linux-openmoko_2.6.22.5.bb : update for kernel, gta02 sound support,

fixed suspend/resume while playing sound. fixed suspend/resume for
vibrator. Moved SCRREV onwards.

5 files changed, 964 insertions, 0 deletions

diff --git a/packages/linux/linux-openmoko/gta-vibro-pwm-suspend.patch b/packages/linux/linux-openmoko/gta-vibro-pwm-suspend.patch
new file mode 100644
index 0000000000..a98f5c8ca5
--- /dev/null
+++ b/packages/linux/linux-openmoko/gta-vibro-pwm-suspend.patch
@@ -0,0 +1,27 @@
+Index: linux-2.6.22/drivers/leds/leds-gta01.c
+===================================================================
+--- linux-2.6.22.orig/drivers/leds/leds-gta01.c	2007-11-09 16:27:03.000000000 +0000
++++ linux-2.6.22/drivers/leds/leds-gta01.c	2007-11-09 16:27:13.000000000 +0000
+@@ -33,6 +33,8 @@
+ 	struct s3c2410_pwm pwm;
+ };
+ 
++static int gta01vib_init_hw(struct gta01_vib_priv *vp);
++
+ static inline struct gta01_vib_priv *pdev_to_vpriv(struct platform_device *dev)
+ {
+ 	return platform_get_drvdata(dev);
+@@ -80,7 +82,13 @@
+ 
+ static int gta01vib_resume(struct platform_device *dev)
+ {
++	struct gta01_vib_priv *vp = pdev_to_vpriv(dev);
++
+ 	led_classdev_resume(&gta01_vib_led);
++
++	if (vp->has_pwm)
++		gta01vib_init_hw(vp);
++
+ 	return 0;
+ }
+ #endif
diff --git a/packages/linux/linux-openmoko/gta02-sound.patch b/packages/linux/linux-openmoko/gta02-sound.patch
new file mode 100644
index 0000000000..66fc850350
--- /dev/null
+++ b/packages/linux/linux-openmoko/gta02-sound.patch
@@ -0,0 +1,754 @@
+Index: linux-2.6.22/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
+===================================================================
+--- /dev/null	1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.22/sound/soc/s3c24xx/neo1973_gta02_wm8753.c	2007-11-09 16:21:35.000000000 +0000
+@@ -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/arch/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_AMP) >> 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-2.6.22/sound/soc/s3c24xx/Kconfig
+===================================================================
+--- linux-2.6.22.orig/sound/soc/s3c24xx/Kconfig	2007-11-09 16:13:04.000000000 +0000
++++ linux-2.6.22/sound/soc/s3c24xx/Kconfig	2007-11-09 16:13:08.000000000 +0000
+@@ -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-2.6.22/sound/soc/s3c24xx/Makefile
+===================================================================
+--- linux-2.6.22.orig/sound/soc/s3c24xx/Makefile	2007-11-09 16:13:04.000000000 +0000
++++ linux-2.6.22/sound/soc/s3c24xx/Makefile	2007-11-09 16:13:08.000000000 +0000
+@@ -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-2.6.22/include/sound/soc-dapm.h
+===================================================================
+--- linux-2.6.22.orig/include/sound/soc-dapm.h	2007-07-09 00:32:17.000000000 +0100
++++ linux-2.6.22/include/sound/soc-dapm.h	2007-11-09 16:13:08.000000000 +0000
+@@ -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-2.6.22/sound/soc/soc-dapm.c
+===================================================================
+--- linux-2.6.22.orig/sound/soc/soc-dapm.c	2007-07-09 00:32:17.000000000 +0100
++++ linux-2.6.22/sound/soc/soc-dapm.c	2007-11-09 16:13:08.000000000 +0000
+@@ -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
+  *
diff --git a/packages/linux/linux-openmoko/iis-suspend.patch b/packages/linux/linux-openmoko/iis-suspend.patch
new file mode 100644
index 0000000000..ab38672463
--- /dev/null
+++ b/packages/linux/linux-openmoko/iis-suspend.patch
@@ -0,0 +1,63 @@
+Index: linux-2.6.22/sound/soc/s3c24xx/s3c24xx-i2s.c
+===================================================================
+--- linux-2.6.22.orig/sound/soc/s3c24xx/s3c24xx-i2s.c	2007-11-09 16:13:04.000000000 +0000
++++ linux-2.6.22/sound/soc/s3c24xx/s3c24xx-i2s.c	2007-11-09 16:17:47.000000000 +0000
+@@ -75,6 +75,10 @@
+ struct s3c24xx_i2s_info {
+ 	void __iomem	*regs;
+ 	struct clk	*iis_clk;
++	u32		iiscon;
++	u32		iismod;
++	u32		iisfcon;
++	u32		iispsr;
+ };
+ static struct s3c24xx_i2s_info s3c24xx_i2s;
+ 
+@@ -404,6 +408,38 @@
+ 	return 0;
+ }
+ 
++#ifdef CONFIG_PM
++int s3c24xx_i2s_suspend(struct platform_device *pdev,
++                struct snd_soc_cpu_dai *cpu_dai)
++{
++        s3c24xx_i2s.iiscon=readl(s3c24xx_i2s.regs + S3C2410_IISCON);
++	s3c24xx_i2s.iismod=readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
++	s3c24xx_i2s.iisfcon=readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
++	s3c24xx_i2s.iispsr=readl(s3c24xx_i2s.regs + S3C2410_IISPSR);
++
++	clk_disable(s3c24xx_i2s.iis_clk);
++
++	return 0;
++}
++
++int s3c24xx_i2s_resume(struct platform_device *pdev,
++                struct snd_soc_cpu_dai *cpu_dai)
++{
++	clk_enable(s3c24xx_i2s.iis_clk);
++
++        writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
++        writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
++        writel(s3c24xx_i2s.iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
++        writel(s3c24xx_i2s.iispsr, s3c24xx_i2s.regs + S3C2410_IISPSR);
++
++	return 0;
++}
++#else
++#define s3c24xx_i2s_suspend NULL
++#define s3c24xx_i2s_resume NULL
++#endif
++
++
+ #define S3C24XX_I2S_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 | \
+@@ -414,6 +450,8 @@
+ 	.id = 0,
+ 	.type = SND_SOC_DAI_I2S,
+ 	.probe = s3c24xx_i2s_probe,
++	.suspend = s3c24xx_i2s_suspend,
++	.resume = s3c24xx_i2s_resume,
+ 	.playback = {
+ 		.channels_min = 2,
+ 		.channels_max = 2,
diff --git a/packages/linux/linux-openmoko/s3c24xx-pcm-suspend.patch b/packages/linux/linux-openmoko/s3c24xx-pcm-suspend.patch
new file mode 100644
index 0000000000..fffc685a4e
--- /dev/null
+++ b/packages/linux/linux-openmoko/s3c24xx-pcm-suspend.patch
@@ -0,0 +1,73 @@
+Index: linux-2.6.22/sound/soc/s3c24xx/s3c24xx-pcm.c
+===================================================================
+--- linux-2.6.22.orig/sound/soc/s3c24xx/s3c24xx-pcm.c	2007-11-09 16:28:43.000000000 +0000
++++ linux-2.6.22/sound/soc/s3c24xx/s3c24xx-pcm.c	2007-11-09 16:33:05.000000000 +0000
+@@ -49,7 +49,9 @@
+ 	.info			= SNDRV_PCM_INFO_INTERLEAVED |
+ 				    SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ 				    SNDRV_PCM_INFO_MMAP |
+-				    SNDRV_PCM_INFO_MMAP_VALID,
++				    SNDRV_PCM_INFO_MMAP_VALID |
++				    SNDRV_PCM_INFO_PAUSE |
++				    SNDRV_PCM_INFO_RESUME,
+ 	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
+ 				    SNDRV_PCM_FMTBIT_U16_LE |
+ 				    SNDRV_PCM_FMTBIT_U8 |
+@@ -176,28 +178,6 @@
+ 		}
+ 	}
+ 
+-	/* channel needs configuring for mem=>device, increment memory addr,
+-	 * sync to pclk, half-word transfers to the IIS-FIFO. */
+-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+-		s3c2410_dma_devconfig(prtd->params->channel,
+-				S3C2410_DMASRC_MEM, S3C2410_DISRCC_INC |
+-				S3C2410_DISRCC_APB, prtd->params->dma_addr);
+-
+-		s3c2410_dma_config(prtd->params->channel,
+-				prtd->params->dma_size,
+-				S3C2410_DCON_SYNC_PCLK | 
+-				S3C2410_DCON_HANDSHAKE);
+-	} else {
+-		s3c2410_dma_config(prtd->params->channel,
+-				prtd->params->dma_size,
+-				S3C2410_DCON_HANDSHAKE | 
+-				S3C2410_DCON_SYNC_PCLK);
+-
+-		s3c2410_dma_devconfig(prtd->params->channel,
+-					S3C2410_DMASRC_HW, 0x3,
+-					prtd->params->dma_addr);
+-	}
+-
+ 	s3c2410_dma_set_buffdone_fn(prtd->params->channel,
+ 				    s3c24xx_audio_buffdone);
+ 
+@@ -246,6 +226,28 @@
+ 	if (!prtd->params)
+ 	 	return 0;
+ 
++	/* channel needs configuring for mem=>device, increment memory addr,
++	 * sync to pclk, half-word transfers to the IIS-FIFO. */
++	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
++		s3c2410_dma_devconfig(prtd->params->channel,
++				S3C2410_DMASRC_MEM, S3C2410_DISRCC_INC |
++				S3C2410_DISRCC_APB, prtd->params->dma_addr);
++
++		s3c2410_dma_config(prtd->params->channel,
++				prtd->params->dma_size,
++				S3C2410_DCON_SYNC_PCLK |
++				S3C2410_DCON_HANDSHAKE);
++	} else {
++		s3c2410_dma_config(prtd->params->channel,
++				prtd->params->dma_size,
++				S3C2410_DCON_HANDSHAKE |
++				S3C2410_DCON_SYNC_PCLK);
++
++		s3c2410_dma_devconfig(prtd->params->channel,
++					S3C2410_DMASRC_HW, 0x3,
++					prtd->params->dma_addr);
++	}
++
+ 	/* flush the DMA channel */
+ 	s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH);
+ 	prtd->dma_loaded = 0;
diff --git a/packages/linux/linux-openmoko/soc-core-suspend.patch b/packages/linux/linux-openmoko/soc-core-suspend.patch
new file mode 100644
index 0000000000..7fb6cc7f7a
--- /dev/null
+++ b/packages/linux/linux-openmoko/soc-core-suspend.patch
@@ -0,0 +1,47 @@
+From 6c868238a5e083dca4d74439a7fd467b5c7726b0 Mon Sep 17 00:00:00 2001
+From: Liam Girdwood <liam@localhost.localdomain>
+Date: Fri, 1 Jun 2007 12:56:55 +0100
+Subject: [PATCH] This fixes a bug whereby PCM's were not being suspended when the rest of the audio subsystem was suspended.
+
+---
+ include/sound/soc.h  |    3 +++
+ sound/soc/soc-core.c |    8 ++++++++
+ 2 files changed, 11 insertions(+), 0 deletions(-)
+
+Index: linux-2.6.22/include/sound/soc.h
+===================================================================
+--- linux-2.6.22.orig/include/sound/soc.h	2007-07-09 00:32:17.000000000 +0100
++++ linux-2.6.22/include/sound/soc.h	2007-11-09 16:13:08.000000000 +0000
+@@ -411,6 +411,9 @@
+ 
+ 	/* codec/machine specific init - e.g. add machine controls */
+ 	int (*init)(struct snd_soc_codec *codec);
++	
++	/* DAI pcm */
++	struct snd_pcm *pcm;
+ };
+ 
+ /* SoC machine */
+Index: linux-2.6.22/sound/soc/soc-core.c
+===================================================================
+--- linux-2.6.22.orig/sound/soc/soc-core.c	2007-07-09 00:32:17.000000000 +0100
++++ linux-2.6.22/sound/soc/soc-core.c	2007-11-09 16:18:45.000000000 +0000
+@@ -639,6 +639,10 @@
+ 			dai->dai_ops.digital_mute(dai, 1);
+ 	}
+ 
++	/* suspend all pcm's */
++	for(i = 0; i < machine->num_links; i++)
++		snd_pcm_suspend_all(machine->dai_link[i].pcm);
++
+ 	if (machine->suspend_pre)
+ 		machine->suspend_pre(pdev, state);
+ 
+@@ -873,6 +877,7 @@
+ 		return ret;
+ 	}
+ 
++	dai_link->pcm = pcm;
+ 	pcm->private_data = rtd;
+ 	soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap;
+ 	soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer;