sound/soc/samsung/h1940_uda1380.c - maze/linux - Git at Google

 /*
  * 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/types.h>
 #include <linux/gpio.h>
 #include <linux/module.h>

 #include <sound/soc.h>
 #include <sound/jack.h>

 #include "regs-iis.h"
 #include <asm/mach-types.h>

 #include <mach/gpio-samsung.h>
 #include "s3c24xx-i2s.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(S3C_GPIO_END + 9, 1);
 	else
 		gpio_set_value(S3C_GPIO_END + 9, 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;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;

 	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
 	snd_soc_dapm_enable_pin(dapm, "Speaker");
 	snd_soc_dapm_enable_pin(dapm, "Mic Jack");

 	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-iis",
 		.codec_name	= "uda1380-codec.0-001a",
 		.ops		= &h1940_ops,
 	},
 };

 static struct snd_soc_card h1940_asoc = {
 	.name = "h1940",
 	.owner = THIS_MODULE,
 	.dai_link = h1940_uda1380_dai,
 	.num_links = ARRAY_SIZE(h1940_uda1380_dai),

 	.dapm_widgets = uda1380_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
 	.dapm_routes = audio_map,
 	.num_dapm_routes = ARRAY_SIZE(audio_map),
 };

 static int __init h1940_init(void)
 {
 	int ret;

 	if (!machine_is_h1940())
 		return -ENODEV;

 	/* configure some gpios */
 	ret = gpio_request(S3C_GPIO_END + 9, "speaker-power");
 	if (ret)
 		goto err_out;

 	ret = gpio_direction_output(S3C_GPIO_END + 9, 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(S3C_GPIO_END + 9);

 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(S3C_GPIO_END + 9);
 }

 module_init(h1940_init);
 module_exit(h1940_exit);

 /* Module information */
 MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
 MODULE_DESCRIPTION("ALSA SoC H1940");
 MODULE_LICENSE("GPL");
	/*
	* 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/types.h>
	#include <linux/gpio.h>
	#include <linux/module.h>

	#include <sound/soc.h>
	#include <sound/jack.h>

	#include "regs-iis.h"
	#include <asm/mach-types.h>

	#include <mach/gpio-samsung.h>
	#include "s3c24xx-i2s.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(S3C_GPIO_END + 9, 1);
	else
	gpio_set_value(S3C_GPIO_END + 9, 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;
	struct snd_soc_dapm_context *dapm = &codec->dapm;
	int err;

	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
	snd_soc_dapm_enable_pin(dapm, "Speaker");
	snd_soc_dapm_enable_pin(dapm, "Mic Jack");

	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-iis",
	.codec_name = "uda1380-codec.0-001a",
	.ops = &h1940_ops,
	},
	};

	static struct snd_soc_card h1940_asoc = {
	.name = "h1940",
	.owner = THIS_MODULE,
	.dai_link = h1940_uda1380_dai,
	.num_links = ARRAY_SIZE(h1940_uda1380_dai),

	.dapm_widgets = uda1380_dapm_widgets,
	.num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
	.dapm_routes = audio_map,
	.num_dapm_routes = ARRAY_SIZE(audio_map),
	};

	static int __init h1940_init(void)
	{
	int ret;

	if (!machine_is_h1940())
	return -ENODEV;

	/* configure some gpios */
	ret = gpio_request(S3C_GPIO_END + 9, "speaker-power");
	if (ret)
	goto err_out;

	ret = gpio_direction_output(S3C_GPIO_END + 9, 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(S3C_GPIO_END + 9);

	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(S3C_GPIO_END + 9);
	}

	module_init(h1940_init);
	module_exit(h1940_exit);

	/* Module information */
	MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
	MODULE_DESCRIPTION("ALSA SoC H1940");
	MODULE_LICENSE("GPL");