sound/soc/codecs/ak4642.c - kernel/panda - Git at Google

 /*
  * 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 not tested.
  * AK4643 is tested.
  */

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

 #define AK4642_VERSION "0.0.1"

 #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

 #define AK4642_CACHEREGNUM 	0x25

 /* 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, -11500, 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),
 };


 /* codec private data */
 struct ak4642_priv {
 	unsigned int sysclk;
 	enum snd_soc_control_type control_type;
 };

 /*
  * ak4642 register cache
  */
 static const u8 ak4642_reg[AK4642_CACHEREGNUM] = {
 	0x00, 0x00, 0x01, 0x00,
 	0x02, 0x00, 0x00, 0x00,
 	0xe1, 0xe1, 0x18, 0x00,
 	0xe1, 0x18, 0x11, 0x08,
 	0x00, 0x00, 0x00, 0x00,
 	0x00, 0x00, 0x00, 0x00,
 	0x00, 0x00, 0x00, 0x00,
 	0x00, 0x00, 0x00, 0x00,
 	0x00, 0x00, 0x00, 0x00,
 	0x00,
 };

 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_update_bits(codec, MD_CTL4, DACH, DACH);
 		snd_soc_update_bits(codec, MD_CTL3, BST1, BST1);
 		snd_soc_write(codec, L_IVC, 0x91); /* volume */
 		snd_soc_write(codec, R_IVC, 0x91); /* volume */
 		snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMMIN | PMDAC,
 						     PMVCM | PMMIN | PMDAC);
 		snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK,	PMHP);
 		snd_soc_update_bits(codec, PW_MGMT2, HPMTN,	HPMTN);
 	} 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_write(codec, SG_SL1, 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, PMVCM | PMADL,
 						     PMVCM | 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) {
 		/* stop headphone output */
 		snd_soc_update_bits(codec, PW_MGMT2, HPMTN,	0);
 		snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK,	0);
 		snd_soc_update_bits(codec, PW_MGMT1, PMMIN | PMDAC, 0);
 		snd_soc_update_bits(codec, MD_CTL3, BST1, 0);
 		snd_soc_update_bits(codec, MD_CTL4, DACH, 0);
 	} 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;
 		break;
 	}
 	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;
 		break;
 	}
 	snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);

 	return 0;
 }

 static 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)
 {
 	snd_soc_cache_sync(codec);
 	return 0;
 }


 static int ak4642_probe(struct snd_soc_codec *codec)
 {
 	struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec);
 	int ret;

 	dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION);

 	ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
 	}

 	snd_soc_add_controls(codec, ak4642_snd_controls,
 			     ARRAY_SIZE(ak4642_snd_controls));

 	return 0;
 }

 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
 	.probe			= ak4642_probe,
 	.resume			= ak4642_resume,
 	.reg_cache_size		= ARRAY_SIZE(ak4642_reg),
 	.reg_word_size		= sizeof(u8),
 	.reg_cache_default	= ak4642_reg,
 };

 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 static __devinit int ak4642_i2c_probe(struct i2c_client *i2c,
 				      const struct i2c_device_id *id)
 {
 	struct ak4642_priv *ak4642;
 	int ret;

 	ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL);
 	if (!ak4642)
 		return -ENOMEM;

 	i2c_set_clientdata(i2c, ak4642);
 	ak4642->control_type = SND_SOC_I2C;

 	ret =  snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_ak4642, &ak4642_dai, 1);
 	if (ret < 0)
 		kfree(ak4642);
 	return ret;
 }

 static __devexit int ak4642_i2c_remove(struct i2c_client *client)
 {
 	snd_soc_unregister_codec(&client->dev);
 	kfree(i2c_get_clientdata(client));
 	return 0;
 }

 static const struct i2c_device_id ak4642_i2c_id[] = {
 	{ "ak4642", 0 },
 	{ "ak4643", 0 },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);

 static struct i2c_driver ak4642_i2c_driver = {
 	.driver = {
 		.name = "ak4642-codec",
 		.owner = THIS_MODULE,
 	},
 	.probe		= ak4642_i2c_probe,
 	.remove		= __devexit_p(ak4642_i2c_remove),
 	.id_table	= ak4642_i2c_id,
 };
 #endif

 static int __init ak4642_modinit(void)
 {
 	int ret = 0;
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	ret = i2c_add_driver(&ak4642_i2c_driver);
 #endif
 	return ret;

 }
 module_init(ak4642_modinit);

 static void __exit ak4642_exit(void)
 {
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	i2c_del_driver(&ak4642_i2c_driver);
 #endif

 }
 module_exit(ak4642_exit);

 MODULE_DESCRIPTION("Soc AK4642 driver");
 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
 MODULE_LICENSE("GPL");
	/*
	* 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 not tested.
	* AK4643 is tested.
	*/

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

	#define AK4642_VERSION "0.0.1"

	#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

	#define AK4642_CACHEREGNUM 0x25

	/* 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, -11500, 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),
	};


	/* codec private data */
	struct ak4642_priv {
	unsigned int sysclk;
	enum snd_soc_control_type control_type;
	};

	/*
	* ak4642 register cache
	*/
	static const u8 ak4642_reg[AK4642_CACHEREGNUM] = {
	0x00, 0x00, 0x01, 0x00,
	0x02, 0x00, 0x00, 0x00,
	0xe1, 0xe1, 0x18, 0x00,
	0xe1, 0x18, 0x11, 0x08,
	0x00, 0x00, 0x00, 0x00,
	0x00, 0x00, 0x00, 0x00,
	0x00, 0x00, 0x00, 0x00,
	0x00, 0x00, 0x00, 0x00,
	0x00, 0x00, 0x00, 0x00,
	0x00,
	};

	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_update_bits(codec, MD_CTL4, DACH, DACH);
	snd_soc_update_bits(codec, MD_CTL3, BST1, BST1);
	snd_soc_write(codec, L_IVC, 0x91); /* volume */
	snd_soc_write(codec, R_IVC, 0x91); /* volume */
	snd_soc_update_bits(codec, PW_MGMT1, PMVCM \| PMMIN \| PMDAC,
	PMVCM \| PMMIN \| PMDAC);
	snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP);
	snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN);
	} 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_write(codec, SG_SL1, 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, PMVCM \| PMADL,
	PMVCM \| 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) {
	/* stop headphone output */
	snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0);
	snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, 0);
	snd_soc_update_bits(codec, PW_MGMT1, PMMIN \| PMDAC, 0);
	snd_soc_update_bits(codec, MD_CTL3, BST1, 0);
	snd_soc_update_bits(codec, MD_CTL4, DACH, 0);
	} 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;
	break;
	}
	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;
	break;
	}
	snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);

	return 0;
	}

	static 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)
	{
	snd_soc_cache_sync(codec);
	return 0;
	}


	static int ak4642_probe(struct snd_soc_codec *codec)
	{
	struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec);
	int ret;

	dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION);

	ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type);
	if (ret < 0) {
	dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
	return ret;
	}

	snd_soc_add_controls(codec, ak4642_snd_controls,
	ARRAY_SIZE(ak4642_snd_controls));

	return 0;
	}

	static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
	.probe = ak4642_probe,
	.resume = ak4642_resume,
	.reg_cache_size = ARRAY_SIZE(ak4642_reg),
	.reg_word_size = sizeof(u8),
	.reg_cache_default = ak4642_reg,
	};

	#if defined(CONFIG_I2C) \|\| defined(CONFIG_I2C_MODULE)
	static __devinit int ak4642_i2c_probe(struct i2c_client *i2c,
	const struct i2c_device_id *id)
	{
	struct ak4642_priv *ak4642;
	int ret;

	ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL);
	if (!ak4642)
	return -ENOMEM;

	i2c_set_clientdata(i2c, ak4642);
	ak4642->control_type = SND_SOC_I2C;

	ret = snd_soc_register_codec(&i2c->dev,
	&soc_codec_dev_ak4642, &ak4642_dai, 1);
	if (ret < 0)
	kfree(ak4642);
	return ret;
	}

	static __devexit int ak4642_i2c_remove(struct i2c_client *client)
	{
	snd_soc_unregister_codec(&client->dev);
	kfree(i2c_get_clientdata(client));
	return 0;
	}

	static const struct i2c_device_id ak4642_i2c_id[] = {
	{ "ak4642", 0 },
	{ "ak4643", 0 },
	{ }
	};
	MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);

	static struct i2c_driver ak4642_i2c_driver = {
	.driver = {
	.name = "ak4642-codec",
	.owner = THIS_MODULE,
	},
	.probe = ak4642_i2c_probe,
	.remove = __devexit_p(ak4642_i2c_remove),
	.id_table = ak4642_i2c_id,
	};
	#endif

	static int __init ak4642_modinit(void)
	{
	int ret = 0;
	#if defined(CONFIG_I2C) \|\| defined(CONFIG_I2C_MODULE)
	ret = i2c_add_driver(&ak4642_i2c_driver);
	#endif
	return ret;

	}
	module_init(ak4642_modinit);

	static void __exit ak4642_exit(void)
	{
	#if defined(CONFIG_I2C) \|\| defined(CONFIG_I2C_MODULE)
	i2c_del_driver(&ak4642_i2c_driver);
	#endif

	}
	module_exit(ak4642_exit);

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