diff options
Diffstat (limited to 'sound/soc/s3c24xx')
| -rw-r--r-- | sound/soc/s3c24xx/Kconfig | 35 | ||||
| -rw-r--r-- | sound/soc/s3c24xx/Makefile | 9 | ||||
| -rw-r--r-- | sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 498 | ||||
| -rw-r--r-- | sound/soc/s3c24xx/s3c-i2s-v2.c | 33 | ||||
| -rw-r--r-- | sound/soc/s3c24xx/s3c2443-ac97.c | 20 | ||||
| -rw-r--r-- | sound/soc/s3c24xx/s3c24xx-ac97.h | 6 | ||||
| -rw-r--r-- | sound/soc/s3c24xx/s3c24xx-i2s.c | 5 | ||||
| -rw-r--r-- | sound/soc/s3c24xx/s3c24xx-pcm.c | 2 | ||||
| -rw-r--r-- | sound/soc/s3c24xx/s3c24xx_simtec.c | 394 | ||||
| -rw-r--r-- | sound/soc/s3c24xx/s3c24xx_simtec.h | 22 | ||||
| -rw-r--r-- | sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 153 | ||||
| -rw-r--r-- | sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 137 | ||||
| -rw-r--r-- | sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 |
13 files changed, 1291 insertions, 25 deletions
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index df494d1e346f..923428fc1adb 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,6 +1,7 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3CXXXX chips" - depends on ARCH_S3C2410 + depends on ARCH_S3C2410 || ARCH_S3C64XX + select S3C64XX_DMA if ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to the S3C24XX AC97 or I2S interfaces. You will also need to @@ -38,6 +39,15 @@ 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 "Audio support for the Openmoko Neo FreeRunner (GTA02)" + depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02 + select SND_S3C24XX_SOC_I2S + select SND_SOC_WM8753 + help + This driver provides audio support for the Openmoko Neo FreeRunner + smartphone. + config SND_S3C24XX_SOC_JIVE_WM8750 tristate "SoC I2S Audio support for Jive" depends on SND_S3C24XX_SOC && MACH_JIVE @@ -57,7 +67,7 @@ config SND_S3C24XX_SOC_SMDK2443_WM9710 config SND_S3C24XX_SOC_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" - depends on SND_S3C24XX_SOC + depends on SND_S3C24XX_SOC && ARCH_S3C2410 select SND_S3C2443_SOC_AC97 select SND_SOC_AC97_CODEC help @@ -66,7 +76,26 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650 config SND_S3C24XX_SOC_S3C24XX_UDA134X tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" - depends on SND_S3C24XX_SOC + depends on SND_S3C24XX_SOC && ARCH_S3C2410 select SND_S3C24XX_SOC_I2S select SND_SOC_L3 select SND_SOC_UDA134X + +config SND_S3C24XX_SOC_SIMTEC + tristate + help + Internal node for common S3C24XX/Simtec suppor + +config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23 + tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC23 + select SND_S3C24XX_SOC_SIMTEC + +config SND_S3C24XX_SOC_SIMTEC_HERMES + tristate "SoC I2S Audio support for Simtec Hermes board" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC3X + select SND_S3C24XX_SOC_SIMTEC diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 07a93a2ebe5f..99f5a7dd3fc6 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -16,12 +16,21 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o +snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o +snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o +snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o +snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o +obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-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-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o + diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c new file mode 100644 index 000000000000..0c52e36ddd87 --- /dev/null +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -0,0 +1,498 @@ +/* + * neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02) + * + * Copyright 2007 Openmoko Inc + * Author: Graeme Gregory <graeme@openmoko.org> + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory <linux@wolfsonmicro.com> + * Copyright 2009 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include <plat/regs-iis.h> + +#include <mach/regs-clock.h> +#include <asm/io.h> +#include <mach/gta02.h> +#include "../codecs/wm8753.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" + +static struct snd_soc_card 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_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_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 = snd_soc_dai_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 = snd_soc_dai_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 = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set codec BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, + WM8753_BCLKDIV, bclk); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(4, 4)); + if (ret < 0) + return ret; + + /* codec PLL input is PCLK/4 */ + ret = snd_soc_dai_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_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_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_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 = snd_soc_dai_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 = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, + 12288000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec PCM division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, + pcmdiv); + if (ret < 0) + return ret; + + /* configue and enable PLL for 12.288MHz output */ + ret = snd_soc_dai_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_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_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; + +/* This has no effect, it exists only to maintain compatibility with + * existing ALSA state files. + */ +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; + else + lm4853_state &= ~LM4853_AMP; + + 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; + gpio_set_value(GTA02_GPIO_HP_IN, 0); + } else { + lm4853_state &= ~LM4853_SPK; + gpio_set_value(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 lm4853_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, + int event) +{ + gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(value)); + + return 0; +} + +static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), + 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 struct snd_soc_dapm_route audio_map[] = { + + /* 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"}, +}; + +static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = { + SOC_DAPM_PIN_SWITCH("Stereo Out"), + SOC_DAPM_PIN_SWITCH("GSM Line Out"), + SOC_DAPM_PIN_SWITCH("GSM Line In"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Spk"), + + /* This has no effect, it exists only to maintain compatibility with + * existing ALSA state files. + */ + 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 err; + + /* set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "OUT4"); + snd_soc_dapm_nc_pin(codec, "LINE1"); + snd_soc_dapm_nc_pin(codec, "LINE2"); + + /* Add neo1973 gta02 specific widgets */ + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); + + /* add neo1973 gta02 specific controls */ + err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls, + ARRAY_SIZE(wm8753_neo1973_gta02_controls)); + + if (err < 0) + return err; + + /* set up neo1973 gta02 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* set endpoints to default off mode */ + snd_soc_dapm_disable_pin(codec, "Stereo Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Handset Mic"); + snd_soc_dapm_disable_pin(codec, "Handset Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* + * BT Codec DAI + */ +static struct snd_soc_dai bt_dai = { + .name = "Bluetooth", + .id = 0, + .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, +}, +}; + +static struct snd_soc_card neo1973_gta02 = { + .name = "neo1973-gta02", + .platform = &s3c24xx_soc_platform, + .dai_link = neo1973_gta02_dai, + .num_links = ARRAY_SIZE(neo1973_gta02_dai), +}; + +static struct snd_soc_device neo1973_gta02_snd_devdata = { + .card = &neo1973_gta02, + .codec_dev = &soc_codec_dev_wm8753, +}; + +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 is supported by this ASoC driver\n"); + return -ENODEV; + } + + /* register bluetooth DAI here */ + ret = snd_soc_register_dai(&bt_dai); + if (ret) + return ret; + + 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); + return ret; + } + + /* Initialise GPIOs used by amp */ + ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN"); + if (ret) { + pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN); + goto err_unregister_device; + } + + ret = gpio_direction_output(GTA02_GPIO_AMP_HP_IN, 1); + if (ret) { + pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN); + goto err_free_gpio_hp_in; + } + + ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT"); + if (ret) { + pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT); + goto err_free_gpio_hp_in; + } + + ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1); + if (ret) { + pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT); + goto err_free_gpio_amp_shut; + } + + return 0; + +err_free_gpio_amp_shut: + gpio_free(GTA02_GPIO_AMP_SHUT); +err_free_gpio_hp_in: + gpio_free(GTA02_GPIO_HP_IN); +err_unregister_device: + platform_device_unregister(neo1973_gta02_snd_device); + return ret; +} +module_init(neo1973_gta02_init); + +static void __exit neo1973_gta02_exit(void) +{ + snd_soc_unregister_dai(&bt_dai); + platform_device_unregister(neo1973_gta02_snd_device); + gpio_free(GTA02_GPIO_HP_IN); + gpio_free(GTA02_GPIO_AMP_SHUT); +} +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/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 1a283170ca92..9bc4aa35caab 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -36,6 +36,7 @@ #include <mach/dma.h> #include "s3c-i2s-v2.h" +#include "s3c24xx-pcm.h" #undef S3C_IIS_V2_SUPPORTED @@ -229,6 +230,8 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); } +#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) + /* * Wait for the LR signal to allow synchronisation to the L/R clock * from the codec. May only be needed for slave mode. @@ -236,19 +239,21 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) { u32 iiscon; - unsigned long timeout = jiffies + msecs_to_jiffies(5); + unsigned long loops = msecs_to_loops(5); pr_debug("Entered %s\n", __func__); - while (1) { + while (--loops) { iiscon = readl(i2s->regs + S3C2412_IISCON); if (iiscon & S3C2412_IISCON_LRINDEX) break; - if (timeout < jiffies) { - printk(KERN_ERR "%s: timeout\n", __func__); - return -ETIMEDOUT; - } + cpu_relax(); + } + + if (!loops) { + printk(KERN_ERR "%s: timeout\n", __func__); + return -ETIMEDOUT; } return 0; @@ -357,19 +362,19 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, #endif #ifdef CONFIG_PLAT_S3C64XX - iismod &= ~0x606; + iismod &= ~(S3C64XX_IISMOD_BLC_MASK | S3C2412_IISMOD_BCLK_MASK); /* Sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: /* 8 bit sample, 16fs BCLK */ - iismod |= 0x2004; + iismod |= (S3C64XX_IISMOD_BLC_8BIT | S3C2412_IISMOD_BCLK_16FS); break; case SNDRV_PCM_FORMAT_S16_LE: /* 16 bit sample, 32fs BCLK */ break; case SNDRV_PCM_FORMAT_S24_LE: /* 24 bit sample, 48fs BCLK */ - iismod |= 0x4002; + iismod |= (S3C64XX_IISMOD_BLC_24BIT | S3C2412_IISMOD_BCLK_48FS); break; } #endif @@ -387,6 +392,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; + int channel = ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); @@ -416,6 +423,14 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, s3c2412_snd_txctrl(i2s, 1); local_irq_restore(irqs); + + /* + * Load the next buffer to DMA to meet the reqirement + * of the auto reload mechanism of S3C24XX. + * This call won't bother S3C64XX. + */ + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + break; case SNDRV_PCM_TRIGGER_STOP: diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 3f03d5ddfacd..fc1beb0930b9 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -47,7 +47,7 @@ static struct s3c24xx_ac97_info s3c24xx_ac97; static DECLARE_COMPLETION(ac97_completion); static u32 codec_ready; -static DECLARE_MUTEX(ac97_mutex); +static DEFINE_MUTEX(ac97_mutex); static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, unsigned short reg) @@ -56,7 +56,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, u32 ac_codec_cmd; u32 stat, addr, data; - down(&ac97_mutex); + mutex_lock(&ac97_mutex); codec_ready = S3C_AC97_GLBSTAT_CODECREADY; ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); @@ -79,7 +79,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, printk(KERN_ERR "s3c24xx-ac97: req addr = %02x," " rep addr = %02x\n", reg, addr); - up(&ac97_mutex); + mutex_unlock(&ac97_mutex); return (unsigned short)data; } @@ -90,7 +90,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, u32 ac_glbctrl; u32 ac_codec_cmd; - down(&ac97_mutex); + mutex_lock(&ac97_mutex); codec_ready = S3C_AC97_GLBSTAT_CODECREADY; ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); @@ -109,7 +109,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - up(&ac97_mutex); + mutex_unlock(&ac97_mutex); } @@ -290,6 +290,9 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int channel = ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); switch (cmd) { @@ -312,6 +315,8 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, } writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + return 0; } @@ -334,6 +339,9 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int channel = ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); switch (cmd) { @@ -349,6 +357,8 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, } writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + return 0; } diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h index a96dcadf28b4..e96f941a810b 100644 --- a/sound/soc/s3c24xx/s3c24xx-ac97.h +++ b/sound/soc/s3c24xx/s3c24xx-ac97.h @@ -20,12 +20,6 @@ #define AC_CMD_ADDR(x) (x << 16) #define AC_CMD_DATA(x) (x & 0xffff) -#ifdef CONFIG_CPU_S3C2440 -#define IRQ_S3C244x_AC97 IRQ_S3C2440_AC97 -#else -#define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97 -#endif - extern struct snd_soc_dai s3c2443_ac97_dai[]; #endif /*S3C24XXAC97_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 556e35f0ab73..40e2c4790f0d 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -279,6 +279,9 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int channel = ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); @@ -296,6 +299,8 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, s3c24xx_snd_rxctrl(1); else s3c24xx_snd_txctrl(1); + + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index eecfa5eba06b..5cbbdc80fde3 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -255,7 +255,6 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->state |= ST_RUNNING; s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: @@ -318,6 +317,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c new file mode 100644 index 000000000000..1966e0d5652d --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c @@ -0,0 +1,394 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <linux/i2c.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +static struct s3c24xx_audio_simtec_pdata *pdata; +static struct clk *xtal_clk; + +static int spk_gain; +static int spk_unmute; + +/** + * speaker_gain_get - read the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_gain; + return 0; +} + +/** + * speaker_gain_set - set the value of the speaker amp gain + * @value: The value to write. + */ +static void speaker_gain_set(int value) +{ + gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); + gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); +} + +/** + * speaker_gain_put - set the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + * + * Note, if the speaker amp is muted, then we do not set a gain value + * as at-least one of the ICs that is fitted will try and power up even + * if the main control is set to off. + */ +static int speaker_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = ucontrol->value.integer.value[0]; + + spk_gain = value; + + if (!spk_unmute) + speaker_gain_set(value); + + return 0; +} + +static const struct snd_kcontrol_new amp_gain_controls[] = { + SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, + speaker_gain_get, speaker_gain_put), +}; + +/** + * spk_unmute_state - set the unmute state of the speaker + * @to: zero to unmute, non-zero to ununmute. + */ +static void spk_unmute_state(int to) +{ + pr_debug("%s: to=%d\n", __func__, to); + + spk_unmute = to; + gpio_set_value(pdata->amp_gpio, to); + + /* if we're umuting, also re-set the gain */ + if (to && pdata->amp_gain[0] > 0) + speaker_gain_set(spk_gain); +} + +/** + * speaker_unmute_get - read the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_unmute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_unmute; + return 0; +} + +/** + * speaker_unmute_put - set the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + */ +static int speaker_unmute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + spk_unmute_state(ucontrol->value.integer.value[0]); + return 0; +} + +/* This is added as a manual control as the speaker amps create clicks + * when their power state is changed, which are far more noticeable than + * anything produced by the CODEC itself. + */ +static const struct snd_kcontrol_new amp_unmute_controls[] = { + SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0, + speaker_unmute_get, speaker_unmute_put), +}; + +void simtec_audio_init(struct snd_soc_codec *codec) +{ + if (pdata->amp_gpio > 0) { + pr_debug("%s: adding amp routes\n", __func__); + + snd_soc_add_controls(codec, amp_unmute_controls, + ARRAY_SIZE(amp_unmute_controls)); + } + + if (pdata->amp_gain[0] > 0) { + pr_debug("%s: adding amp controls\n", __func__); + snd_soc_add_controls(codec, amp_gain_controls, + ARRAY_SIZE(amp_gain_controls)); + } +} +EXPORT_SYMBOL_GPL(simtec_audio_init); + +#define CODEC_CLOCK 12000000 + +/** + * simtec_hw_params - update hardware parameters + * @substream: The audio substream instance. + * @params: The parameters requested. + * + * Update the codec data routing and configuration settings + * from the supplied data. + */ +static int simtec_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_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set the CODEC as the bus clock master, I2S */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + + /* Set the CODEC as the bus clock master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret) { + pr_err( "%s: failed setting codec sysclk\n", __func__); + return ret; + } + + if (pdata->use_mpllin) { + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, + 0, SND_SOC_CLOCK_OUT); + + if (ret) { + pr_err("%s: failed to set MPLLin as clksrc\n", + __func__); + return ret; + } + } + + if (pdata->output_cdclk) { + int cdclk_scale; + + cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; + cdclk_scale--; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + cdclk_scale); + } + + return 0; +} + +static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) +{ + /* call any board supplied startup code, this currently only + * covers the bast/vr1000 which have a CPLD in the way of the + * LRCLK */ + if (pd->startup) + pd->startup(); + + return 0; +} + +static struct snd_soc_ops simtec_snd_ops = { + .hw_params = simtec_hw_params, +}; + +/** + * attach_gpio_amp - get and configure the necessary gpios + * @dev: The device we're probing. + * @pd: The platform data supplied by the board. + * + * If there is a GPIO based amplifier attached to the board, claim + * the necessary GPIO lines for it, and set default values. + */ +static int attach_gpio_amp(struct device *dev, + struct s3c24xx_audio_simtec_pdata *pd) +{ + int ret; + + /* attach gpio amp gain (if any) */ + if (pdata->amp_gain[0] > 0) { + ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain0\n"); + return ret; + } + + ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain1\n"); + gpio_free(pdata->amp_gain[0]); + return ret; + } + + gpio_direction_output(pd->amp_gain[0], 0); + gpio_direction_output(pd->amp_gain[1], 0); + } + + /* note, curently we assume GPA0 isn't valid amp */ + if (pdata->amp_gpio > 0) { + ret = gpio_request(pd->amp_gpio, "gpio-amp"); + if (ret) { + dev_err(dev, "cannot get amp gpio %d (%d)\n", + pd->amp_gpio, ret); + goto err_amp; + } + + /* set the amp off at startup */ + spk_unmute_state(0); + } + + return 0; + +err_amp: + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + return ret; +} + +static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) +{ + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + if (pd->amp_gpio > 0) + gpio_free(pd->amp_gpio); +} + +#ifdef CONFIG_PM +int simtec_audio_resume(struct device *dev) +{ + simtec_call_startup(pdata); + return 0; +} + +struct dev_pm_ops simtec_audio_pmops = { + .resume = simtec_audio_resume, +}; +EXPORT_SYMBOL_GPL(simtec_audio_pmops); +#endif + +int __devinit simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev) +{ + struct platform_device *snd_dev; + int ret; + + socdev->card->dai_link->ops = &simtec_snd_ops; + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "no platform data supplied\n"); + return -EINVAL; + } + + simtec_call_startup(pdata); + + xtal_clk = clk_get(&pdev->dev, "xtal"); + if (IS_ERR(xtal_clk)) { + dev_err(&pdev->dev, "could not get clkout0\n"); + return -EINVAL; + } + + dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); + + ret = attach_gpio_amp(&pdev->dev, pdata); + if (ret) + goto err_clk; + + snd_dev = platform_device_alloc("soc-audio", -1); + if (!snd_dev) { + dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); + ret = -ENOMEM; + goto err_gpio; + } + + platform_set_drvdata(snd_dev, socdev); + socdev->dev = &snd_dev->dev; + + ret = platform_device_add(snd_dev); + if (ret) { + dev_err(&pdev->dev, "failed to add soc-audio dev\n"); + goto err_pdev; + } + + platform_set_drvdata(pdev, snd_dev); + return 0; + +err_pdev: + platform_device_put(snd_dev); + +err_gpio: + detach_gpio_amp(pdata); + +err_clk: + clk_put(xtal_clk); + return ret; +} +EXPORT_SYMBOL_GPL(simtec_audio_core_probe); + +int __devexit simtec_audio_remove(struct platform_device *pdev) +{ + struct platform_device *snd_dev = platform_get_drvdata(pdev); + + platform_device_unregister(snd_dev); + + detach_gpio_amp(pdata); + clk_put(xtal_clk); + return 0; +} +EXPORT_SYMBOL_GPL(simtec_audio_remove); + +MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h new file mode 100644 index 000000000000..2714203af161 --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_simtec.h @@ -0,0 +1,22 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.h + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +extern void simtec_audio_init(struct snd_soc_codec *codec); + +extern int simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev); + +extern int simtec_audio_remove(struct platform_device *pdev); + +#ifdef CONFIG_PM +extern struct dev_pm_ops simtec_audio_pmops; +#define simtec_audio_pm &simtec_audio_pmops +#else +#define simtec_audio_pm NULL +#endif diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c new file mode 100644 index 000000000000..8346bd96eaf5 --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c @@ -0,0 +1,153 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic3x.h" + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_LINE("GSM Out", NULL), + SND_SOC_DAPM_LINE("GSM In", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_LINE("ZV", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */ + + { "Headphone Jack", NULL, "HPLOUT" }, + { "Headphone Jack", NULL, "HPLCOM" }, + { "Headphone Jack", NULL, "HPROUT" }, + { "Headphone Jack", NULL, "HPRCOM" }, + + /* ZV connected to Line1 */ + + { "LINE1L", NULL, "ZV" }, + { "LINE1R", NULL, "ZV" }, + + /* Line In connected to Line2 */ + + { "LINE2L", NULL, "Line In" }, + { "LINE2R", NULL, "Line In" }, + + /* Microphone connected to MIC3R and MIC_BIAS */ + + { "MIC3L", NULL, "Mic Jack" }, + + /* GSM connected to MONO_LOUT and MIC3L (in) */ + + { "GSM Out", NULL, "MONO_LOUT" }, + { "MIC3L", NULL, "GSM In" }, + + /* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are + * not using the DAPM to power it up and down as there it makes + * a click when powering up. */ +}; + +/** + * simtec_hermes_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_hermes_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct aic3x_setup_data codec_setup = { +}; + +static struct snd_soc_dai_link simtec_dai_aic33 = { + .name = "tlv320aic33", + .stream_name = "TLV320AIC33", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &aic3x_dai, + .init = simtec_hermes_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic33 = { + .name = "Simtec-Hermes", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic33, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic33 = { + .card = &snd_soc_machine_simtec_aic33, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &codec_setup, +}; + +static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) +{ + dev_info(&pd->dev, "probing....\n"); + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33); +} + +static struct platform_driver simtec_audio_hermes_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-hermes-snd", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_hermes_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); + +static int __init simtec_hermes_modinit(void) +{ + return platform_driver_register(&simtec_audio_hermes_platdrv); +} + +static void __exit simtec_hermes_modexit(void) +{ + platform_driver_unregister(&simtec_audio_hermes_platdrv); +} + +module_init(simtec_hermes_modinit); +module_exit(simtec_hermes_modexit); + +MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c new file mode 100644 index 000000000000..25797e096175 --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c @@ -0,0 +1,137 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic23.h" + +/* supported machines: + * + * Machine Connections AMP + * ------- ----------- --- + * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) + * VR1000 HPOUT, LIN None + * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) + */ + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + { "Headphone Jack", NULL, "LHPOUT"}, + { "Headphone Jack", NULL, "RHPOUT"}, + + { "Line Out", NULL, "LOUT" }, + { "Line Out", NULL, "ROUT" }, + + { "LLINEIN", NULL, "Line In"}, + { "RLINEIN", NULL, "Line In"}, + + { "MICIN", NULL, "Mic Jack"}, +}; + +/** + * simtec_tlv320aic23_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link simtec_dai_aic23 = { + .name = "tlv320aic23", + .stream_name = "TLV320AIC23", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &tlv320aic23_dai, + .init = simtec_tlv320aic23_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic23 = { + .name = "Simtec", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic23, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic23 = { + .card = &snd_soc_machine_simtec_aic23, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) +{ + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23); +} + +static struct platform_driver simtec_audio_tlv320aic23_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-tlv320aic23", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_tlv320aic23_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); + +static int __init simtec_tlv320aic23_modinit(void) +{ + return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); +} + +static void __exit simtec_tlv320aic23_modexit(void) +{ + platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); +} + +module_init(simtec_tlv320aic23_modinit); +module_exit(simtec_tlv320aic23_modexit); + +MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c index 8e79a416db57..c215d32d6322 100644 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -67,7 +67,7 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) { int ret = 0; #ifdef ENFORCE_RATES - struct snd_pcm_runtime *runtime = substream->runtime;; + struct snd_pcm_runtime *runtime = substream->runtime; #endif mutex_lock(&clk_lock); |