Seungwhan Youn wrote:
> 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.
Ok, I will modify as below codes,
- ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, 0);
+ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK,
+ 0, SND_SOC_CLOCK_IN);
>
>> + 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?
No, there isn't special reason. I will modify id from '0' to '-1'.
- aquila_snd_device = platform_device_alloc("soc-audio", 0);
+ aquila_snd_device = platform_device_alloc("soc-audio", -1);
>> + 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
>>
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