path: root/packages/linux/linux-ezx-2.6.23/patches/ezx-asoc.patch

Index: linux-2.6.23/sound/soc/codecs/pcap2.c
===================================================================
--- /dev/null	1970-01-01 00:00:00.000000000 +0000
+++ linux-2.6.23/sound/soc/codecs/pcap2.c	2007-10-22 22:28:06.000000000 +0200
@@ -0,0 +1,796 @@
+/*
+ * pcap2.c - PCAP2 ASIC Audio driver
+ *
+ * 	Copyright (C) 2007 Daniel Ribeiro <drwyrm@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/delay.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/ezx-pcap.h>
+#include <asm/arch/ezx.h>
+#include <asm/arch/hardware.h>
+
+#include "pcap2.h"
+
+#define AUDIO_NAME "pcap2-codec"
+#define PCAP2_VERSION "0.1"
+
+extern int ezx_pcap_write(u_int8_t, u_int32_t);
+extern int ezx_pcap_read(u_int8_t, u_int32_t *);
+static struct snd_soc_device *pcap2_codec_socdev;
+
+/*
+ * Debug
+ */
+
+//#define PCAP2_DEBUG
+
+#ifdef PCAP2_DEBUG
+#define dbg(format, arg...) \
+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
+#else
+#define dbg(format, arg...)
+#endif
+
+#define err(format, arg...) \
+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
+#define info(format, arg...) \
+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
+#define warn(format, arg...) \
+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
+
+#define dump_registers()	pcap2_codec_read(NULL, 13); \
+				pcap2_codec_read(NULL, 12); \
+				pcap2_codec_read(NULL, 11); \
+				pcap2_codec_read(NULL, 26);
+
+
+
+
+/*
+ * ASoC limits register value to 16 bits and pcap uses 32 bit registers
+ * to work around this, we get 16 bits from low, mid or high positions.
+ * ASoC limits register number to 8 bits we use 0x1f for register
+ * number and 0xe0 for register offset. -WM
+ */
+static int pcap2_codec_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	unsigned int tmp;
+
+	ezx_pcap_read((reg & 0x1f), &tmp);
+
+	if (reg & SL) {
+		tmp &= 0xffff0000;
+		tmp |= (value & 0xffff);
+	}
+	else if (reg & SM) {
+		tmp &= 0xff0000ff;
+		tmp |= ((value << 8) & 0x00ffff00);
+	}
+	else if (reg & SH) {
+		tmp &= 0xffff;
+		tmp |= ((value << 16) & 0xffff0000);
+	}
+	else
+		tmp = value;
+
+	dbg("codec_write reg=%x, rval=%08x, fval=%08x", reg, tmp,  value);
+	ezx_pcap_write((reg & 0x1f), tmp);
+	return 0;
+
+}
+
+static unsigned int pcap2_codec_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+	unsigned int tmp, ret;
+
+	ezx_pcap_read((reg & 0x1f), &tmp);
+	ret = tmp;
+	if (reg & SL)
+		ret = (tmp & 0xffff);
+	else if (reg & SM)
+		ret = ((tmp >> 8) & 0xffff);
+	else if (reg & SH)
+		ret = ((tmp >> 16) & 0xffff);
+
+	dbg("codec_read  reg=%x, rval=%08x, fval=%08x", reg, tmp, ret);
+	return(ret);
+
+}
+
+static const char *pcap2_output_select[] = {"2ch", "2->1ch", "2->1ch -3db", "2->1ch -6db"};
+
+static const struct soc_enum pcap2_enum[] = {
+SOC_ENUM_SINGLE((PCAP2_OUTPUT_AMP|SH), 3, 4, pcap2_output_select),
+};
+
+static const struct snd_kcontrol_new pcap2_input_mixer_controls[] = {
+SOC_DAPM_SINGLE("A3 Switch", (PCAP2_INPUT_AMP|SL), 6, 1, 0),
+SOC_DAPM_SINGLE("A5 Switch", (PCAP2_INPUT_AMP|SL), 8, 1, 0),
+};
+
+static const struct snd_kcontrol_new pcap2_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("A1 Switch", (PCAP2_OUTPUT_AMP|SL), 0, 1, 0),
+SOC_DAPM_SINGLE("A2 Switch", (PCAP2_OUTPUT_AMP|SL), 1, 1, 0),
+SOC_DAPM_SINGLE("AR Switch", (PCAP2_OUTPUT_AMP|SL), 5, 1, 0),
+SOC_DAPM_SINGLE("AL Switch", (PCAP2_OUTPUT_AMP|SL), 6, 1, 0),
+};
+
+/* pcap2 codec non DAPM controls */
+static const struct snd_kcontrol_new pcap2_codec_snd_controls[] = {
+SOC_SINGLE("Output gain", (PCAP2_OUTPUT_AMP|SM),  5, 15, 0),
+SOC_SINGLE("Input gain", (PCAP2_INPUT_AMP|SL),   0, 31, 0),
+};
+
+static const struct snd_kcontrol_new pcap2_codec_dm_mux_control[] = {
+	SOC_DAPM_ENUM("Output Mode",	pcap2_enum[0]),
+};
+
+/* add non dapm controls */
+static int pcap2_codec_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(pcap2_codec_snd_controls); i++) {
+		if ((err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&pcap2_codec_snd_controls[i],codec, NULL))) < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+/* pcap2 codec DAPM controls */
+static const struct snd_soc_dapm_widget pcap2_codec_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("ST_DAC", "ST_DAC playback", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("CDC_DAC", "CDC_DAC playback", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_ADC("CDC_ADC", "CDC_DAC capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_PGA("PGA_ST", (PCAP2_OUTPUT_AMP|SL), 9, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PGA_CDC", (PCAP2_OUTPUT_AMP|SL), 8, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PGA_R", (PCAP2_OUTPUT_AMP|SL), 11, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PGA_L", (PCAP2_OUTPUT_AMP|SL), 12, 0, NULL, 0),
+	SND_SOC_DAPM_MUX("Downmixer", SND_SOC_NOPM, 0, 0, pcap2_codec_dm_mux_control),
+	SND_SOC_DAPM_PGA("PGA_A1CTRL", (PCAP2_OUTPUT_AMP|SH), 1, 1, NULL, 0),
+	SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, &pcap2_output_mixer_controls[0], ARRAY_SIZE(pcap2_output_mixer_controls)),
+	SND_SOC_DAPM_OUTPUT("A1"), /* Earpiece */
+	SND_SOC_DAPM_OUTPUT("A2"), /* LoudSpeaker */
+	SND_SOC_DAPM_OUTPUT("AR"), /* headset right */
+	SND_SOC_DAPM_OUTPUT("AL"), /* headset left */
+
+	SND_SOC_DAPM_MICBIAS("BIAS1", (PCAP2_INPUT_AMP|SL), 10, 0),
+	SND_SOC_DAPM_MICBIAS("BIAS2", (PCAP2_INPUT_AMP|SL), 11, 0),
+	SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, &pcap2_input_mixer_controls[0], ARRAY_SIZE(pcap2_input_mixer_controls)),
+	SND_SOC_DAPM_INPUT("A3"), /* Headset Mic */
+	SND_SOC_DAPM_INPUT("A5"), /* Builtin Mic */
+};
+
+static const char *audio_map[][3] = {
+	{ "A1", NULL, "Output Mixer" },
+	{ "A2", NULL, "Output Mixer" },
+	{ "AR", NULL, "Output Mixer" },
+	{ "AL", NULL, "Output Mixer" },
+
+	{ "Output Mixer", "A1 Switch", "PGA_A1CTRL" },
+	{ "Output Mixer", "A2 Switch", "Downmixer" },
+	{ "Output Mixer", "AR Switch", "PGA_R" },
+	{ "Output Mixer", "AL Switch", "PGA_L" },
+
+	{ "PGA_A1CTRL", NULL, "Downmixer" },
+
+	{ "Downmixer", "2->1ch", "PGA_L" },
+	{ "Downmixer", "2->1ch", "PGA_R" },
+	{ "Downmixer", "2->1ch -3db", "PGA_L" },
+	{ "Downmixer", "2->1ch -3db", "PGA_R" },
+	{ "Downmixer", "2->1ch -6db", "PGA_L" },
+	{ "Downmixer", "2->1ch -6db", "PGA_R" },
+	{ "Downmixer", "2ch", "PGA_R" },
+
+	{ "PGA_R", NULL, "PGA_ST" },
+	{ "PGA_L", NULL, "PGA_ST" },
+	{ "PGA_R", NULL, "PGA_CDC" },
+
+	{ "PGA_ST", NULL, "ST_DAC" },
+	{ "PGA_CDC", NULL, "CDC_DAC" },
+
+	/* input path */
+	{ "BIAS1", NULL, "A3" },
+	{ "BIAS2", NULL, "A5" },
+
+	{ "Input Mixer", "A3 Switch", "BIAS1" },
+	{ "Input Mixer", "A5 Switch", "BIAS2" },
+
+	{ "PGA_R", NULL, "Input Mixer" },
+
+	{ "PGA_CDC", NULL, "PGA_R" },
+	{ "CDC_ADC", NULL, "PGA_CDC" },
+
+	/* terminator */
+	{NULL, NULL, NULL},
+};
+
+static int pcap2_codec_add_widgets(struct snd_soc_codec *codec)
+{
+	int i;
+
+	for(i = 0; i < ARRAY_SIZE(pcap2_codec_dapm_widgets); i++) {
+		snd_soc_dapm_new_control(codec, &pcap2_codec_dapm_widgets[i]);
+	}
+
+	/* set up audio path interconnects */
+	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]);
+	}
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+static int pcap2_codec_dapm_event(struct snd_soc_codec *codec, int event)
+{
+	unsigned int input = pcap2_codec_read(codec, PCAP2_INPUT_AMP);
+
+	input &= ~PCAP2_INPUT_AMP_LOWPWR;
+
+	switch (event) {
+	case SNDRV_CTL_POWER_D0:
+	case SNDRV_CTL_POWER_D1:
+	case SNDRV_CTL_POWER_D2:
+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+		dbg("dapm: ON\n");
+		break;
+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+		input |= PCAP2_INPUT_AMP_LOWPWR;
+		dbg("dapm: OFF\n");
+		break;
+	}
+	codec->dapm_state = event;
+	pcap2_codec_write(codec, PCAP2_INPUT_AMP, input);
+	return 0;
+}
+
+static int pcap2_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_codec *codec = codec_dai->codec;
+	unsigned int tmp;
+
+	if (codec_dai->id == PCAP2_STEREO_DAI) {
+		tmp = pcap2_codec_read(codec, PCAP2_ST_DAC);
+
+		tmp &= ~PCAP2_ST_DAC_RATE_MASK;
+		switch(params_rate(params)) {
+		case 8000:
+			break;
+		case 11025:
+			tmp |= PCAP2_ST_DAC_RATE_11025;
+			break;
+		case 12000:
+			tmp |= PCAP2_ST_DAC_RATE_12000;
+			break;
+		case 16000:
+			tmp |= PCAP2_ST_DAC_RATE_16000;
+			break;
+		case 22050:
+			tmp |= PCAP2_ST_DAC_RATE_22050;
+			break;
+		case 24000:
+			tmp |= PCAP2_ST_DAC_RATE_24000;
+			break;
+		case 32000:
+			tmp |= PCAP2_ST_DAC_RATE_32000;
+			break;
+		case 44100:
+			tmp |= PCAP2_ST_DAC_RATE_44100;
+			break;
+		case 48000:
+			tmp |= PCAP2_ST_DAC_RATE_48000;
+			break;
+		default:
+			return -EINVAL;
+		}
+		tmp |= PCAP2_ST_DAC_RESET_DF;
+		pcap2_codec_write(codec, PCAP2_ST_DAC, tmp);
+	}
+	else {
+		tmp = pcap2_codec_read(codec, PCAP2_CODEC);
+
+		tmp &= ~PCAP2_CODEC_RATE_MASK;
+		switch(params_rate(params)) {
+		case 8000:
+			break;
+		case 16000:
+			tmp |= PCAP2_CODEC_RATE_16000;
+			break;
+		default:
+			return -EINVAL;
+		}
+		tmp |= PCAP2_CODEC_RESET_DF;
+		pcap2_codec_write(codec, PCAP2_CODEC, tmp);
+	}
+
+	return 0;
+}
+
+static int pcap2_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;
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct snd_soc_dapm_widget *w;
+	unsigned int tmp;
+
+	if (codec_dai->id == PCAP2_STEREO_DAI) {
+		snd_soc_dapm_set_endpoint(codec, "ST_DAC", 0);
+		tmp = pcap2_codec_read(codec, PCAP2_ST_DAC);
+		tmp &= ~(PCAP2_ST_DAC_EN | PCAP2_ST_DAC_CLK_EN);
+		pcap2_codec_write(codec, PCAP2_ST_DAC, tmp);
+	}
+	else {
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			snd_soc_dapm_set_endpoint(codec, "CDC_DAC", 0);
+		else
+			snd_soc_dapm_set_endpoint(codec, "CDC_ADC", 0);
+		list_for_each_entry(w, &codec->dapm_widgets, list) {
+			if ((!strcmp(w->name, "CDC_DAC") || !strcmp(w->name, "CDC_ADC")) && w->connected)
+				goto in_use;
+		}
+		tmp = pcap2_codec_read(codec, PCAP2_CODEC);
+		tmp &= ~(PCAP2_CODEC_EN | PCAP2_CODEC_CLK_EN);
+		pcap2_codec_write(codec, PCAP2_CODEC, tmp);
+	}
+in_use:
+	snd_soc_dapm_sync_endpoints(codec);
+
+	return 0;
+}
+
+static int pcap2_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	unsigned int tmp;
+	if (codec_dai->id == PCAP2_STEREO_DAI) {
+		/* ST_DAC */
+
+		tmp = pcap2_codec_read(codec, PCAP2_ST_DAC);
+
+		tmp &= ~PCAP2_ST_DAC_CLKSEL_MASK;
+		switch (clk_id) {
+		case PCAP2_CLK_AP:
+			tmp |= PCAP2_ST_DAC_CLKSEL_AP;
+			break;
+		case PCAP2_CLK_BP:
+			break;
+		default:
+			return -ENODEV;
+		}
+
+		tmp &= ~PCAP2_ST_DAC_CLK_MASK;
+		switch (freq) {
+		case 13000000:
+			break;
+/*		case 15M36:
+			tmp |= PCAP2_ST_DAC_CLK_15M36;
+			break;
+		case 16M8:
+			tmp |= PCAP2_ST_DAC_CLK_16M8;
+			break;
+		case 19M44:
+			tmp |= PCAP2_ST_DAC_CLK_19M44;
+			break;
+*/		case 26000000:
+			tmp |= PCAP2_ST_DAC_CLK_26M;
+			break;
+/*		case EXT_MCLK:
+			tmp |= PCAP2_ST_DAC_CLK_MCLK;
+			break;
+		case FSYNC:
+			tmp |= PCAP2_ST_DAC_CLK_FSYNC;
+			break;
+		case BITCLK:
+			tmp |= PCAP2_ST_DAC_CLK_BITCLK;
+			break;
+*/		default:
+			return -EINVAL;
+		}
+		pcap2_codec_write(codec, PCAP2_ST_DAC, tmp);
+	}
+	else {
+		/* MONO_DAC */
+		tmp = pcap2_codec_read(codec, PCAP2_CODEC);
+
+		tmp &= ~PCAP2_CODEC_CLKSEL_MASK;
+		switch (clk_id) {
+		case PCAP2_CLK_AP:
+			tmp |= PCAP2_CODEC_CLKSEL_AP;
+			break;
+		case PCAP2_CLK_BP:
+			break;
+		default:
+			return -ENODEV;
+		}
+
+		tmp &= ~PCAP2_CODEC_CLK_MASK;
+		switch (freq) {
+		case 13000000:
+			break;
+/*		case 15M36:
+			tmp |= PCAP2_CODEC_CLK_15M36;
+			break;
+		case 16M8:
+			tmp |= PCAP2_CODEC_CLK_16M8;
+			break;
+		case 19M44:
+			tmp |= PCAP2_CODEC_CLK_19M44;
+			break;
+*/		case 26000000:
+			tmp |= PCAP2_CODEC_CLK_26M;
+			break;
+		default:
+			return -EINVAL;
+		}
+		pcap2_codec_write(codec, PCAP2_CODEC, tmp);
+	}
+	return 0;
+}
+
+static int pcap2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	unsigned int tmp = 0;
+
+	if (codec_dai->id == PCAP2_STEREO_DAI) {
+		/* ST_DAC */
+
+		/* disable CODEC */
+		pcap2_codec_write(codec, PCAP2_CODEC, 0);
+
+		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+		case SND_SOC_DAIFMT_CBM_CFM:
+			break;
+		case SND_SOC_DAIFMT_CBS_CFS:
+			tmp |= 0x1;
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+		case SND_SOC_DAIFMT_I2S:
+			tmp |= 0x4000;
+			break;
+/*		case SND_SOC_NET:
+			tmp |= 0x2000;
+			break;
+*/		case SND_SOC_DAIFMT_DSP_B:
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_IB_IF:
+			break;
+		case SND_SOC_DAIFMT_NB_NF:
+			tmp |= 0x60000;
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			tmp |= 0x40000;
+			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			tmp |= 0x20000;
+			break;
+		}
+		/* set dai to AP */
+		tmp |= 0x1000;
+
+		/* set BCLK */
+		tmp |= 0x18000;
+
+		pcap2_codec_write(codec, PCAP2_ST_DAC, tmp);
+	}
+	else {
+		/* MONO_DAC */
+
+		/* disable ST_DAC */
+		pcap2_codec_write(codec, PCAP2_ST_DAC, 0);
+
+		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+		case SND_SOC_DAIFMT_CBM_CFM:
+			break;
+		case SND_SOC_DAIFMT_CBS_CFS:
+			tmp |= 0x2;
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+		case SND_SOC_DAIFMT_DSP_B:
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_IB_IF:
+			break;
+		case SND_SOC_DAIFMT_NB_NF:
+			tmp |= 0x600;
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			tmp |= 0x400;
+			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			tmp |= 0x200;
+			break;
+		}
+		if (codec_dai->id == PCAP2_MONO_DAI)
+			/* set dai to AP */
+			tmp |= 0x8000;
+
+		tmp |= 0x5; /* IHF / OHF */
+
+		pcap2_codec_write(codec, PCAP2_CODEC, tmp);
+	}
+	return 0;
+}
+
+static int pcap2_prepare(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;
+	struct snd_soc_codec *codec = codec_dai->codec;
+	unsigned int tmp;
+	/* FIXME enable clock only if codec is master */
+	if (codec_dai->id == PCAP2_STEREO_DAI) {
+		snd_soc_dapm_set_endpoint(codec, "ST_DAC", 1);
+		snd_soc_dapm_set_endpoint(codec, "CDC_DAC", 0);
+		snd_soc_dapm_set_endpoint(codec, "CDC_ADC", 0);
+		tmp = pcap2_codec_read(codec, PCAP2_ST_DAC);
+		tmp |= (PCAP2_ST_DAC_EN | PCAP2_ST_DAC_CLK_EN);
+		pcap2_codec_write(codec, PCAP2_ST_DAC, tmp);
+	}
+	else {
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			snd_soc_dapm_set_endpoint(codec, "CDC_DAC", 1);
+		else
+			snd_soc_dapm_set_endpoint(codec, "CDC_ADC", 1);
+		snd_soc_dapm_set_endpoint(codec, "ST_DAC", 0);
+		tmp = pcap2_codec_read(codec, PCAP2_CODEC);
+		tmp |= (PCAP2_CODEC_EN | PCAP2_CODEC_CLK_EN);
+		pcap2_codec_write(codec, PCAP2_CODEC, tmp);
+	}
+	snd_soc_dapm_sync_endpoints(codec);
+	mdelay(1);
+#ifdef PCAP2_DEBUG
+	dump_registers();
+#endif
+	return 0;
+}
+
+/*
+ * Define codec DAI.
+ */
+struct snd_soc_codec_dai pcap2_dai[] = {
+{
+	.name = "PCAP2 MONO",
+	.id = 0,
+	.playback = {
+		.stream_name = "CDC_DAC playback",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000),
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "CDC_DAC capture",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000),
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = {
+		.prepare = pcap2_prepare,
+		.hw_params = pcap2_hw_params,
+		.hw_free = pcap2_hw_free,
+	},
+	.dai_ops = {
+//		.digital_mute = pcap2_mute,
+		.set_fmt = pcap2_set_dai_fmt,
+		.set_sysclk = pcap2_set_dai_sysclk,
+	},
+},
+{
+	.name = "PCAP2 STEREO",
+	.id = 1,
+	.playback = {
+		.stream_name = "ST_DAC playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+			SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |
+			SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+			SNDRV_PCM_RATE_48000),
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = { /* FIXME: PCAP support this?? */
+		.stream_name = "ST_DAC capture",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+			SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |
+			SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+			SNDRV_PCM_RATE_48000),
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = {
+		.prepare = pcap2_prepare,
+		.hw_params = pcap2_hw_params,
+		.hw_free = pcap2_hw_free,
+	},
+	.dai_ops = {
+//		.digital_mute = pcap2_mute,
+		.set_fmt = pcap2_set_dai_fmt,
+		.set_sysclk = pcap2_set_dai_sysclk,
+	},
+},
+{
+	.name = "PCAP2 BP",
+	.id = 2,
+	.playback = {
+		.stream_name = "BP playback",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = {
+		.prepare = pcap2_prepare,
+		.hw_params = pcap2_hw_params,
+		.hw_free = pcap2_hw_free,
+	},
+	.dai_ops = {
+//		.digital_mute = pcap2_mute,
+		.set_fmt = pcap2_set_dai_fmt,
+		.set_sysclk = pcap2_set_dai_sysclk,
+	},
+},
+};
+EXPORT_SYMBOL_GPL(pcap2_dai);
+
+static int pcap2_codec_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	dbg("pcap2_codec_suspend");
+	pcap2_codec_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	return 0;
+}
+
+static int pcap2_codec_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	dbg("pcap2_codec_resume");
+	pcap2_codec_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	pcap2_codec_dapm_event(codec, codec->suspend_dapm_state);
+	return 0;
+}
+
+/*
+ * initialise the PCAP2 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int pcap2_codec_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret = 0;
+
+	dbg("pcap2_codec_init");
+	codec->name = "PCAP2 Audio";
+	codec->owner = THIS_MODULE;
+	codec->read = pcap2_codec_read;
+	codec->write = pcap2_codec_write;
+	codec->dapm_event = pcap2_codec_dapm_event;
+	codec->dai = pcap2_dai;
+	codec->num_dai = ARRAY_SIZE(pcap2_dai);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		return ret;
+	}
+	/* power on device */
+	pcap2_codec_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	/* set the update bits */
+
+	pcap2_codec_add_controls(codec);
+	pcap2_codec_add_widgets(codec);
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		snd_soc_free_pcms(socdev);
+		snd_soc_dapm_free(socdev);
+	}
+
+	return ret;
+}
+
+static int pcap2_codec_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct pcap2_codec_setup_data *setup;
+	struct snd_soc_codec *codec;
+	int ret = 0;
+	info("PCAP2 Audio Codec %s", PCAP2_VERSION);
+
+	setup = socdev->codec_data;
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	socdev->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	pcap2_codec_socdev = socdev;
+
+	ret = pcap2_codec_init(socdev);
+	return ret;
+}
+
+/* power down chip and remove */
+static int pcap2_codec_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	if (codec->control_data)
+		pcap2_codec_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	kfree(codec);
+
+	return 0;
+}
+
+/* codec device ops */
+struct snd_soc_codec_device soc_codec_dev_pcap2 = {
+	.probe = 	pcap2_codec_probe,
+	.remove = 	pcap2_codec_remove,
+	.suspend = 	pcap2_codec_suspend,
+	.resume =	pcap2_codec_resume,
+};
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_pcap2);
+
+MODULE_DESCRIPTION("ASoC PCAP2 codec");
+MODULE_AUTHOR("Daniel Ribeiro");
+MODULE_LICENSE("GPL");
Index: linux-2.6.23/sound/soc/codecs/pcap2.h
===================================================================
--- /dev/null	1970-01-01 00:00:00.000000000 +0000
+++ linux-2.6.23/sound/soc/codecs/pcap2.h	2007-10-22 22:28:06.000000000 +0200
@@ -0,0 +1,81 @@
+/*
+ * 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 _PCAP2_H
+#define _PCAP2_H
+
+/* 16 bit reads/writes on pcap registers (ugly workaround) */
+#define SL (1 << 5)	/* lower 16 bits */
+#define SM (1 << 6)	/* mid 16 bits */
+#define SH (1 << 7)	/* higher 16 bits */
+
+/* PCAP2 register space */
+#define PCAP2_CODEC			0x0b
+#define PCAP2_OUTPUT_AMP		0x0c
+#define PCAP2_ST_DAC			0x0d
+#define PCAP2_INPUT_AMP			0x1a
+
+#define PCAP2_MONO_DAI			0
+#define PCAP2_STEREO_DAI		1
+#define PCAP2_BP_DAI			2
+
+#define PCAP2_CLK_BP			0
+#define PCAP2_CLK_AP			1
+
+#define PCAP2_CODEC_EN			0x2000
+#define PCAP2_CODEC_CLK_EN		0x1000
+#define PCAP2_CODEC_RESET_DF		0x800
+#define PCAP2_CODEC_RATE_MASK		0x4000
+#define PCAP2_CODEC_RATE_8000		0x0
+#define PCAP2_CODEC_RATE_16000		0x4000
+#define PCAP2_CODEC_CLKSEL_MASK		0x10000
+#define PCAP2_CODEC_CLKSEL_AP		0x10000
+#define PCAP2_CODEC_CLKSEL_BP		0x0
+#define PCAP2_CODEC_CLK_MASK		0x1c0
+#define PCAP2_CODEC_CLK_13M		0x0
+#define PCAP2_CODEC_CLK_15M36		0x40
+#define PCAP2_CODEC_CLK_16M8		0x80
+#define PCAP2_CODEC_CLK_19M44		0xc0
+#define PCAP2_CODEC_CLK_26M		0x100
+
+#define PCAP2_ST_DAC_EN			0x80
+#define PCAP2_ST_DAC_CLK_EN		0x20
+#define PCAP2_ST_DAC_RESET_DF		0x40
+#define PCAP2_ST_DAC_RATE_MASK		0xf00
+#define PCAP2_ST_DAC_RATE_8000		0x0
+#define PCAP2_ST_DAC_RATE_11025		0x100
+#define PCAP2_ST_DAC_RATE_12000		0x200
+#define PCAP2_ST_DAC_RATE_16000		0x300
+#define PCAP2_ST_DAC_RATE_22050		0x400
+#define PCAP2_ST_DAC_RATE_24000		0x500
+#define PCAP2_ST_DAC_RATE_32000		0x600
+#define PCAP2_ST_DAC_RATE_44100		0x700
+#define PCAP2_ST_DAC_RATE_48000		0x800
+#define PCAP2_ST_DAC_CLKSEL_MASK	0x80000
+#define PCAP2_ST_DAC_CLKSEL_AP		0x80000
+#define PCAP2_ST_DAC_CLKSEL_BP		0x0
+#define PCAP2_ST_DAC_CLK_MASK		0x1c
+#define PCAP2_ST_DAC_CLK_13M		0x0
+#define PCAP2_ST_DAC_CLK_15M36		0x4
+#define PCAP2_ST_DAC_CLK_16M8		0x8
+#define PCAP2_ST_DAC_CLK_19M44		0xc
+#define PCAP2_ST_DAC_CLK_26M		0x10
+#define PCAP2_ST_DAC_CLK_MCLK		0x14
+#define PCAP2_ST_DAC_CLK_FSYNC		0x18
+#define PCAP2_ST_DAC_CLK_BITCLK		0x1c
+
+#define PCAP2_INPUT_AMP_LOWPWR		0x80000
+#define PCAP2_INPUT_AMP_V2EN2		0x200000
+
+#define PCAP2_OUTPUT_AMP_PGAR_EN	0x800
+#define PCAP2_OUTPUT_AMP_PGAL_EN	0x1000
+#define PCAP2_OUTPUT_AMP_CDC_SW		0x100
+#define PCAP2_OUTPUT_AMP_ST_DAC_SW	0x200
+
+extern struct snd_soc_codec_dai pcap2_dai[];
+extern struct snd_soc_codec_device soc_codec_dev_pcap2;
+
+#endif
Index: linux-2.6.23/sound/soc/pxa/Kconfig
===================================================================
--- linux-2.6.23.orig/sound/soc/pxa/Kconfig	2007-10-22 22:27:11.000000000 +0200
+++ linux-2.6.23/sound/soc/pxa/Kconfig	2007-10-22 22:29:23.000000000 +0200
@@ -57,3 +57,12 @@
 	help
 	  Say Y if you want to add support for SoC audio on Sharp
 	  Zaurus SL-C6000x models (Tosa).
+
+config SND_PXA2XX_SOC_EZX
+	tristate "SoC Audio support for EZX"
+	depends on SND_PXA2XX_SOC && PXA_EZX
+	select SND_PXA2XX_SOC_SSP
+	select SND_SOC_PCAP2
+	help
+	  Say Y if you want to add support for SoC audio on
+	  Motorola EZX Phones (a780/e680).
Index: linux-2.6.23/sound/soc/pxa/ezx.c
===================================================================
--- /dev/null	1970-01-01 00:00:00.000000000 +0000
+++ linux-2.6.23/sound/soc/pxa/ezx.c	2007-10-22 22:28:06.000000000 +0200
@@ -0,0 +1,349 @@
+/*
+ * ezx.c - Machine specific code for EZX phones
+ *
+ *	Copyright (C) 2007 Daniel Ribeiro <drwyrm@gmail.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 <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/arch/pxa-regs.h>
+#include <asm/arch/hardware.h>
+
+#include <asm/arch/ezx-pcap.h>
+
+#include "../codecs/pcap2.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ssp.h"
+
+#define GPIO_HW_ATTENUATE_A780  96
+
+static struct snd_soc_codec *control_codec;
+
+static void ezx_ext_control(struct snd_soc_codec *codec)
+{
+	if (ezx_pcap_read_bit(pbit(PCAP_REG_PSTAT, PCAP_IRQ_A1)))
+		snd_soc_dapm_set_endpoint(codec, "Headset", 1);
+	else
+		snd_soc_dapm_set_endpoint(codec, "Headset", 0);
+	if (ezx_pcap_read_bit(pbit(PCAP_REG_PSTAT, PCAP_IRQ_MB2)))
+		snd_soc_dapm_set_endpoint(codec, "External Mic", 1);
+	else
+		snd_soc_dapm_set_endpoint(codec, "External Mic", 0);
+
+	snd_soc_dapm_sync_endpoints(codec);
+}
+
+static irqreturn_t jack_irq(int irq, void *data)
+{
+	ezx_ext_control(control_codec);
+	return IRQ_HANDLED;
+}
+
+
+/*
+ * Alsa operations
+ * Only implement the required operations for your platform.
+ * These operations are specific to the machine only.
+ */
+
+ /*
+ * Called by ALSA when a PCM substream is opened, private data can be allocated.
+ */
+static int ezx_machine_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 */
+	ezx_ext_control(codec);
+	return 0;
+}
+
+/*
+ * Called by ALSA when the hardware params are set by application. This
+ * function can also be called multiple times and can allocate buffers
+ * (using snd_pcm_lib_* ). It's non-atomic.
+ */
+static int ezx_machine_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;
+	int ret;
+
+	/* set codec DAI configuration */
+	if (codec_dai->id == PCAP2_STEREO_DAI)
+		ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
+			SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	else
+		ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
+			SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
+	if(ret < 0)
+		return ret;
+
+	/* Turn on clock output on CLK_PIO */
+	OSCC |= 0x8;
+
+	/* set clock source */
+	ret = codec_dai->dai_ops.set_sysclk(codec_dai, PCAP2_CLK_AP,
+					13000000, SND_SOC_CLOCK_IN);
+	if(ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B |
+			SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	ret = cpu_dai->dai_ops.set_tristate(cpu_dai, 0);
+	if (ret < 0)
+		return ret;
+
+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai,PXA2XX_SSP_CLK_EXT,
+						0, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+/*
+ * Free's resources allocated by hw_params, can be called multiple times
+ */
+static int ezx_machine_hw_free(struct snd_pcm_substream *substream)
+{
+	OSCC &= ~0x8; /* turn off clock output on CLK_PIO */
+
+	return 0;
+}
+
+static int ezx_machine_prepare(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;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	if (codec_dai->id == PCAP2_STEREO_DAI) {
+		/* override pxa2xx-ssp sample size for stereo/network mode */
+		SSCR0_P(cpu_dai->id+1) &= ~(SSCR0_DSS | SSCR0_EDSS);
+		SSCR0_P(cpu_dai->id+1) |= (SSCR0_EDSS | SSCR0_DataSize(16));
+	}
+	return 0;
+}
+
+/* machine Alsa PCM operations */
+static struct snd_soc_ops ezx_ops = {
+	.startup = ezx_machine_startup,
+	.prepare = ezx_machine_prepare,
+	.hw_free = ezx_machine_hw_free,
+	.hw_params = ezx_machine_hw_params,
+};
+
+static int bp_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;
+	int ret = 0;
+	/* set codec DAI configuration */
+	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
+		SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
+	if(ret < 0)
+		return ret;
+
+	/* set clock source */
+	ret = codec_dai->dai_ops.set_sysclk(codec_dai, PCAP2_CLK_BP,
+					13000000, SND_SOC_CLOCK_IN);
+
+	return ret;
+}
+
+
+
+/* machine dapm widgets */
+static const struct snd_soc_dapm_widget ezx_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headset", NULL),
+	SND_SOC_DAPM_SPK("Earpiece", NULL),
+	SND_SOC_DAPM_SPK("Loudspeaker", NULL),
+	SND_SOC_DAPM_MIC("Built-in Mic", NULL),
+	SND_SOC_DAPM_MIC("External Mic", NULL),
+};
+
+/* machine audio map (connections to the codec pins) */
+static const char *audio_map[][3] = {
+	{ "Headset", NULL, "AR" },
+	{ "Headset", NULL, "AL" },
+	{ "Earpiece", NULL, "A1" },
+	{ "Loudspeaker", NULL, "A2" },
+
+	{ "Built-in Mic", NULL, "A5" },
+	{ "External Mic", NULL, "A3" },
+
+	{NULL, NULL, NULL},
+};
+
+/*
+ * Initialise the machine audio subsystem.
+ */
+static int ezx_machine_init(struct snd_soc_codec *codec)
+{
+	int i;
+	/* mark unused codec pins as NC */
+//	snd_soc_dapm_set_endpoint(codec, "FIXME", 0);
+	control_codec = codec;
+
+        /* Add ezx specific controls */
+//	for (i = 0; i < ARRAY_SIZE(ezx_controls); i++) {
+//		if ((err = snd_ctl_add(codec->card, snd_soc_cnew(&ezx_controls[i], codec, NULL))) < 0)
+//			return err;
+//	}
+
+	/* Add ezx specific widgets */
+	for(i = 0; i < ARRAY_SIZE(ezx_dapm_widgets); i++) {
+		snd_soc_dapm_new_control(codec, &ezx_dapm_widgets[i]);
+	}
+	/* Set up ezx specific audio path interconnects */
+	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]);
+	}
+
+	/* synchronise subsystem */
+	snd_soc_dapm_sync_endpoints(codec);
+	return 0;
+}
+
+static struct snd_soc_cpu_dai bp_dai =
+{
+	.name = "Baseband",
+	.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,
+	},
+	.ops = {
+//		.startup = bp_startup,
+//		.shutdown = bp_shutdown,
+		.hw_params = bp_hw_params,
+//		.hw_free = bp_hw_free,
+	},
+};
+
+/* template digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ezx_dai[] = {
+{
+	.name = "PCAP2 STEREO",
+	.stream_name = "stereo playback",
+	.cpu_dai = &pxa_ssp_dai[PXA2XX_DAI_SSP3],
+	.codec_dai = &pcap2_dai[PCAP2_STEREO_DAI],
+	.init = ezx_machine_init,
+	.ops = &ezx_ops,
+},
+{
+	.name = "PCAP2 MONO",
+	.stream_name = "mono playback",
+	.cpu_dai = &pxa_ssp_dai[PXA2XX_DAI_SSP3],
+	.codec_dai = &pcap2_dai[PCAP2_MONO_DAI],
+//	.init = ezx_machine_init, /* the stereo call already registered our controls */
+	.ops = &ezx_ops,
+},
+{
+	.name = "PCAP2 BP",
+	.stream_name = "BP Audio",
+	.cpu_dai = &bp_dai,
+	.codec_dai = &pcap2_dai[PCAP2_BP_DAI],
+},
+};
+
+/* template audio machine driver */
+static struct snd_soc_machine snd_soc_machine_ezx = {
+	.name = "Motorola EZX",
+//	.probe
+//	.remove
+//	.suspend_pre
+//	.resume_post
+	.dai_link = ezx_dai,
+	.num_links = ARRAY_SIZE(ezx_dai),
+};
+
+/* template audio subsystem */
+static struct snd_soc_device ezx_snd_devdata = {
+	.machine = &snd_soc_machine_ezx,
+	.platform = &pxa2xx_soc_platform,
+	.codec_dev = &soc_codec_dev_pcap2,
+};
+
+static struct platform_device *ezx_snd_device;
+
+static int __init ezx_init(void)
+{
+	int ret;
+	ezx_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!ezx_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(ezx_snd_device, &ezx_snd_devdata);
+	ezx_snd_devdata.dev = &ezx_snd_device->dev;
+	ret = platform_device_add(ezx_snd_device);
+
+	if (ret)
+		platform_device_put(ezx_snd_device);
+	/* configure gpio for ssp3 */
+	pxa_gpio_mode(GPIO83_SFRM3_MD);	/* SFRM */
+	pxa_gpio_mode(GPIO81_STXD3_MD);	/* TXD  */
+	pxa_gpio_mode(GPIO52_SCLK3_MD);	/* SCLK */
+	pxa_gpio_mode(GPIO89_SRXD3_MD);	/* RXD  */
+
+	/* configure gpio for ssp2 */
+	pxa_gpio_mode(37 | GPIO_IN);	/* SFRM */
+	pxa_gpio_mode(38 | GPIO_IN);	/* TXD  */
+	pxa_gpio_mode(22 | GPIO_IN);	/* SCLK */
+	pxa_gpio_mode(88 | GPIO_IN);	/* RXD  */
+
+	pxa_gpio_mode(GPIO_HW_ATTENUATE_A780 | GPIO_OUT);
+	pxa_gpio_set_value(GPIO_HW_ATTENUATE_A780, 1);
+
+	/* request jack irq */
+	request_irq(EZX_IRQ_HEADJACK, &jack_irq, IRQF_DISABLED, "headphone jack", NULL);
+	request_irq(EZX_IRQ_MIC, &jack_irq, IRQF_DISABLED, "mic jack", NULL);
+
+	return ret;
+}
+
+static void __exit ezx_exit(void)
+{
+	free_irq(EZX_IRQ_HEADJACK, NULL);
+	free_irq(EZX_IRQ_MIC, NULL);
+	platform_device_unregister(ezx_snd_device);
+}
+
+module_init(ezx_init);
+module_exit(ezx_exit);
+
Index: linux-2.6.23/sound/soc/codecs/Makefile
===================================================================
--- linux-2.6.23.orig/sound/soc/codecs/Makefile	2007-10-10 09:38:42.000000000 +0200
+++ linux-2.6.23/sound/soc/codecs/Makefile	2007-10-22 22:30:09.000000000 +0200
@@ -3,9 +3,11 @@
 snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
 snd-soc-wm9712-objs := wm9712.o
+snd-soc-pcap2-objs := pcap2.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_WM8731)	+= snd-soc-wm8731.o
 obj-$(CONFIG_SND_SOC_WM8750)	+= snd-soc-wm8750.o
 obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
 obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
+obj-$(CONFIG_SND_SOC_PCAP2)	+= snd-soc-pcap2.o
Index: linux-2.6.23/sound/soc/codecs/Kconfig
===================================================================
--- linux-2.6.23.orig/sound/soc/codecs/Kconfig	2007-10-10 09:38:42.000000000 +0200
+++ linux-2.6.23/sound/soc/codecs/Kconfig	2007-10-22 22:28:06.000000000 +0200
@@ -17,3 +17,7 @@
 config SND_SOC_WM9712
 	tristate
 	depends on SND_SOC
+
+config SND_SOC_PCAP2
+	tristate
+	depends on SND_SOC && EZX_PCAP
Index: linux-2.6.23/sound/soc/pxa/Makefile
===================================================================
--- linux-2.6.23.orig/sound/soc/pxa/Makefile	2007-10-22 22:27:11.000000000 +0200
+++ linux-2.6.23/sound/soc/pxa/Makefile	2007-10-22 22:28:06.000000000 +0200
@@ -14,9 +14,10 @@
 snd-soc-poodle-objs := poodle.o
 snd-soc-tosa-objs := tosa.o
 snd-soc-spitz-objs := spitz.o
+snd-soc-ezx-objs := ezx.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
 obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
-
+obj-$(CONFIG_SND_PXA2XX_SOC_EZX) += snd-soc-ezx.o