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authormokopatches <mokopatches@openmoko.org>2008-11-19 17:03:15 +0000
committerwarmcat <andy@warmcat.com>2008-11-19 17:03:15 +0000
commit5915ea88b0b6ada7fc5234239feec5d4960887df (patch)
tree7ff043677dd2bac531fe096eab2aed70a3491732 /sound
parentf30f7d454ad77de110b89e90cc27351a302ba651 (diff)
gta02-sound.patch
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/s3c24xx/Kconfig9
-rw-r--r--sound/soc/s3c24xx/Makefile3
-rw-r--r--sound/soc/s3c24xx/neo1973_gta02_wm8753.c667
-rw-r--r--sound/soc/soc-dapm.c24
4 files changed, 703 insertions, 0 deletions
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index b9f2353effe..06385721bcd 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -26,6 +26,15 @@ 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 0aa5fb0b970..f154cb142a2 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -13,7 +13,10 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
+snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
obj-$(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_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.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 00000000000..f32cba319cd
--- /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 7351db9606e..6a41d200487 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1525,6 +1525,30 @@ int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin)
EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status);
/**
+ * 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
*