[alsa-devel] [PATCH 1/3] ASoC: multi-component - Add Aquila sound driver
Seungwhan Youn
claude.youn at gmail.com
Tue Jul 20 08:07:55 CEST 2010
Hi,
On Tue, Jul 20, 2010 at 2:28 PM, Chanwoo Choi <cw00.choi at samsung.com> 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 at samsung.com>
> Signed-off-by: Joonyoung Shim <jy0922.shim at samsung.com>
> Signed-off-by: Kyungmin Park <kyungmin.park at samsung.com>
> ---
> 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 at samsung.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.
> + *
> + */
> +
> +#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 at samsung.com>");
> +MODULE_LICENSE("GPL");
> --
> 1.6.3.3
>
>
>
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