MERGE-via-pending-tracking-hist-MERGE-via-stable-tracking-MERGE-via-mokopatches-tracking-fix-stray-endmenu-patch-1232632040-1232632141

pending-tracking-hist top was MERGE-via-stable-tracking-MERGE-via-mokopatches-tracking-fix-stray-endmenu-patch-1232632040-1232632141 / fdf777a63bcb59e0dfd78bfe2c6242e01f6d4eb9 ... parent commitmessage:
From: merge <null@invalid>
MERGE-via-stable-tracking-hist-MERGE-via-mokopatches-tracking-fix-stray-endmenu-patch-1232632040

stable-tracking-hist top was MERGE-via-mokopatches-tracking-fix-stray-endmenu-patch-1232632040 / 90463bfd2d5a3c8b52f6e6d71024a00e052b0ced ... parent commitmessage:
From: merge <null@invalid>
MERGE-via-mokopatches-tracking-hist-fix-stray-endmenu-patch

mokopatches-tracking-hist top was fix-stray-endmenu-patch / 3630e0be570de8057e7f8d2fe501ed353cdf34e6 ... parent commitmessage:
From: Andy Green <andy@openmoko.com>
fix-stray-endmenu.patch

Signed-off-by: Andy Green <andy@openmoko.com>

15 files changed, 1599 insertions, 59 deletions

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index f8c1cdd940a..f82e1069947 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -21,6 +21,9 @@ config SND_PXA2XX_SOC_AC97
 config SND_PXA2XX_SOC_I2S
 	tristate
 
+config SND_PXA_SOC_SSP
+	tristate
+
 config SND_PXA2XX_SOC_CORGI
 	tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
 	depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
@@ -75,3 +78,22 @@ config SND_PXA2XX_SOC_EM_X270
 	help
 	  Say Y if you want to add support for SoC audio on
 	  CompuLab EM-x270.
+
+config SND_PXA2XX_SOC_PALM27X
+	bool "SoC Audio support for Palm T|X, T5 and LifeDrive"
+	depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || MACH_PALMT5)
+	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 or LifeDrive handheld computer.
+
+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.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 5bc8edf9dca..08a9f279772 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -2,10 +2,12 @@
 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
 
 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
 
 # PXA Machine Support
 snd-soc-corgi-objs := corgi.o
@@ -14,6 +16,8 @@ snd-soc-tosa-objs := tosa.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
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -21,3 +25,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
 obj-$(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_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 2718eaf7895..1ba25a55952 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -108,15 +108,11 @@ static int corgi_startup(struct snd_pcm_substream *substream)
 }
 
 /* we need to unmute the HP at shutdown as the mute burns power on corgi */
-static int corgi_shutdown(struct snd_pcm_substream *substream)
+static void corgi_shutdown(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
-
 	/* set = unmute headphone */
 	gpio_set_value(CORGI_GPIO_MUTE_L, 1);
 	gpio_set_value(CORGI_GPIO_MUTE_R, 1);
-	return 0;
 }
 
 static int corgi_hw_params(struct snd_pcm_substream *substream,
@@ -314,8 +310,9 @@ static struct snd_soc_dai_link corgi_dai = {
 };
 
 /* corgi audio machine driver */
-static struct snd_soc_machine snd_soc_machine_corgi = {
+static struct snd_soc_card snd_soc_corgi = {
 	.name = "Corgi",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = &corgi_dai,
 	.num_links = 1,
 };
@@ -328,8 +325,7 @@ static struct wm8731_setup_data corgi_wm8731_setup = {
 
 /* corgi audio subsystem */
 static struct snd_soc_device corgi_snd_devdata = {
-	.machine = &snd_soc_machine_corgi,
-	.platform = &pxa2xx_soc_platform,
+	.card = &snd_soc_corgi,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &corgi_wm8731_setup,
 };
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 6781c5be242..2e3386dfa0f 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -29,7 +29,7 @@
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
-static struct snd_soc_machine e800;
+static struct snd_soc_card e800;
 
 static struct snd_soc_dai_link e800_dai[] = {
 {
@@ -40,15 +40,15 @@ static struct snd_soc_dai_link e800_dai[] = {
 },
 };
 
-static struct snd_soc_machine e800 = {
+static struct snd_soc_card e800 = {
 	.name = "Toshiba e800",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = e800_dai,
 	.num_links = ARRAY_SIZE(e800_dai),
 };
 
 static struct snd_soc_device e800_snd_devdata = {
-	.machine = &e800,
-	.platform = &pxa2xx_soc_platform,
+	.card = &e800,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index e6ff6929ab4..fe4a729ea64 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -23,7 +23,6 @@
 #include <linux/moduleparam.h>
 #include <linux/device.h>
 
-#include <sound/driver.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
@@ -53,15 +52,15 @@ static struct snd_soc_dai_link em_x270_dai[] = {
 	},
 };
 
-static struct snd_soc_machine em_x270 = {
+static struct snd_soc_card em_x270 = {
 	.name = "EM-X270",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = em_x270_dai,
 	.num_links = ARRAY_SIZE(em_x270_dai),
 };
 
 static struct snd_soc_device em_x270_snd_devdata = {
-	.machine = &em_x270,
-	.platform = &pxa2xx_soc_platform,
+	.card = &em_x270,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
new file mode 100644
index 00000000000..4a9cf3083af
--- /dev/null
+++ b/sound/soc/pxa/palm27x.c
@@ -0,0 +1,269 @@
+/*
+ * 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 <linux/interrupt.h>
+#include <linux/irq.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+#include <mach/palmasoc.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static int palm27x_jack_func = 1;
+static int palm27x_spk_func = 1;
+static int palm27x_ep_gpio = -1;
+
+static void palm27x_ext_control(struct snd_soc_codec *codec)
+{
+	if (!palm27x_spk_func)
+		snd_soc_dapm_enable_pin(codec, "Speaker");
+	else
+		snd_soc_dapm_disable_pin(codec, "Speaker");
+
+	if (!palm27x_jack_func)
+		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+	snd_soc_dapm_sync(codec);
+}
+
+static int palm27x_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->codec;
+
+	/* check the jack status at stream startup */
+	palm27x_ext_control(codec);
+	return 0;
+}
+
+static struct snd_soc_ops palm27x_ops = {
+	.startup = palm27x_startup,
+};
+
+static irqreturn_t palm27x_interrupt(int irq, void *v)
+{
+	palm27x_spk_func = gpio_get_value(palm27x_ep_gpio);
+	palm27x_jack_func = !palm27x_spk_func;
+	return IRQ_HANDLED;
+}
+
+static int palm27x_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = palm27x_jack_func;
+	return 0;
+}
+
+static int palm27x_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (palm27x_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	palm27x_jack_func = ucontrol->value.integer.value[0];
+	palm27x_ext_control(codec);
+	return 1;
+}
+
+static int palm27x_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = palm27x_spk_func;
+	return 0;
+}
+
+static int palm27x_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (palm27x_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	palm27x_spk_func = ucontrol->value.integer.value[0];
+	palm27x_ext_control(codec);
+	return 1;
+}
+
+/* PalmTX machine dapm widgets */
+static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", 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 */
+	{"Speaker", NULL, "LOUT2"},
+	{"Speaker", NULL, "ROUT2"},
+};
+
+static const char *jack_function[] = {"Headphone", "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum palm27x_enum[] = {
+	SOC_ENUM_SINGLE_EXT(2, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new palm27x_controls[] = {
+	SOC_ENUM_EXT("Jack Function", palm27x_enum[0], palm27x_get_jack,
+		palm27x_set_jack),
+	SOC_ENUM_EXT("Speaker Function", palm27x_enum[1], palm27x_get_spk,
+		palm27x_set_spk),
+};
+
+static int palm27x_ac97_init(struct snd_soc_codec *codec)
+{
+	int i, err;
+
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "MONOOUT");
+
+	/* add palm27x specific controls */
+	for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&palm27x_controls[i],
+						codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* add palm27x specific widgets */
+	snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
+				ARRAY_SIZE(palm27x_dapm_widgets));
+
+	/* set up palm27x specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+	return 0;
+}
+
+static struct snd_soc_dai_link palm27x_dai[] = {
+{
+	.name = "AC97 HiFi",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+	.init = palm27x_ac97_init,
+	.ops = &palm27x_ops,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+	.ops = &palm27x_ops,
+},
+};
+
+static struct snd_soc_card palm27x_asoc = {
+	.name = "Palm/PXA27x",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = palm27x_dai,
+	.num_links = ARRAY_SIZE(palm27x_dai),
+};
+
+static struct snd_soc_device palm27x_snd_devdata = {
+	.card = &palm27x_asoc,
+	.codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *palm27x_snd_device;
+
+static int __init palm27x_asoc_init(void)
+{
+	int ret;
+
+	if (!(machine_is_palmtx() || machine_is_palmt5() ||
+		machine_is_palmld()))
+		return -ENODEV;
+
+	ret = gpio_request(palm27x_ep_gpio, "Headphone Jack");
+	if (ret)
+		return ret;
+	ret = gpio_direction_input(palm27x_ep_gpio);
+	if (ret)
+		goto err_alloc;
+
+	if (request_irq(gpio_to_irq(palm27x_ep_gpio), palm27x_interrupt,
+			IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
+			"Headphone jack", NULL))
+		goto err_alloc;
+
+	palm27x_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!palm27x_snd_device) {
+		ret = -ENOMEM;
+		goto err_dev;
+	}
+
+	platform_set_drvdata(palm27x_snd_device, &palm27x_snd_devdata);
+	palm27x_snd_devdata.dev = &palm27x_snd_device->dev;
+	ret = platform_device_add(palm27x_snd_device);
+
+	if (ret != 0)
+		goto put_device;
+
+	return 0;
+
+put_device:
+	platform_device_put(palm27x_snd_device);
+err_dev:
+	free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
+err_alloc:
+	gpio_free(palm27x_ep_gpio);
+
+	return ret;
+}
+
+static void __exit palm27x_asoc_exit(void)
+{
+	free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
+	gpio_free(palm27x_ep_gpio);
+	platform_device_unregister(palm27x_snd_device);
+}
+
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data)
+{
+	palm27x_ep_gpio = data->jack_gpio;
+}
+
+module_init(palm27x_asoc_init);
+module_exit(palm27x_asoc_exit);
+
+/* 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
index 4d9930c5278..6e9827189ff 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,8 +276,9 @@ static struct snd_soc_dai_link poodle_dai = {
 };
 
 /* poodle audio machine driver */
-static struct snd_soc_machine snd_soc_machine_poodle = {
+static struct snd_soc_card snd_soc_poodle = {
 	.name = "Poodle",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = &poodle_dai,
 	.num_links = 1,
 };
@@ -290,8 +291,7 @@ static struct wm8731_setup_data poodle_wm8731_setup = {
 
 /* poodle audio subsystem */
 static struct snd_soc_device poodle_snd_devdata = {
-	.machine = &snd_soc_machine_poodle,
-	.platform = &pxa2xx_soc_platform,
+	.card = &snd_soc_poodle,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &poodle_wm8731_setup,
 };
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
new file mode 100644
index 00000000000..73cb6b4c2f2
--- /dev/null
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -0,0 +1,931 @@
+#define DEBUG
+/*
+ * 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/platform_device.h>
+#include <linux/clk.h>
+#include <linux/io.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 <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/regs-ssp.h>
+#include <mach/audio.h>
+#include <mach/ssp.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa-ssp.h"
+
+/*
+ * SSP audio private data
+ */
+struct ssp_priv {
+	struct ssp_dev dev;
+	unsigned int sysclk;
+	int dai_fmt;
+#ifdef CONFIG_PM
+	struct ssp_state state;
+#endif
+};
+
+#define PXA2xx_SSP1_BASE	0x41000000
+#define PXA27x_SSP2_BASE	0x41700000
+#define PXA27x_SSP3_BASE	0x41900000
+#define PXA3xx_SSP4_BASE	0x41a00000
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = {
+	.name			= "SSP1 PCM Mono out",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(14),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = {
+	.name			= "SSP1 PCM Mono in",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(13),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = {
+	.name			= "SSP1 PCM Stereo out",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(14),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = {
+	.name			= "SSP1 PCM Stereo in",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(13),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = {
+	.name			= "SSP2 PCM Mono out",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(16),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = {
+	.name			= "SSP2 PCM Mono in",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(15),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = {
+	.name			= "SSP2 PCM Stereo out",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(16),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = {
+	.name			= "SSP2 PCM Stereo in",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(15),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = {
+	.name			= "SSP3 PCM Mono out",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = {
+	.name			= "SSP3 PCM Mono in",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = {
+	.name			= "SSP3 PCM Stereo out",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = {
+	.name			= "SSP3 PCM Stereo in",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = {
+	.name			= "SSP4 PCM Mono out",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = {
+	.name			= "SSP4 PCM Mono in",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = {
+	.name			= "SSP4 PCM Stereo out",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = {
+	.name			= "SSP4 PCM Stereo in",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static void dump_registers(struct ssp_device *ssp)
+{
+	dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n",
+		 ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1),
+		 ssp_read_reg(ssp, SSTO));
+
+	dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n",
+		 ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR),
+		 ssp_read_reg(ssp, SSACD));
+}
+
+static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = {
+	{
+		&pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in,
+		&pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in,
+	},
+	{
+		&pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in,
+		&pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in,
+	},
+	{
+		&pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in,
+		&pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in,
+	},
+	{
+		&pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in,
+		&pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in,
+	},
+};
+
+static int pxa_ssp_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->dai->cpu_dai;
+	struct ssp_priv *priv = cpu_dai->private_data;
+	int ret = 0;
+
+	if (!cpu_dai->active) {
+		ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ);
+		if (ret < 0)
+			return ret;
+		ssp_disable(&priv->dev);
+	}
+	return ret;
+}
+
+static void pxa_ssp_shutdown(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->dai->cpu_dai;
+	struct ssp_priv *priv = cpu_dai->private_data;
+
+	if (!cpu_dai->active) {
+		ssp_disable(&priv->dev);
+		ssp_exit(&priv->dev);
+	}
+}
+
+#ifdef CONFIG_PM
+
+static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+
+	if (!cpu_dai->active)
+		return 0;
+
+	ssp_save_state(&priv->dev, &priv->state);
+	clk_disable(priv->dev.ssp->clk);
+	return 0;
+}
+
+static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+
+	if (!cpu_dai->active)
+		return 0;
+
+	clk_enable(priv->dev.ssp->clk);
+	ssp_restore_state(&priv->dev, &priv->state);
+	ssp_enable(&priv->dev);
+
+	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 ssp_set_scr(struct ssp_dev *dev, u32 div)
+{
+	struct ssp_device *ssp = dev->ssp;
+	u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR;
+
+	ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(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 = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int val;
+
+	u32 sscr0 = ssp_read_reg(ssp, SSCR0) &
+		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+
+	dev_dbg(&ssp->pdev->dev,
+		"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\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 (cpu_is_pxa25x())
+			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;
+		ssp_set_scr(&priv->dev, 1);
+		sscr0 |= SSCR0_ADC;
+		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 (!cpu_is_pxa3xx())
+		clk_disable(priv->dev.ssp->clk);
+	val = ssp_read_reg(ssp, SSCR0) | sscr0;
+	ssp_write_reg(ssp, SSCR0, val);
+	if (!cpu_is_pxa3xx())
+		clk_enable(priv->dev.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 = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int val;
+
+	switch (div_id) {
+	case PXA_SSP_AUDIO_DIV_ACDS:
+		val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div);
+		ssp_write_reg(ssp, SSACD, val);
+		break;
+	case PXA_SSP_AUDIO_DIV_SCDB:
+		val = ssp_read_reg(ssp, SSACD);
+		val &= ~SSACD_SCDB;
+#if defined(CONFIG_PXA3xx)
+		if (cpu_is_pxa3xx())
+			val &= ~SSACD_SCDX8;
+#endif
+		switch (div) {
+		case PXA_SSP_CLK_SCDB_1:
+			val |= SSACD_SCDB;
+			break;
+		case PXA_SSP_CLK_SCDB_4:
+			break;
+#if defined(CONFIG_PXA3xx)
+		case PXA_SSP_CLK_SCDB_8:
+			if (cpu_is_pxa3xx())
+				val |= SSACD_SCDX8;
+			else
+				return -EINVAL;
+			break;
+#endif
+		default:
+			return -EINVAL;
+		}
+		ssp_write_reg(ssp, SSACD, val);
+		break;
+	case PXA_SSP_DIV_SCR:
+		ssp_set_scr(&priv->dev, 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, unsigned int freq_in, unsigned int freq_out)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70;
+
+#if defined(CONFIG_PXA3xx)
+	if (cpu_is_pxa3xx())
+		ssp_write_reg(ssp, SSACDD, 0);
+#endif
+
+	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:
+#ifdef CONFIG_PXA3xx
+		/* PXA3xx has a clock ditherer which can be used to generate
+		 * a wider range of frequencies - calculate a value for it.
+		 */
+		if (cpu_is_pxa3xx()) {
+			u32 val;
+			u64 tmp = 19968;
+			tmp *= 1000000;
+			do_div(tmp, freq_out);
+			val = tmp;
+
+			val = (val << 16) | 64;;
+			ssp_write_reg(ssp, SSACDD, val);
+
+			ssacd |= (0x6 << 4);
+
+			dev_dbg(&ssp->pdev->dev,
+				"Using SSACDD %x to supply %dHz\n",
+				val, freq_out);
+			break;
+		}
+#endif
+
+		return -EINVAL;
+	}
+
+	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 mask, int slots)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 sscr0;
+
+	sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7);
+
+	/* set number of active slots */
+	sscr0 |= SSCR0_SlotsPerFrm(slots);
+	ssp_write_reg(ssp, SSCR0, sscr0);
+
+	/* set active slot mask */
+	ssp_write_reg(ssp, SSTSA, mask);
+	ssp_write_reg(ssp, SSRSA, mask);
+	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 = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 sscr1;
+
+	sscr1 = ssp_read_reg(ssp, SSCR1);
+	if (tristate)
+		sscr1 &= ~SSCR1_TTE;
+	else
+		sscr1 |= SSCR1_TTE;
+	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 = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 sscr0;
+	u32 sscr1;
+	u32 sspsp;
+
+	/* reset port settings */
+	sscr0 = ssp_read_reg(ssp, SSCR0) &
+		(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+	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;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		sscr1 |= SSCR1_SCLKDIR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	ssp_write_reg(ssp, SSCR0, sscr0);
+	ssp_write_reg(ssp, SSCR1, sscr1);
+	ssp_write_reg(ssp, SSPSP, sspsp);
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		sscr0 |= SSCR0_MOD | SSCR0_PSP;
+		sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
+
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			sspsp |= SSPSP_FSRT;
+			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			sspsp |= SSPSP_SFRMP | SSPSP_FSRT;
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			sspsp |= SSPSP_SFRMP;
+			break;
+		default:
+			return -EINVAL;
+		}
+		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;
+
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			sspsp |= SSPSP_SFRMP;
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	ssp_write_reg(ssp, SSCR0, sscr0);
+	ssp_write_reg(ssp, SSCR1, sscr1);
+	ssp_write_reg(ssp, SSPSP, sspsp);
+
+	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 *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int dma = 0, chn = params_channels(params);
+	u32 sscr0;
+	u32 sspsp;
+	int width = snd_pcm_format_physical_width(params_format(params));
+
+	/* select correct DMA params */
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		dma = 1; /* capture DMA offset is 1,3 */
+	if (chn == 2)
+		dma += 2; /* stereo DMA offset is 2, mono is 0 */
+	cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
+
+	dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
+
+	/* we can only change the settings if the port is not in use */
+	if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
+		return 0;
+
+	/* clear selected SSP bits */
+	sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS);
+	ssp_write_reg(ssp, SSCR0, sscr0);
+
+	/* bit size */
+	sscr0 = ssp_read_reg(ssp, SSCR0);
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+#ifdef CONFIG_PXA3xx
+		if (cpu_is_pxa3xx())
+			sscr0 |= SSCR0_FPCKE;
+#endif
+		sscr0 |= SSCR0_DataSize(16);
+		if (params_channels(params) > 1)
+			sscr0 |= SSCR0_EDSS;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8));
+		/* we must be in network mode (2 slots) for 24 bit stereo */
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16));
+		/* we must be in network mode (2 slots) for 32 bit stereo */
+		break;
+	}
+	ssp_write_reg(ssp, SSCR0, sscr0);
+
+	switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		/* Cleared when the DAI format is set */
+		sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width);
+		ssp_write_reg(ssp, SSPSP, sspsp);
+		break;
+	default:
+		break;
+	}
+
+	/* We always use a network mode so we always require TDM slots
+	 * - complain loudly and fail if they've not been set up yet.
+	 */
+	if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+		dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
+		return -EINVAL;
+	}
+
+	dump_registers(ssp);
+
+	return 0;
+}
+
+static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret = 0;
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int val;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_RESUME:
+		ssp_enable(&priv->dev);
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val |= SSCR1_TSRE;
+		else
+			val |= SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		val = ssp_read_reg(ssp, SSSR);
+		ssp_write_reg(ssp, SSSR, val);
+		break;
+	case SNDRV_PCM_TRIGGER_START:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val |= SSCR1_TSRE;
+		else
+			val |= SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		ssp_enable(&priv->dev);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val &= ~SSCR1_TSRE;
+		else
+			val &= ~SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		ssp_disable(&priv->dev);
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val &= ~SSCR1_TSRE;
+		else
+			val &= ~SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	dump_registers(ssp);
+
+	return ret;
+}
+
+static int pxa_ssp_probe(struct platform_device *pdev,
+			    struct snd_soc_dai *dai)
+{
+	struct ssp_priv *priv;
+	int ret;
+
+	priv = kzalloc(sizeof(struct ssp_priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->dev.ssp = ssp_request(dai->id, "SoC audio");
+	if (priv->dev.ssp == NULL) {
+		ret = -ENODEV;
+		goto err_priv;
+	}
+
+	dai->private_data = priv;
+
+	return 0;
+
+err_priv:
+	kfree(priv);
+	return ret;
+}
+
+static void pxa_ssp_remove(struct platform_device *pdev,
+			      struct snd_soc_dai *dai)
+{
+	struct ssp_priv *priv = dai->private_data;
+	ssp_free(priv->dev.ssp);
+}
+
+#define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+			  SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |	\
+			  SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |	\
+			  SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			    SNDRV_PCM_FMTBIT_S24_LE |	\
+			    SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai pxa_ssp_dai[] = {
+	{
+		.name = "pxa2xx-ssp1",
+		.id = 0,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			 .channels_min = 1,
+			 .channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.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,
+		},
+	},
+	{	.name = "pxa2xx-ssp2",
+		.id = 1,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.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,
+		},
+	},
+	{
+		.name = "pxa2xx-ssp3",
+		.id = 2,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.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,
+		},
+	},
+	{
+		.name = "pxa2xx-ssp4",
+		.id = 3,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.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,
+		},
+	},
+};
+EXPORT_SYMBOL_GPL(pxa_ssp_dai);
+
+static int __init pxa_ssp_init(void)
+{
+	return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+}
+module_init(pxa_ssp_init);
+
+static void __exit pxa_ssp_exit(void)
+{
+	snd_soc_unregister_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+}
+module_exit(pxa_ssp_exit);
+
+/* 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 00000000000..91deadd5567
--- /dev/null
+++ b/sound/soc/pxa/pxa-ssp.h
@@ -0,0 +1,47 @@
+/*
+ * 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
+
+extern struct snd_soc_dai pxa_ssp_dai[4];
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index a7a3a9c5c6f..812c2b4d3e0 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -21,6 +21,7 @@
 
 #include <mach/hardware.h>
 #include <mach/pxa-regs.h>
+#include <mach/regs-ac97.h>
 
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
@@ -87,14 +88,12 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
 };
 
 #ifdef CONFIG_PM
-static int pxa2xx_ac97_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_ac97_suspend(struct snd_soc_dai *dai)
 {
 	return pxa2xx_ac97_hw_suspend();
 }
 
-static int pxa2xx_ac97_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_ac97_resume(struct snd_soc_dai *dai)
 {
 	return pxa2xx_ac97_hw_resume();
 }
@@ -117,7 +116,8 @@ static void pxa2xx_ac97_remove(struct platform_device *pdev,
 }
 
 static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				 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->dai->cpu_dai;
@@ -131,7 +131,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
 }
 
 static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				     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->dai->cpu_dai;
@@ -145,7 +146,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
 }
 
 static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				     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->dai->cpu_dai;
@@ -170,7 +172,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97",
 	.id = 0,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.probe = pxa2xx_ac97_probe,
 	.remove = pxa2xx_ac97_remove,
 	.suspend = pxa2xx_ac97_suspend,
@@ -193,7 +195,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97-aux",
 	.id = 1,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "AC97 Aux Playback",
 		.channels_min = 1,
@@ -212,7 +214,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97-mic",
 	.id = 2,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.capture = {
 		.stream_name = "AC97 Mic Capture",
 		.channels_min = 1,
@@ -227,6 +229,18 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 EXPORT_SYMBOL_GPL(pxa_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
+static int __init pxa_ac97_init(void)
+{
+	return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+module_init(pxa_ac97_init);
+
+static void __exit pxa_ac97_exit(void)
+{
+	snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+module_exit(pxa_ac97_exit);
+
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index e758034db5c..517991fb109 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -121,7 +121,8 @@ static struct pxa2xx_gpio gpio_bus[] = {
 	},
 };
 
-static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
+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->dai->cpu_dai;
@@ -187,7 +188,8 @@ static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 }
 
 static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				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->dai->cpu_dai;
@@ -248,7 +250,8 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			      struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -269,7 +272,8 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
 	return ret;
 }
 
-static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
+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;
@@ -289,8 +293,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
 }
 
 #ifdef CONFIG_PM
-static int pxa2xx_i2s_suspend(struct platform_device *dev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
 {
 	if (!dai->active)
 		return 0;
@@ -307,8 +310,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev,
 	return 0;
 }
 
-static int pxa2xx_i2s_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
 {
 	if (!dai->active)
 		return 0;
@@ -336,7 +338,6 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai pxa_i2s_dai = {
 	.name = "pxa2xx-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.suspend = pxa2xx_i2s_suspend,
 	.resume = pxa2xx_i2s_resume,
 	.playback = {
@@ -353,8 +354,7 @@ struct snd_soc_dai pxa_i2s_dai = {
 		.startup = pxa2xx_i2s_startup,
 		.shutdown = pxa2xx_i2s_shutdown,
 		.trigger = pxa2xx_i2s_trigger,
-		.hw_params = pxa2xx_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = pxa2xx_i2s_hw_params,
 		.set_fmt = pxa2xx_i2s_set_dai_fmt,
 		.set_sysclk = pxa2xx_i2s_set_dai_sysclk,
 	},
@@ -364,12 +364,23 @@ EXPORT_SYMBOL_GPL(pxa_i2s_dai);
 
 static int pxa2xx_i2s_probe(struct platform_device *dev)
 {
+	int ret;
+
 	clk_i2s = clk_get(&dev->dev, "I2SCLK");
-	return IS_ERR(clk_i2s) ? PTR_ERR(clk_i2s) : 0;
+	if (IS_ERR(clk_i2s))
+		return PTR_ERR(clk_i2s);
+
+	pxa_i2s_dai.dev = &dev->dev;
+	ret = snd_soc_register_dai(&pxa_i2s_dai);
+	if (ret != 0)
+		clk_put(clk_i2s);
+
+	return ret;
 }
 
 static int __devexit pxa2xx_i2s_remove(struct platform_device *dev)
 {
+	snd_soc_unregister_dai(&pxa_i2s_dai);
 	clk_put(clk_i2s);
 	clk_i2s = ERR_PTR(-ENOENT);
 	return 0;
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index afcd892cd2f..53b9fb127a6 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -61,15 +61,15 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 
 	__pxa2xx_pcm_hw_free(substream);
 
-	if (prtd->dma_ch) {
+	if (prtd->dma_ch >= 0) {
 		pxa_free_dma(prtd->dma_ch);
-		prtd->dma_ch = 0;
+		prtd->dma_ch = -1;
 	}
 
 	return 0;
 }
 
-struct snd_pcm_ops pxa2xx_pcm_ops = {
+static struct snd_pcm_ops pxa2xx_pcm_ops = {
 	.open		= __pxa2xx_pcm_open,
 	.close		= __pxa2xx_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
@@ -118,6 +118,18 @@ struct snd_soc_platform pxa2xx_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
 
+static int __init pxa2xx_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&pxa2xx_soc_platform);
+}
+module_init(pxa2xx_soc_platform_init);
+
+static void __exit pxa2xx_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&pxa2xx_soc_platform);
+}
+module_exit(pxa2xx_soc_platform_exit);
+
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d307b6757e9..a3b9e6bdf97 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -319,8 +319,9 @@ static struct snd_soc_dai_link spitz_dai = {
 };
 
 /* spitz audio machine driver */
-static struct snd_soc_machine snd_soc_machine_spitz = {
+static struct snd_soc_card snd_soc_spitz = {
 	.name = "Spitz",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = &spitz_dai,
 	.num_links = 1,
 };
@@ -333,8 +334,7 @@ static struct wm8750_setup_data spitz_wm8750_setup = {
 
 /* spitz audio subsystem */
 static struct snd_soc_device spitz_snd_devdata = {
-	.machine = &snd_soc_machine_spitz,
-	.platform = &pxa2xx_soc_platform,
+	.card = &snd_soc_spitz,
 	.codec_dev = &soc_codec_dev_wm8750,
 	.codec_data = &spitz_wm8750_setup,
 };
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index afefe41b8c4..c77194f74c9 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -38,7 +38,7 @@
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
-static struct snd_soc_machine tosa;
+static struct snd_soc_card tosa;
 
 #define TOSA_HP        0
 #define TOSA_MIC_INT   1
@@ -230,15 +230,37 @@ static struct snd_soc_dai_link tosa_dai[] = {
 },
 };
 
-static struct snd_soc_machine tosa = {
+static int tosa_probe(struct platform_device *dev)
+{
+	int ret;
+
+	ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
+	if (ret)
+		return ret;
+	ret = gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
+	if (ret)
+		gpio_free(TOSA_GPIO_L_MUTE);
+
+	return ret;
+}
+
+static int tosa_remove(struct platform_device *dev)
+{
+	gpio_free(TOSA_GPIO_L_MUTE);
+	return 0;
+}
+
+static struct snd_soc_card tosa = {
 	.name = "Tosa",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = tosa_dai,
 	.num_links = ARRAY_SIZE(tosa_dai),
+	.probe = tosa_probe,
+	.remove = tosa_remove,
 };
 
 static struct snd_soc_device tosa_snd_devdata = {
-	.machine = &tosa,
-	.platform = &pxa2xx_soc_platform,
+	.card = &tosa,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
@@ -251,11 +273,6 @@ static int __init tosa_init(void)
 	if (!machine_is_tosa())
 		return -ENODEV;
 
-	ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
-	if (ret)
-		return ret;
-	gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
-
 	tosa_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!tosa_snd_device) {
 		ret = -ENOMEM;
@@ -272,15 +289,12 @@ static int __init tosa_init(void)
 	platform_device_put(tosa_snd_device);
 
 err_alloc:
-	gpio_free(TOSA_GPIO_L_MUTE);
-
 	return ret;
 }
 
 static void __exit tosa_exit(void)
 {
 	platform_device_unregister(tosa_snd_device);
-	gpio_free(TOSA_GPIO_L_MUTE);
 }
 
 module_init(tosa_init);
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
new file mode 100644
index 00000000000..f8e9ecd589d
--- /dev/null
+++ b/sound/soc/pxa/zylonite.c
@@ -0,0 +1,219 @@
+/*
+ * 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/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/wm9713.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+#include "pxa-ssp.h"
+
+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_codec *codec)
+{
+	/* Currently we only support use of the AC97 clock here.  If
+	 * CLK_POUT is selected by SW15 then the clock API will need
+	 * to be used to request and enable it here.
+	 */
+
+	snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
+				  ARRAY_SIZE(zylonite_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* Static setup for now */
+	snd_soc_dapm_enable_pin(codec, "Headphone");
+	snd_soc_dapm_enable_pin(codec, "Headset Earpiece");
+
+	snd_soc_dapm_sync(codec);
+	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->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int pll_out = 0;
+	unsigned int acds = 0;
+	unsigned int wm9713_div = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+		wm9713_div = 12;
+		pll_out = 2048000;
+		break;
+	case 16000:
+		wm9713_div = 6;
+		pll_out = 4096000;
+		break;
+	case 48000:
+	default:
+		wm9713_div = 2;
+		pll_out = 12288000;
+		acds = 1;
+		break;
+	}
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai,
+				       params_channels(params),
+				       params_channels(params));
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
+	if (ret < 0)
+		return ret;
+
+	/* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs
+	 * to be set instead.
+	 */
+	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",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
+	.init = zylonite_wm9713_init,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
+},
+{
+	.name = "WM9713 Voice",
+	.stream_name = "WM9713 Voice",
+	.cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3],
+	.codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE],
+	.ops = &zylonite_voice_ops,
+},
+};
+
+static struct snd_soc_card zylonite = {
+	.name = "Zylonite",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = zylonite_dai,
+	.num_links = ARRAY_SIZE(zylonite_dai),
+};
+
+static struct snd_soc_device zylonite_snd_ac97_devdata = {
+	.card = &zylonite,
+	.codec_dev = &soc_codec_dev_wm9713,
+};
+
+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_snd_ac97_devdata);
+	zylonite_snd_ac97_devdata.dev = &zylonite_snd_ac97_device->dev;
+
+	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");