Index: linux-2.6.24-rc7/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
===================================================================
--- /dev/null
+++ linux-2.6.24-rc7/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
@@ -0,0 +1,667 @@
+/*
+ * neo1973_gta02_wm8753.c  --  SoC audio for Neo1973
+ *
+ * Copyright 2007 OpenMoko Inc
+ * Author: Graeme Gregory <graeme@openmoko.org>
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory <linux@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.
+ *
+ *  Revision history
+ *    06th Nov 2007   Changed from GTA01 to GTA02
+ *    20th Jan 2007   Initial version.
+ *    05th Feb 2007   Rename all to Neo1973
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/i2c.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/hardware/scoop.h>
+#include <asm/plat-s3c24xx/regs-iis.h>
+#include <asm/arch/regs-clock.h>
+#include <asm/arch/regs-gpio.h>
+#include <asm/hardware.h>
+#include <asm/arch/audio.h>
+#include <asm/io.h>
+#include <asm/arch/spi-gpio.h>
+#include <asm/arch/regs-gpioj.h>
+#include <asm/arch/gta02.h>
+#include "../codecs/wm8753.h"
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+
+/* define the scenarios */
+#define NEO_AUDIO_OFF			0
+#define NEO_GSM_CALL_AUDIO_HANDSET	1
+#define NEO_GSM_CALL_AUDIO_HEADSET	2
+#define NEO_GSM_CALL_AUDIO_BLUETOOTH	3
+#define NEO_STEREO_TO_SPEAKERS		4
+#define NEO_STEREO_TO_HEADPHONES	5
+#define NEO_CAPTURE_HANDSET		6
+#define NEO_CAPTURE_HEADSET		7
+#define NEO_CAPTURE_BLUETOOTH		8
+#define NEO_STEREO_TO_HANDSET_SPK	9
+
+static struct snd_soc_machine 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_codec_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_cpu_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 = codec_dai->dai_ops.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 = cpu_dai->dai_ops.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 = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set MCLK division for sample rate */
+	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+		S3C2410_IISMOD_32FS );
+	if (ret < 0)
+		return ret;
+
+	/* set codec BCLK division for sample rate */
+	ret = codec_dai->dai_ops.set_clkdiv(codec_dai,
+					WM8753_BCLKDIV, bclk);
+	if (ret < 0)
+		return ret;
+
+	/* set prescaler division for sample rate */
+	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+		S3C24XX_PRESCALE(4,4));
+	if (ret < 0)
+		return ret;
+
+	/* codec PLL input is PCLK/4 */
+	ret = codec_dai->dai_ops.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_codec_dai *codec_dai = rtd->dai->codec_dai;
+
+	/* disable the PLL */
+	return codec_dai->dai_ops.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_codec_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 = codec_dai->dai_ops.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 = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK,
+		12288000, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set codec PCM division for sample rate */
+	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV,
+					pcmdiv);
+	if (ret < 0)
+		return ret;
+
+	/* configue and enable PLL for 12.288MHz output */
+	ret = codec_dai->dai_ops.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_codec_dai *codec_dai = rtd->dai->codec_dai;
+
+	/* disable the PLL */
+	return codec_dai->dai_ops.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=0;
+
+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;
+		s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT,0);
+	} else {
+		lm4853_state &= ~LM4853_AMP;
+		s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT,1);
+	}
+
+	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;
+		s3c2410_gpio_setpin(GTA02_GPIO_HP_IN,0);
+	} else {
+		lm4853_state &= ~LM4853_SPK;
+		s3c2410_gpio_setpin(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 neo1973_gta02_set_stereo_out(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int val = ucontrol->value.integer.value[0];
+
+	snd_soc_dapm_set_endpoint(codec, "Stereo Out", val);
+
+	snd_soc_dapm_sync_endpoints(codec);
+
+	return 0;
+}
+
+static int neo1973_gta02_get_stereo_out(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.integer.value[0] =
+		snd_soc_dapm_get_endpoint(codec, "Stereo Out");
+
+	return 0;
+}
+
+
+static int neo1973_gta02_set_gsm_out(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int val = ucontrol->value.integer.value[0];
+
+	snd_soc_dapm_set_endpoint(codec, "GSM Line Out", val);
+
+	snd_soc_dapm_sync_endpoints(codec);
+
+	return 0;
+}
+
+static int neo1973_gta02_get_gsm_out(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.integer.value[0] =
+		snd_soc_dapm_get_endpoint(codec, "GSM Line Out");
+
+	return 0;
+}
+
+static int neo1973_gta02_set_gsm_in(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int val = ucontrol->value.integer.value[0];
+
+	snd_soc_dapm_set_endpoint(codec, "GSM Line In", val);
+
+	snd_soc_dapm_sync_endpoints(codec);
+
+	return 0;
+}
+
+static int neo1973_gta02_get_gsm_in(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.integer.value[0] =
+		snd_soc_dapm_get_endpoint(codec, "GSM Line In");
+
+	return 0;
+}
+
+static int neo1973_gta02_set_headset_mic(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int val = ucontrol->value.integer.value[0];
+
+	snd_soc_dapm_set_endpoint(codec, "Headset Mic", val);
+
+	snd_soc_dapm_sync_endpoints(codec);
+
+	return 0;
+}
+
+static int neo1973_gta02_get_headset_mic(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.integer.value[0] =
+		snd_soc_dapm_get_endpoint(codec, "Headset Mic");
+
+	return 0;
+}
+
+static int neo1973_gta02_set_handset_mic(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int val = ucontrol->value.integer.value[0];
+
+	snd_soc_dapm_set_endpoint(codec, "Handset Mic", val);
+
+	snd_soc_dapm_sync_endpoints(codec);
+
+	return 0;
+}
+
+static int neo1973_gta02_get_handset_mic(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.integer.value[0] =
+		snd_soc_dapm_get_endpoint(codec, "Handset Mic");
+
+	return 0;
+}
+
+static int neo1973_gta02_set_handset_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int val = ucontrol->value.integer.value[0];
+
+	snd_soc_dapm_set_endpoint(codec, "Handset Spk", val);
+
+	snd_soc_dapm_sync_endpoints(codec);
+
+	return 0;
+}
+
+static int neo1973_gta02_get_handset_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.integer.value[0] =
+		snd_soc_dapm_get_endpoint(codec, "Handset Spk");
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("Stereo Out", NULL),
+	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+	SND_SOC_DAPM_LINE("GSM Line In", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Handset Mic", NULL),
+	SND_SOC_DAPM_SPK("Handset Spk", NULL),
+};
+
+
+/* example machine audio_mapnections */
+static const char* audio_map[][3] = {
+
+	/* 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"},
+
+	{NULL, NULL, NULL},
+};
+
+static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = {
+	SOC_SINGLE_EXT("DAPM Stereo Out Switch", 0, 0, 1, 0,
+		neo1973_gta02_get_stereo_out,
+		neo1973_gta02_set_stereo_out),
+	SOC_SINGLE_EXT("DAPM GSM Line Out Switch", 1, 0, 1, 0,
+		neo1973_gta02_get_gsm_out,
+		neo1973_gta02_set_gsm_out),
+	SOC_SINGLE_EXT("DAPM GSM Line In Switch", 2, 0, 1, 0,
+		neo1973_gta02_get_gsm_in,
+		neo1973_gta02_set_gsm_in),
+	SOC_SINGLE_EXT("DAPM Headset Mic Switch", 3, 0, 1, 0,
+		neo1973_gta02_get_headset_mic,
+		neo1973_gta02_set_headset_mic),
+	SOC_SINGLE_EXT("DAPM Handset Mic Switch", 4, 0, 1, 0,
+		neo1973_gta02_get_handset_mic,
+		neo1973_gta02_set_handset_mic),
+	SOC_SINGLE_EXT("DAPM Handset Spk Switch", 5, 0, 1, 0,
+		neo1973_gta02_get_handset_spk,
+		neo1973_gta02_set_handset_spk),
+	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 i, err;
+
+	/* set up NC codec pins */
+	snd_soc_dapm_set_endpoint(codec, "OUT3",  0);
+	snd_soc_dapm_set_endpoint(codec, "OUT4",  0);
+	snd_soc_dapm_set_endpoint(codec, "LINE1", 0);
+	snd_soc_dapm_set_endpoint(codec, "LINE2", 0);
+
+	/* Add neo1973 gta02 specific widgets */
+	for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
+		snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
+
+	/* add neo1973 gta02 specific controls */
+	for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_gta02_controls); i++) {
+		err = snd_ctl_add(codec->card,
+			snd_soc_cnew(&wm8753_neo1973_gta02_controls[i],
+			codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* set up neo1973 gta02 specific audio path audio_mapnects */
+	for (i = 0; audio_map[i][0] != NULL; i++) {
+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
+			audio_map[i][1], audio_map[i][2]);
+	}
+
+	/* set endpoints to default off mode */
+	snd_soc_dapm_set_endpoint(codec, "Stereo Out",  0);
+	snd_soc_dapm_set_endpoint(codec, "GSM Line Out",0);
+	snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+	snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+	snd_soc_dapm_set_endpoint(codec, "Handset Mic", 0);
+	snd_soc_dapm_set_endpoint(codec, "Handset Spk", 0);
+
+	snd_soc_dapm_sync_endpoints(codec);
+	return 0;
+}
+
+/*
+ * BT Codec DAI
+ */
+static struct snd_soc_cpu_dai bt_dai =
+{	.name = "Bluetooth",
+	.id = 0,
+	.type = SND_SOC_DAI_PCM,
+	.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,
+},
+};
+
+#ifdef CONFIG_PM
+int neo1973_gta02_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT, 1);
+
+	return 0;
+}
+
+int neo1973_gta02_resume(struct platform_device *pdev)
+{
+	if(lm4853_state & LM4853_AMP)
+		s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT, 0);
+
+	return 0;
+}
+#else
+#define neo1973_gta02_suspend NULL
+#define neo1973_gta02_resume NULL
+#endif
+
+static struct snd_soc_machine neo1973_gta02 = {
+	.name = "neo1973-gta02",
+	.suspend_pre = neo1973_gta02_suspend,
+	.resume_post = neo1973_gta02_resume,
+	.dai_link = neo1973_gta02_dai,
+	.num_links = ARRAY_SIZE(neo1973_gta02_dai),
+};
+
+static struct wm8753_setup_data neo1973_gta02_wm8753_setup = {
+	.i2c_address = 0x1a,
+};
+
+static struct snd_soc_device neo1973_gta02_snd_devdata = {
+	.machine = &neo1973_gta02,
+	.platform = &s3c24xx_soc_platform,
+	.codec_dev = &soc_codec_dev_wm8753,
+	.codec_data = &neo1973_gta02_wm8753_setup,
+};
+
+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 hardware supported by ASoc driver\n");
+		return -ENODEV;
+	}
+
+	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);
+
+	/* Initialise GPIOs used by amp */
+	s3c2410_gpio_cfgpin(GTA02_GPIO_HP_IN, S3C2410_GPIO_OUTPUT);
+	s3c2410_gpio_cfgpin(GTA02_GPIO_AMP_SHUT, S3C2410_GPIO_OUTPUT);
+
+	/* Amp off by default */
+	s3c2410_gpio_setpin(GTA02_GPIO_AMP_SHUT, 1);
+
+	/* Speaker off by default */
+	s3c2410_gpio_setpin(GTA02_GPIO_HP_IN, 1);
+
+	return ret;
+}
+
+static void __exit neo1973_gta02_exit(void)
+{
+	platform_device_unregister(neo1973_gta02_snd_device);
+}
+
+module_init(neo1973_gta02_init);
+module_exit(neo1973_gta02_exit);
+
+/* Module information */
+MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org");
+MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02");
+MODULE_LICENSE("GPL");
+
Index: linux-2.6.24-rc7/sound/soc/s3c24xx/Kconfig
===================================================================
--- linux-2.6.24-rc7.orig/sound/soc/s3c24xx/Kconfig
+++ linux-2.6.24-rc7/sound/soc/s3c24xx/Kconfig
@@ -25,6 +25,15 @@
 	  Say Y if you want to add support for SoC audio on smdk2440
 	  with the WM8753.
 
+config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753
+	tristate "SoC I2S Audio support for NEO1973 GTA02 - WM8753"
+	depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02
+	select SND_S3C24XX_SOC_I2S
+	select SND_SOC_WM8753
+	help
+	  Say Y if you want to add support for SoC audio on neo1973 gta02
+	  with the WM8753 codec
+
 config SND_S3C24XX_SOC_SMDK2443_WM9710
 	tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
 	depends on SND_S3C24XX_SOC && MACH_SMDK2443
Index: linux-2.6.24-rc7/sound/soc/s3c24xx/Makefile
===================================================================
--- linux-2.6.24-rc7.orig/sound/soc/s3c24xx/Makefile
+++ linux-2.6.24-rc7/sound/soc/s3c24xx/Makefile
@@ -10,6 +10,9 @@
 # S3C24XX Machine Support
 snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
 snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
+snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
 
 obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
 obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
+obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o
+
Index: linux-2.6.24-rc7/include/sound/soc-dapm.h
===================================================================
--- linux-2.6.24-rc7.orig/include/sound/soc-dapm.h
+++ linux-2.6.24-rc7/include/sound/soc-dapm.h
@@ -206,6 +206,8 @@
 /* dapm audio endpoint control */
 int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
 	char *pin, int status);
+int snd_soc_dapm_get_endpoint(struct snd_soc_codec *codec,
+	char *pin);
 int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec);
 
 /* dapm widget types */
Index: linux-2.6.24-rc7/sound/soc/soc-dapm.c
===================================================================
--- linux-2.6.24-rc7.orig/sound/soc/soc-dapm.c
+++ linux-2.6.24-rc7/sound/soc/soc-dapm.c
@@ -1305,6 +1305,30 @@
 EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint);
 
 /**
+ * snd_soc_dapm_get_endpoint - get audio endpoint status
+ * @codec: audio codec
+ * @endpoint: audio signal endpoint (or start point)
+ *
+ * Get audio endpoint status - connected or disconnected.
+ *
+ * Returns status
+ */
+int snd_soc_dapm_get_endpoint(struct snd_soc_codec *codec,
+	char *endpoint)
+{
+	struct snd_soc_dapm_widget *w;
+
+	list_for_each_entry(w, &codec->dapm_widgets, list) {
+		if (!strcmp(w->name, endpoint)) {
+			return w->connected;
+		}
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint);
+
+/**
  * snd_soc_dapm_free - free dapm resources
  * @socdev: SoC device
  *
