Initial import

1 files changed, 642 insertions, 0 deletions

diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
new file mode 100644
index 000000000..13585e88f
--- /dev/null
+++ b/sound/soc/codecs/ak4642.c
@@ -0,0 +1,642 @@
+/*
+ * 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*/
+
+/* SG_SL2 */
+#define LOPS		(1 << 6) /* Stero Line-out Power Save Mode */
+
+/* TIMER */
+#define ZTM(param)	((param & 0x3) << 4) /* ALC Zero 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)
+
+struct ak4642_drvdata {
+	const struct regmap_config *regmap_config;
+	int extended_frequencies;
+};
+
+struct ak4642_priv {
+	const struct ak4642_drvdata *drvdata;
+};
+
+/*
+ * 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),
+	SOC_SINGLE("ALC Capture Switch", ALC_CTL1, 5, 1, 0),
+	SOC_SINGLE("ALC Capture ZC Switch", ALC_CTL1, 4, 1, 1),
+};
+
+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),
+};
+
+/* event handlers */
+static int ak4642_lout_event(struct snd_soc_dapm_widget *w,
+			     struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMD:
+	case SND_SOC_DAPM_PRE_PMU:
+		/* Power save mode ON */
+		snd_soc_update_bits(codec, SG_SL2, LOPS, LOPS);
+		break;
+	case SND_SOC_DAPM_POST_PMU:
+	case SND_SOC_DAPM_POST_PMD:
+		/* Power save mode OFF */
+		mdelay(300);
+		snd_soc_update_bits(codec, SG_SL2, LOPS, 0);
+		break;
+	}
+
+	return 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_E("LINEOUT Mixer", PW_MGMT1, 3, 0,
+			   &ak4642_lout_mixer_controls[0],
+			   ARRAY_SIZE(ak4642_lout_mixer_controls),
+			   ak4642_lout_event,
+			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+			   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+	/* DAC */
+	SND_SOC_DAPM_DAC("DAC", NULL, 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"},
+
+	{ "DAC", NULL, "Playback" },
+};
+
+/*
+ * 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;
+	struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+	u8 pll;
+	int extended_freq = 0;
+
+	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;
+	case 19200000:
+		pll = PLL3;
+		extended_freq = 1;
+		break;
+	case 13000000:
+		pll = PLL3 | PLL2 | PLL1;
+		extended_freq = 1;
+		break;
+	case 26000000:
+		pll = PLL3 | PLL2 | PLL1 | PLL0;
+		extended_freq = 1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (extended_freq && !priv->drvdata->extended_frequencies)
+		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 = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE },
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.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 struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
+	.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 const struct ak4642_drvdata ak4642_drvdata = {
+	.regmap_config = &ak4642_regmap,
+};
+
+static const struct ak4642_drvdata ak4643_drvdata = {
+	.regmap_config = &ak4642_regmap,
+};
+
+static const struct ak4642_drvdata ak4648_drvdata = {
+	.regmap_config = &ak4648_regmap,
+	.extended_frequencies = 1,
+};
+
+static const 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 ak4642_drvdata *drvdata = NULL;
+	struct regmap *regmap;
+	struct ak4642_priv *priv;
+
+	if (np) {
+		const struct of_device_id *of_id;
+
+		of_id = of_match_device(ak4642_of_match, &i2c->dev);
+		if (of_id)
+			drvdata = of_id->data;
+	} else {
+		drvdata = (const struct ak4642_drvdata *)id->driver_data;
+	}
+
+	if (!drvdata) {
+		dev_err(&i2c->dev, "Unknown device type\n");
+		return -EINVAL;
+	}
+
+	priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->drvdata = drvdata;
+
+	i2c_set_clientdata(i2c, priv);
+
+	regmap = devm_regmap_init_i2c(i2c, drvdata->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 const struct of_device_id ak4642_of_match[] = {
+	{ .compatible = "asahi-kasei,ak4642",	.data = &ak4642_drvdata},
+	{ .compatible = "asahi-kasei,ak4643",	.data = &ak4643_drvdata},
+	{ .compatible = "asahi-kasei,ak4648",	.data = &ak4648_drvdata},
+	{},
+};
+MODULE_DEVICE_TABLE(of, ak4642_of_match);
+
+static const struct i2c_device_id ak4642_i2c_id[] = {
+	{ "ak4642", (kernel_ulong_t)&ak4642_drvdata },
+	{ "ak4643", (kernel_ulong_t)&ak4643_drvdata },
+	{ "ak4648", (kernel_ulong_t)&ak4648_drvdata },
+	{ }
+};
+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");