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author | Mirko Vogt <mirko@openwrt.org> | 2008-12-12 11:58:53 +0000 |
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committer | Mirko Vogt <mirko@openwrt.org> | 2008-12-12 11:58:53 +0000 |
commit | 614683faf8029100802db06a825648d0b6490285 (patch) | |
tree | 7401b135dc7ce24ff0175e67e0f2ce7f96296ff0 /target/linux/s3c24xx/patches-2.6.24/1037-gta02-sound.patch.patch | |
parent | 4a018d2445c5f249179ff82c8fffb0e3b717f738 (diff) | |
download | master-31e0f0ae-614683faf8029100802db06a825648d0b6490285.tar.gz master-31e0f0ae-614683faf8029100802db06a825648d0b6490285.tar.bz2 master-31e0f0ae-614683faf8029100802db06a825648d0b6490285.zip |
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.patch | 772 |
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 + |