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authorMirko Vogt <mirko@openwrt.org>2008-12-12 11:58:53 +0000
committerMirko Vogt <mirko@openwrt.org>2008-12-12 11:58:53 +0000
commit614683faf8029100802db06a825648d0b6490285 (patch)
tree7401b135dc7ce24ff0175e67e0f2ce7f96296ff0 /target/linux/s3c24xx/patches-2.6.24/1037-gta02-sound.patch.patch
parent4a018d2445c5f249179ff82c8fffb0e3b717f738 (diff)
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changed Makefile and profiles, added patches for kernel 2.6.24 (stable-branch of Openmoko)
SVN-Revision: 13613
Diffstat (limited to 'target/linux/s3c24xx/patches-2.6.24/1037-gta02-sound.patch.patch')
-rw-r--r--target/linux/s3c24xx/patches-2.6.24/1037-gta02-sound.patch.patch772
1 files changed, 772 insertions, 0 deletions
diff --git a/target/linux/s3c24xx/patches-2.6.24/1037-gta02-sound.patch.patch b/target/linux/s3c24xx/patches-2.6.24/1037-gta02-sound.patch.patch
new file mode 100644
index 0000000000..b1721e7516
--- /dev/null
+++ b/target/linux/s3c24xx/patches-2.6.24/1037-gta02-sound.patch.patch
@@ -0,0 +1,772 @@
+From c7b72aecf04a0a51e04f7d9346757c7068e0a74d Mon Sep 17 00:00:00 2001
+From: mokopatches <mokopatches@openmoko.org>
+Date: Fri, 4 Apr 2008 11:35:32 +0100
+Subject: [PATCH] gta02-sound.patch
+
+---
+ include/sound/soc-dapm.h | 2 +
+ sound/soc/s3c24xx/Kconfig | 9 +
+ sound/soc/s3c24xx/Makefile | 3 +
+ sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 667 ++++++++++++++++++++++++++++++
+ sound/soc/soc-dapm.c | 24 ++
+ 5 files changed, 705 insertions(+), 0 deletions(-)
+ create mode 100644 sound/soc/s3c24xx/neo1973_gta02_wm8753.c
+
+diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
+index 2b1ae8e..204be0b 100644
+--- a/include/sound/soc-dapm.h
++++ b/include/sound/soc-dapm.h
+@@ -206,6 +206,8 @@ int snd_soc_dapm_sys_add(struct device *dev);
+ /* 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 */
+diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
+index 5632a2e..4146ac4 100644
+--- a/sound/soc/s3c24xx/Kconfig
++++ b/sound/soc/s3c24xx/Kconfig
+@@ -25,6 +25,15 @@ config SND_S3C24XX_SOC_NEO1973_WM8753
+ Say Y if you want to add support for SoC audio on smdk2440
+ with the WM8753.
+
++config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753
++ tristate "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
+diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
+index 13c92f0..e356071 100644
+--- a/sound/soc/s3c24xx/Makefile
++++ b/sound/soc/s3c24xx/Makefile
+@@ -10,6 +10,9 @@ obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o
+ # S3C24XX Machine Support
+ snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
+ snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
++snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
+
+ obj-$(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
++
+diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
+new file mode 100644
+index 0000000..f32cba3
+--- /dev/null
++++ b/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");
++
+diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
+index 29a546f..84fa860 100644
+--- a/sound/soc/soc-dapm.c
++++ b/sound/soc/soc-dapm.c
+@@ -1305,6 +1305,30 @@ int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
+ 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
+ *
+--
+1.5.6.5
+