Hi, On Tue, Jul 20, 2010 at 2:28 PM, Chanwoo Choi <cw00.choi@xxxxxxxxxxx> wrote: > This patch add sound support for the Aquila board based on S5PC110. > > The Aquila board is based on Samsung SoC(S5PC110) and include > WM8994 codec over I2S to support sound. This uses the I2Sv4 driver > compatible with I2Sv5 to run sound. > > The kind of jack is below states : > * SND_JACK_HEADPHONE > * SND_JACK_HEADSET > * SND_JACK_MECHANICAL > : When TV-OUT cable is inserted on Aquila board, > the TV-OUT cable isn't connected to television. > * SND_JACK_AVOUT > : When TV-OUT cable is inserted on Aquila board, > the TV-OUT cable is connected to television. > > Signed-off-by: Chanwoo Choi <cw00.choi@xxxxxxxxxxx> > Signed-off-by: Joonyoung Shim <jy0922.shim@xxxxxxxxxxx> > Signed-off-by: Kyungmin Park <kyungmin.park@xxxxxxxxxxx> > --- > sound/soc/s3c24xx/Kconfig | 9 + > sound/soc/s3c24xx/Makefile | 2 + > sound/soc/s3c24xx/aquila_wm8994.c | 300 +++++++++++++++++++++++++++++++++++++ > 3 files changed, 311 insertions(+), 0 deletions(-) > create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c > > diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig > index 213963a..c09b402 100644 > --- a/sound/soc/s3c24xx/Kconfig > +++ b/sound/soc/s3c24xx/Kconfig > @@ -131,3 +131,12 @@ config SND_S3C64XX_SOC_SMARTQ > depends on SND_S3C24XX_SOC && MACH_SMARTQ > select SND_S3C64XX_SOC_I2S > select SND_SOC_WM8750 > + > +config SND_S5PC110_SOC_AQUILA_WM8994 > + tristate "SoC I2S Audio support for AQUILA - WM8994" > + depends on SND_S3C24XX_SOC && MACH_AQUILA > + select SND_S3C64XX_SOC_I2S_V4 > + select SND_SOC_WM8994 > + help > + Say Y if you want to add support for SoC audio on aquila > + with the WM8994. > diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile > index 50172c3..02dd12c 100644 > --- a/sound/soc/s3c24xx/Makefile > +++ b/sound/soc/s3c24xx/Makefile > @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o > snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o > snd-soc-smdk-wm9713-objs := smdk_wm9713.o > snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o > +snd-soc-aquila-wm8994-objs := aquila_wm8994.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 > @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 > obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o > obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o > obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o > +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o > diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c > new file mode 100644 > index 0000000..6ff5068 > --- /dev/null > +++ b/sound/soc/s3c24xx/aquila_wm8994.c > @@ -0,0 +1,300 @@ > +/* > + * aquila_wm8994.c > + * > + * Copyright (C) 2010 Samsung Electronics Co.Ltd > + * Author: Chanwoo Choi <cw00.choi@xxxxxxxxxxx> > + * > + * 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/io.h> > +#include <linux/platform_device.h> > +#include <sound/soc.h> > +#include <sound/soc-dapm.h> > +#include <sound/jack.h> > +#include <asm/mach-types.h> > +#include <mach/gpio.h> > +#include <mach/regs-clock.h> > + > +#include <linux/mfd/wm8994/core.h> > +#include <linux/mfd/wm8994/registers.h> > +#include "../codecs/wm8994.h" > +#include "s3c-dma.h" > +#include "s3c64xx-i2s.h" > + > +#define WM8994_DAI_HIFI 0 > +#define WM8994_DAI_VOICE 1 > + > +static struct snd_soc_card aquila; > +static struct platform_device *aquila_snd_device; > + > +/* 3.5 pie jack */ > +static struct snd_soc_jack jack; > + > +/* 3.5 pie jack detection DAPM pins */ > +static struct snd_soc_jack_pin jack_pins[] = { > + { > + .pin = "Headset Mic", > + .mask = SND_JACK_MICROPHONE, > + }, { > + .pin = "Headset Stereophone", > + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | > + SND_JACK_AVOUT, > + }, > +}; > + > +/* 3.5 pie jack detection gpios */ > +static struct snd_soc_jack_gpio jack_gpios[] = { > + { > + .gpio = S5PV210_GPH0(6), > + .name = "DET_3.5", > + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | > + SND_JACK_AVOUT, > + .debounce_time = 200, > + }, > +}; > + > +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { > + SND_SOC_DAPM_SPK("Ext Spk", NULL), > + SND_SOC_DAPM_SPK("Ext Rcv", NULL), > + SND_SOC_DAPM_HP("Headset Stereophone", NULL), > + SND_SOC_DAPM_MIC("Headset Mic", NULL), > + SND_SOC_DAPM_MIC("Main Mic", NULL), > + SND_SOC_DAPM_MIC("2nd Mic", NULL), > + SND_SOC_DAPM_LINE("Radio In", NULL), > +}; > + > +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { > + {"Ext Spk", NULL, "SPKOUTLP"}, > + {"Ext Spk", NULL, "SPKOUTLN"}, > + > + {"Ext Rcv", NULL, "HPOUT2N"}, > + {"Ext Rcv", NULL, "HPOUT2P"}, > + > + {"Headset Stereophone", NULL, "HPOUT1L"}, > + {"Headset Stereophone", NULL, "HPOUT1R"}, > + > + {"IN1RN", NULL, "Headset Mic"}, > + {"IN1RP", NULL, "Headset Mic"}, > + > + {"IN1RN", NULL, "2nd Mic"}, > + {"IN1RP", NULL, "2nd Mic"}, > + > + {"IN1LN", NULL, "Main Mic"}, > + {"IN1LP", NULL, "Main Mic"}, > + > + {"IN2LN", NULL, "Radio In"}, > + {"IN2RN", NULL, "Radio In"}, > +}; > + > +static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd) > +{ > + struct snd_soc_codec *codec = rtd->codec; > + int ret; > + > + /* add aquila specific widgets */ > + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, > + ARRAY_SIZE(aquila_dapm_widgets)); > + > + /* set up aquila specific audio routes */ > + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, > + ARRAY_SIZE(aquila_dapm_routes)); > + > + /* set endpoints to not connected */ > + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); > + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); > + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); > + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); > + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); > + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); > + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); > + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); > + > + snd_soc_dapm_sync(codec); > + > + /* Headset jack detection */ > + ret = snd_soc_jack_new(&aquila, "Headset Jack", > + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, > + &jack); > + if (ret) > + return ret; > + > + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); > + if (ret) > + return ret; > + > + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); > + if (ret) > + return ret; > + > + return 0; > +} > + > +static int aquila_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->codec_dai; > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + unsigned int pll_out = 24000000; > + int ret = 0; > + > + /* set the 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 cpu system clock */ > + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0); I think that this is not good to write hard-coded parameters. > + if (ret < 0) > + return ret; > + > + /* 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 the codec FLL */ > + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, > + params_rate(params) * 256); > + if (ret < 0) > + return ret; > + > + /* set the codec system clock */ > + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, > + params_rate(params) * 256, SND_SOC_CLOCK_IN); > + if (ret < 0) > + return ret; > + > + return 0; > +} > + > +static struct snd_soc_ops aquila_hifi_ops = { > + .hw_params = aquila_hifi_hw_params, > +}; > + > +static int aquila_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->codec_dai; > + unsigned int pll_out = 24000000; > + int ret = 0; > + > + if (params_rate(params) != 8000) > + return -EINVAL; > + > + /* set codec DAI configuration */ > + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | > + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); > + if (ret < 0) > + return ret; > + > + /* set the codec FLL */ > + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, > + params_rate(params) * 256); > + if (ret < 0) > + return ret; > + > + /* set the codec system clock */ > + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, > + params_rate(params) * 256, SND_SOC_CLOCK_IN); > + if (ret < 0) > + return ret; > + > + return 0; > +} > + > +static struct snd_soc_dai_driver voice_dai = { > + .name = "Voice", > + .playback = { > + .channels_min = 1, > + .channels_max = 2, > + .rates = SNDRV_PCM_RATE_8000, > + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, > + .capture = { > + .channels_min = 1, > + .channels_max = 2, > + .rates = SNDRV_PCM_RATE_8000, > + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, > +}; > + > +static struct snd_soc_ops aquila_voice_ops = { > + .hw_params = aquila_voice_hw_params, > +}; > + > +static struct snd_soc_dai_link aquila_dai[] = { > +{ > + .name = "WM8994", > + .stream_name = "WM8994 HiFi", > + .cpu_dai_drv = &s3c64xx_i2s_v4_dai, > + .codec_dai_drv = &wm8994_dai[WM8994_DAI_HIFI], > + .codec_dai_id = WM8994_DAI_HIFI, > + .platform_drv = &s3c24xx_soc_platform, > + .codec_drv = &soc_codec_dev_wm8994, > + .init = aquila_wm8994_init, > + .ops = &aquila_hifi_ops, > +}, { > + .name = "WM8994 Voice", > + .stream_name = "Voice", > + .cpu_dai_drv = &voice_dai, > + .codec_dai_drv = &wm8994_dai[WM8994_DAI_VOICE], > + .codec_dai_id = WM8994_DAI_VOICE, > + .platform_drv = &s3c24xx_soc_platform, > + .codec_drv = &soc_codec_dev_wm8994, > + .ops = &aquila_voice_ops, > +}, > +}; > + > +static struct snd_soc_card aquila = { > + .name = "aquila", > + .dai_link = aquila_dai, > + .num_links = ARRAY_SIZE(aquila_dai), > +}; > + > +static int __init aquila_init(void) > +{ > + int ret; > + > + if (!machine_is_aquila()) > + return -ENODEV; > + > + aquila_snd_device = platform_device_alloc("soc-audio", 0); Really need this allocate as a '0' not "-1"? Is there any reason for? > + if (!aquila_snd_device) > + return -ENOMEM; > + > + /* register voice DAI here */ > + ret = snd_soc_register_dai(&aquila_snd_device->dev, > + 0, &voice_dai); > + if (ret) > + return ret; > + > + platform_set_drvdata(aquila_snd_device, &aquila); > + ret = platform_device_add(aquila_snd_device); > + > + if (ret) > + platform_device_put(aquila_snd_device); > + > + return ret; > +} > + > +static void __exit aquila_exit(void) > +{ > + platform_device_unregister(aquila_snd_device); > +} > + > +module_init(aquila_init); > +module_exit(aquila_exit); > + > +/* Module information */ > +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); > +MODULE_AUTHOR("Chanwoo Choi <cw00.choi@xxxxxxxxxxx>"); > +MODULE_LICENSE("GPL"); > -- > 1.6.3.3 > > > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel@xxxxxxxxxxxxxxxx > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel