Re: [PATCH v2 1/1] ASoC: Add HP iPAQ H1940 support

[Marek Vasut <marek.vasut@gmail.com>]

Dne Ne 12. září 2010 14:18:47 Vasily Khoruzhick napsal(a):
> Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
> Tested-by: Arnaud Patard <arnaud.patard@rtp-net.org>

have my:
Reviewed-by: Marek Vasut <marek.vasut@gmail.com>
> ---
>  sound/soc/s3c24xx/Kconfig         |    8 +
>  sound/soc/s3c24xx/Makefile        |    2 +
>  sound/soc/s3c24xx/h1940_uda1380.c |  297
> +++++++++++++++++++++++++++++++++++++ 3 files changed, 307 insertions(+),
> 0 deletions(-)
>  create mode 100644 sound/soc/s3c24xx/h1940_uda1380.c
> 
> diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
> index 7d8235d..6b50509 100644
> --- a/sound/soc/s3c24xx/Kconfig
> +++ b/sound/soc/s3c24xx/Kconfig
> @@ -118,6 +118,14 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES
>  	select SND_SOC_TLV320AIC3X
>  	select SND_S3C24XX_SOC_SIMTEC
> 
> +config SND_S3C24XX_SOC_H1940_UDA1380
> +	tristate "Audio support for the HP iPAQ H1940"
> +	depends on SND_S3C24XX_SOC && ARCH_H1940
> +	select SND_S3C24XX_SOC_I2S
> +	select SND_SOC_UDA1380
> +	help
> +	  This driver provides audio support for HP iPAQ h1940 PDA.
> +
>  config SND_S3C24XX_SOC_RX1950_UDA1380
>  	tristate "Audio support for the HP iPAQ RX1950"
>  	depends on SND_S3C24XX_SOC && MACH_RX1950
> diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
> index dd412a9..33a7c68 100644
> --- a/sound/soc/s3c24xx/Makefile
> +++ b/sound/soc/s3c24xx/Makefile
> @@ -28,6 +28,7 @@ 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
>  snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o
> +snd-soc-h1940-uda1380-objs := h1940_uda1380.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
> @@ -44,6 +45,7 @@ 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-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) +=
> snd-soc-s3c24xx-simtec-tlv320aic23.o
> obj-$(CONFIG_SND_S3C24XX_SOC_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o
> +obj-$(CONFIG_SND_S3C24XX_SOC_H1940_UDA1380) += snd-soc-h1940-uda1380.o
> obj-$(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
> diff --git a/sound/soc/s3c24xx/h1940_uda1380.c
> b/sound/soc/s3c24xx/h1940_uda1380.c new file mode 100644
> index 0000000..5dbc0ea
> --- /dev/null
> +++ b/sound/soc/s3c24xx/h1940_uda1380.c
> @@ -0,0 +1,297 @@
> +/*
> + * h1940-uda1380.c  --  ALSA Soc Audio Layer
> + *
> + * Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
> + * Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com>
> + *
> + * Based on version from Arnaud Patard <arnaud.patard@rtp-net.org>
> + *
> + * 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/platform_device.h>
> +#include <linux/i2c.h>
> +#include <linux/gpio.h>
> +
> +#include <sound/soc.h>
> +#include <sound/soc-dapm.h>
> +#include <sound/uda1380.h>
> +#include <sound/jack.h>
> +
> +#include <plat/regs-iis.h>
> +
> +#include <mach/h1940-latch.h>
> +
> +#include <asm/mach-types.h>
> +
> +#include "s3c-dma.h"
> +#include "s3c24xx-i2s.h"
> +#include "../codecs/uda1380.h"
> +
> +static unsigned int rates[] = {
> +	11025,
> +	22050,
> +	44100,
> +};
> +
> +static struct snd_pcm_hw_constraint_list hw_rates = {
> +	.count = ARRAY_SIZE(rates),
> +	.list = rates,
> +	.mask = 0,
> +};
> +
> +static struct snd_soc_jack hp_jack;
> +
> +static struct snd_soc_jack_pin hp_jack_pins[] = {
> +	{
> +		.pin	= "Headphone Jack",
> +		.mask	= SND_JACK_HEADPHONE,
> +	},
> +	{
> +		.pin	= "Speaker",
> +		.mask	= SND_JACK_HEADPHONE,
> +		.invert	= 1,
> +	},
> +};
> +
> +static struct snd_soc_jack_gpio hp_jack_gpios[] = {
> +	{
> +		.gpio			= S3C2410_GPG(4),
> +		.name			= "hp-gpio",
> +		.report			= SND_JACK_HEADPHONE,
> +		.invert			= 1,
> +		.debounce_time		= 200,
> +	},
> +};
> +
> +static int h1940_startup(struct snd_pcm_substream *substream)
> +{
> +	struct snd_pcm_runtime *runtime = substream->runtime;
> +
> +	runtime->hw.rate_min = hw_rates.list[0];
> +	runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
> +	runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
> +
> +	return snd_pcm_hw_constraint_list(runtime, 0,
> +					SNDRV_PCM_HW_PARAM_RATE,
> +					&hw_rates);
> +}
> +
> +static int h1940_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 *cpu_dai = rtd->cpu_dai;
> +	struct snd_soc_dai *codec_dai = rtd->codec_dai;
> +	int div;
> +	int ret;
> +	unsigned int rate = params_rate(params);
> +
> +	switch (rate) {
> +	case 11025:
> +	case 22050:
> +	case 44100:
> +		div = s3c24xx_i2s_get_clockrate() / (384 * rate);
> +		if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
> +			div++;
> +		break;
> +	default:
> +		dev_err(&rtd->dev, "%s: rate %d is not supported\n",
> +			__func__, rate);
> +		return -EINVAL;
> +	}
> +
> +	/* set codec DAI configuration */
> +	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
> +		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
> +	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_CBS_CFS);
> +	if (ret < 0)
> +		return ret;
> +
> +	/* select clock source */
> +	ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
> +			SND_SOC_CLOCK_OUT);
> +	if (ret < 0)
> +		return ret;
> +
> +	/* set MCLK division for sample rate */
> +	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
> +		S3C2410_IISMOD_384FS);
> +	if (ret < 0)
> +		return ret;
> +
> +	/* set BCLK division for sample rate */
> +	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
> +		S3C2410_IISMOD_32FS);
> +	if (ret < 0)
> +		return ret;
> +
> +	/* set prescaler division for sample rate */
> +	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
> +		S3C24XX_PRESCALE(div, div));
> +	if (ret < 0)
> +		return ret;
> +
> +	return 0;
> +}
> +
> +static struct snd_soc_ops h1940_ops = {
> +	.startup	= h1940_startup,
> +	.hw_params	= h1940_hw_params,
> +};
> +
> +static int h1940_spk_power(struct snd_soc_dapm_widget *w,
> +				struct snd_kcontrol *kcontrol, int event)
> +{
> +	if (SND_SOC_DAPM_EVENT_ON(event))
> +		gpio_set_value(H1940_LATCH_AUDIO_POWER, 1);
> +	else
> +		gpio_set_value(H1940_LATCH_AUDIO_POWER, 0);
> +
> +	return 0;
> +}
> +
> +/* h1940 machine dapm widgets */
> +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
> +	SND_SOC_DAPM_HP("Headphone Jack", NULL),
> +	SND_SOC_DAPM_MIC("Mic Jack", NULL),
> +	SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
> +};
> +
> +/* h1940 machine audio_map */
> +static const struct snd_soc_dapm_route audio_map[] = {
> +	/* headphone connected to VOUTLHP, VOUTRHP */
> +	{"Headphone Jack", NULL, "VOUTLHP"},
> +	{"Headphone Jack", NULL, "VOUTRHP"},
> +
> +	/* ext speaker connected to VOUTL, VOUTR  */
> +	{"Speaker", NULL, "VOUTL"},
> +	{"Speaker", NULL, "VOUTR"},
> +
> +	/* mic is connected to VINM */
> +	{"VINM", NULL, "Mic Jack"},
> +};
> +
> +static struct platform_device *s3c24xx_snd_device;
> +
> +static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
> +{
> +	struct snd_soc_codec *codec = rtd->codec;
> +	int err;
> +
> +	/* Add h1940 specific widgets */
> +	err = snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
> +				  ARRAY_SIZE(uda1380_dapm_widgets));
> +	if (err)
> +		return err;
> +
> +	/* Set up h1940 specific audio path audio_mapnects */
> +	err = snd_soc_dapm_add_routes(codec, audio_map,
> +				      ARRAY_SIZE(audio_map));
> +	if (err)
> +		return err;
> +
> +	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
> +	snd_soc_dapm_enable_pin(codec, "Speaker");
> +	snd_soc_dapm_enable_pin(codec, "Mic Jack");
> +
> +	snd_soc_dapm_sync(codec);
> +
> +	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
> +		&hp_jack);
> +
> +	snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
> +		hp_jack_pins);
> +
> +	snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
> +		hp_jack_gpios);
> +
> +	return 0;
> +}
> +
> +/* s3c24xx digital audio interface glue - connects codec <--> CPU */
> +static struct snd_soc_dai_link h1940_uda1380_dai[] = {
> +	{
> +		.name		= "uda1380",
> +		.stream_name	= "UDA1380 Duplex",
> +		.cpu_dai_name	= "s3c24xx-iis",
> +		.codec_dai_name	= "uda1380-hifi",
> +		.init		= h1940_uda1380_init,
> +		.platform_name	= "s3c24xx-pcm-audio",
> +		.codec_name	= "uda1380-codec.0-001a",
> +		.ops		= &h1940_ops,
> +	},
> +};
> +
> +static struct snd_soc_card h1940_asoc = {
> +	.name = "h1940",
> +	.dai_link = h1940_uda1380_dai,
> +	.num_links = ARRAY_SIZE(h1940_uda1380_dai),
> +};
> +
> +static int __init h1940_init(void)
> +{
> +	int ret;
> +
> +	if (!machine_is_h1940())
> +		return -ENODEV;
> +
> +	/* configure some gpios */
> +	ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power");
> +	if (ret)
> +		goto err_out;
> +
> +	ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0);
> +	if (ret)
> +		goto err_gpio;
> +
> +	s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
> +	if (!s3c24xx_snd_device) {
> +		ret = -ENOMEM;
> +		goto err_gpio;
> +	}
> +
> +	platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc);
> +	ret = platform_device_add(s3c24xx_snd_device);
> +
> +	if (ret)
> +		goto err_plat;
> +
> +	return 0;
> +
> +err_plat:
> +	platform_device_put(s3c24xx_snd_device);
> +err_gpio:
> +	gpio_free(H1940_LATCH_AUDIO_POWER);
> +
> +err_out:
> +	return ret;
> +}
> +
> +static void __exit h1940_exit(void)
> +{
> +	platform_device_unregister(s3c24xx_snd_device);
> +	snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
> +		hp_jack_gpios);
> +	gpio_free(H1940_LATCH_AUDIO_POWER);
> +}
> +
> +module_init(h1940_init);
> +module_exit(h1940_exit);
> +
> +/* Module information */
> +MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
> +MODULE_DESCRIPTION("ALSA SoC H1940");
> +MODULE_LICENSE("GPL");
> +MODULE_ALIAS("platform:soc-audio");
_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel