11 files changed, 1280 insertions, 12 deletions

diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index df494d1e346..923428fc1ad 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,6 +1,7 @@
 config SND_S3C24XX_SOC
 	tristate "SoC Audio for the Samsung S3CXXXX chips"
-	depends on ARCH_S3C2410
+	depends on ARCH_S3C2410 || ARCH_S3C64XX
+	select S3C64XX_DMA if ARCH_S3C64XX
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the S3C24XX AC97 or I2S interfaces. You will also need to
@@ -38,6 +39,15 @@ config SND_S3C24XX_SOC_NEO1973_WM8753
 	  Say Y if you want to add support for SoC audio on smdk2440
 	  with the WM8753.
 
+config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753
+	tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)"
+	depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02
+	select SND_S3C24XX_SOC_I2S
+	select SND_SOC_WM8753
+	help
+	  This driver provides audio support for the Openmoko Neo FreeRunner
+	  smartphone.
+	  
 config SND_S3C24XX_SOC_JIVE_WM8750
 	tristate "SoC I2S Audio support for Jive"
 	depends on SND_S3C24XX_SOC && MACH_JIVE
@@ -57,7 +67,7 @@ config SND_S3C24XX_SOC_SMDK2443_WM9710
 
 config SND_S3C24XX_SOC_LN2440SBC_ALC650
 	tristate "SoC AC97 Audio support for LN2440SBC - ALC650"
-	depends on SND_S3C24XX_SOC
+	depends on SND_S3C24XX_SOC && ARCH_S3C2410
 	select SND_S3C2443_SOC_AC97
 	select SND_SOC_AC97_CODEC
 	help
@@ -66,7 +76,26 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650
 
 config SND_S3C24XX_SOC_S3C24XX_UDA134X
 	tristate "SoC I2S Audio support UDA134X wired to a S3C24XX"
-       	depends on SND_S3C24XX_SOC
+       	depends on SND_S3C24XX_SOC && ARCH_S3C2410
        	select SND_S3C24XX_SOC_I2S
 	select SND_SOC_L3
        	select SND_SOC_UDA134X
+
+config SND_S3C24XX_SOC_SIMTEC
+	tristate
+	help
+	  Internal node for common S3C24XX/Simtec suppor
+
+config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23
+	tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards"
+	depends on SND_S3C24XX_SOC && ARCH_S3C2410
+	select SND_S3C24XX_SOC_I2S
+	select SND_SOC_TLV320AIC23
+	select SND_S3C24XX_SOC_SIMTEC
+
+config SND_S3C24XX_SOC_SIMTEC_HERMES
+	tristate "SoC I2S Audio support for Simtec Hermes board"
+	depends on SND_S3C24XX_SOC && ARCH_S3C2410
+	select SND_S3C24XX_SOC_I2S
+	select SND_SOC_TLV320AIC3X
+	select SND_S3C24XX_SOC_SIMTEC
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 07a93a2ebe5..99f5a7dd3fc 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -16,12 +16,21 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o
 # S3C24XX Machine Support
 snd-soc-jive-wm8750-objs := jive_wm8750.o
 snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
+snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
 snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
 snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
 snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
+snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o
+snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o
+snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o
 
 obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
 obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
+obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o
 obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
 obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
 obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
+obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o
+obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o
+obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o
+
diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
new file mode 100644
index 00000000000..0c52e36ddd8
--- /dev/null
+++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
@@ -0,0 +1,498 @@
+/*
+ * neo1973_gta02_wm8753.c  --  SoC audio for Openmoko Freerunner(GTA02)
+ *
+ * Copyright 2007 Openmoko Inc
+ * Author: Graeme Gregory <graeme@openmoko.org>
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory <linux@wolfsonmicro.com>
+ * Copyright 2009 Wolfson Microelectronics
+ *
+ *  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 <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+
+#include <plat/regs-iis.h>
+
+#include <mach/regs-clock.h>
+#include <asm/io.h>
+#include <mach/gta02.h>
+#include "../codecs/wm8753.h"
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+
+static struct snd_soc_card neo1973_gta02;
+
+static int neo1973_gta02_hifi_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, bclk = 0;
+	int ret = 0;
+	unsigned long iis_clkrate;
+
+	iis_clkrate = s3c24xx_i2s_get_clockrate();
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+		pll_out = 12288000;
+		break;
+	case 48000:
+		bclk = WM8753_BCLK_DIV_4;
+		pll_out = 12288000;
+		break;
+	case 96000:
+		bclk = WM8753_BCLK_DIV_2;
+		pll_out = 12288000;
+		break;
+	case 11025:
+		bclk = WM8753_BCLK_DIV_16;
+		pll_out = 11289600;
+		break;
+	case 22050:
+		bclk = WM8753_BCLK_DIV_8;
+		pll_out = 11289600;
+		break;
+	case 44100:
+		bclk = WM8753_BCLK_DIV_4;
+		pll_out = 11289600;
+		break;
+	case 88200:
+		bclk = WM8753_BCLK_DIV_2;
+		pll_out = 11289600;
+		break;
+	}
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set MCLK division for sample rate */
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+		S3C2410_IISMOD_32FS);
+	if (ret < 0)
+		return ret;
+
+	/* set codec BCLK division for sample rate */
+	ret = snd_soc_dai_set_clkdiv(codec_dai,
+					WM8753_BCLKDIV, bclk);
+	if (ret < 0)
+		return ret;
+
+	/* set prescaler division for sample rate */
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+		S3C24XX_PRESCALE(4, 4));
+	if (ret < 0)
+		return ret;
+
+	/* codec PLL input is PCLK/4 */
+	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
+		iis_clkrate / 4, pll_out);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+	/* disable the PLL */
+	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
+}
+
+/*
+ * Neo1973 WM8753 HiFi DAI opserations.
+ */
+static struct snd_soc_ops neo1973_gta02_hifi_ops = {
+	.hw_params = neo1973_gta02_hifi_hw_params,
+	.hw_free = neo1973_gta02_hifi_hw_free,
+};
+
+static int neo1973_gta02_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;
+	unsigned int pcmdiv = 0;
+	int ret = 0;
+	unsigned long iis_clkrate;
+
+	iis_clkrate = s3c24xx_i2s_get_clockrate();
+
+	if (params_rate(params) != 8000)
+		return -EINVAL;
+	if (params_channels(params) != 1)
+		return -EINVAL;
+
+	pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
+
+	/* todo: gg check mode (DSP_B) against CSR datasheet */
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK,
+		12288000, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set codec PCM division for sample rate */
+	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV,
+					pcmdiv);
+	if (ret < 0)
+		return ret;
+
+	/* configue and enable PLL for 12.288MHz output */
+	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2,
+		iis_clkrate / 4, 12288000);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+	/* disable the PLL */
+	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
+}
+
+static struct snd_soc_ops neo1973_gta02_voice_ops = {
+	.hw_params = neo1973_gta02_voice_hw_params,
+	.hw_free = neo1973_gta02_voice_hw_free,
+};
+
+#define LM4853_AMP 1
+#define LM4853_SPK 2
+
+static u8 lm4853_state;
+
+/* This has no effect, it exists only to maintain compatibility with
+ * existing ALSA state files.
+ */
+static int lm4853_set_state(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	int val = ucontrol->value.integer.value[0];
+
+	if (val)
+		lm4853_state |= LM4853_AMP;
+	else
+		lm4853_state &= ~LM4853_AMP;
+
+	return 0;
+}
+
+static int lm4853_get_state(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP;
+
+	return 0;
+}
+
+static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	int val = ucontrol->value.integer.value[0];
+
+	if (val) {
+		lm4853_state |= LM4853_SPK;
+		gpio_set_value(GTA02_GPIO_HP_IN, 0);
+	} else {
+		lm4853_state &= ~LM4853_SPK;
+		gpio_set_value(GTA02_GPIO_HP_IN, 1);
+	}
+
+	return 0;
+}
+
+static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1;
+
+	return 0;
+}
+
+static int lm4853_event(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *k,
+			int event)
+{
+	gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(value));
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
+	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("Handset Mic", NULL),
+	SND_SOC_DAPM_SPK("Handset Spk", NULL),
+};
+
+
+/* example machine audio_mapnections */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* Connections to the lm4853 amp */
+	{"Stereo Out", NULL, "LOUT1"},
+	{"Stereo Out", NULL, "ROUT1"},
+
+	/* Connections to the GSM Module */
+	{"GSM Line Out", NULL, "MONO1"},
+	{"GSM Line Out", NULL, "MONO2"},
+	{"RXP", NULL, "GSM Line In"},
+	{"RXN", NULL, "GSM Line In"},
+
+	/* Connections to Headset */
+	{"MIC1", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Headset Mic"},
+
+	/* Call Mic */
+	{"MIC2", NULL, "Mic Bias"},
+	{"MIC2N", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Handset Mic"},
+
+	/* Call Speaker */
+	{"Handset Spk", NULL, "LOUT2"},
+	{"Handset Spk", NULL, "ROUT2"},
+
+	/* Connect the ALC pins */
+	{"ACIN", NULL, "ACOP"},
+};
+
+static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Stereo Out"),
+	SOC_DAPM_PIN_SWITCH("GSM Line Out"),
+	SOC_DAPM_PIN_SWITCH("GSM Line In"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Handset Mic"),
+	SOC_DAPM_PIN_SWITCH("Handset Spk"),
+
+	/* This has no effect, it exists only to maintain compatibility with
+	 * existing ALSA state files.
+	 */
+	SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0,
+		lm4853_get_state,
+		lm4853_set_state),
+	SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0,
+		lm4853_get_spk,
+		lm4853_set_spk),
+};
+
+/*
+ * This is an example machine initialisation for a wm8753 connected to a
+ * neo1973 GTA02.
+ */
+static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
+{
+	int err;
+
+	/* set up NC codec pins */
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "OUT4");
+	snd_soc_dapm_nc_pin(codec, "LINE1");
+	snd_soc_dapm_nc_pin(codec, "LINE2");
+
+	/* Add neo1973 gta02 specific widgets */
+	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+				  ARRAY_SIZE(wm8753_dapm_widgets));
+
+	/* add neo1973 gta02 specific controls */
+	err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls,
+		ARRAY_SIZE(wm8753_neo1973_gta02_controls));
+
+	if (err < 0)
+		return err;
+
+	/* set up neo1973 gta02 specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* set endpoints to default off mode */
+	snd_soc_dapm_disable_pin(codec, "Stereo Out");
+	snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+	snd_soc_dapm_disable_pin(codec, "GSM Line In");
+	snd_soc_dapm_disable_pin(codec, "Headset Mic");
+	snd_soc_dapm_disable_pin(codec, "Handset Mic");
+	snd_soc_dapm_disable_pin(codec, "Handset Spk");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+/*
+ * BT Codec DAI
+ */
+static struct snd_soc_dai bt_dai = {
+	.name = "Bluetooth",
+	.id = 0,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+};
+
+static struct snd_soc_dai_link neo1973_gta02_dai[] = {
+{ /* Hifi Playback - for similatious use with voice below */
+	.name = "WM8753",
+	.stream_name = "WM8753 HiFi",
+	.cpu_dai = &s3c24xx_i2s_dai,
+	.codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
+	.init = neo1973_gta02_wm8753_init,
+	.ops = &neo1973_gta02_hifi_ops,
+},
+{ /* Voice via BT */
+	.name = "Bluetooth",
+	.stream_name = "Voice",
+	.cpu_dai = &bt_dai,
+	.codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
+	.ops = &neo1973_gta02_voice_ops,
+},
+};
+
+static struct snd_soc_card neo1973_gta02 = {
+	.name = "neo1973-gta02",
+	.platform = &s3c24xx_soc_platform,
+	.dai_link = neo1973_gta02_dai,
+	.num_links = ARRAY_SIZE(neo1973_gta02_dai),
+};
+
+static struct snd_soc_device neo1973_gta02_snd_devdata = {
+	.card = &neo1973_gta02,
+	.codec_dev = &soc_codec_dev_wm8753,
+};
+
+static struct platform_device *neo1973_gta02_snd_device;
+
+static int __init neo1973_gta02_init(void)
+{
+	int ret;
+
+	if (!machine_is_neo1973_gta02()) {
+		printk(KERN_INFO
+		       "Only GTA02 is supported by this ASoC driver\n");
+		return -ENODEV;
+	}
+
+	/* register bluetooth DAI here */
+	ret = snd_soc_register_dai(&bt_dai);
+	if (ret)
+		return ret;
+
+	neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!neo1973_gta02_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(neo1973_gta02_snd_device,
+			&neo1973_gta02_snd_devdata);
+	neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev;
+	ret = platform_device_add(neo1973_gta02_snd_device);
+
+	if (ret) {
+		platform_device_put(neo1973_gta02_snd_device);
+		return ret;
+	}
+
+	/* Initialise GPIOs used by amp */
+	ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN");
+	if (ret) {
+		pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN);
+		goto err_unregister_device;
+	}
+
+	ret = gpio_direction_output(GTA02_GPIO_AMP_HP_IN, 1);
+	if (ret) {
+		pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN);
+		goto err_free_gpio_hp_in;
+	}
+
+	ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT");
+	if (ret) {
+		pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT);
+		goto err_free_gpio_hp_in;
+	}
+
+	ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1);
+	if (ret) {
+		pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT);
+		goto err_free_gpio_amp_shut;
+	}
+
+	return 0;
+
+err_free_gpio_amp_shut:
+	gpio_free(GTA02_GPIO_AMP_SHUT);
+err_free_gpio_hp_in:
+	gpio_free(GTA02_GPIO_HP_IN);
+err_unregister_device:
+	platform_device_unregister(neo1973_gta02_snd_device);
+	return ret;
+}
+module_init(neo1973_gta02_init);
+
+static void __exit neo1973_gta02_exit(void)
+{
+	snd_soc_unregister_dai(&bt_dai);
+	platform_device_unregister(neo1973_gta02_snd_device);
+	gpio_free(GTA02_GPIO_HP_IN);
+	gpio_free(GTA02_GPIO_AMP_SHUT);
+}
+module_exit(neo1973_gta02_exit);
+
+/* Module information */
+MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org");
+MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 1a283170ca9..aa7af0b8d42 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -36,6 +36,7 @@
 #include <mach/dma.h>
 
 #include "s3c-i2s-v2.h"
+#include "s3c24xx-pcm.h"
 
 #undef S3C_IIS_V2_SUPPORTED
 
@@ -357,19 +358,19 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 #endif
 
 #ifdef CONFIG_PLAT_S3C64XX
-	iismod &= ~0x606;
+	iismod &= ~(S3C64XX_IISMOD_BLC_MASK | S3C2412_IISMOD_BCLK_MASK);
 	/* Sample size */
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
 		/* 8 bit sample, 16fs BCLK */
-		iismod |= 0x2004;
+		iismod |= (S3C64XX_IISMOD_BLC_8BIT | S3C2412_IISMOD_BCLK_16FS);
 		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		/* 16 bit sample, 32fs BCLK */
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
 		/* 24 bit sample, 48fs BCLK */
-		iismod |= 0x4002;
+		iismod |= (S3C64XX_IISMOD_BLC_24BIT | S3C2412_IISMOD_BCLK_48FS);
 		break;
 	}
 #endif
@@ -387,6 +388,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
 	unsigned long irqs;
 	int ret = 0;
+	int channel = ((struct s3c24xx_pcm_dma_params *)
+		  rtd->dai->cpu_dai->dma_data)->channel;
 
 	pr_debug("Entered %s\n", __func__);
 
@@ -416,6 +419,14 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 			s3c2412_snd_txctrl(i2s, 1);
 
 		local_irq_restore(irqs);
+
+		/*
+		 * Load the next buffer to DMA to meet the reqirement
+		 * of the auto reload mechanism of S3C24XX.
+		 * This call won't bother S3C64XX.
+		 */
+		s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 3f03d5ddfac..fc1beb0930b 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -47,7 +47,7 @@ static struct s3c24xx_ac97_info s3c24xx_ac97;
 
 static DECLARE_COMPLETION(ac97_completion);
 static u32 codec_ready;
-static DECLARE_MUTEX(ac97_mutex);
+static DEFINE_MUTEX(ac97_mutex);
 
 static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97,
 	unsigned short reg)
@@ -56,7 +56,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97,
 	u32 ac_codec_cmd;
 	u32 stat, addr, data;
 
-	down(&ac97_mutex);
+	mutex_lock(&ac97_mutex);
 
 	codec_ready = S3C_AC97_GLBSTAT_CODECREADY;
 	ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
@@ -79,7 +79,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97,
 		printk(KERN_ERR "s3c24xx-ac97: req addr = %02x,"
 				" rep addr = %02x\n", reg, addr);
 
-	up(&ac97_mutex);
+	mutex_unlock(&ac97_mutex);
 
 	return (unsigned short)data;
 }
@@ -90,7 +90,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 	u32 ac_glbctrl;
 	u32 ac_codec_cmd;
 
-	down(&ac97_mutex);
+	mutex_lock(&ac97_mutex);
 
 	codec_ready = S3C_AC97_GLBSTAT_CODECREADY;
 	ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
@@ -109,7 +109,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 	ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ;
 	writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
 
-	up(&ac97_mutex);
+	mutex_unlock(&ac97_mutex);
 
 }
 
@@ -290,6 +290,9 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
 				struct snd_soc_dai *dai)
 {
 	u32 ac_glbctrl;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int channel = ((struct s3c24xx_pcm_dma_params *)
+		  rtd->dai->cpu_dai->dma_data)->channel;
 
 	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
 	switch (cmd) {
@@ -312,6 +315,8 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
 	}
 	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
 
+	s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+
 	return 0;
 }
 
@@ -334,6 +339,9 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
 				    int cmd, struct snd_soc_dai *dai)
 {
 	u32 ac_glbctrl;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int channel = ((struct s3c24xx_pcm_dma_params *)
+		  rtd->dai->cpu_dai->dma_data)->channel;
 
 	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
 	switch (cmd) {
@@ -349,6 +357,8 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
 	}
 	writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
 
+	s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+
 	return 0;
 }
 
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 556e35f0ab7..40e2c4790f0 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -279,6 +279,9 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 			       struct snd_soc_dai *dai)
 {
 	int ret = 0;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int channel = ((struct s3c24xx_pcm_dma_params *)
+		  rtd->dai->cpu_dai->dma_data)->channel;
 
 	pr_debug("Entered %s\n", __func__);
 
@@ -296,6 +299,8 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 			s3c24xx_snd_rxctrl(1);
 		else
 			s3c24xx_snd_txctrl(1);
+
+		s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index eecfa5eba06..5cbbdc80fde 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -255,7 +255,6 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		prtd->state |= ST_RUNNING;
 		s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START);
-		s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
@@ -318,6 +317,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
 
 	pr_debug("Entered %s\n", __func__);
 
+	snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
 	snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
 
 	prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL);
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c
new file mode 100644
index 00000000000..1966e0d5652
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.c
@@ -0,0 +1,394 @@
+/* sound/soc/s3c24xx/s3c24xx_simtec.c
+ *
+ * Copyright 2009 Simtec Electronics
+ *
+ * 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/platform_device.h>
+#include <linux/gpio.h>
+#include <linux/clk.h>
+#include <linux/i2c.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <plat/audio-simtec.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "s3c24xx_simtec.h"
+
+static struct s3c24xx_audio_simtec_pdata *pdata;
+static struct clk *xtal_clk;
+
+static int spk_gain;
+static int spk_unmute;
+
+/**
+ * speaker_gain_get - read the speaker gain setting.
+ * @kcontrol: The control for the speaker gain.
+ * @ucontrol: The value that needs to be updated.
+ *
+ * Read the value for the AMP gain control.
+ */
+static int speaker_gain_get(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = spk_gain;
+	return 0;
+}
+
+/**
+ * speaker_gain_set - set the value of the speaker amp gain
+ * @value: The value to write.
+ */
+static void speaker_gain_set(int value)
+{
+	gpio_set_value_cansleep(pdata->amp_gain[0], value & 1);
+	gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1);
+}
+
+/**
+ * speaker_gain_put - set the speaker gain setting.
+ * @kcontrol: The control for the speaker gain.
+ * @ucontrol: The value that needs to be set.
+ *
+ * Set the value of the speaker gain from the specified
+ * @ucontrol setting.
+ *
+ * Note, if the speaker amp is muted, then we do not set a gain value
+ * as at-least one of the ICs that is fitted will try and power up even
+ * if the main control is set to off.
+ */
+static int speaker_gain_put(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	int value = ucontrol->value.integer.value[0];
+
+	spk_gain = value;
+
+	if (!spk_unmute)
+		speaker_gain_set(value);
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new amp_gain_controls[] = {
+	SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0,
+		       speaker_gain_get, speaker_gain_put),
+};
+
+/**
+ * spk_unmute_state - set the unmute state of the speaker
+ * @to: zero to unmute, non-zero to ununmute.
+ */
+static void spk_unmute_state(int to)
+{
+	pr_debug("%s: to=%d\n", __func__, to);
+
+	spk_unmute = to;
+	gpio_set_value(pdata->amp_gpio, to);
+
+	/* if we're umuting, also re-set the gain */
+	if (to && pdata->amp_gain[0] > 0)
+		speaker_gain_set(spk_gain);
+}
+
+/**
+ * speaker_unmute_get - read the speaker unmute setting.
+ * @kcontrol: The control for the speaker gain.
+ * @ucontrol: The value that needs to be updated.
+ *
+ * Read the value for the AMP gain control.
+ */
+static int speaker_unmute_get(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = spk_unmute;
+	return 0;
+}
+
+/**
+ * speaker_unmute_put - set the speaker unmute setting.
+ * @kcontrol: The control for the speaker gain.
+ * @ucontrol: The value that needs to be set.
+ *
+ * Set the value of the speaker gain from the specified
+ * @ucontrol setting.
+ */
+static int speaker_unmute_put(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	spk_unmute_state(ucontrol->value.integer.value[0]);
+	return 0;
+}
+
+/* This is added as a manual control as the speaker amps create clicks
+ * when their power state is changed, which are far more noticeable than
+ * anything produced by the CODEC itself.
+ */
+static const struct snd_kcontrol_new amp_unmute_controls[] = {
+	SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0,
+		       speaker_unmute_get, speaker_unmute_put),
+};
+
+void simtec_audio_init(struct snd_soc_codec *codec)
+{
+	if (pdata->amp_gpio > 0) {
+		pr_debug("%s: adding amp routes\n", __func__);
+
+		snd_soc_add_controls(codec, amp_unmute_controls,
+				     ARRAY_SIZE(amp_unmute_controls));
+	}
+
+	if (pdata->amp_gain[0] > 0) {
+		pr_debug("%s: adding amp controls\n", __func__);
+		snd_soc_add_controls(codec, amp_gain_controls,
+				     ARRAY_SIZE(amp_gain_controls));
+	}
+}
+EXPORT_SYMBOL_GPL(simtec_audio_init);
+
+#define CODEC_CLOCK 12000000
+
+/**
+ * simtec_hw_params - update hardware parameters
+ * @substream: The audio substream instance.
+ * @params: The parameters requested.
+ *
+ * Update the codec data routing and configuration  settings
+ * from the supplied data.
+ */
+static int simtec_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;
+	int ret;
+
+	/* Set the CODEC as the bus clock master, I2S */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret) {
+		pr_err("%s: failed set cpu dai format\n", __func__);
+		return ret;
+	}
+
+	/* Set the CODEC as the bus clock master */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret) {
+		pr_err("%s: failed set codec dai format\n", __func__);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+				     CODEC_CLOCK, SND_SOC_CLOCK_IN);
+	if (ret) {
+		pr_err( "%s: failed setting codec sysclk\n", __func__);
+		return ret;
+	}
+
+	if (pdata->use_mpllin) {
+		ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL,
+					     0, SND_SOC_CLOCK_OUT);
+
+		if (ret) {
+			pr_err("%s: failed to set MPLLin as clksrc\n",
+			       __func__);
+			return ret;
+		}
+	}
+
+	if (pdata->output_cdclk) {
+		int cdclk_scale;
+
+		cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK;
+		cdclk_scale--;
+
+		ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+					     cdclk_scale);
+	}
+
+	return 0;
+}
+
+static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd)
+{
+	/* call any board supplied startup code, this currently only
+	 * covers the bast/vr1000 which have a CPLD in the way of the
+	 * LRCLK */
+	if (pd->startup)
+		pd->startup();
+
+	return 0;
+}
+
+static struct snd_soc_ops simtec_snd_ops = {
+	.hw_params	= simtec_hw_params,
+};
+
+/**
+ * attach_gpio_amp - get and configure the necessary gpios
+ * @dev: The device we're probing.
+ * @pd: The platform data supplied by the board.
+ *
+ * If there is a GPIO based amplifier attached to the board, claim
+ * the necessary GPIO lines for it, and set default values.
+ */
+static int attach_gpio_amp(struct device *dev,
+			   struct s3c24xx_audio_simtec_pdata *pd)
+{
+	int ret;
+
+	/* attach gpio amp gain (if any) */
+	if (pdata->amp_gain[0] > 0) {
+		ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0");
+		if (ret) {
+			dev_err(dev, "cannot get amp gpio gain0\n");
+			return ret;
+		}
+
+		ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1");
+		if (ret) {
+			dev_err(dev, "cannot get amp gpio gain1\n");
+			gpio_free(pdata->amp_gain[0]);
+			return ret;
+		}
+
+		gpio_direction_output(pd->amp_gain[0], 0);
+		gpio_direction_output(pd->amp_gain[1], 0);
+	}
+
+	/* note, curently we assume GPA0 isn't valid amp */
+	if (pdata->amp_gpio > 0) {
+		ret = gpio_request(pd->amp_gpio, "gpio-amp");
+		if (ret) {
+			dev_err(dev, "cannot get amp gpio %d (%d)\n",
+				pd->amp_gpio, ret);
+			goto err_amp;
+		}
+
+		/* set the amp off at startup */
+		spk_unmute_state(0);
+	}
+
+	return 0;
+
+err_amp:
+	if (pd->amp_gain[0] > 0) {
+		gpio_free(pd->amp_gain[0]);
+		gpio_free(pd->amp_gain[1]);
+	}
+
+	return ret;
+}
+
+static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd)
+{
+	if (pd->amp_gain[0] > 0) {
+		gpio_free(pd->amp_gain[0]);
+		gpio_free(pd->amp_gain[1]);
+	}
+
+	if (pd->amp_gpio > 0)
+		gpio_free(pd->amp_gpio);
+}
+
+#ifdef CONFIG_PM
+int simtec_audio_resume(struct device *dev)
+{
+	simtec_call_startup(pdata);
+	return 0;
+}
+
+struct dev_pm_ops simtec_audio_pmops = {
+	.resume	= simtec_audio_resume,
+};
+EXPORT_SYMBOL_GPL(simtec_audio_pmops);
+#endif
+
+int __devinit simtec_audio_core_probe(struct platform_device *pdev,
+				      struct snd_soc_device *socdev)
+{
+	struct platform_device *snd_dev;
+	int ret;
+
+	socdev->card->dai_link->ops = &simtec_snd_ops;
+
+	pdata = pdev->dev.platform_data;
+	if (!pdata) {
+		dev_err(&pdev->dev, "no platform data supplied\n");
+		return -EINVAL;
+	}
+
+	simtec_call_startup(pdata);
+
+	xtal_clk = clk_get(&pdev->dev, "xtal");
+	if (IS_ERR(xtal_clk)) {
+		dev_err(&pdev->dev, "could not get clkout0\n");
+		return -EINVAL;
+	}
+
+	dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk));
+
+	ret = attach_gpio_amp(&pdev->dev, pdata);
+	if (ret)
+		goto err_clk;
+
+	snd_dev = platform_device_alloc("soc-audio", -1);
+	if (!snd_dev) {
+		dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n");
+		ret = -ENOMEM;
+		goto err_gpio;
+	}
+
+	platform_set_drvdata(snd_dev, socdev);
+	socdev->dev = &snd_dev->dev;
+
+	ret = platform_device_add(snd_dev);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to add soc-audio dev\n");
+		goto err_pdev;
+	}
+
+	platform_set_drvdata(pdev, snd_dev);
+	return 0;
+
+err_pdev:
+	platform_device_put(snd_dev);
+
+err_gpio:
+	detach_gpio_amp(pdata);
+
+err_clk:
+	clk_put(xtal_clk);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(simtec_audio_core_probe);
+
+int __devexit simtec_audio_remove(struct platform_device *pdev)
+{
+	struct platform_device *snd_dev = platform_get_drvdata(pdev);
+
+	platform_device_unregister(snd_dev);
+
+	detach_gpio_amp(pdata);
+	clk_put(xtal_clk);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(simtec_audio_remove);
+
+MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h
new file mode 100644
index 00000000000..2714203af16
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.h
@@ -0,0 +1,22 @@
+/* sound/soc/s3c24xx/s3c24xx_simtec.h
+ *
+ * Copyright 2009 Simtec Electronics
+ *
+ * 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.
+*/
+
+extern void simtec_audio_init(struct snd_soc_codec *codec);
+
+extern int simtec_audio_core_probe(struct platform_device *pdev,
+				   struct snd_soc_device *socdev);
+
+extern int simtec_audio_remove(struct platform_device *pdev);
+
+#ifdef CONFIG_PM
+extern struct dev_pm_ops simtec_audio_pmops;
+#define simtec_audio_pm &simtec_audio_pmops
+#else
+#define simtec_audio_pm NULL
+#endif
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
new file mode 100644
index 00000000000..8346bd96eaf
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
@@ -0,0 +1,153 @@
+/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
+ *
+ * Copyright 2009 Simtec Electronics
+ *
+ * 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/clk.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <plat/audio-simtec.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "s3c24xx_simtec.h"
+
+#include "../codecs/tlv320aic3x.h"
+
+static const struct snd_soc_dapm_widget dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("GSM Out", NULL),
+	SND_SOC_DAPM_LINE("GSM In", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+	SND_SOC_DAPM_LINE("ZV", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route base_map[] = {
+	/* Headphone connected to HP{L,R}OUT and HP{L,R}COM */
+
+	{ "Headphone Jack", NULL, "HPLOUT" },
+	{ "Headphone Jack", NULL, "HPLCOM" },
+	{ "Headphone Jack", NULL, "HPROUT" },
+	{ "Headphone Jack", NULL, "HPRCOM" },
+
+	/* ZV connected to Line1 */
+
+	{ "LINE1L", NULL, "ZV" },
+	{ "LINE1R", NULL, "ZV" },
+
+	/* Line In connected to Line2 */
+
+	{ "LINE2L", NULL, "Line In" },
+	{ "LINE2R", NULL, "Line In" },
+
+	/* Microphone connected to MIC3R and MIC_BIAS */
+
+	{ "MIC3L", NULL, "Mic Jack" },
+
+	/* GSM connected to MONO_LOUT and MIC3L (in) */
+
+	{ "GSM Out", NULL, "MONO_LOUT" },
+	{ "MIC3L", NULL, "GSM In" },
+
+	/* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are
+	 * not using the DAPM to power it up and down as there it makes
+	 * a click when powering up. */
+};
+
+/**
+ * simtec_hermes_init - initialise and add controls
+ * @codec; The codec instance to attach to.
+ *
+ * Attach our controls and configure the necessary codec
+ * mappings for our sound card instance.
+*/
+static int simtec_hermes_init(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, dapm_widgets,
+				  ARRAY_SIZE(dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map));
+
+	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	snd_soc_dapm_enable_pin(codec, "Line In");
+	snd_soc_dapm_enable_pin(codec, "Line Out");
+	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+	simtec_audio_init(codec);
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct aic3x_setup_data codec_setup = {
+};
+
+static struct snd_soc_dai_link simtec_dai_aic33 = {
+	.name		= "tlv320aic33",
+	.stream_name	= "TLV320AIC33",
+	.cpu_dai	= &s3c24xx_i2s_dai,
+	.codec_dai	= &aic3x_dai,
+	.init		= simtec_hermes_init,
+};
+
+/* simtec audio machine driver */
+static struct snd_soc_card snd_soc_machine_simtec_aic33 = {
+	.name		= "Simtec-Hermes",
+	.platform	= &s3c24xx_soc_platform,
+	.dai_link	= &simtec_dai_aic33,
+	.num_links	= 1,
+};
+
+/* simtec audio subsystem */
+static struct snd_soc_device simtec_snd_devdata_aic33 = {
+	.card		= &snd_soc_machine_simtec_aic33,
+	.codec_dev	= &soc_codec_dev_aic3x,
+	.codec_data	= &codec_setup,
+};
+
+static int __devinit simtec_audio_hermes_probe(struct platform_device *pd)
+{
+	dev_info(&pd->dev, "probing....\n");
+	return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33);
+}
+
+static struct platform_driver simtec_audio_hermes_platdrv = {
+	.driver	= {
+		.owner	= THIS_MODULE,
+		.name	= "s3c24xx-simtec-hermes-snd",
+		.pm	= simtec_audio_pm,
+	},
+	.probe	= simtec_audio_hermes_probe,
+	.remove	= __devexit_p(simtec_audio_remove),
+};
+
+MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd");
+
+static int __init simtec_hermes_modinit(void)
+{
+	return platform_driver_register(&simtec_audio_hermes_platdrv);
+}
+
+static void __exit simtec_hermes_modexit(void)
+{
+	platform_driver_unregister(&simtec_audio_hermes_platdrv);
+}
+
+module_init(simtec_hermes_modinit);
+module_exit(simtec_hermes_modexit);
+
+MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC Simtec Audio support");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
new file mode 100644
index 00000000000..25797e09617
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
@@ -0,0 +1,137 @@
+/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
+ *
+ * Copyright 2009 Simtec Electronics
+ *
+ * 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/clk.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <plat/audio-simtec.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "s3c24xx_simtec.h"
+
+#include "../codecs/tlv320aic23.h"
+
+/* supported machines:
+ *
+ * Machine	Connections		AMP
+ * -------	-----------		---
+ * BAST		MIC, HPOUT, LOUT, LIN	TPA2001D1 (HPOUTL,R) (gain hardwired)
+ * VR1000	HPOUT, LIN		None
+ * VR2000	LIN, LOUT, MIC, HP	LM4871 (HPOUTL,R)
+ * DePicture	LIN, LOUT, MIC, HP	LM4871 (HPOUTL,R)
+ * Anubis	LIN, LOUT, MIC, HP	TPA2001D1 (HPOUTL,R)
+ */
+
+static const struct snd_soc_dapm_widget dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route base_map[] = {
+	{ "Headphone Jack", NULL, "LHPOUT"},
+	{ "Headphone Jack", NULL, "RHPOUT"},
+
+	{ "Line Out", NULL, "LOUT" },
+	{ "Line Out", NULL, "ROUT" },
+
+	{ "LLINEIN", NULL, "Line In"},
+	{ "RLINEIN", NULL, "Line In"},
+
+	{ "MICIN", NULL, "Mic Jack"},
+};
+
+/**
+ * simtec_tlv320aic23_init - initialise and add controls
+ * @codec; The codec instance to attach to.
+ *
+ * Attach our controls and configure the necessary codec
+ * mappings for our sound card instance.
+*/
+static int simtec_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, dapm_widgets,
+				  ARRAY_SIZE(dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map));
+
+	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	snd_soc_dapm_enable_pin(codec, "Line In");
+	snd_soc_dapm_enable_pin(codec, "Line Out");
+	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+	simtec_audio_init(codec);
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link simtec_dai_aic23 = {
+	.name		= "tlv320aic23",
+	.stream_name	= "TLV320AIC23",
+	.cpu_dai	= &s3c24xx_i2s_dai,
+	.codec_dai	= &tlv320aic23_dai,
+	.init		= simtec_tlv320aic23_init,
+};
+
+/* simtec audio machine driver */
+static struct snd_soc_card snd_soc_machine_simtec_aic23 = {
+	.name		= "Simtec",
+	.platform	= &s3c24xx_soc_platform,
+	.dai_link	= &simtec_dai_aic23,
+	.num_links	= 1,
+};
+
+/* simtec audio subsystem */
+static struct snd_soc_device simtec_snd_devdata_aic23 = {
+	.card		= &snd_soc_machine_simtec_aic23,
+	.codec_dev	= &soc_codec_dev_tlv320aic23,
+};
+
+static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd)
+{
+	return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23);
+}
+
+static struct platform_driver simtec_audio_tlv320aic23_platdrv = {
+	.driver	= {
+		.owner	= THIS_MODULE,
+		.name	= "s3c24xx-simtec-tlv320aic23",
+		.pm	= simtec_audio_pm,
+	},
+	.probe	= simtec_audio_tlv320aic23_probe,
+	.remove	= __devexit_p(simtec_audio_remove),
+};
+
+MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23");
+
+static int __init simtec_tlv320aic23_modinit(void)
+{
+	return platform_driver_register(&simtec_audio_tlv320aic23_platdrv);
+}
+
+static void __exit simtec_tlv320aic23_modexit(void)
+{
+	platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv);
+}
+
+module_init(simtec_tlv320aic23_modinit);
+module_exit(simtec_tlv320aic23_modexit);
+
+MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC Simtec Audio support");
+MODULE_LICENSE("GPL");