[deliverable/linux.git] / sound / soc / samsung / h1940_uda1380.c

/*
 * 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;

	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;
	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;
	}

	/* 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)
{
	snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE,
		&hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));

	snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
		hp_jack_gpios);

	return 0;
}

static int h1940_uda1380_card_remove(struct snd_soc_card *card)
{
	snd_soc_jack_free_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",
		.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
				  SND_SOC_DAIFMT_CBS_CFS,
		.ops		= &h1940_ops,
	},
};

static struct snd_soc_card h1940_asoc = {
	.name = "h1940",
	.owner = THIS_MODULE,
	.remove = h1940_uda1380_card_remove,
	.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);
	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");
Commit	Line	Data
1957668b VK	1	/*
	2	* h1940-uda1380.c -- ALSA Soc Audio Layer
	3	*
	4	* Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
	5	* Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com>
	6	*
	7	* Based on version from Arnaud Patard <arnaud.patard@rtp-net.org>
	8	*
	9	* This program is free software; you can redistribute it and/or modify it
	10	* under the terms of the GNU General Public License as published by the
	11	* Free Software Foundation; either version 2 of the License, or (at your
	12	* option) any later version.
	13	*
	14	*/
	15
fd049755	16	#include <linux/types.h>
1957668b	17	#include <linux/gpio.h>
da155d5b	18	#include <linux/module.h>
1957668b VK	19
1957668b VK	20	#include <sound/soc.h>
1957668b VK	21	#include <sound/jack.h>
1957668b VK	22
5d229ce5	23	#include "regs-iis.h"
1957668b VK	24	#include <asm/mach-types.h>
1957668b VK	25
abffae64	26	#include <mach/gpio-samsung.h>
1957668b	27	#include "s3c24xx-i2s.h"
1957668b VK	28
	29	static unsigned int rates[] = {
	30	11025,
	31	22050,
	32	44100,
	33	};
	34
	35	static struct snd_pcm_hw_constraint_list hw_rates = {
	36	.count = ARRAY_SIZE(rates),
	37	.list = rates,
	38	.mask = 0,
	39	};
	40
	41	static struct snd_soc_jack hp_jack;
	42
	43	static struct snd_soc_jack_pin hp_jack_pins[] = {
	44	{
	45	.pin = "Headphone Jack",
	46	.mask = SND_JACK_HEADPHONE,
	47	},
	48	{
	49	.pin = "Speaker",
	50	.mask = SND_JACK_HEADPHONE,
	51	.invert = 1,
	52	},
	53	};
	54
	55	static struct snd_soc_jack_gpio hp_jack_gpios[] = {
	56	{
	57	.gpio = S3C2410_GPG(4),
	58	.name = "hp-gpio",
	59	.report = SND_JACK_HEADPHONE,
	60	.invert = 1,
	61	.debounce_time = 200,
	62	},
	63	};
	64
	65	static int h1940_startup(struct snd_pcm_substream *substream)
	66	{
	67	struct snd_pcm_runtime *runtime = substream->runtime;
	68
1957668b VK	69	return snd_pcm_hw_constraint_list(runtime, 0,
	70	SNDRV_PCM_HW_PARAM_RATE,
	71	&hw_rates);
	72	}
	73
	74	static int h1940_hw_params(struct snd_pcm_substream *substream,
	75	struct snd_pcm_hw_params *params)
	76	{
	77	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	78	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
1957668b VK	79	int div;
	80	int ret;
	81	unsigned int rate = params_rate(params);
	82
	83	switch (rate) {
	84	case 11025:
	85	case 22050:
	86	case 44100:
	87	div = s3c24xx_i2s_get_clockrate() / (384 * rate);
	88	if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
	89	div++;
	90	break;
	91	default:
2046d453	92	dev_err(rtd->dev, "%s: rate %d is not supported\n",
1957668b VK	93	__func__, rate);
	94	return -EINVAL;
	95	}
	96
1957668b VK	97	/* select clock source */
	98	ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
	99	SND_SOC_CLOCK_OUT);
	100	if (ret < 0)
	101	return ret;
	102
	103	/* set MCLK division for sample rate */
	104	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
	105	S3C2410_IISMOD_384FS);
	106	if (ret < 0)
	107	return ret;
	108
	109	/* set BCLK division for sample rate */
	110	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
	111	S3C2410_IISMOD_32FS);
	112	if (ret < 0)
	113	return ret;
	114
	115	/* set prescaler division for sample rate */
	116	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
	117	S3C24XX_PRESCALE(div, div));
	118	if (ret < 0)
	119	return ret;
	120
	121	return 0;
	122	}
	123
	124	static struct snd_soc_ops h1940_ops = {
	125	.startup = h1940_startup,
	126	.hw_params = h1940_hw_params,
	127	};
	128
	129	static int h1940_spk_power(struct snd_soc_dapm_widget *w,
	130	struct snd_kcontrol *kcontrol, int event)
	131	{
	132	if (SND_SOC_DAPM_EVENT_ON(event))
232910d6	133	gpio_set_value(S3C_GPIO_END + 9, 1);
1957668b	134	else
232910d6	135	gpio_set_value(S3C_GPIO_END + 9, 0);
1957668b VK	136
	137	return 0;
	138	}
	139
	140	/* h1940 machine dapm widgets */
	141	static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
	142	SND_SOC_DAPM_HP("Headphone Jack", NULL),
	143	SND_SOC_DAPM_MIC("Mic Jack", NULL),
	144	SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
	145	};
	146
	147	/* h1940 machine audio_map */
	148	static const struct snd_soc_dapm_route audio_map[] = {
	149	/* headphone connected to VOUTLHP, VOUTRHP */
	150	{"Headphone Jack", NULL, "VOUTLHP"},
	151	{"Headphone Jack", NULL, "VOUTRHP"},
	152
	153	/* ext speaker connected to VOUTL, VOUTR */
	154	{"Speaker", NULL, "VOUTL"},
	155	{"Speaker", NULL, "VOUTR"},
	156
	157	/* mic is connected to VINM */
	158	{"VINM", NULL, "Mic Jack"},
	159	};
	160
	161	static struct platform_device *s3c24xx_snd_device;
	162
	163	static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
	164	{
dfe11f28 LPC	165	snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE,
dfe11f28 LPC	166	&hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
1957668b VK	167
	168	snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
	169	hp_jack_gpios);
	170
	171	return 0;
	172	}
	173
16088cb6	174	static int h1940_uda1380_card_remove(struct snd_soc_card *card)
e1d4d3c8 SW	175	{
	176	snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
	177	hp_jack_gpios);
	178
	179	return 0;
	180	}
	181
1957668b VK	182	/* s3c24xx digital audio interface glue - connects codec <--> CPU */
	183	static struct snd_soc_dai_link h1940_uda1380_dai[] = {
	184	{
	185	.name = "uda1380",
	186	.stream_name = "UDA1380 Duplex",
	187	.cpu_dai_name = "s3c24xx-iis",
	188	.codec_dai_name = "uda1380-hifi",
	189	.init = h1940_uda1380_init,
a08485d8	190	.platform_name = "s3c24xx-iis",
1957668b	191	.codec_name = "uda1380-codec.0-001a",
5cc10b9b LPC	192	.dai_fmt = SND_SOC_DAIFMT_I2S \| SND_SOC_DAIFMT_NB_NF \|
5cc10b9b LPC	193	SND_SOC_DAIFMT_CBS_CFS,
1957668b VK	194	.ops = &h1940_ops,
	195	},
	196	};
	197
	198	static struct snd_soc_card h1940_asoc = {
	199	.name = "h1940",
095d79dc	200	.owner = THIS_MODULE,
e1d4d3c8	201	.remove = h1940_uda1380_card_remove,
1957668b VK	202	.dai_link = h1940_uda1380_dai,
1957668b VK	203	.num_links = ARRAY_SIZE(h1940_uda1380_dai),
6119d016 MB	204
	205	.dapm_widgets = uda1380_dapm_widgets,
	206	.num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
	207	.dapm_routes = audio_map,
	208	.num_dapm_routes = ARRAY_SIZE(audio_map),
1957668b VK	209	};
	210
	211	static int __init h1940_init(void)
	212	{
	213	int ret;
	214
	215	if (!machine_is_h1940())
	216	return -ENODEV;
	217
	218	/* configure some gpios */
232910d6	219	ret = gpio_request(S3C_GPIO_END + 9, "speaker-power");
1957668b VK	220	if (ret)
	221	goto err_out;
	222
232910d6	223	ret = gpio_direction_output(S3C_GPIO_END + 9, 0);
1957668b VK	224	if (ret)
	225	goto err_gpio;
	226
	227	s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
	228	if (!s3c24xx_snd_device) {
	229	ret = -ENOMEM;
	230	goto err_gpio;
	231	}
	232
	233	platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc);
	234	ret = platform_device_add(s3c24xx_snd_device);
	235
	236	if (ret)
	237	goto err_plat;
	238
	239	return 0;
	240
	241	err_plat:
	242	platform_device_put(s3c24xx_snd_device);
	243	err_gpio:
232910d6	244	gpio_free(S3C_GPIO_END + 9);
1957668b VK	245
	246	err_out:
	247	return ret;
	248	}
	249
	250	static void __exit h1940_exit(void)
	251	{
	252	platform_device_unregister(s3c24xx_snd_device);
232910d6	253	gpio_free(S3C_GPIO_END + 9);
1957668b VK	254	}
	255
	256	module_init(h1940_init);
	257	module_exit(h1940_exit);
	258
	259	/* Module information */
	260	MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
	261	MODULE_DESCRIPTION("ALSA SoC H1940");
	262	MODULE_LICENSE("GPL");