 Marek Vasut <marek.vasut@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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+#include <linux/platform_data/asoc-palm27x.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-ac97.h"
+
+static struct snd_soc_jack hs_jack;
+
+/* Headphones jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{
+		.pin    = "Headphone Jack",
+		.mask   = SND_JACK_HEADPHONE,
+	},
+};
+
+/* Headphones jack detection gpios */
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+	[0] = {
+		/* gpio is set on per-platform basis */
+		.name           = "hp-gpio",
+		.report         = SND_JACK_HEADPHONE,
+		.debounce_time	= 200,
+	},
+};
+
+/* Palm27x machine dapm widgets */
+static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Ext. Speaker", NULL),
+	SND_SOC_DAPM_MIC("Ext. Microphone", NULL),
+};
+
+/* PalmTX audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* headphone connected to HPOUTL, HPOUTR */
+	{"Headphone Jack", NULL, "HPOUTL"},
+	{"Headphone Jack", NULL, "HPOUTR"},
+
+	/* ext speaker connected to ROUT2, LOUT2 */
+	{"Ext. Speaker", NULL, "LOUT2"},
+	{"Ext. Speaker", NULL, "ROUT2"},
+
+	/* mic connected to MIC1 */
+	{"MIC1", NULL, "Ext. Microphone"},
+};
+
+static struct snd_soc_card palm27x_asoc;
+
+static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd)
+{
+	int err;
+
+	/* Jack detection API stuff */
+	err = snd_soc_card_jack_new(rtd->card, "Headphone Jack",
+				    SND_JACK_HEADPHONE, &hs_jack, hs_jack_pins,
+				    ARRAY_SIZE(hs_jack_pins));
+	if (err)
+		return err;
+
+	err = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+				hs_jack_gpios);
+
+	return err;
+}
+
+static struct snd_soc_dai_link palm27x_dai[] = {
+{
+	.name = "AC97 HiFi",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai_name = "pxa2xx-ac97",
+	.codec_dai_name =  "wm9712-hifi",
+	.codec_name = "wm9712-codec",
+	.platform_name = "pxa-pcm-audio",
+	.init = palm27x_ac97_init,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.cpu_dai_name = "pxa2xx-ac97-aux",
+	.codec_dai_name = "wm9712-aux",
+	.codec_name = "wm9712-codec",
+	.platform_name = "pxa-pcm-audio",
+},
+};
+
+static struct snd_soc_card palm27x_asoc = {
+	.name = "Palm/PXA27x",
+	.owner = THIS_MODULE,
+	.dai_link = palm27x_dai,
+	.num_links = ARRAY_SIZE(palm27x_dai),
+	.dapm_widgets = palm27x_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(palm27x_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+	.fully_routed = true,
+};
+
+static int palm27x_asoc_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	if (!(machine_is_palmtx() || machine_is_palmt5() ||
+		machine_is_palmld() || machine_is_palmte2()))
+		return -ENODEV;
+
+	if (!pdev->dev.platform_data) {
+		dev_err(&pdev->dev, "please supply platform_data\n");
+		return -ENODEV;
+	}
+
+	hs_jack_gpios[0].gpio = ((struct palm27x_asoc_info *)
+			(pdev->dev.platform_data))->jack_gpio;
+
+	palm27x_asoc.dev = &pdev->dev;
+
+	ret = snd_soc_register_card(&palm27x_asoc);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+	return ret;
+}
+
+static int palm27x_asoc_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&palm27x_asoc);
+	return 0;
+}
+
+static struct platform_driver palm27x_wm9712_driver = {
+	.probe		= palm27x_asoc_probe,
+	.remove		= palm27x_asoc_remove,
+	.driver		= {
+		.name		= "palm27x-asoc",
+		.pm     = &snd_soc_pm_ops,
+	},
+};
+
+module_platform_driver(palm27x_wm9712_driver);
+
+/* Module information */
+MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
new file mode 100644
index 000000000..0fce8c420
--- /dev/null
+++ b/sound/soc/pxa/poodle.c
@@ -0,0 +1,311 @@
+/*
+ * poodle.c  --  SoC audio for Poodle
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/i2c.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <asm/hardware/locomo.h>
+#include <mach/poodle.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm8731.h"
+#include "pxa2xx-i2s.h"
+
+#define POODLE_HP        1
+#define POODLE_HP_OFF    0
+#define POODLE_SPK_ON    1
+#define POODLE_SPK_OFF   0
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define POODLE_AUDIO_CLOCK 12288000
+
+static int poodle_jack_func;
+static int poodle_spk_func;
+
+static void poodle_ext_control(struct snd_soc_dapm_context *dapm)
+{
+	/* set up jack connection */
+	if (poodle_jack_func == POODLE_HP) {
+		/* set = unmute headphone */
+		locomo_gpio_write(&poodle_locomo_device.dev,
+			POODLE_LOCOMO_GPIO_MUTE_L, 1);
+		locomo_gpio_write(&poodle_locomo_device.dev,
+			POODLE_LOCOMO_GPIO_MUTE_R, 1);
+		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	} else {
+		locomo_gpio_write(&poodle_locomo_device.dev,
+			POODLE_LOCOMO_GPIO_MUTE_L, 0);
+		locomo_gpio_write(&poodle_locomo_device.dev,
+			POODLE_LOCOMO_GPIO_MUTE_R, 0);
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+	}
+
+	/* set the enpoints to their new connetion states */
+	if (poodle_spk_func == POODLE_SPK_ON)
+		snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+
+	/* signal a DAPM event */
+	snd_soc_dapm_sync(dapm);
+}
+
+static int poodle_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* check the jack status at stream startup */
+	poodle_ext_control(&rtd->card->dapm);
+
+	return 0;
+}
+
+/* we need to unmute the HP at shutdown as the mute burns power on poodle */
+static void poodle_shutdown(struct snd_pcm_substream *substream)
+{
+	/* set = unmute headphone */
+	locomo_gpio_write(&poodle_locomo_device.dev,
+		POODLE_LOCOMO_GPIO_MUTE_L, 1);
+	locomo_gpio_write(&poodle_locomo_device.dev,
+		POODLE_LOCOMO_GPIO_MUTE_R, 1);
+}
+
+static int poodle_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 clk = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk = 11289600;
+		break;
+	}
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as input (unused) */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops poodle_ops = {
+	.startup = poodle_startup,
+	.hw_params = poodle_hw_params,
+	.shutdown = poodle_shutdown,
+};
+
+static int poodle_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = poodle_jack_func;
+	return 0;
+}
+
+static int poodle_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card =  snd_kcontrol_chip(kcontrol);
+
+	if (poodle_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	poodle_jack_func = ucontrol->value.integer.value[0];
+	poodle_ext_control(&card->dapm);
+	return 1;
+}
+
+static int poodle_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = poodle_spk_func;
+	return 0;
+}
+
+static int poodle_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card =  snd_kcontrol_chip(kcontrol);
+
+	if (poodle_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	poodle_spk_func = ucontrol->value.integer.value[0];
+	poodle_ext_control(&card->dapm);
+	return 1;
+}
+
+static int poodle_amp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		locomo_gpio_write(&poodle_locomo_device.dev,
+			POODLE_LOCOMO_GPIO_AMP_ON, 0);
+	else
+		locomo_gpio_write(&poodle_locomo_device.dev,
+			POODLE_LOCOMO_GPIO_AMP_ON, 1);
+
+	return 0;
+}
+
+/* poodle machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", NULL),
+SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event),
+};
+
+/* Corgi machine connections to the codec pins */
+static const struct snd_soc_dapm_route poodle_audio_map[] = {
+
+	/* headphone connected to LHPOUT1, RHPOUT1 */
+	{"Headphone Jack", NULL, "LHPOUT"},
+	{"Headphone Jack", NULL, "RHPOUT"},
+
+	/* speaker connected to LOUT, ROUT */
+	{"Ext Spk", NULL, "ROUT"},
+	{"Ext Spk", NULL, "LOUT"},
+};
+
+static const char *jack_function[] = {"Off", "Headphone"};
+static const char *spk_function[] = {"Off", "On"};
+static const struct soc_enum poodle_enum[] = {
+	SOC_ENUM_SINGLE_EXT(2, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
+	SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack,
+		poodle_set_jack),
+	SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk,
+		poodle_set_spk),
+};
+
+/*
+ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
+ */
+static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_dapm_nc_pin(dapm, "LLINEIN");
+	snd_soc_dapm_nc_pin(dapm, "RLINEIN");
+
+	return 0;
+}
+
+/* poodle digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link poodle_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731",
+	.cpu_dai_name = "pxa2xx-i2s",
+	.codec_dai_name = "wm8731-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm8731.0-001b",
+	.init = poodle_wm8731_init,
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBS_CFS,
+	.ops = &poodle_ops,
+};
+
+/* poodle audio machine driver */
+static struct snd_soc_card poodle = {
+	.name = "Poodle",
+	.dai_link = &poodle_dai,
+	.num_links = 1,
+	.owner = THIS_MODULE,
+
+	.controls = wm8731_poodle_controls,
+	.num_controls = ARRAY_SIZE(wm8731_poodle_controls),
+	.dapm_widgets = wm8731_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets),
+	.dapm_routes = poodle_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(poodle_audio_map),
+};
+
+static int poodle_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &poodle;
+	int ret;
+
+	locomo_gpio_set_dir(&poodle_locomo_device.dev,
+		POODLE_LOCOMO_GPIO_AMP_ON, 0);
+	/* should we mute HP at startup - burning power ?*/
+	locomo_gpio_set_dir(&poodle_locomo_device.dev,
+		POODLE_LOCOMO_GPIO_MUTE_L, 0);
+	locomo_gpio_set_dir(&poodle_locomo_device.dev,
+		POODLE_LOCOMO_GPIO_MUTE_R, 0);
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+	return ret;
+}
+
+static int poodle_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static struct platform_driver poodle_driver = {
+	.driver		= {
+		.name	= "poodle-audio",
+		.pm     = &snd_soc_pm_ops,
+	},
+	.probe		= poodle_probe,
+	.remove		= poodle_remove,
+};
+
+module_platform_driver(poodle_driver);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Poodle");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:poodle-audio");
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
new file mode 100644
index 000000000..fbe2e93d6
--- /dev/null
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -0,0 +1,841 @@
+/*
+ * pxa-ssp.c  --  ALSA Soc Audio Layer
+ *
+ * Copyright 2005,2008 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ *         Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ *  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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ * TODO:
+ *  o Test network mode for > 16bit sample size
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <linux/io.h>
+#include <linux/pxa2xx_ssp.h>
+#include <linux/of.h>
+#include <linux/dmaengine.h>
+
+#include <asm/irq.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "../../arm/pxa2xx-pcm.h"
+#include "pxa-ssp.h"
+
+/*
+ * SSP audio private data
+ */
+struct ssp_priv {
+	struct ssp_device *ssp;
+	unsigned int sysclk;
+	int dai_fmt;
+#ifdef CONFIG_PM
+	uint32_t	cr0;
+	uint32_t	cr1;
+	uint32_t	to;
+	uint32_t	psp;
+#endif
+};
+
+static void dump_registers(struct ssp_device *ssp)
+{
+	dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n",
+		 pxa_ssp_read_reg(ssp, SSCR0), pxa_ssp_read_reg(ssp, SSCR1),
+		 pxa_ssp_read_reg(ssp, SSTO));
+
+	dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n",
+		 pxa_ssp_read_reg(ssp, SSPSP), pxa_ssp_read_reg(ssp, SSSR),
+		 pxa_ssp_read_reg(ssp, SSACD));
+}
+
+static void pxa_ssp_enable(struct ssp_device *ssp)
+{
+	uint32_t sscr0;
+
+	sscr0 = __raw_readl(ssp->mmio_base + SSCR0) | SSCR0_SSE;
+	__raw_writel(sscr0, ssp->mmio_base + SSCR0);
+}
+
+static void pxa_ssp_disable(struct ssp_device *ssp)
+{
+	uint32_t sscr0;
+
+	sscr0 = __raw_readl(ssp->mmio_base + SSCR0) & ~SSCR0_SSE;
+	__raw_writel(sscr0, ssp->mmio_base + SSCR0);
+}
+
+static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4,
+			int out, struct snd_dmaengine_dai_dma_data *dma)
+{
+	dma->addr_width = width4 ? DMA_SLAVE_BUSWIDTH_4_BYTES :
+				   DMA_SLAVE_BUSWIDTH_2_BYTES;
+	dma->maxburst = 16;
+	dma->addr = ssp->phys_base + SSDR;
+}
+
+static int pxa_ssp_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *ssp = priv->ssp;
+	struct snd_dmaengine_dai_dma_data *dma;
+	int ret = 0;
+
+	if (!cpu_dai->active) {
+		clk_prepare_enable(ssp->clk);
+		pxa_ssp_disable(ssp);
+	}
+
+	dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL);
+	if (!dma)
+		return -ENOMEM;
+
+	dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+				&ssp->drcmr_tx : &ssp->drcmr_rx;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma);
+
+	return ret;
+}
+
+static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *ssp = priv->ssp;
+
+	if (!cpu_dai->active) {
+		pxa_ssp_disable(ssp);
+		clk_disable_unprepare(ssp->clk);
+	}
+
+	kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
+	snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
+}
+
+#ifdef CONFIG_PM
+
+static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *ssp = priv->ssp;
+
+	if (!cpu_dai->active)
+		clk_prepare_enable(ssp->clk);
+
+	priv->cr0 = __raw_readl(ssp->mmio_base + SSCR0);
+	priv->cr1 = __raw_readl(ssp->mmio_base + SSCR1);
+	priv->to  = __raw_readl(ssp->mmio_base + SSTO);
+	priv->psp = __raw_readl(ssp->mmio_base + SSPSP);
+
+	pxa_ssp_disable(ssp);
+	clk_disable_unprepare(ssp->clk);
+	return 0;
+}
+
+static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *ssp = priv->ssp;
+	uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE;
+
+	clk_prepare_enable(ssp->clk);
+
+	__raw_writel(sssr, ssp->mmio_base + SSSR);
+	__raw_writel(priv->cr0 & ~SSCR0_SSE, ssp->mmio_base + SSCR0);
+	__raw_writel(priv->cr1, ssp->mmio_base + SSCR1);
+	__raw_writel(priv->to,  ssp->mmio_base + SSTO);
+	__raw_writel(priv->psp, ssp->mmio_base + SSPSP);
+
+	if (cpu_dai->active)
+		pxa_ssp_enable(ssp);
+	else
+		clk_disable_unprepare(ssp->clk);
+
+	return 0;
+}
+
+#else
+#define pxa_ssp_suspend	NULL
+#define pxa_ssp_resume	NULL
+#endif
+
+/**
+ * ssp_set_clkdiv - set SSP clock divider
+ * @div: serial clock rate divider
+ */
+static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div)
+{
+	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
+
+	if (ssp->type == PXA25x_SSP) {
+		sscr0 &= ~0x0000ff00;
+		sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
+	} else {
+		sscr0 &= ~0x000fff00;
+		sscr0 |= (div - 1) << 8;     /* 1..4096 */
+	}
+	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
+}
+
+/**
+ * pxa_ssp_get_clkdiv - get SSP clock divider
+ */
+static u32 pxa_ssp_get_scr(struct ssp_device *ssp)
+{
+	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
+	u32 div;
+
+	if (ssp->type == PXA25x_SSP)
+		div = ((sscr0 >> 8) & 0xff) * 2 + 2;
+	else
+		div = ((sscr0 >> 8) & 0xfff) + 1;
+	return div;
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+	int clk_id, unsigned int freq, int dir)
+{
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *ssp = priv->ssp;
+	int val;
+
+	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) &
+		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
+
+	dev_dbg(&ssp->pdev->dev,
+		"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n",
+		cpu_dai->id, clk_id, freq);
+
+	switch (clk_id) {
+	case PXA_SSP_CLK_NET_PLL:
+		sscr0 |= SSCR0_MOD;
+		break;
+	case PXA_SSP_CLK_PLL:
+		/* Internal PLL is fixed */
+		if (ssp->type == PXA25x_SSP)
+			priv->sysclk = 1843200;
+		else
+			priv->sysclk = 13000000;
+		break;
+	case PXA_SSP_CLK_EXT:
+		priv->sysclk = freq;
+		sscr0 |= SSCR0_ECS;
+		break;
+	case PXA_SSP_CLK_NET:
+		priv->sysclk = freq;
+		sscr0 |= SSCR0_NCS | SSCR0_MOD;
+		break;
+	case PXA_SSP_CLK_AUDIO:
+		priv->sysclk = 0;
+		pxa_ssp_set_scr(ssp, 1);
+		sscr0 |= SSCR0_ACS;
+		break;
+	default:
+		return -ENODEV;
+	}
+
+	/* The SSP clock must be disabled when changing SSP clock mode
+	 * on PXA2xx.  On PXA3xx it must be enabled when doing so. */
+	if (ssp->type != PXA3xx_SSP)
+		clk_disable_unprepare(ssp->clk);
+	val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0;
+	pxa_ssp_write_reg(ssp, SSCR0, val);
+	if (ssp->type != PXA3xx_SSP)
+		clk_prepare_enable(ssp->clk);
+
+	return 0;
+}
+
+/*
+ * Set the SSP clock dividers.
+ */
+static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+	int div_id, int div)
+{
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *ssp = priv->ssp;
+	int val;
+
+	switch (div_id) {
+	case PXA_SSP_AUDIO_DIV_ACDS:
+		val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div);
+		pxa_ssp_write_reg(ssp, SSACD, val);
+		break;
+	case PXA_SSP_AUDIO_DIV_SCDB:
+		val = pxa_ssp_read_reg(ssp, SSACD);
+		val &= ~SSACD_SCDB;
+		if (ssp->type == PXA3xx_SSP)
+			val &= ~SSACD_SCDX8;
+		switch (div) {
+		case PXA_SSP_CLK_SCDB_1:
+			val |= SSACD_SCDB;
+			break;
+		case PXA_SSP_CLK_SCDB_4:
+			break;
+		case PXA_SSP_CLK_SCDB_8:
+			if (ssp->type == PXA3xx_SSP)
+				val |= SSACD_SCDX8;
+			else
+				return -EINVAL;
+			break;
+		default:
+			return -EINVAL;
+		}
+		pxa_ssp_write_reg(ssp, SSACD, val);
+		break;
+	case PXA_SSP_DIV_SCR:
+		pxa_ssp_set_scr(ssp, div);
+		break;
+	default:
+		return -ENODEV;
+	}
+
+	return 0;
+}
+
+/*
+ * Configure the PLL frequency pxa27x and (afaik - pxa320 only)
+ */
+static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
+	int source, unsigned int freq_in, unsigned int freq_out)
+{
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *ssp = priv->ssp;
+	u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70;
+
+	if (ssp->type == PXA3xx_SSP)
+		pxa_ssp_write_reg(ssp, SSACDD, 0);
+
+	switch (freq_out) {
+	case 5622000:
+		break;
+	case 11345000:
+		ssacd |= (0x1 << 4);
+		break;
+	case 12235000:
+		ssacd |= (0x2 << 4);
+		break;
+	case 14857000:
+		ssacd |= (0x3 << 4);
+		break;
+	case 32842000:
+		ssacd |= (0x4 << 4);
+		break;
+	case 48000000:
+		ssacd |= (0x5 << 4);
+		break;
+	case 0:
+		/* Disable */
+		break;
+
+	default:
+		/* PXA3xx has a clock ditherer which can be used to generate
+		 * a wider range of frequencies - calculate a value for it.
+		 */
+		if (ssp->type == PXA3xx_SSP) {
+			u32 val;
+			u64 tmp = 19968;
+			tmp *= 1000000;
+			do_div(tmp, freq_out);
+			val = tmp;
+
+			val = (val << 16) | 64;
+			pxa_ssp_write_reg(ssp, SSACDD, val);
+
+			ssacd |= (0x6 << 4);
+
+			dev_dbg(&ssp->pdev->dev,
+				"Using SSACDD %x to supply %uHz\n",
+				val, freq_out);
+			break;
+		}
+
+		return -EINVAL;
+	}
+
+	pxa_ssp_write_reg(ssp, SSACD, ssacd);
+
+	return 0;
+}
+
+/*
+ * Set the active slots in TDM/Network mode
+ */
+static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
+	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
+{
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *ssp = priv->ssp;
+	u32 sscr0;
+
+	sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
+	sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS);
+
+	/* set slot width */
+	if (slot_width > 16)
+		sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16);
+	else
+		sscr0 |= SSCR0_DataSize(slot_width);
+
+	if (slots > 1) {
+		/* enable network mode */
+		sscr0 |= SSCR0_MOD;
+
+		/* set number of active slots */
+		sscr0 |= SSCR0_SlotsPerFrm(slots);
+
+		/* set active slot mask */
+		pxa_ssp_write_reg(ssp, SSTSA, tx_mask);
+		pxa_ssp_write_reg(ssp, SSRSA, rx_mask);
+	}
+	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
+
+	return 0;
+}
+
+/*
+ * Tristate the SSP DAI lines
+ */
+static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai,
+	int tristate)
+{
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *ssp = priv->ssp;
+	u32 sscr1;
+
+	sscr1 = pxa_ssp_read_reg(ssp, SSCR1);
+	if (tristate)
+		sscr1 &= ~SSCR1_TTE;
+	else
+		sscr1 |= SSCR1_TTE;
+	pxa_ssp_write_reg(ssp, SSCR1, sscr1);
+
+	return 0;
+}
+
+/*
+ * Set up the SSP DAI format.
+ * The SSP Port must be inactive before calling this function as the
+ * physical interface format is changed.
+ */
+static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+		unsigned int fmt)
+{
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *ssp = priv->ssp;
+	u32 sscr0, sscr1, sspsp, scfr;
+
+	/* check if we need to change anything at all */
+	if (priv->dai_fmt == fmt)
+		return 0;
+
+	/* we can only change the settings if the port is not in use */
+	if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) {
+		dev_err(&ssp->pdev->dev,
+			"can't change hardware dai format: stream is in use");
+		return -EINVAL;
+	}
+
+	/* reset port settings */
+	sscr0 = pxa_ssp_read_reg(ssp, SSCR0) &
+		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
+	sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7);
+	sspsp = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR | SSCR1_SCFR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		sscr1 |= SSCR1_SCLKDIR | SSCR1_SCFR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		sspsp |= SSPSP_SFRMP;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		sspsp |= SSPSP_SCMODE(2);
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		sscr0 |= SSCR0_PSP;
+		sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
+		/* See hw_params() */
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A:
+		sspsp |= SSPSP_FSRT;
+	case SND_SOC_DAIFMT_DSP_B:
+		sscr0 |= SSCR0_MOD | SSCR0_PSP;
+		sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
+	pxa_ssp_write_reg(ssp, SSCR1, sscr1);
+	pxa_ssp_write_reg(ssp, SSPSP, sspsp);
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+	case SND_SOC_DAIFMT_CBM_CFS:
+		scfr = pxa_ssp_read_reg(ssp, SSCR1) | SSCR1_SCFR;
+		pxa_ssp_write_reg(ssp, SSCR1, scfr);
+
+		while (pxa_ssp_read_reg(ssp, SSSR) & SSSR_BSY)
+			cpu_relax();
+		break;
+	}
+
+	dump_registers(ssp);
+
+	/* Since we are configuring the timings for the format by hand
+	 * we have to defer some things until hw_params() where we
+	 * know parameters like the sample size.
+	 */
+	priv->dai_fmt = fmt;
+
+	return 0;
+}
+
+/*
+ * Set the SSP audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *ssp = priv->ssp;
+	int chn = params_channels(params);
+	u32 sscr0;
+	u32 sspsp;
+	int width = snd_pcm_format_physical_width(params_format(params));
+	int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf;
+	struct snd_dmaengine_dai_dma_data *dma_data;
+
+	dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream);
+
+	/* Network mode with one active slot (ttsa == 1) can be used
+	 * to force 16-bit frame width on the wire (for S16_LE), even
+	 * with two channels. Use 16-bit DMA transfers for this case.
+	 */
+	pxa_ssp_set_dma_params(ssp,
+		((chn == 2) && (ttsa != 1)) || (width == 32),
+		substream->stream == SNDRV_PCM_STREAM_PLAYBACK, dma_data);
+
+	/* we can only change the settings if the port is not in use */
+	if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
+		return 0;
+
+	/* clear selected SSP bits */
+	sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS);
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		if (ssp->type == PXA3xx_SSP)
+			sscr0 |= SSCR0_FPCKE;
+		sscr0 |= SSCR0_DataSize(16);
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8));
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16));
+		break;
+	}
+	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
+
+	switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+	       sspsp = pxa_ssp_read_reg(ssp, SSPSP);
+
+		if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) {
+			/* This is a special case where the bitclk is 64fs
+			* and we're not dealing with 2*32 bits of audio
+			* samples.
+			*
+			* The SSP values used for that are all found out by
+			* trying and failing a lot; some of the registers
+			* needed for that mode are only available on PXA3xx.
+			*/
+			if (ssp->type != PXA3xx_SSP)
+				return -EINVAL;
+
+			sspsp |= SSPSP_SFRMWDTH(width * 2);
+			sspsp |= SSPSP_SFRMDLY(width * 4);
+			sspsp |= SSPSP_EDMYSTOP(3);
+			sspsp |= SSPSP_DMYSTOP(3);
+			sspsp |= SSPSP_DMYSTRT(1);
+		} else {
+			/* The frame width is the width the LRCLK is
+			 * asserted for; the delay is expressed in
+			 * half cycle units.  We need the extra cycle
+			 * because the data starts clocking out one BCLK
+			 * after LRCLK changes polarity.
+			 */
+			sspsp |= SSPSP_SFRMWDTH(width + 1);
+			sspsp |= SSPSP_SFRMDLY((width + 1) * 2);
+			sspsp |= SSPSP_DMYSTRT(1);
+		}
+
+		pxa_ssp_write_reg(ssp, SSPSP, sspsp);
+		break;
+	default:
+		break;
+	}
+
+	/* When we use a network mode, we always require TDM slots
+	 * - complain loudly and fail if they've not been set up yet.
+	 */
+	if ((sscr0 & SSCR0_MOD) && !ttsa) {
+		dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
+		return -EINVAL;
+	}
+
+	dump_registers(ssp);
+
+	return 0;
+}
+
+static void pxa_ssp_set_running_bit(struct snd_pcm_substream *substream,
+				    struct ssp_device *ssp, int value)
+{
+	uint32_t sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
+	uint32_t sscr1 = pxa_ssp_read_reg(ssp, SSCR1);
+	uint32_t sspsp = pxa_ssp_read_reg(ssp, SSPSP);
+	uint32_t sssr = pxa_ssp_read_reg(ssp, SSSR);
+
+	if (value && (sscr0 & SSCR0_SSE))
+		pxa_ssp_write_reg(ssp, SSCR0, sscr0 & ~SSCR0_SSE);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (value)
+			sscr1 |= SSCR1_TSRE;
+		else
+			sscr1 &= ~SSCR1_TSRE;
+	} else {
+		if (value)
+			sscr1 |= SSCR1_RSRE;
+		else
+			sscr1 &= ~SSCR1_RSRE;
+	}
+
+	pxa_ssp_write_reg(ssp, SSCR1, sscr1);
+
+	if (value) {
+		pxa_ssp_write_reg(ssp, SSSR, sssr);
+		pxa_ssp_write_reg(ssp, SSPSP, sspsp);
+		pxa_ssp_write_reg(ssp, SSCR0, sscr0 | SSCR0_SSE);
+	}
+}
+
+static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *cpu_dai)
+{
+	int ret = 0;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *ssp = priv->ssp;
+	int val;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_RESUME:
+		pxa_ssp_enable(ssp);
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		pxa_ssp_set_running_bit(substream, ssp, 1);
+		val = pxa_ssp_read_reg(ssp, SSSR);
+		pxa_ssp_write_reg(ssp, SSSR, val);
+		break;
+	case SNDRV_PCM_TRIGGER_START:
+		pxa_ssp_set_running_bit(substream, ssp, 1);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		pxa_ssp_set_running_bit(substream, ssp, 0);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		pxa_ssp_disable(ssp);
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		pxa_ssp_set_running_bit(substream, ssp, 0);
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	dump_registers(ssp);
+
+	return ret;
+}
+
+static int pxa_ssp_probe(struct snd_soc_dai *dai)
+{
+	struct device *dev = dai->dev;
+	struct ssp_priv *priv;
+	int ret;
+
+	priv = kzalloc(sizeof(struct ssp_priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	if (dev->of_node) {
+		struct device_node *ssp_handle;
+
+		ssp_handle = of_parse_phandle(dev->of_node, "port", 0);
+		if (!ssp_handle) {
+			dev_err(dev, "unable to get 'port' phandle\n");
+			ret = -ENODEV;
+			goto err_priv;
+		}
+
+		priv->ssp = pxa_ssp_request_of(ssp_handle, "SoC audio");
+		if (priv->ssp == NULL) {
+			ret = -ENODEV;
+			goto err_priv;
+		}
+	} else {
+		priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
+		if (priv->ssp == NULL) {
+			ret = -ENODEV;
+			goto err_priv;
+		}
+	}
+
+	priv->dai_fmt = (unsigned int) -1;
+	snd_soc_dai_set_drvdata(dai, priv);
+
+	return 0;
+
+err_priv:
+	kfree(priv);
+	return ret;
+}
+
+static int pxa_ssp_remove(struct snd_soc_dai *dai)
+{
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+	pxa_ssp_free(priv->ssp);
+	kfree(priv);
+	return 0;
+}
+
+#define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+			  SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |	\
+			  SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |	\
+			  SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 |	\
+			  SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static const struct snd_soc_dai_ops pxa_ssp_dai_ops = {
+	.startup	= pxa_ssp_startup,
+	.shutdown	= pxa_ssp_shutdown,
+	.trigger	= pxa_ssp_trigger,
+	.hw_params	= pxa_ssp_hw_params,
+	.set_sysclk	= pxa_ssp_set_dai_sysclk,
+	.set_clkdiv	= pxa_ssp_set_dai_clkdiv,
+	.set_pll	= pxa_ssp_set_dai_pll,
+	.set_fmt	= pxa_ssp_set_dai_fmt,
+	.set_tdm_slot	= pxa_ssp_set_dai_tdm_slot,
+	.set_tristate	= pxa_ssp_set_dai_tristate,
+};
+
+static struct snd_soc_dai_driver pxa_ssp_dai = {
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			 .channels_min = 1,
+			 .channels_max = 8,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = &pxa_ssp_dai_ops,
+};
+
+static const struct snd_soc_component_driver pxa_ssp_component = {
+	.name		= "pxa-ssp",
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id pxa_ssp_of_ids[] = {
+	{ .compatible = "mrvl,pxa-ssp-dai" },
+	{}
+};
+#endif
+
+static int asoc_ssp_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_component(&pdev->dev, &pxa_ssp_component,
+					  &pxa_ssp_dai, 1);
+}
+
+static int asoc_ssp_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_component(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver asoc_ssp_driver = {
+	.driver = {
+		.name = "pxa-ssp-dai",
+		.of_match_table = of_match_ptr(pxa_ssp_of_ids),
+	},
+
+	.probe = asoc_ssp_probe,
+	.remove = asoc_ssp_remove,
+};
+
+module_platform_driver(asoc_ssp_driver);
+
+/* Module information */
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h
new file mode 100644
index 000000000..bc79da221
--- /dev/null
+++ b/sound/soc/pxa/pxa-ssp.h
@@ -0,0 +1,45 @@
+/*
+ * ASoC PXA SSP port support
+ *
+ * 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.
+ */
+
+#ifndef _PXA_SSP_H
+#define _PXA_SSP_H
+
+/* pxa DAI SSP IDs */
+#define PXA_DAI_SSP1			0
+#define PXA_DAI_SSP2			1
+#define PXA_DAI_SSP3			2
+#define PXA_DAI_SSP4			3
+
+/* SSP clock sources */
+#define PXA_SSP_CLK_PLL	0
+#define PXA_SSP_CLK_EXT	1
+#define PXA_SSP_CLK_NET	2
+#define PXA_SSP_CLK_AUDIO	3
+#define PXA_SSP_CLK_NET_PLL	4
+
+/* SSP audio dividers */
+#define PXA_SSP_AUDIO_DIV_ACDS		0
+#define PXA_SSP_AUDIO_DIV_SCDB		1
+#define PXA_SSP_DIV_SCR				2
+
+/* SSP ACDS audio dividers values */
+#define PXA_SSP_CLK_AUDIO_DIV_1		0
+#define PXA_SSP_CLK_AUDIO_DIV_2		1
+#define PXA_SSP_CLK_AUDIO_DIV_4		2
+#define PXA_SSP_CLK_AUDIO_DIV_8		3
+#define PXA_SSP_CLK_AUDIO_DIV_16	4
+#define PXA_SSP_CLK_AUDIO_DIV_32	5
+
+/* SSP divider bypass */
+#define PXA_SSP_CLK_SCDB_4		0
+#define PXA_SSP_CLK_SCDB_1		1
+#define PXA_SSP_CLK_SCDB_8		2
+
+#define PXA_SSP_PLL_OUT  0
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
new file mode 100644
index 000000000..1f6054650
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -0,0 +1,274 @@
+/*
+ * linux/sound/pxa2xx-ac97.c -- AC97 support for the Intel PXA2xx chip.
+ *
+ * Author:	Nicolas Pitre
+ * Created:	Dec 02, 2004
+ * Copyright:	MontaVista Software 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.
+ */
+
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/dmaengine.h>
+
+#include <sound/core.h>
+#include <sound/ac97_codec.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
+
+#include <mach/hardware.h>
+#include <mach/regs-ac97.h>
+#include <mach/audio.h>
+
+#include "pxa2xx-ac97.h"
+
+static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+	pxa2xx_ac97_try_warm_reset(ac97);
+
+	pxa2xx_ac97_finish_reset(ac97);
+}
+
+static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+	pxa2xx_ac97_try_cold_reset(ac97);
+
+	pxa2xx_ac97_finish_reset(ac97);
+}
+
+static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
+	.read	= pxa2xx_ac97_read,
+	.write	= pxa2xx_ac97_write,
+	.warm_reset	= pxa2xx_ac97_warm_reset,
+	.reset	= pxa2xx_ac97_cold_reset,
+};
+
+static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
+	.addr		= __PREG(PCDR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.maxburst	= 32,
+	.filter_data	= &pxa2xx_ac97_pcm_stereo_in_req,
+};
+
+static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
+	.addr		= __PREG(PCDR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.maxburst	= 32,
+	.filter_data	= &pxa2xx_ac97_pcm_stereo_out_req,
+};
+
+static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = {
+	.addr		= __PREG(MODR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
+	.maxburst	= 16,
+	.filter_data	= &pxa2xx_ac97_pcm_aux_mono_out_req,
+};
+
+static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = {
+	.addr		= __PREG(MODR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
+	.maxburst	= 16,
+	.filter_data	= &pxa2xx_ac97_pcm_aux_mono_in_req,
+};
+
+static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = {
+	.addr		= __PREG(MCDR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_2_BYTES,
+	.maxburst	= 16,
+	.filter_data	= &pxa2xx_ac97_pcm_aux_mic_mono_req,
+};
+
+static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *cpu_dai)
+{
+	struct snd_dmaengine_dai_dma_data *dma_data;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dma_data = &pxa2xx_ac97_pcm_stereo_out;
+	else
+		dma_data = &pxa2xx_ac97_pcm_stereo_in;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
+
+	return 0;
+}
+
+static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *cpu_dai)
+{
+	struct snd_dmaengine_dai_dma_data *dma_data;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
+	else
+		dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
+
+	return 0;
+}
+
+static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *cpu_dai)
+{
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return -ENODEV;
+	else
+		snd_soc_dai_set_dma_data(cpu_dai, substream,
+					 &pxa2xx_ac97_pcm_mic_mono_in);
+
+	return 0;
+}
+
+#define PXA2XX_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+		SNDRV_PCM_RATE_48000)
+
+static const struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = {
+	.hw_params	= pxa2xx_ac97_hw_params,
+};
+
+static const struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = {
+	.hw_params	= pxa2xx_ac97_hw_aux_params,
+};
+
+static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = {
+	.hw_params	= pxa2xx_ac97_hw_mic_params,
+};
+
+/*
+ * There is only 1 physical AC97 interface for pxa2xx, but it
+ * has extra fifo's that can be used for aux DACs and ADCs.
+ */
+static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = {
+{
+	.name = "pxa2xx-ac97",
+	.bus_control = true,
+	.playback = {
+		.stream_name = "AC97 Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = PXA2XX_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.stream_name = "AC97 Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = PXA2XX_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = &pxa_ac97_hifi_dai_ops,
+},
+{
+	.name = "pxa2xx-ac97-aux",
+	.bus_control = true,
+	.playback = {
+		.stream_name = "AC97 Aux Playback",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = PXA2XX_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.stream_name = "AC97 Aux Capture",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = PXA2XX_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = &pxa_ac97_aux_dai_ops,
+},
+{
+	.name = "pxa2xx-ac97-mic",
+	.bus_control = true,
+	.capture = {
+		.stream_name = "AC97 Mic Capture",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = PXA2XX_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = &pxa_ac97_mic_dai_ops,
+},
+};
+
+static const struct snd_soc_component_driver pxa_ac97_component = {
+	.name		= "pxa-ac97",
+};
+
+static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	if (pdev->id != -1) {
+		dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n");
+		return -ENXIO;
+	}
+
+	ret = pxa2xx_ac97_hw_probe(pdev);
+	if (ret) {
+		dev_err(&pdev->dev, "PXA2xx AC97 hw probe error (%d)\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops);
+	if (ret != 0)
+		return ret;
+
+	/* Punt most of the init to the SoC probe; we may need the machine
+	 * driver to do interesting things with the clocking to get us up
+	 * and running.
+	 */
+	return snd_soc_register_component(&pdev->dev, &pxa_ac97_component,
+					  pxa_ac97_dai_driver, ARRAY_SIZE(pxa_ac97_dai_driver));
+}
+
+static int pxa2xx_ac97_dev_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_component(&pdev->dev);
+	snd_soc_set_ac97_ops(NULL);
+	pxa2xx_ac97_hw_remove(pdev);
+	return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+static int pxa2xx_ac97_dev_suspend(struct device *dev)
+{
+	return pxa2xx_ac97_hw_suspend();
+}
+
+static int pxa2xx_ac97_dev_resume(struct device *dev)
+{
+	return pxa2xx_ac97_hw_resume();
+}
+
+static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops,
+		pxa2xx_ac97_dev_suspend, pxa2xx_ac97_dev_resume);
+#endif
+
+static struct platform_driver pxa2xx_ac97_driver = {
+	.probe		= pxa2xx_ac97_dev_probe,
+	.remove		= pxa2xx_ac97_dev_remove,
+	.driver		= {
+		.name	= "pxa2xx-ac97",
+#ifdef CONFIG_PM_SLEEP
+		.pm	= &pxa2xx_ac97_pm_ops,
+#endif
+	},
+};
+
+module_platform_driver(pxa2xx_ac97_driver);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h
new file mode 100644
index 000000000..a49c21ba3
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-ac97.h
@@ -0,0 +1,17 @@
+/*
+ * linux/sound/soc/pxa/pxa2xx-ac97.h
+ *
+ * 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.
+ */
+
+#ifndef _PXA2XX_AC97_H
+#define _PXA2XX_AC97_H
+
+/* pxa2xx DAI ID's */
+#define PXA2XX_DAI_AC97_HIFI	0
+#define PXA2XX_DAI_AC97_AUX		1
+#define PXA2XX_DAI_AC97_MIC		2
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
new file mode 100644
index 000000000..e68290c15
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -0,0 +1,407 @@
+/*
+ * pxa2xx-i2s.c  --  ALSA Soc Audio Layer
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ *         lrg@slimlogic.co.uk
+ *
+ *  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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
+
+#include <mach/hardware.h>
+#include <mach/audio.h>
+
+#include "pxa2xx-i2s.h"
+
+/*
+ * I2S Controller Register and Bit Definitions
+ */
+#define SACR0		__REG(0x40400000)  /* Global Control Register */
+#define SACR1		__REG(0x40400004)  /* Serial Audio I 2 S/MSB-Justified Control Register */
+#define SASR0		__REG(0x4040000C)  /* Serial Audio I 2 S/MSB-Justified Interface and FIFO Status Register */
+#define SAIMR		__REG(0x40400014)  /* Serial Audio Interrupt Mask Register */
+#define SAICR		__REG(0x40400018)  /* Serial Audio Interrupt Clear Register */
+#define SADIV		__REG(0x40400060)  /* Audio Clock Divider Register. */
+#define SADR		__REG(0x40400080)  /* Serial Audio Data Register (TX and RX FIFO access Register). */
+
+#define SACR0_RFTH(x)	((x) << 12)	/* Rx FIFO Interrupt or DMA Trigger Threshold */
+#define SACR0_TFTH(x)	((x) << 8)	/* Tx FIFO Interrupt or DMA Trigger Threshold */
+#define SACR0_STRF	(1 << 5)	/* FIFO Select for EFWR Special Function */
+#define SACR0_EFWR	(1 << 4)	/* Enable EFWR Function  */
+#define SACR0_RST	(1 << 3)	/* FIFO, i2s Register Reset */
+#define SACR0_BCKD	(1 << 2) 	/* Bit Clock Direction */
+#define SACR0_ENB	(1 << 0)	/* Enable I2S Link */
+#define SACR1_ENLBF	(1 << 5)	/* Enable Loopback */
+#define SACR1_DRPL	(1 << 4) 	/* Disable Replaying Function */
+#define SACR1_DREC	(1 << 3)	/* Disable Recording Function */
+#define SACR1_AMSL	(1 << 0)	/* Specify Alternate Mode */
+
+#define SASR0_I2SOFF	(1 << 7)	/* Controller Status */
+#define SASR0_ROR	(1 << 6)	/* Rx FIFO Overrun */
+#define SASR0_TUR	(1 << 5)	/* Tx FIFO Underrun */
+#define SASR0_RFS	(1 << 4)	/* Rx FIFO Service Request */
+#define SASR0_TFS	(1 << 3)	/* Tx FIFO Service Request */
+#define SASR0_BSY	(1 << 2)	/* I2S Busy */
+#define SASR0_RNE	(1 << 1)	/* Rx FIFO Not Empty */
+#define SASR0_TNF	(1 << 0) 	/* Tx FIFO Not Empty */
+
+#define SAICR_ROR	(1 << 6)	/* Clear Rx FIFO Overrun Interrupt */
+#define SAICR_TUR	(1 << 5)	/* Clear Tx FIFO Underrun Interrupt */
+
+#define SAIMR_ROR	(1 << 6)	/* Enable Rx FIFO Overrun Condition Interrupt */
+#define SAIMR_TUR	(1 << 5)	/* Enable Tx FIFO Underrun Condition Interrupt */
+#define SAIMR_RFS	(1 << 4)	/* Enable Rx FIFO Service Interrupt */
+#define SAIMR_TFS	(1 << 3)	/* Enable Tx FIFO Service Interrupt */
+
+struct pxa_i2s_port {
+	u32 sadiv;
+	u32 sacr0;
+	u32 sacr1;
+	u32 saimr;
+	int master;
+	u32 fmt;
+};
+static struct pxa_i2s_port pxa_i2s;
+static struct clk *clk_i2s;
+static int clk_ena = 0;
+
+static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3;
+static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = {
+	.addr		= __PREG(SADR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.maxburst	= 32,
+	.filter_data	= &pxa2xx_i2s_pcm_stereo_out_req,
+};
+
+static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2;
+static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = {
+	.addr		= __PREG(SADR),
+	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
+	.maxburst	= 32,
+	.filter_data	= &pxa2xx_i2s_pcm_stereo_in_req,
+};
+
+static int pxa2xx_i2s_startup(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;
+
+	if (IS_ERR(clk_i2s))
+		return PTR_ERR(clk_i2s);
+
+	if (!cpu_dai->active)
+		SACR0 = 0;
+
+	return 0;
+}
+
+/* wait for I2S controller to be ready */
+static int pxa_i2s_wait(void)
+{
+	int i;
+
+	/* flush the Rx FIFO */
+	for(i = 0; i < 16; i++)
+		SADR;
+	return 0;
+}
+
+static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+		unsigned int fmt)
+{
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		pxa_i2s.fmt = 0;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		pxa_i2s.fmt = SACR1_AMSL;
+		break;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		pxa_i2s.master = 1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		pxa_i2s.master = 0;
+		break;
+	default:
+		break;
+	}
+	return 0;
+}
+
+static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	if (clk_id != PXA2XX_I2S_SYSCLK)
+		return -ENODEV;
+
+	return 0;
+}
+
+static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_dmaengine_dai_dma_data *dma_data;
+
+	if (WARN_ON(IS_ERR(clk_i2s)))
+		return -EINVAL;
+	clk_prepare_enable(clk_i2s);
+	clk_ena = 1;
+	pxa_i2s_wait();
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dma_data = &pxa2xx_i2s_pcm_stereo_out;
+	else
+		dma_data = &pxa2xx_i2s_pcm_stereo_in;
+
+	snd_soc_dai_set_dma_data(dai, substream, dma_data);
+
+	/* is port used by another stream */
+	if (!(SACR0 & SACR0_ENB)) {
+		SACR0 = 0;
+		if (pxa_i2s.master)
+			SACR0 |= SACR0_BCKD;
+
+		SACR0 |= SACR0_RFTH(14) | SACR0_TFTH(1);
+		SACR1 |= pxa_i2s.fmt;
+	}
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		SAIMR |= SAIMR_TFS;
+	else
+		SAIMR |= SAIMR_RFS;
+
+	switch (params_rate(params)) {
+	case 8000:
+		SADIV = 0x48;
+		break;
+	case 11025:
+		SADIV = 0x34;
+		break;
+	case 16000:
+		SADIV = 0x24;
+		break;
+	case 22050:
+		SADIV = 0x1a;
+		break;
+	case 44100:
+		SADIV = 0xd;
+		break;
+	case 48000:
+		SADIV = 0xc;
+		break;
+	case 96000: /* not in manual and possibly slightly inaccurate */
+		SADIV = 0x6;
+		break;
+	}
+
+	return 0;
+}
+
+static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			      struct snd_soc_dai *dai)
+{
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			SACR1 &= ~SACR1_DRPL;
+		else
+			SACR1 &= ~SACR1_DREC;
+		SACR0 |= SACR0_ENB;
+		break;
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		break;
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		SACR1 |= SACR1_DRPL;
+		SAIMR &= ~SAIMR_TFS;
+	} else {
+		SACR1 |= SACR1_DREC;
+		SAIMR &= ~SAIMR_RFS;
+	}
+
+	if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) {
+		SACR0 &= ~SACR0_ENB;
+		pxa_i2s_wait();
+		if (clk_ena) {
+			clk_disable_unprepare(clk_i2s);
+			clk_ena = 0;
+		}
+	}
+}
+
+#ifdef CONFIG_PM
+static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
+{
+	/* store registers */
+	pxa_i2s.sacr0 = SACR0;
+	pxa_i2s.sacr1 = SACR1;
+	pxa_i2s.saimr = SAIMR;
+	pxa_i2s.sadiv = SADIV;
+
+	/* deactivate link */
+	SACR0 &= ~SACR0_ENB;
+	pxa_i2s_wait();
+	return 0;
+}
+
+static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
+{
+	pxa_i2s_wait();
+
+	SACR0 = pxa_i2s.sacr0 & ~SACR0_ENB;
+	SACR1 = pxa_i2s.sacr1;
+	SAIMR = pxa_i2s.saimr;
+	SADIV = pxa_i2s.sadiv;
+
+	SACR0 = pxa_i2s.sacr0;
+
+	return 0;
+}
+
+#else
+#define pxa2xx_i2s_suspend	NULL
+#define pxa2xx_i2s_resume	NULL
+#endif
+
+static int pxa2xx_i2s_probe(struct snd_soc_dai *dai)
+{
+	clk_i2s = clk_get(dai->dev, "I2SCLK");
+	if (IS_ERR(clk_i2s))
+		return PTR_ERR(clk_i2s);
+
+	/*
+	 * PXA Developer's Manual:
+	 * If SACR0[ENB] is toggled in the middle of a normal operation,
+	 * the SACR0[RST] bit must also be set and cleared to reset all
+	 * I2S controller registers.
+	 */
+	SACR0 = SACR0_RST;
+	SACR0 = 0;
+	/* Make sure RPL and REC are disabled */
+	SACR1 = SACR1_DRPL | SACR1_DREC;
+	/* Along with FIFO servicing */
+	SAIMR &= ~(SAIMR_RFS | SAIMR_TFS);
+
+	return 0;
+}
+
+static int  pxa2xx_i2s_remove(struct snd_soc_dai *dai)
+{
+	clk_put(clk_i2s);
+	clk_i2s = ERR_PTR(-ENOENT);
+	return 0;
+}
+
+#define PXA2XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+
+static const struct snd_soc_dai_ops pxa_i2s_dai_ops = {
+	.startup	= pxa2xx_i2s_startup,
+	.shutdown	= pxa2xx_i2s_shutdown,
+	.trigger	= pxa2xx_i2s_trigger,
+	.hw_params	= pxa2xx_i2s_hw_params,
+	.set_fmt	= pxa2xx_i2s_set_dai_fmt,
+	.set_sysclk	= pxa2xx_i2s_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver pxa_i2s_dai = {
+	.probe = pxa2xx_i2s_probe,
+	.remove = pxa2xx_i2s_remove,
+	.suspend = pxa2xx_i2s_suspend,
+	.resume = pxa2xx_i2s_resume,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = PXA2XX_I2S_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = PXA2XX_I2S_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = &pxa_i2s_dai_ops,
+	.symmetric_rates = 1,
+};
+
+static const struct snd_soc_component_driver pxa_i2s_component = {
+	.name		= "pxa-i2s",
+};
+
+static int pxa2xx_i2s_drv_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_component(&pdev->dev, &pxa_i2s_component,
+					  &pxa_i2s_dai, 1);
+}
+
+static int pxa2xx_i2s_drv_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_component(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver pxa2xx_i2s_driver = {
+	.probe = pxa2xx_i2s_drv_probe,
+	.remove = pxa2xx_i2s_drv_remove,
+
+	.driver = {
+		.name = "pxa2xx-i2s",
+	},
+};
+
+static int __init pxa2xx_i2s_init(void)
+{
+	clk_i2s = ERR_PTR(-ENOENT);
+	return platform_driver_register(&pxa2xx_i2s_driver);
+}
+
+static void __exit pxa2xx_i2s_exit(void)
+{
+	platform_driver_unregister(&pxa2xx_i2s_driver);
+}
+
+module_init(pxa2xx_i2s_init);
+module_exit(pxa2xx_i2s_exit);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
+MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa2xx-i2s");
diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h
new file mode 100644
index 000000000..070f3c605
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-i2s.h
@@ -0,0 +1,18 @@
+/*
+ * linux/sound/soc/pxa/pxa2xx-i2s.h
+ *
+ * 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.
+ */
+
+#ifndef _PXA2XX_I2S_H
+#define _PXA2XX_I2S_H
+
+/* pxa2xx DAI ID's */
+#define PXA2XX_DAI_I2S			0
+
+/* I2S clock */
+#define PXA2XX_I2S_SYSCLK		0
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
new file mode 100644
index 000000000..a51c9da66
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -0,0 +1,157 @@
+/*
+ * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip
+ *
+ * Author:	Nicolas Pitre
+ * Created:	Nov 30, 2004
+ * Copyright:	(C) 2004 MontaVista Software, 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.
+ */
+
+#include <linux/dma-mapping.h>
+#include <linux/module.h>
+#include <linux/dmaengine.h>
+#include <linux/of.h>
+
+#include <mach/dma.h>
+
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "../../arm/pxa2xx-pcm.h"
+
+static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct pxa2xx_runtime_data *prtd = runtime->private_data;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_dmaengine_dai_dma_data *dma;
+	int ret;
+
+	dma = 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)
+		return 0;
+
+	/* this may get called several times by oss emulation
+	 * with different params */
+	if (prtd->params == NULL) {
+		prtd->params = dma;
+		ret = pxa_request_dma("name", DMA_PRIO_LOW,
+			      pxa2xx_pcm_dma_irq, substream);
+		if (ret < 0)
+			return ret;
+		prtd->dma_ch = ret;
+	} else if (prtd->params != dma) {
+		pxa_free_dma(prtd->dma_ch);
+		prtd->params = dma;
+		ret = pxa_request_dma("name", DMA_PRIO_LOW,
+			      pxa2xx_pcm_dma_irq, substream);
+		if (ret < 0)
+			return ret;
+		prtd->dma_ch = ret;
+	}
+
+	return __pxa2xx_pcm_hw_params(substream, params);
+}
+
+static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+
+	__pxa2xx_pcm_hw_free(substream);
+
+	if (prtd->dma_ch >= 0) {
+		pxa_free_dma(prtd->dma_ch);
+		prtd->dma_ch = -1;
+		prtd->params = NULL;
+	}
+
+	return 0;
+}
+
+static struct snd_pcm_ops pxa2xx_pcm_ops = {
+	.open		= __pxa2xx_pcm_open,
+	.close		= __pxa2xx_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= pxa2xx_pcm_hw_params,
+	.hw_free	= pxa2xx_pcm_hw_free,
+	.prepare	= __pxa2xx_pcm_prepare,
+	.trigger	= pxa2xx_pcm_trigger,
+	.pointer	= pxa2xx_pcm_pointer,
+	.mmap		= pxa2xx_pcm_mmap,
+};
+
+static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_pcm *pcm = rtd->pcm;
+	int ret;
+
+	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+	if (ret)
+		return ret;
+
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto out;
+	}
+
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto out;
+	}
+ out:
+	return ret;
+}
+
+static struct snd_soc_platform_driver pxa2xx_soc_platform = {
+	.ops 	= &pxa2xx_pcm_ops,
+	.pcm_new	= pxa2xx_soc_pcm_new,
+	.pcm_free	= pxa2xx_pcm_free_dma_buffers,
+};
+
+static int pxa2xx_soc_platform_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_platform(&pdev->dev, &pxa2xx_soc_platform);
+}
+
+static int pxa2xx_soc_platform_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+	return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id snd_soc_pxa_audio_match[] = {
+	{ .compatible   = "mrvl,pxa-pcm-audio" },
+	{ }
+};
+#endif
+
+static struct platform_driver pxa_pcm_driver = {
+	.driver = {
+		.name = "pxa-pcm-audio",
+		.of_match_table = of_match_ptr(snd_soc_pxa_audio_match),
+	},
+
+	.probe = pxa2xx_soc_platform_probe,
+	.remove = pxa2xx_soc_platform_remove,
+};
+
+module_platform_driver(pxa_pcm_driver);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c
new file mode 100644
index 000000000..552b76300
--- /dev/null
+++ b/sound/soc/pxa/raumfeld.c
@@ -0,0 +1,320 @@
+/*
+ * raumfeld_audio.c  --  SoC audio for Raumfeld audio devices
+ *
+ * Copyright (c) 2009 Daniel Mack <daniel@caiaq.de>
+ *
+ * based on code from:
+ *
+ *    Wolfson Microelectronics PLC.
+ *    Openedhand Ltd.
+ *    Liam Girdwood <lrg@slimlogic.co.uk>
+ *    Richard Purdie <richard@openedhand.com>
+ *
+ * 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;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+
+#include "pxa-ssp.h"
+
+#define GPIO_SPDIF_RESET	(38)
+#define GPIO_MCLK_RESET		(111)
+#define GPIO_CODEC_RESET	(120)
+
+static struct i2c_client *max9486_client;
+static struct i2c_board_info max9486_hwmon_info = {
+	I2C_BOARD_INFO("max9485", 0x63),
+};
+
+#define MAX9485_MCLK_FREQ_112896 0x22
+#define MAX9485_MCLK_FREQ_122880 0x23
+#define MAX9485_MCLK_FREQ_225792 0x32
+#define MAX9485_MCLK_FREQ_245760 0x33
+
+static void set_max9485_clk(char clk)
+{
+	i2c_master_send(max9486_client, &clk, 1);
+}
+
+static void raumfeld_enable_audio(bool en)
+{
+	if (en) {
+		gpio_set_value(GPIO_MCLK_RESET, 1);
+
+		/* wait some time to let the clocks become stable */
+		msleep(100);
+
+		gpio_set_value(GPIO_SPDIF_RESET, 1);
+		gpio_set_value(GPIO_CODEC_RESET, 1);
+	} else {
+		gpio_set_value(GPIO_MCLK_RESET, 0);
+		gpio_set_value(GPIO_SPDIF_RESET, 0);
+		gpio_set_value(GPIO_CODEC_RESET, 0);
+	}
+}
+
+/* CS4270 */
+static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	/* set freq to 0 to enable all possible codec sample rates */
+	return snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
+}
+
+static void raumfeld_cs4270_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	/* set freq to 0 to enable all possible codec sample rates */
+	snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
+}
+
+static int raumfeld_cs4270_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 clk = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 44100:
+		set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+		clk = 11289600;
+		break;
+	case 48000:
+		set_max9485_clk(MAX9485_MCLK_FREQ_122880);
+		clk = 12288000;
+		break;
+	case 88200:
+		set_max9485_clk(MAX9485_MCLK_FREQ_225792);
+		clk = 22579200;
+		break;
+	case 96000:
+		set_max9485_clk(MAX9485_MCLK_FREQ_245760);
+		clk = 24576000;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0);
+	if (ret < 0)
+		return ret;
+
+	/* setup the CPU DAI */
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops raumfeld_cs4270_ops = {
+	.startup = raumfeld_cs4270_startup,
+	.shutdown = raumfeld_cs4270_shutdown,
+	.hw_params = raumfeld_cs4270_hw_params,
+};
+
+static int raumfeld_analog_suspend(struct snd_soc_card *card)
+{
+	raumfeld_enable_audio(false);
+	return 0;
+}
+
+static int raumfeld_analog_resume(struct snd_soc_card *card)
+{
+	raumfeld_enable_audio(true);
+	return 0;
+}
+
+/* AK4104 */
+
+static int raumfeld_ak4104_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 *cpu_dai = rtd->cpu_dai;
+	int ret = 0, clk = 0;
+
+	switch (params_rate(params)) {
+	case 44100:
+		set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+		clk = 11289600;
+		break;
+	case 48000:
+		set_max9485_clk(MAX9485_MCLK_FREQ_122880);
+		clk = 12288000;
+		break;
+	case 88200:
+		set_max9485_clk(MAX9485_MCLK_FREQ_225792);
+		clk = 22579200;
+		break;
+	case 96000:
+		set_max9485_clk(MAX9485_MCLK_FREQ_245760);
+		clk = 24576000;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* setup the CPU DAI */
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops raumfeld_ak4104_ops = {
+	.hw_params = raumfeld_ak4104_hw_params,
+};
+
+#define DAI_LINK_CS4270		\
+{							\
+	.name		= "CS4270",			\
+	.stream_name	= "CS4270",			\
+	.cpu_dai_name	= "pxa-ssp-dai.0",		\
+	.platform_name	= "pxa-pcm-audio",		\
+	.codec_dai_name	= "cs4270-hifi",		\
+	.codec_name	= "cs4270.0-0048",	\
+	.dai_fmt	= SND_SOC_DAIFMT_I2S |		\
+			  SND_SOC_DAIFMT_NB_NF |        \
+			  SND_SOC_DAIFMT_CBS_CFS,       \
+	.ops		= &raumfeld_cs4270_ops,		\
+}
+
+#define DAI_LINK_AK4104		\
+{							\
+	.name		= "ak4104",			\
+	.stream_name	= "Playback",			\
+	.cpu_dai_name	= "pxa-ssp-dai.1",		\
+	.codec_dai_name	= "ak4104-hifi",		\
+	.platform_name	= "pxa-pcm-audio",		\
+	.dai_fmt	= SND_SOC_DAIFMT_I2S |		\
+			  SND_SOC_DAIFMT_NB_NF |	\
+			  SND_SOC_DAIFMT_CBS_CFS,       \
+	.ops		= &raumfeld_ak4104_ops,		\
+	.codec_name	= "spi0.0",			\
+}
+
+static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] =
+{
+	DAI_LINK_CS4270,
+	DAI_LINK_AK4104,
+};
+
+static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] =
+{
+	DAI_LINK_CS4270,
+};
+
+static struct snd_soc_card snd_soc_raumfeld_connector = {
+	.name		= "Raumfeld Connector",
+	.owner		= THIS_MODULE,
+	.dai_link	= snd_soc_raumfeld_connector_dai,
+	.num_links	= ARRAY_SIZE(snd_soc_raumfeld_connector_dai),
+	.suspend_post	= raumfeld_analog_suspend,
+	.resume_pre	= raumfeld_analog_resume,
+};
+
+static struct snd_soc_card snd_soc_raumfeld_speaker = {
+	.name		= "Raumfeld Speaker",
+	.owner		= THIS_MODULE,
+	.dai_link	= snd_soc_raumfeld_speaker_dai,
+	.num_links	= ARRAY_SIZE(snd_soc_raumfeld_speaker_dai),
+	.suspend_post	= raumfeld_analog_suspend,
+	.resume_pre	= raumfeld_analog_resume,
+};
+
+static struct platform_device *raumfeld_audio_device;
+
+static int __init raumfeld_audio_init(void)
+{
+	int ret;
+
+	if (!machine_is_raumfeld_speaker() &&
+	    !machine_is_raumfeld_connector())
+		return 0;
+
+	max9486_client = i2c_new_device(i2c_get_adapter(0),
+					&max9486_hwmon_info);
+
+	if (!max9486_client)
+		return -ENOMEM;
+
+	set_max9485_clk(MAX9485_MCLK_FREQ_122880);
+
+	/* Register analog device */
+	raumfeld_audio_device = platform_device_alloc("soc-audio", 0);
+	if (!raumfeld_audio_device)
+		return -ENOMEM;
+
+	if (machine_is_raumfeld_speaker())
+		platform_set_drvdata(raumfeld_audio_device,
+				     &snd_soc_raumfeld_speaker);
+
+	if (machine_is_raumfeld_connector())
+		platform_set_drvdata(raumfeld_audio_device,
+				     &snd_soc_raumfeld_connector);
+
+	ret = platform_device_add(raumfeld_audio_device);
+	if (ret < 0) {
+		platform_device_put(raumfeld_audio_device);
+		return ret;
+	}
+
+	raumfeld_enable_audio(true);
+	return 0;
+}
+
+static void __exit raumfeld_audio_exit(void)
+{
+	raumfeld_enable_audio(false);
+
+	platform_device_unregister(raumfeld_audio_device);
+
+	i2c_unregister_device(max9486_client);
+
+	gpio_free(GPIO_MCLK_RESET);
+	gpio_free(GPIO_CODEC_RESET);
+	gpio_free(GPIO_SPDIF_RESET);
+}
+
+module_init(raumfeld_audio_init);
+module_exit(raumfeld_audio_exit);
+
+/* Module information */
+MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
+MODULE_DESCRIPTION("Raumfeld audio SoC");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
new file mode 100644
index 000000000..461123ad5
--- /dev/null
+++ b/sound/soc/pxa/spitz.c
@@ -0,0 +1,346 @@
+/*
+ * spitz.c  --  SoC audio for Sharp SL-Cxx00 models Spitz, Borzoi and Akita
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/spitz.h>
+#include "../codecs/wm8750.h"
+#include "pxa2xx-i2s.h"
+
+#define SPITZ_HP        0
+#define SPITZ_MIC       1
+#define SPITZ_LINE      2
+#define SPITZ_HEADSET   3
+#define SPITZ_HP_OFF    4
+#define SPITZ_SPK_ON    0
+#define SPITZ_SPK_OFF   1
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define SPITZ_AUDIO_CLOCK 12288000
+
+static int spitz_jack_func;
+static int spitz_spk_func;
+static int spitz_mic_gpio;
+
+static void spitz_ext_control(struct snd_soc_dapm_context *dapm)
+{
+	snd_soc_dapm_mutex_lock(dapm);
+
+	if (spitz_spk_func == SPITZ_SPK_ON)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
+
+	/* set up jack connection */
+	switch (spitz_jack_func) {
+	case SPITZ_HP:
+		/* enable and unmute hp jack, disable mic bias */
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+		gpio_set_value(SPITZ_GPIO_MUTE_L, 1);
+		gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
+		break;
+	case SPITZ_MIC:
+		/* enable mic jack and bias, mute hp */
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+		gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+		gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
+		break;
+	case SPITZ_LINE:
+		/* enable line jack, disable mic bias and mute hp */
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack");
+		gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+		gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
+		break;
+	case SPITZ_HEADSET:
+		/* enable and unmute headset jack enable mic bias, mute L hp */
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
+		gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+		gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
+		break;
+	case SPITZ_HP_OFF:
+
+		/* jack removed, everything off */
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+		gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+		gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
+		break;
+	}
+
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static int spitz_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* check the jack status at stream startup */
+	spitz_ext_control(&rtd->card->dapm);
+
+	return 0;
+}
+
+static int spitz_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 clk = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk = 11289600;
+		break;
+	}
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as input (unused) */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops spitz_ops = {
+	.startup = spitz_startup,
+	.hw_params = spitz_hw_params,
+};
+
+static int spitz_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = spitz_jack_func;
+	return 0;
+}
+
+static int spitz_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+	if (spitz_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	spitz_jack_func = ucontrol->value.integer.value[0];
+	spitz_ext_control(&card->dapm);
+	return 1;
+}
+
+static int spitz_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = spitz_spk_func;
+	return 0;
+}
+
+static int spitz_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+	if (spitz_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	spitz_spk_func = ucontrol->value.integer.value[0];
+	spitz_ext_control(&card->dapm);
+	return 1;
+}
+
+static int spitz_mic_bias(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
+{
+	gpio_set_value_cansleep(spitz_mic_gpio, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+/* spitz machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_LINE("Line Jack", NULL),
+
+	/* headset is a mic and mono headphone */
+	SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Spitz machine audio_map */
+static const struct snd_soc_dapm_route spitz_audio_map[] = {
+
+	/* headphone connected to LOUT1, ROUT1 */
+	{"Headphone Jack", NULL, "LOUT1"},
+	{"Headphone Jack", NULL, "ROUT1"},
+
+	/* headset connected to ROUT1 and LINPUT1 with bias (def below) */
+	{"Headset Jack", NULL, "ROUT1"},
+
+	/* ext speaker connected to LOUT2, ROUT2  */
+	{"Ext Spk", NULL , "ROUT2"},
+	{"Ext Spk", NULL , "LOUT2"},
+
+	/* mic is connected to input 1 - with bias */
+	{"LINPUT1", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Mic Jack"},
+
+	/* line is connected to input 1 - no bias */
+	{"LINPUT1", NULL, "Line Jack"},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+	"Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum spitz_enum[] = {
+	SOC_ENUM_SINGLE_EXT(5, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
+	SOC_ENUM_EXT("Jack Function", spitz_enum[0], spitz_get_jack,
+		spitz_set_jack),
+	SOC_ENUM_EXT("Speaker Function", spitz_enum[1], spitz_get_spk,
+		spitz_set_spk),
+};
+
+/* spitz digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link spitz_dai = {
+	.name = "wm8750",
+	.stream_name = "WM8750",
+	.cpu_dai_name = "pxa2xx-i2s",
+	.codec_dai_name = "wm8750-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm8750.0-001b",
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBS_CFS,
+	.ops = &spitz_ops,
+};
+
+/* spitz audio machine driver */
+static struct snd_soc_card snd_soc_spitz = {
+	.name = "Spitz",
+	.owner = THIS_MODULE,
+	.dai_link = &spitz_dai,
+	.num_links = 1,
+
+	.controls = wm8750_spitz_controls,
+	.num_controls = ARRAY_SIZE(wm8750_spitz_controls),
+	.dapm_widgets = wm8750_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets),
+	.dapm_routes = spitz_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(spitz_audio_map),
+	.fully_routed = true,
+};
+
+static int spitz_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &snd_soc_spitz;
+	int ret;
+
+	if (machine_is_akita())
+		spitz_mic_gpio = AKITA_GPIO_MIC_BIAS;
+	else
+		spitz_mic_gpio = SPITZ_GPIO_MIC_BIAS;
+
+	ret = gpio_request(spitz_mic_gpio, "MIC GPIO");
+	if (ret)
+		goto err1;
+
+	ret = gpio_direction_output(spitz_mic_gpio, 0);
+	if (ret)
+		goto err2;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+		goto err2;
+	}
+
+	return 0;
+
+err2:
+	gpio_free(spitz_mic_gpio);
+err1:
+	return ret;
+}
+
+static int spitz_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+	gpio_free(spitz_mic_gpio);
+	return 0;
+}
+
+static struct platform_driver spitz_driver = {
+	.driver		= {
+		.name	= "spitz-audio",
+		.pm     = &snd_soc_pm_ops,
+	},
+	.probe		= spitz_probe,
+	.remove		= spitz_remove,
+};
+
+module_platform_driver(spitz_driver);
+
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Spitz");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:spitz-audio");
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
new file mode 100644
index 000000000..cb49284e8
--- /dev/null
+++ b/sound/soc/pxa/tosa.c
@@ -0,0 +1,280 @@
+/*
+ * tosa.c  --  SoC audio for Tosa
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ * GPIO's
+ *  1 - Jack Insertion
+ *  5 - Hookswitch (headset answer/hang up switch)
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <mach/tosa.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-ac97.h"
+
+#define TOSA_HP        0
+#define TOSA_MIC_INT   1
+#define TOSA_HEADSET   2
+#define TOSA_HP_OFF    3
+#define TOSA_SPK_ON    0
+#define TOSA_SPK_OFF   1
+
+static int tosa_jack_func;
+static int tosa_spk_func;
+
+static void tosa_ext_control(struct snd_soc_dapm_context *dapm)
+{
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	/* set up jack connection */
+	switch (tosa_jack_func) {
+	case TOSA_HP:
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+		break;
+	case TOSA_MIC_INT:
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Mic (Internal)");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+		break;
+	case TOSA_HEADSET:
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
+		break;
+	}
+
+	if (tosa_spk_func == TOSA_SPK_ON)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static int tosa_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* check the jack status at stream startup */
+	tosa_ext_control(&rtd->card->dapm);
+
+	return 0;
+}
+
+static struct snd_soc_ops tosa_ops = {
+	.startup = tosa_startup,
+};
+
+static int tosa_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = tosa_jack_func;
+	return 0;
+}
+
+static int tosa_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+	if (tosa_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	tosa_jack_func = ucontrol->value.integer.value[0];
+	tosa_ext_control(&card->dapm);
+	return 1;
+}
+
+static int tosa_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = tosa_spk_func;
+	return 0;
+}
+
+static int tosa_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+	if (tosa_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	tosa_spk_func = ucontrol->value.integer.value[0];
+	tosa_ext_control(&card->dapm);
+	return 1;
+}
+
+/* tosa dapm event handlers */
+static int tosa_hp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 :0);
+	return 0;
+}
+
+/* tosa machine dapm widgets */
+static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event),
+SND_SOC_DAPM_HP("Headset Jack", NULL),
+SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* tosa audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* headphone connected to HPOUTL, HPOUTR */
+	{"Headphone Jack", NULL, "HPOUTL"},
+	{"Headphone Jack", NULL, "HPOUTR"},
+
+	/* ext speaker connected to LOUT2, ROUT2 */
+	{"Speaker", NULL, "LOUT2"},
+	{"Speaker", NULL, "ROUT2"},
+
+	/* internal mic is connected to mic1, mic2 differential - with bias */
+	{"MIC1", NULL, "Mic Bias"},
+	{"MIC2", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Mic (Internal)"},
+
+	/* headset is connected to HPOUTR, and LINEINR with bias */
+	{"Headset Jack", NULL, "HPOUTR"},
+	{"LINEINR", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Headset Jack"},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+	"Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum tosa_enum[] = {
+	SOC_ENUM_SINGLE_EXT(5, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new tosa_controls[] = {
+	SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack,
+		tosa_set_jack),
+	SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk,
+		tosa_set_spk),
+};
+
+static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_dapm_nc_pin(dapm, "OUT3");
+	snd_soc_dapm_nc_pin(dapm, "MONOOUT");
+
+	return 0;
+}
+
+static struct snd_soc_dai_link tosa_dai[] = {
+{
+	.name = "AC97",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai_name = "pxa2xx-ac97",
+	.codec_dai_name = "wm9712-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm9712-codec",
+	.init = tosa_ac97_init,
+	.ops = &tosa_ops,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.cpu_dai_name = "pxa2xx-ac97-aux",
+	.codec_dai_name = "wm9712-aux",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm9712-codec",
+	.ops = &tosa_ops,
+},
+};
+
+static struct snd_soc_card tosa = {
+	.name = "Tosa",
+	.owner = THIS_MODULE,
+	.dai_link = tosa_dai,
+	.num_links = ARRAY_SIZE(tosa_dai),
+
+	.controls = tosa_controls,
+	.num_controls = ARRAY_SIZE(tosa_controls),
+	.dapm_widgets = tosa_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tosa_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static int tosa_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &tosa;
+	int ret;
+
+	ret = gpio_request_one(TOSA_GPIO_L_MUTE, GPIOF_OUT_INIT_LOW,
+			       "Headphone Jack");
+	if (ret)
+		return ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+		gpio_free(TOSA_GPIO_L_MUTE);
+	}
+	return ret;
+}
+
+static int tosa_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	gpio_free(TOSA_GPIO_L_MUTE);
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static struct platform_driver tosa_driver = {
+	.driver		= {
+		.name	= "tosa-audio",
+		.pm     = &snd_soc_pm_ops,
+	},
+	.probe		= tosa_probe,
+	.remove		= tosa_remove,
+};
+
+module_platform_driver(tosa_driver);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Tosa");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:tosa-audio");
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
new file mode 100644
index 000000000..1753c7d9e
--- /dev/null
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -0,0 +1,163 @@
+/*
+ * linux/sound/soc/pxa/ttc_dkb.c
+ *
+ * Copyright (C) 2012 Marvell International Ltd.
+ *
+ * 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; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <sound/pcm_params.h>
+#include "../codecs/88pm860x-codec.h"
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{ .pin = "Headset Stereophone",	.mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+	{ .pin = "Headset Mic 2",	.mask = SND_JACK_MICROPHONE, },
+};
+
+/* ttc machine dapm widgets */
+static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+	SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* ttc machine audio map */
+static const struct snd_soc_dapm_route ttc_audio_map[] = {
+	{"Headset Stereophone", NULL, "HS1"},
+	{"Headset Stereophone", NULL, "HS2"},
+
+	{"Ext Speaker", NULL, "LSP"},
+	{"Ext Speaker", NULL, "LSN"},
+
+	{"Lineout Out 1", NULL, "LINEOUT1"},
+	{"Lineout Out 2", NULL, "LINEOUT2"},
+
+	{"MIC1P", NULL, "Mic1 Bias"},
+	{"MIC1N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Ext Mic 1"},
+
+	{"MIC2P", NULL, "Mic1 Bias"},
+	{"MIC2N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Headset Mic 2"},
+
+	{"MIC3P", NULL, "Mic3 Bias"},
+	{"MIC3N", NULL, "Mic3 Bias"},
+	{"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+
+	/* Headset jack detection */
+	snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE |
+			      SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+			      &hs_jack, hs_jack_pins, ARRAY_SIZE(hs_jack_pins));
+	snd_soc_card_jack_new(rtd->card, "Microphone Jack", SND_JACK_MICROPHONE,
+			      &mic_jack, mic_jack_pins,
+			      ARRAY_SIZE(mic_jack_pins));
+
+	/* headphone, microphone detection & headset short detection */
+	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+			      SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+	pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+
+	return 0;
+}
+
+/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = {
+{
+	 .name = "88pm860x i2s",
+	 .stream_name = "audio playback",
+	 .codec_name = "88pm860x-codec",
+	 .platform_name = "mmp-pcm-audio",
+	 .cpu_dai_name = "pxa-ssp-dai.1",
+	 .codec_dai_name = "88pm860x-i2s",
+	 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBM_CFM,
+	 .init = ttc_pm860x_init,
+},
+};
+
+/* ttc/td audio machine driver */
+static struct snd_soc_card ttc_dkb_card = {
+	.name = "ttc-dkb-hifi",
+	.owner = THIS_MODULE,
+	.dai_link = ttc_pm860x_hifi_dai,
+	.num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),
+
+	.dapm_widgets = ttc_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets),
+	.dapm_routes = ttc_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(ttc_audio_map),
+};
+
+static int ttc_dkb_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &ttc_dkb_card;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+
+	return ret;
+}
+
+static int ttc_dkb_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver ttc_dkb_driver = {
+	.driver		= {
+		.name	= "ttc-dkb-audio",
+		.pm     = &snd_soc_pm_ops,
+	},
+	.probe		= ttc_dkb_probe,
+	.remove		= ttc_dkb_remove,
+};
+
+module_platform_driver(ttc_dkb_driver);
+
+/* Module information */
+MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC TTC DKB");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:ttc-dkb-audio");
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
new file mode 100644
index 000000000..bcbfbe830
--- /dev/null
+++ b/sound/soc/pxa/z2.c
@@ -0,0 +1,227 @@
+/*
+ * linux/sound/soc/pxa/z2.c
+ *
+ * SoC Audio driver for Aeronix Zipit Z2
+ *
+ * Copyright (C) 2009 Ken McGuire <kenm@desertweyr.com>
+ * Copyright (C) 2010 Marek Vasut <marek.vasut@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.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.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/audio.h>
+#include <mach/z2.h>
+
+#include "../codecs/wm8750.h"
+#include "pxa2xx-i2s.h"
+
+static struct snd_soc_card snd_soc_z2;
+
+static int z2_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 clk = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk = 11289600;
+		break;
+	}
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as input (unused) */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_jack hs_jack;
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{
+		.pin	= "Mic Jack",
+		.mask	= SND_JACK_MICROPHONE,
+	},
+	{
+		.pin	= "Headphone Jack",
+		.mask	= SND_JACK_HEADPHONE,
+	},
+	{
+		.pin    = "Ext Spk",
+		.mask   = SND_JACK_HEADPHONE,
+		.invert = 1
+	},
+};
+
+/* Headset jack detection gpios */
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+	{
+		.gpio		= GPIO37_ZIPITZ2_HEADSET_DETECT,
+		.name		= "hsdet-gpio",
+		.report		= SND_JACK_HEADSET,
+		.debounce_time	= 200,
+		.invert		= 1,
+	},
+};
+
+/* z2 machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+
+	/* headset is a mic and mono headphone */
+	SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Z2 machine audio_map */
+static const struct snd_soc_dapm_route z2_audio_map[] = {
+
+	/* headphone connected to LOUT1, ROUT1 */
+	{"Headphone Jack", NULL, "LOUT1"},
+	{"Headphone Jack", NULL, "ROUT1"},
+
+	/* ext speaker connected to LOUT2, ROUT2  */
+	{"Ext Spk", NULL , "ROUT2"},
+	{"Ext Spk", NULL , "LOUT2"},
+
+	/* mic is connected to R input 2 - with bias */
+	{"RINPUT2", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Mic Jack"},
+};
+
+/*
+ * Logic for a wm8750 as connected on a Z2 Device
+ */
+static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	/* NC codec pins */
+	snd_soc_dapm_disable_pin(dapm, "LINPUT3");
+	snd_soc_dapm_disable_pin(dapm, "RINPUT3");
+	snd_soc_dapm_disable_pin(dapm, "OUT3");
+	snd_soc_dapm_disable_pin(dapm, "MONO1");
+
+	/* Jack detection API stuff */
+	ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET,
+				    &hs_jack, hs_jack_pins,
+				    ARRAY_SIZE(hs_jack_pins));
+	if (ret)
+		goto err;
+
+	ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+				hs_jack_gpios);
+	if (ret)
+		goto err;
+
+	return 0;
+
+err:
+	return ret;
+}
+
+static struct snd_soc_ops z2_ops = {
+	.hw_params = z2_hw_params,
+};
+
+/* z2 digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link z2_dai = {
+	.name		= "wm8750",
+	.stream_name	= "WM8750",
+	.cpu_dai_name	= "pxa2xx-i2s",
+	.codec_dai_name	= "wm8750-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name	= "wm8750.0-001b",
+	.init		= z2_wm8750_init,
+	.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			  SND_SOC_DAIFMT_CBS_CFS,
+	.ops		= &z2_ops,
+};
+
+/* z2 audio machine driver */
+static struct snd_soc_card snd_soc_z2 = {
+	.name		= "Z2",
+	.owner		= THIS_MODULE,
+	.dai_link	= &z2_dai,
+	.num_links	= 1,
+
+	.dapm_widgets = wm8750_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets),
+	.dapm_routes = z2_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(z2_audio_map),
+};
+
+static struct platform_device *z2_snd_device;
+
+static int __init z2_init(void)
+{
+	int ret;
+
+	if (!machine_is_zipit2())
+		return -ENODEV;
+
+	z2_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!z2_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(z2_snd_device, &snd_soc_z2);
+	ret = platform_device_add(z2_snd_device);
+
+	if (ret)
+		platform_device_put(z2_snd_device);
+
+	return ret;
+}
+
+static void __exit z2_exit(void)
+{
+	platform_device_unregister(z2_snd_device);
+}
+
+module_init(z2_init);
+module_exit(z2_exit);
+
+MODULE_AUTHOR("Ken McGuire <kenm@desertweyr.com>, "
+		"Marek Vasut <marek.vasut@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC ZipitZ2");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
new file mode 100644
index 000000000..8f301c72e
--- /dev/null
+++ b/sound/soc/pxa/zylonite.c
@@ -0,0 +1,275 @@
+/*
+ * zylonite.c  --  SoC audio for Zylonite
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * 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; either version 2 of the
+ * License, or (at your option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "../codecs/wm9713.h"
+#include "pxa2xx-ac97.h"
+#include "pxa-ssp.h"
+
+/*
+ * There is a physical switch SW15 on the board which changes the MCLK
+ * for the WM9713 between the standard AC97 master clock and the
+ * output of the CLK_POUT signal from the PXA.
+ */
+static int clk_pout;
+module_param(clk_pout, int, 0);
+MODULE_PARM_DESC(clk_pout, "Use CLK_POUT as WM9713 MCLK (SW15 on board).");
+
+static struct clk *pout;
+
+static struct snd_soc_card zylonite;
+
+static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Microphone", NULL),
+	SND_SOC_DAPM_MIC("Handset Microphone", NULL),
+	SND_SOC_DAPM_SPK("Multiactor", NULL),
+	SND_SOC_DAPM_SPK("Headset Earpiece", NULL),
+};
+
+/* Currently supported audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* Headphone output connected to HPL/HPR */
+	{ "Headphone", NULL,  "HPL" },
+	{ "Headphone", NULL,  "HPR" },
+
+	/* On-board earpiece */
+	{ "Headset Earpiece", NULL, "OUT3" },
+
+	/* Headphone mic */
+	{ "MIC2A", NULL, "Mic Bias" },
+	{ "Mic Bias", NULL, "Headset Microphone" },
+
+	/* On-board mic */
+	{ "MIC1", NULL, "Mic Bias" },
+	{ "Mic Bias", NULL, "Handset Microphone" },
+
+	/* Multiactor differentially connected over SPKL/SPKR */
+	{ "Multiactor", NULL, "SPKL" },
+	{ "Multiactor", NULL, "SPKR" },
+};
+
+static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd)
+{
+	if (clk_pout)
+		snd_soc_dai_set_pll(rtd->codec_dai, 0, 0,
+				    clk_get_rate(pout), 0);
+
+	return 0;
+}
+
+static int zylonite_voice_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 pll_out = 0;
+	unsigned int wm9713_div = 0;
+	int ret = 0;
+	int rate = params_rate(params);
+	int width = snd_pcm_format_physical_width(params_format(params));
+
+	/* Only support ratios that we can generate neatly from the AC97
+	 * based master clock - in particular, this excludes 44.1kHz.
+	 * In most applications the voice DAC will be used for telephony
+	 * data so multiples of 8kHz will be the common case.
+	 */
+	switch (rate) {
+	case 8000:
+		wm9713_div = 12;
+		break;
+	case 16000:
+		wm9713_div = 6;
+		break;
+	case 48000:
+		wm9713_div = 2;
+		break;
+	default:
+		/* Don't support OSS emulation */
+		return -EINVAL;
+	}
+
+	/* Add 1 to the width for the leading clock cycle */
+	pll_out = rate * (width + 1) * 8;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out);
+	if (ret < 0)
+		return ret;
+
+	if (clk_pout)
+		ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
+					     WM9713_PCMDIV(wm9713_div));
+	else
+		ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
+					     WM9713_PCMDIV(wm9713_div));
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops zylonite_voice_ops = {
+	.hw_params = zylonite_voice_hw_params,
+};
+
+static struct snd_soc_dai_link zylonite_dai[] = {
+{
+	.name = "AC97",
+	.stream_name = "AC97 HiFi",
+	.codec_name = "wm9713-codec",
+	.platform_name = "pxa-pcm-audio",
+	.cpu_dai_name = "pxa2xx-ac97",
+	.codec_dai_name = "wm9713-hifi",
+	.init = zylonite_wm9713_init,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.codec_name = "wm9713-codec",
+	.platform_name = "pxa-pcm-audio",
+	.cpu_dai_name = "pxa2xx-ac97-aux",
+	.codec_dai_name = "wm9713-aux",
+},
+{
+	.name = "WM9713 Voice",
+	.stream_name = "WM9713 Voice",
+	.codec_name = "wm9713-codec",
+	.platform_name = "pxa-pcm-audio",
+	.cpu_dai_name = "pxa-ssp-dai.2",
+	.codec_dai_name = "wm9713-voice",
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBS_CFS,
+	.ops = &zylonite_voice_ops,
+},
+};
+
+static int zylonite_probe(struct snd_soc_card *card)
+{
+	int ret;
+
+	if (clk_pout) {
+		pout = clk_get(NULL, "CLK_POUT");
+		if (IS_ERR(pout)) {
+			dev_err(card->dev, "Unable to obtain CLK_POUT: %ld\n",
+				PTR_ERR(pout));
+			return PTR_ERR(pout);
+		}
+
+		ret = clk_enable(pout);
+		if (ret != 0) {
+			dev_err(card->dev, "Unable to enable CLK_POUT: %d\n",
+				ret);
+			clk_put(pout);
+			return ret;
+		}
+
+		dev_dbg(card->dev, "MCLK enabled at %luHz\n",
+			clk_get_rate(pout));
+	}
+
+	return 0;
+}
+
+static int zylonite_remove(struct snd_soc_card *card)
+{
+	if (clk_pout) {
+		clk_disable(pout);
+		clk_put(pout);
+	}
+
+	return 0;
+}
+
+static int zylonite_suspend_post(struct snd_soc_card *card)
+{
+	if (clk_pout)
+		clk_disable(pout);
+
+	return 0;
+}
+
+static int zylonite_resume_pre(struct snd_soc_card *card)
+{
+	int ret = 0;
+
+	if (clk_pout) {
+		ret = clk_enable(pout);
+		if (ret != 0)
+			dev_err(card->dev, "Unable to enable CLK_POUT: %d\n",
+				ret);
+	}
+
+	return ret;
+}
+
+static struct snd_soc_card zylonite = {
+	.name = "Zylonite",
+	.owner = THIS_MODULE,
+	.probe = &zylonite_probe,
+	.remove = &zylonite_remove,
+	.suspend_post = &zylonite_suspend_post,
+	.resume_pre = &zylonite_resume_pre,
+	.dai_link = zylonite_dai,
+	.num_links = ARRAY_SIZE(zylonite_dai),
+
+	.dapm_widgets = zylonite_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(zylonite_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static struct platform_device *zylonite_snd_ac97_device;
+
+static int __init zylonite_init(void)
+{
+	int ret;
+
+	zylonite_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+	if (!zylonite_snd_ac97_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(zylonite_snd_ac97_device, &zylonite);
+
+	ret = platform_device_add(zylonite_snd_ac97_device);
+	if (ret != 0)
+		platform_device_put(zylonite_snd_ac97_device);
+
+	return ret;
+}
+
+static void __exit zylonite_exit(void)
+{
+	platform_device_unregister(zylonite_snd_ac97_device);
+}
+
+module_init(zylonite_init);
+module_exit(zylonite_exit);
+
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("ALSA SoC WM9713 Zylonite");
+MODULE_LICENSE("GPL");