/*
 * ak4642.c  --  AK4642/AK4643 ALSA Soc Audio driver
 *
 * Copyright (C) 2009 Renesas Solutions Corp.
 * Kuninori Morimoto <morimoto.kuninori@renesas.com>
 *
 * Based on wm8731.c by Richard Purdie
 * Based on ak4535.c by Richard Purdie
 * Based on wm8753.c by Liam Girdwood
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License version 2 as
 * published by the Free Software Foundation.
 */

/* ** CAUTION **
 *
 * This is very simple driver.
 * It can use headphone output / stereo input only
 *
 * AK4642 is tested.
 * AK4643 is tested.
 * AK4648 is tested.
 */

#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/slab.h>
#include <linux/of_device.h>
#include <linux/module.h>
#include <linux/regmap.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>

#define PW_MGMT1	0x00
#define PW_MGMT2	0x01
#define SG_SL1		0x02
#define SG_SL2		0x03
#define MD_CTL1		0x04
#define MD_CTL2		0x05
#define TIMER		0x06
#define ALC_CTL1	0x07
#define ALC_CTL2	0x08
#define L_IVC		0x09
#define L_DVC		0x0a
#define ALC_CTL3	0x0b
#define R_IVC		0x0c
#define R_DVC		0x0d
#define MD_CTL3		0x0e
#define MD_CTL4		0x0f
#define PW_MGMT3	0x10
#define DF_S		0x11
#define FIL3_0		0x12
#define FIL3_1		0x13
#define FIL3_2		0x14
#define FIL3_3		0x15
#define EQ_0		0x16
#define EQ_1		0x17
#define EQ_2		0x18
#define EQ_3		0x19
#define EQ_4		0x1a
#define EQ_5		0x1b
#define FIL1_0		0x1c
#define FIL1_1		0x1d
#define FIL1_2		0x1e
#define FIL1_3		0x1f
#define PW_MGMT4	0x20
#define MD_CTL5		0x21
#define LO_MS		0x22
#define HP_MS		0x23
#define SPK_MS		0x24

/* PW_MGMT1*/
#define PMVCM		(1 << 6) /* VCOM Power Management */
#define PMMIN		(1 << 5) /* MIN Input Power Management */
#define PMDAC		(1 << 2) /* DAC Power Management */
#define PMADL		(1 << 0) /* MIC Amp Lch and ADC Lch Power Management */

/* PW_MGMT2 */
#define HPMTN		(1 << 6)
#define PMHPL		(1 << 5)
#define PMHPR		(1 << 4)
#define MS		(1 << 3) /* master/slave select */
#define MCKO		(1 << 1)
#define PMPLL		(1 << 0)

#define PMHP_MASK	(PMHPL | PMHPR)
#define PMHP		PMHP_MASK

/* PW_MGMT3 */
#define PMADR		(1 << 0) /* MIC L / ADC R Power Management */

/* SG_SL1 */
#define MINS		(1 << 6) /* Switch from MIN to Speaker */
#define DACL		(1 << 4) /* Switch from DAC to Stereo or Receiver */
#define PMMP		(1 << 2) /* MPWR pin Power Management */
#define MGAIN0		(1 << 0) /* MIC amp gain*/

/* TIMER */
#define ZTM(param)	((param & 0x3) << 4) /* ALC Zoro Crossing TimeOut */
#define WTM(param)	(((param & 0x4) << 4) | ((param & 0x3) << 2))

/* ALC_CTL1 */
#define ALC		(1 << 5) /* ALC Enable */
#define LMTH0		(1 << 0) /* ALC Limiter / Recovery Level */

/* MD_CTL1 */
#define PLL3		(1 << 7)
#define PLL2		(1 << 6)
#define PLL1		(1 << 5)
#define PLL0		(1 << 4)
#define PLL_MASK	(PLL3 | PLL2 | PLL1 | PLL0)

#define BCKO_MASK	(1 << 3)
#define BCKO_64		BCKO_MASK

#define DIF_MASK	(3 << 0)
#define DSP		(0 << 0)
#define RIGHT_J		(1 << 0)
#define LEFT_J		(2 << 0)
#define I2S		(3 << 0)

/* MD_CTL2 */
#define FS0		(1 << 0)
#define FS1		(1 << 1)
#define FS2		(1 << 2)
#define FS3		(1 << 5)
#define FS_MASK		(FS0 | FS1 | FS2 | FS3)

/* MD_CTL3 */
#define BST1		(1 << 3)

/* MD_CTL4 */
#define DACH		(1 << 0)

/*
 * Playback Volume (table 39)
 *
 * max : 0x00 : +12.0 dB
 *       ( 0.5 dB step )
 * min : 0xFE : -115.0 dB
 * mute: 0xFF
 */
static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);

static const struct snd_kcontrol_new ak4642_snd_controls[] = {

	SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
			 0, 0xFF, 1, out_tlv),
};

static const struct snd_kcontrol_new ak4642_headphone_control =
	SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);

static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
	SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
};

static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {

	/* Outputs */
	SND_SOC_DAPM_OUTPUT("HPOUTL"),
	SND_SOC_DAPM_OUTPUT("HPOUTR"),
	SND_SOC_DAPM_OUTPUT("LINEOUT"),

	SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
	SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
	SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
			    &ak4642_headphone_control),

	SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),

	SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
			   &ak4642_lout_mixer_controls[0],
			   ARRAY_SIZE(ak4642_lout_mixer_controls)),

	/* DAC */
	SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0),
};

static const struct snd_soc_dapm_route ak4642_intercon[] = {

	/* Outputs */
	{"HPOUTL", NULL, "HPL Out"},
	{"HPOUTR", NULL, "HPR Out"},
	{"LINEOUT", NULL, "LINEOUT Mixer"},

	{"HPL Out", NULL, "Headphone Enable"},
	{"HPR Out", NULL, "Headphone Enable"},

	{"Headphone Enable", "Switch", "DACH"},

	{"DACH", NULL, "DAC"},

	{"LINEOUT Mixer", "DACL", "DAC"},
};

/*
 * ak4642 register cache
 */
static const struct reg_default ak4642_reg[] = {
	{  0, 0x00 }, {  1, 0x00 }, {  2, 0x01 }, {  3, 0x00 },
	{  4, 0x02 }, {  5, 0x00 }, {  6, 0x00 }, {  7, 0x00 },
	{  8, 0xe1 }, {  9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
	{ 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0x08 },
	{ 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
	{ 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
	{ 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
	{ 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
	{ 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
	{ 36, 0x00 },
};

static const struct reg_default ak4648_reg[] = {
	{  0, 0x00 }, {  1, 0x00 }, {  2, 0x01 }, {  3, 0x00 },
	{  4, 0x02 }, {  5, 0x00 }, {  6, 0x00 }, {  7, 0x00 },
	{  8, 0xe1 }, {  9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
	{ 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0xb8 },
	{ 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
	{ 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
	{ 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
	{ 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
	{ 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
	{ 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 0x08 },
};

static int ak4642_dai_startup(struct snd_pcm_substream *substream,
			      struct snd_soc_dai *dai)
{
	int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
	struct snd_soc_codec *codec = dai->codec;

	if (is_play) {
		/*
		 * start headphone output
		 *
		 * PLL, Master Mode
		 * Audio I/F Format :MSB justified (ADC & DAC)
		 * Bass Boost Level : Middle
		 *
		 * This operation came from example code of
		 * "ASAHI KASEI AK4642" (japanese) manual p97.
		 */
		snd_soc_write(codec, L_IVC, 0x91); /* volume */
		snd_soc_write(codec, R_IVC, 0x91); /* volume */
	} else {
		/*
		 * start stereo input
		 *
		 * PLL Master Mode
		 * Audio I/F Format:MSB justified (ADC & DAC)
		 * Pre MIC AMP:+20dB
		 * MIC Power On
		 * ALC setting:Refer to Table 35
		 * ALC bit=“1”
		 *
		 * This operation came from example code of
		 * "ASAHI KASEI AK4642" (japanese) manual p94.
		 */
		snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0);
		snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3));
		snd_soc_write(codec, ALC_CTL1, ALC | LMTH0);
		snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL);
		snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR);
	}

	return 0;
}

static void ak4642_dai_shutdown(struct snd_pcm_substream *substream,
			       struct snd_soc_dai *dai)
{
	int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
	struct snd_soc_codec *codec = dai->codec;

	if (is_play) {
	} else {
		/* stop stereo input */
		snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0);
		snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0);
		snd_soc_update_bits(codec, ALC_CTL1, ALC, 0);
	}
}

static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
	int clk_id, unsigned int freq, int dir)
{
	struct snd_soc_codec *codec = codec_dai->codec;
	u8 pll;

	switch (freq) {
	case 11289600:
		pll = PLL2;
		break;
	case 12288000:
		pll = PLL2 | PLL0;
		break;
	case 12000000:
		pll = PLL2 | PLL1;
		break;
	case 24000000:
		pll = PLL2 | PLL1 | PLL0;
		break;
	case 13500000:
		pll = PLL3 | PLL2;
		break;
	case 27000000:
		pll = PLL3 | PLL2 | PLL0;
		break;
	default:
		return -EINVAL;
	}
	snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll);

	return 0;
}

static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
	struct snd_soc_codec *codec = dai->codec;
	u8 data;
	u8 bcko;

	data = MCKO | PMPLL; /* use MCKO */
	bcko = 0;

	/* set master/slave audio interface */
	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
	case SND_SOC_DAIFMT_CBM_CFM:
		data |= MS;
		bcko = BCKO_64;
		break;
	case SND_SOC_DAIFMT_CBS_CFS:
		break;
	default:
		return -EINVAL;
	}
	snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
	snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);

	/* format type */
	data = 0;
	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
	case SND_SOC_DAIFMT_LEFT_J:
		data = LEFT_J;
		break;
	case SND_SOC_DAIFMT_I2S:
		data = I2S;
		break;
	/* FIXME
	 * Please add RIGHT_J / DSP support here
	 */
	default:
		return -EINVAL;
	}
	snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data);

	return 0;
}

static int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
				struct snd_pcm_hw_params *params,
				struct snd_soc_dai *dai)
{
	struct snd_soc_codec *codec = dai->codec;
	u8 rate;

	switch (params_rate(params)) {
	case 7350:
		rate = FS2;
		break;
	case 8000:
		rate = 0;
		break;
	case 11025:
		rate = FS2 | FS0;
		break;
	case 12000:
		rate = FS0;
		break;
	case 14700:
		rate = FS2 | FS1;
		break;
	case 16000:
		rate = FS1;
		break;
	case 22050:
		rate = FS2 | FS1 | FS0;
		break;
	case 24000:
		rate = FS1 | FS0;
		break;
	case 29400:
		rate = FS3 | FS2 | FS1;
		break;
	case 32000:
		rate = FS3 | FS1;
		break;
	case 44100:
		rate = FS3 | FS2 | FS1 | FS0;
		break;
	case 48000:
		rate = FS3 | FS1 | FS0;
		break;
	default:
		return -EINVAL;
	}
	snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);

	return 0;
}

static int ak4642_set_bias_level(struct snd_soc_codec *codec,
				 enum snd_soc_bias_level level)
{
	switch (level) {
	case SND_SOC_BIAS_OFF:
		snd_soc_write(codec, PW_MGMT1, 0x00);
		break;
	default:
		snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM);
		break;
	}
	codec->dapm.bias_level = level;

	return 0;
}

static const struct snd_soc_dai_ops ak4642_dai_ops = {
	.startup	= ak4642_dai_startup,
	.shutdown	= ak4642_dai_shutdown,
	.set_sysclk	= ak4642_dai_set_sysclk,
	.set_fmt	= ak4642_dai_set_fmt,
	.hw_params	= ak4642_dai_hw_params,
};

static struct snd_soc_dai_driver ak4642_dai = {
	.name = "ak4642-hifi",
	.playback = {
		.stream_name = "Playback",
		.channels_min = 1,
		.channels_max = 2,
		.rates = SNDRV_PCM_RATE_8000_48000,
		.formats = SNDRV_PCM_FMTBIT_S16_LE },
	.capture = {
		.stream_name = "Capture",
		.channels_min = 1,
		.channels_max = 2,
		.rates = SNDRV_PCM_RATE_8000_48000,
		.formats = SNDRV_PCM_FMTBIT_S16_LE },
	.ops = &ak4642_dai_ops,
	.symmetric_rates = 1,
};

static int ak4642_resume(struct snd_soc_codec *codec)
{
	struct regmap *regmap = dev_get_regmap(codec->dev, NULL);

	regcache_mark_dirty(regmap);
	regcache_sync(regmap);
	return 0;
}


static int ak4642_probe(struct snd_soc_codec *codec)
{
	int ret;

	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
	if (ret < 0) {
		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
		return ret;
	}

	ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

	return 0;
}

static int ak4642_remove(struct snd_soc_codec *codec)
{
	ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF);
	return 0;
}

static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
	.probe			= ak4642_probe,
	.remove			= ak4642_remove,
	.resume			= ak4642_resume,
	.set_bias_level		= ak4642_set_bias_level,
	.controls		= ak4642_snd_controls,
	.num_controls		= ARRAY_SIZE(ak4642_snd_controls),
	.dapm_widgets		= ak4642_dapm_widgets,
	.num_dapm_widgets	= ARRAY_SIZE(ak4642_dapm_widgets),
	.dapm_routes		= ak4642_intercon,
	.num_dapm_routes	= ARRAY_SIZE(ak4642_intercon),
};

static const struct regmap_config ak4642_regmap = {
	.reg_bits		= 8,
	.val_bits		= 8,
	.max_register		= ARRAY_SIZE(ak4642_reg) + 1,
	.reg_defaults		= ak4642_reg,
	.num_reg_defaults	= ARRAY_SIZE(ak4642_reg),
};

static const struct regmap_config ak4648_regmap = {
	.reg_bits		= 8,
	.val_bits		= 8,
	.max_register		= ARRAY_SIZE(ak4648_reg) + 1,
	.reg_defaults		= ak4648_reg,
	.num_reg_defaults	= ARRAY_SIZE(ak4648_reg),
};

static struct of_device_id ak4642_of_match[];
static int ak4642_i2c_probe(struct i2c_client *i2c,
			    const struct i2c_device_id *id)
{
	struct device_node *np = i2c->dev.of_node;
	const struct regmap_config *regmap_config = NULL;
	struct regmap *regmap;

	if (np) {
		const struct of_device_id *of_id;

		of_id = of_match_device(ak4642_of_match, &i2c->dev);
		if (of_id)
			regmap_config = of_id->data;
	} else {
		regmap_config = (const struct regmap_config *)id->driver_data;
	}

	if (!regmap_config) {
		dev_err(&i2c->dev, "Unknown device type\n");
		return -EINVAL;
	}

	regmap = devm_regmap_init_i2c(i2c, regmap_config);
	if (IS_ERR(regmap))
		return PTR_ERR(regmap);

	return snd_soc_register_codec(&i2c->dev,
				      &soc_codec_dev_ak4642, &ak4642_dai, 1);
}

static int ak4642_i2c_remove(struct i2c_client *client)
{
	snd_soc_unregister_codec(&client->dev);
	return 0;
}

static struct of_device_id ak4642_of_match[] = {
	{ .compatible = "asahi-kasei,ak4642",	.data = &ak4642_regmap},
	{ .compatible = "asahi-kasei,ak4643",	.data = &ak4642_regmap},
	{ .compatible = "asahi-kasei,ak4648",	.data = &ak4648_regmap},
	{},
};
MODULE_DEVICE_TABLE(of, ak4642_of_match);

static const struct i2c_device_id ak4642_i2c_id[] = {
	{ "ak4642", (kernel_ulong_t)&ak4642_regmap },
	{ "ak4643", (kernel_ulong_t)&ak4642_regmap },
	{ "ak4648", (kernel_ulong_t)&ak4648_regmap },
	{ }
};
MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);

static struct i2c_driver ak4642_i2c_driver = {
	.driver = {
		.name = "ak4642-codec",
		.owner = THIS_MODULE,
		.of_match_table = ak4642_of_match,
	},
	.probe		= ak4642_i2c_probe,
	.remove		= ak4642_i2c_remove,
	.id_table	= ak4642_i2c_id,
};

module_i2c_driver(ak4642_i2c_driver);

MODULE_DESCRIPTION("Soc AK4642 driver");
MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
MODULE_LICENSE("GPL");