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

#include <plat/regs-iis.h>
#include <mach/h1940-latch.h>
#include <asm/mach-types.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(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;
	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	= "samsung-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),

	.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(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");
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 VK	17	#include <linux/gpio.h>
	18
	19	#include <sound/soc.h>
1957668b VK	20	#include <sound/jack.h>
	21
	22	#include <plat/regs-iis.h>
1957668b	23	#include <mach/h1940-latch.h>
1957668b VK	24	#include <asm/mach-types.h>
1957668b VK	25
1957668b	26	#include "s3c24xx-i2s.h"
1957668b VK	27
	28	static unsigned int rates[] = {
	29	11025,
	30	22050,
	31	44100,
	32	};
	33
	34	static struct snd_pcm_hw_constraint_list hw_rates = {
	35	.count = ARRAY_SIZE(rates),
	36	.list = rates,
	37	.mask = 0,
	38	};
	39
	40	static struct snd_soc_jack hp_jack;
	41
	42	static struct snd_soc_jack_pin hp_jack_pins[] = {
	43	{
	44	.pin = "Headphone Jack",
	45	.mask = SND_JACK_HEADPHONE,
	46	},
	47	{
	48	.pin = "Speaker",
	49	.mask = SND_JACK_HEADPHONE,
	50	.invert = 1,
	51	},
	52	};
	53
	54	static struct snd_soc_jack_gpio hp_jack_gpios[] = {
	55	{
	56	.gpio = S3C2410_GPG(4),
	57	.name = "hp-gpio",
	58	.report = SND_JACK_HEADPHONE,
	59	.invert = 1,
	60	.debounce_time = 200,
	61	},
	62	};
	63
	64	static int h1940_startup(struct snd_pcm_substream *substream)
	65	{
	66	struct snd_pcm_runtime *runtime = substream->runtime;
	67
	68	runtime->hw.rate_min = hw_rates.list[0];
	69	runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
	70	runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
	71
	72	return snd_pcm_hw_constraint_list(runtime, 0,
	73	SNDRV_PCM_HW_PARAM_RATE,
	74	&hw_rates);
	75	}
	76
	77	static int h1940_hw_params(struct snd_pcm_substream *substream,
	78	struct snd_pcm_hw_params *params)
	79	{
	80	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	81	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
	82	struct snd_soc_dai *codec_dai = rtd->codec_dai;
	83	int div;
	84	int ret;
	85	unsigned int rate = params_rate(params);
	86
	87	switch (rate) {
	88	case 11025:
	89	case 22050:
	90	case 44100:
91	div = s3c24xx_i2s_get_clockrate() / (384 * rate);
92	if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
93	div++;
94	break;
95	default:
96	dev_err(&rtd->dev, "%s: rate %d is not supported\n",
97	__func__, rate);
98	return -EINVAL;
99	}
100
101	/* set codec DAI configuration */
102	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S \|
103	SND_SOC_DAIFMT_NB_NF \| SND_SOC_DAIFMT_CBS_CFS);
104	if (ret < 0)
105	return ret;
106
107	/* set cpu DAI configuration */
108	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S \|
109	SND_SOC_DAIFMT_NB_NF \| SND_SOC_DAIFMT_CBS_CFS);
110	if (ret < 0)
111	return ret;
112
113	/* select clock source */
114	ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
115	SND_SOC_CLOCK_OUT);
116	if (ret < 0)
117	return ret;
118
119	/* set MCLK division for sample rate */
120	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
121	S3C2410_IISMOD_384FS);
122	if (ret < 0)
123	return ret;
124
125	/* set BCLK division for sample rate */
126	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
127	S3C2410_IISMOD_32FS);
128	if (ret < 0)
129	return ret;
130
131	/* set prescaler division for sample rate */
132	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
133	S3C24XX_PRESCALE(div, div));
134	if (ret < 0)
135	return ret;
136
137	return 0;
138	}
139
140	static struct snd_soc_ops h1940_ops = {
141	.startup = h1940_startup,
142	.hw_params = h1940_hw_params,
143	};
144
145	static int h1940_spk_power(struct snd_soc_dapm_widget *w,
146	struct snd_kcontrol *kcontrol, int event)
147	{
148	if (SND_SOC_DAPM_EVENT_ON(event))
149	gpio_set_value(H1940_LATCH_AUDIO_POWER, 1);
150	else
151	gpio_set_value(H1940_LATCH_AUDIO_POWER, 0);
152
153	return 0;
154	}
155
156	/* h1940 machine dapm widgets */
157	static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
158	SND_SOC_DAPM_HP("Headphone Jack", NULL),
159	SND_SOC_DAPM_MIC("Mic Jack", NULL),
160	SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
161	};
162
163	/* h1940 machine audio_map */
164	static const struct snd_soc_dapm_route audio_map[] = {
165	/* headphone connected to VOUTLHP, VOUTRHP */
166	{"Headphone Jack", NULL, "VOUTLHP"},
167	{"Headphone Jack", NULL, "VOUTRHP"},
168
169	/* ext speaker connected to VOUTL, VOUTR */
170	{"Speaker", NULL, "VOUTL"},
171	{"Speaker", NULL, "VOUTR"},
172
173	/* mic is connected to VINM */
174	{"VINM", NULL, "Mic Jack"},
175	};
176
177	static struct platform_device *s3c24xx_snd_device;
178
179	static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
180	{
181	struct snd_soc_codec *codec = rtd->codec;
182	struct snd_soc_dapm_context *dapm = &codec->dapm;
183	int err;
184
1957668b VK	185	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
	186	snd_soc_dapm_enable_pin(dapm, "Speaker");
	187	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
	188
1957668b VK	189	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
	190	&hp_jack);
	191
	192	snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
	193	hp_jack_pins);
	194
	195	snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
	196	hp_jack_gpios);
	197
	198	return 0;
	199	}
	200
	201	/* s3c24xx digital audio interface glue - connects codec <--> CPU */
	202	static struct snd_soc_dai_link h1940_uda1380_dai[] = {
	203	{
	204	.name = "uda1380",
	205	.stream_name = "UDA1380 Duplex",
	206	.cpu_dai_name = "s3c24xx-iis",
	207	.codec_dai_name = "uda1380-hifi",
	208	.init = h1940_uda1380_init,
	209	.platform_name = "samsung-audio",
	210	.codec_name = "uda1380-codec.0-001a",
	211	.ops = &h1940_ops,
	212	},
	213	};
	214
	215	static struct snd_soc_card h1940_asoc = {
	216	.name = "h1940",
	217	.dai_link = h1940_uda1380_dai,
	218	.num_links = ARRAY_SIZE(h1940_uda1380_dai),
6119d016 MB	219
	220	.dapm_widgets = uda1380_dapm_widgets,
	221	.num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
	222	.dapm_routes = audio_map,
	223	.num_dapm_routes = ARRAY_SIZE(audio_map),
1957668b VK	224	};
	225
	226	static int __init h1940_init(void)
	227	{
	228	int ret;
	229
	230	if (!machine_is_h1940())
	231	return -ENODEV;
	232
	233	/* configure some gpios */
	234	ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power");
	235	if (ret)
	236	goto err_out;
	237
	238	ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0);
	239	if (ret)
	240	goto err_gpio;
	241
	242	s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
	243	if (!s3c24xx_snd_device) {
	244	ret = -ENOMEM;
	245	goto err_gpio;
	246	}
	247
	248	platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc);
	249	ret = platform_device_add(s3c24xx_snd_device);
	250
	251	if (ret)
	252	goto err_plat;
	253
	254	return 0;
	255
	256	err_plat:
	257	platform_device_put(s3c24xx_snd_device);
	258	err_gpio:
	259	gpio_free(H1940_LATCH_AUDIO_POWER);
	260
	261	err_out:
	262	return ret;
	263	}
	264
	265	static void __exit h1940_exit(void)
	266	{
	267	platform_device_unregister(s3c24xx_snd_device);
	268	snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
	269	hp_jack_gpios);
	270	gpio_free(H1940_LATCH_AUDIO_POWER);
	271	}
	272
	273	module_init(h1940_init);
	274	module_exit(h1940_exit);
	275
	276	/* Module information */
	277	MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
	278	MODULE_DESCRIPTION("ALSA SoC H1940");
	279	MODULE_LICENSE("GPL");