Copy the GTA02 support and edit out the items that are

not present in the GTA03 (such as the external AMP) and
bind to the S3C64XX I2S audio.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>

3 files changed, 579 insertions, 1 deletions

diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index b845699dedd..11dcdfa663f 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -52,6 +52,15 @@ config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753
 	  Say Y if you want to add support for SoC audio on neo1973 gta02
 	  with the WM8753 codec
 
+config SND_S3C24XX_SOC_OM_GTA03_WM8753
+	tristate "SoC I2S Audio support for OM GTA03 - WM8753"
+	depends on SND_S3C24XX_SOC && MACH_OPENMOKO_GTA03
+	select SND_S3C64XX_SOC_I2S
+	select SND_SOC_WM8753
+	help
+	  Say Y if you want support for SoC audio on Openmoko GTA03
+	  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
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index de985410444..c1ff0e4bcde 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -20,6 +20,7 @@ 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-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
+snd-soc-om-gta03-wm8753-objs := om_gta03_wm8753.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
@@ -27,4 +28,4 @@ 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_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o
-
+obj-$(CONFIG_SND_S3C24XX_SOC_OM_GTA03_WM8753) += snd-soc-om-gta03-wm8753.o
diff --git a/sound/soc/s3c24xx/om_gta03_wm8753.c b/sound/soc/s3c24xx/om_gta03_wm8753.c
new file mode 100644
index 00000000000..60879b8bc80
--- /dev/null
+++ b/sound/soc/s3c24xx/om_gta03_wm8753.c
@@ -0,0 +1,568 @@
+/*
+ * om_gta03_wm8753.c  --  SoC audio for GTA03
+ *
+ * Based on neo1973_gta02_wm8753
+ *
+ * Copyright 2009 Openmoko Inc
+ * Author: Ben Dooks <ben@simtec.co.uk>
+ * 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.
+ */
+
+#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/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 <plat/regs-s3c2412-iis.h>
+
+#include "../codecs/wm8753.h"
+#include "s3c24xx-pcm.h"
+#include "s3c64xx-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_card om_gta03;
+
+static int om_gta03_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 = s3c64xx_i2s_get_clockrate(cpu_dai);
+
+	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;
+
+#if 0
+	/* do not think we need to set this if the cpu is not the bitclk
+	 * master */
+	/* 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;
+#endif
+
+	/* 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, S3C64XX_DIV_PRESCALER, 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 om_gta03_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);
+}
+
+/*
+ * GTA03 WM8753 HiFi DAI opserations.
+ */
+static struct snd_soc_ops om_gta03_hifi_ops = {
+	.hw_params	= om_gta03_hifi_hw_params,
+	.hw_free	= om_gta03_hifi_hw_free,
+};
+
+static int om_gta03_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 = s3c64xx_i2s_get_clockrate(rtd->dai->cpu_dai);
+
+	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 om_gta03_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 om_gta03_voice_ops = {
+	.hw_params	= om_gta03_voice_hw_params,
+	.hw_free	= om_gta03_voice_hw_free,
+};
+
+static int om_gta03_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(codec);
+
+	return 0;
+}
+
+static int om_gta03_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 om_gta03_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(codec);
+
+	return 0;
+}
+
+static int om_gta03_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 om_gta03_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(codec);
+
+	return 0;
+}
+
+static int om_gta03_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 om_gta03_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(codec);
+
+	return 0;
+}
+
+static int om_gta03_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 om_gta03_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(codec);
+
+	return 0;
+}
+
+static int om_gta03_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 om_gta03_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(codec);
+
+	return 0;
+}
+
+static int om_gta03_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 struct snd_soc_dapm_route audio_map[] = {
+
+	{"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_om_gta03_controls[] = {
+	SOC_SINGLE_EXT("DAPM Stereo Out Switch", 0, 0, 1, 0,
+		om_gta03_get_stereo_out,
+		om_gta03_set_stereo_out),
+	SOC_SINGLE_EXT("DAPM GSM Line Out Switch", 1, 0, 1, 0,
+		om_gta03_get_gsm_out,
+		om_gta03_set_gsm_out),
+	SOC_SINGLE_EXT("DAPM GSM Line In Switch", 2, 0, 1, 0,
+		om_gta03_get_gsm_in,
+		om_gta03_set_gsm_in),
+	SOC_SINGLE_EXT("DAPM Headset Mic Switch", 3, 0, 1, 0,
+		om_gta03_get_headset_mic,
+		om_gta03_set_headset_mic),
+	SOC_SINGLE_EXT("DAPM Handset Mic Switch", 4, 0, 1, 0,
+		om_gta03_get_handset_mic,
+		om_gta03_set_handset_mic),
+	SOC_SINGLE_EXT("DAPM Handset Spk Switch", 5, 0, 1, 0,
+		om_gta03_get_handset_spk,
+		om_gta03_set_handset_spk),
+};
+
+/*
+ * This is an example machine initialisation for a wm8753 connected to a
+ * neo1973 GTA02.
+ */
+static int om_gta03_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 */
+	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+				  ARRAY_SIZE(wm8753_dapm_widgets));
+
+	/* add neo1973 gta02 specific controls */
+	for (i = 0; i < ARRAY_SIZE(wm8753_om_gta03_controls); i++) {
+		err = snd_ctl_add(codec->card,
+			snd_soc_cnew(&wm8753_om_gta03_controls[i],
+			codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* set up neo1973 gta02 specific audio path audio_mapnects */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* 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(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 om_gta03_dai[] = {
+	{ /* Hifi Playback - for similatious use with voice below */
+		.name		= "WM8753",
+		.stream_name	= "WM8753 HiFi",
+		.cpu_dai	= &s3c64xx_i2s_dai,
+		.codec_dai	= &wm8753_dai[WM8753_DAI_HIFI],
+		.init		= om_gta03_wm8753_init,
+		.ops		= &om_gta03_hifi_ops,
+	},
+	{ /* Voice via BT */
+		.name		= "Bluetooth",
+		.stream_name	= "Voice",
+		.cpu_dai	= &bt_dai,
+		.codec_dai	= &wm8753_dai[WM8753_DAI_VOICE],
+		.ops		= &om_gta03_voice_ops,
+	},
+};
+
+static struct snd_soc_card om_gta03 = {
+	.name		= "om-gta03",
+	.platform	= &s3c24xx_soc_platform,
+	.dai_link	= om_gta03_dai,
+	.num_links	= ARRAY_SIZE(om_gta03_dai),
+};
+
+/* Audio private data */
+static struct wm8753_setup_data soc_codec_data_wm8753_gta02 = {
+	.i2c_bus = 0,
+	.i2c_address = 0x1a,
+};
+
+static struct snd_soc_device om_gta03_snd_devdata = {
+	.card		= &om_gta03,
+	.codec_dev	= &soc_codec_dev_wm8753,
+	.codec_data	= &soc_codec_data_wm8753_gta02,
+};
+
+static struct platform_device *om_gta03_snd_device;
+
+static int __init om_gta03_init(void)
+{
+	int ret;
+
+	if (!machine_is_openmoko_gta03()) {
+		printk(KERN_INFO "Only GTA03 supported by ASoC driver\n");
+		return -ENODEV;
+	}
+
+	/* register bluetooth DAI here */
+	ret = snd_soc_register_dai(&bt_dai);
+	if (ret)
+		return ret;
+
+	om_gta03_snd_device = platform_device_alloc("soc-audio", 0);
+	if (!om_gta03_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(om_gta03_snd_device, &om_gta03_snd_devdata);
+	om_gta03_snd_devdata.dev = &om_gta03_snd_device->dev;
+	ret = platform_device_add(om_gta03_snd_device);
+
+	if (ret) {
+		platform_device_put(om_gta03_snd_device);
+		return ret;
+	}
+
+	return ret;
+}
+
+static void __exit om_gta03_exit(void)
+{
+	platform_device_unregister(om_gta03_snd_device);
+}
+
+module_init(om_gta03_init);
+module_exit(om_gta03_exit);
+
+/* Module information */
+MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org; Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC WM8753 OM GTA03");
+MODULE_LICENSE("GPL");