linux/sound/soc/codecs/ak4642.c
Rob Herring 340d79a14d
ASoC: Explicitly include correct DT includes
The DT of_device.h and of_platform.h date back to the separate
of_platform_bus_type before it was merged into the regular platform bus.
As part of that merge prepping Arm DT support 13 years ago, they
"temporarily" include each other. They also include platform_device.h
and of.h. As a result, there's a pretty much random mix of those include
files used throughout the tree. In order to detangle these headers and
replace the implicit includes with struct declarations, users need to
explicitly include the correct includes.

Acked-by: Jernej Skrabec <jernej.skrabec@gmail.com>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Claudiu Beznea <claudiu.beznea@tuxon.dev> # for at91
Signed-off-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20231006-dt-asoc-header-cleanups-v3-1-13a4f0f7fee6@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
2023-10-09 13:13:56 +01:00

696 lines
17 KiB
C

// SPDX-License-Identifier: GPL-2.0
//
// 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
/* ** 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/clk.h>
#include <linux/clk-provider.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/slab.h>
#include <linux/of.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 /* The maximum valid register for ak4642 */
#define PW_MGMT4 0x20
#define MD_CTL5 0x21
#define LO_MS 0x22
#define HP_MS 0x23
#define SPK_MS 0x24 /* The maximum valid register for ak4643 */
#define EQ_FBEQAB 0x25
#define EQ_FBEQCD 0x26
#define EQ_FBEQE 0x27 /* The maximum valid register for ak4648 */
/* 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 FSs(val) (((val & 0x7) << 0) | ((val & 0x8) << 2))
#define PSs(val) ((val & 0x3) << 6)
/* 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;
struct clk *mcko;
};
/*
* 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_component *component = snd_soc_dapm_to_component(w->dapm);
switch (event) {
case SND_SOC_DAPM_PRE_PMD:
case SND_SOC_DAPM_PRE_PMU:
/* Power save mode ON */
snd_soc_component_update_bits(component, SG_SL2, LOPS, LOPS);
break;
case SND_SOC_DAPM_POST_PMU:
case SND_SOC_DAPM_POST_PMD:
/* Power save mode OFF */
msleep(300);
snd_soc_component_update_bits(component, 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 ak4643_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 },
};
/* The default settings for 0x0 ~ 0x1f registers are the same for ak4642
and ak4643. So we reuse the ak4643 reg_default for ak4642.
The valid registers for ak4642 are 0x0 ~ 0x1f which is a subset of ak4643,
so define NUM_AK4642_REG_DEFAULTS for ak4642.
*/
#define ak4642_reg ak4643_reg
#define NUM_AK4642_REG_DEFAULTS (FIL1_3 + 1)
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_component *component = dai->component;
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_component_write(component, L_IVC, 0x91); /* volume */
snd_soc_component_write(component, 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_component_update_bits(component, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0);
snd_soc_component_write(component, TIMER, ZTM(0x3) | WTM(0x3));
snd_soc_component_write(component, ALC_CTL1, ALC | LMTH0);
snd_soc_component_update_bits(component, PW_MGMT1, PMADL, PMADL);
snd_soc_component_update_bits(component, 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_component *component = dai->component;
if (is_play) {
} else {
/* stop stereo input */
snd_soc_component_update_bits(component, PW_MGMT1, PMADL, 0);
snd_soc_component_update_bits(component, PW_MGMT3, PMADR, 0);
snd_soc_component_update_bits(component, 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_component *component = codec_dai->component;
struct ak4642_priv *priv = snd_soc_component_get_drvdata(component);
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_component_update_bits(component, MD_CTL1, PLL_MASK, pll);
return 0;
}
static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_component *component = dai->component;
u8 data;
u8 bcko;
data = MCKO | PMPLL; /* use MCKO */
bcko = 0;
/* set clocking for audio interface */
switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
case SND_SOC_DAIFMT_CBP_CFP:
data |= MS;
bcko = BCKO_64;
break;
case SND_SOC_DAIFMT_CBC_CFC:
break;
default:
return -EINVAL;
}
snd_soc_component_update_bits(component, PW_MGMT2, MS | MCKO | PMPLL, data);
snd_soc_component_update_bits(component, 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_component_update_bits(component, MD_CTL1, DIF_MASK, data);
return 0;
}
static int ak4642_set_mcko(struct snd_soc_component *component,
u32 frequency)
{
static const u32 fs_list[] = {
[0] = 8000,
[1] = 12000,
[2] = 16000,
[3] = 24000,
[4] = 7350,
[5] = 11025,
[6] = 14700,
[7] = 22050,
[10] = 32000,
[11] = 48000,
[14] = 29400,
[15] = 44100,
};
static const u32 ps_list[] = {
[0] = 256,
[1] = 128,
[2] = 64,
[3] = 32
};
int ps, fs;
for (ps = 0; ps < ARRAY_SIZE(ps_list); ps++) {
for (fs = 0; fs < ARRAY_SIZE(fs_list); fs++) {
if (frequency == ps_list[ps] * fs_list[fs]) {
snd_soc_component_write(component, MD_CTL2,
PSs(ps) | FSs(fs));
return 0;
}
}
}
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_component *component = dai->component;
struct ak4642_priv *priv = snd_soc_component_get_drvdata(component);
u32 rate = clk_get_rate(priv->mcko);
if (!rate)
rate = params_rate(params) * 256;
return ak4642_set_mcko(component, rate);
}
static int ak4642_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
switch (level) {
case SND_SOC_BIAS_OFF:
snd_soc_component_write(component, PW_MGMT1, 0x00);
break;
default:
snd_soc_component_update_bits(component, PW_MGMT1, PMVCM, PMVCM);
break;
}
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_rate = 1,
};
static int ak4642_suspend(struct snd_soc_component *component)
{
struct regmap *regmap = dev_get_regmap(component->dev, NULL);
regcache_cache_only(regmap, true);
regcache_mark_dirty(regmap);
return 0;
}
static int ak4642_resume(struct snd_soc_component *component)
{
struct regmap *regmap = dev_get_regmap(component->dev, NULL);
regcache_cache_only(regmap, false);
regcache_sync(regmap);
return 0;
}
static int ak4642_probe(struct snd_soc_component *component)
{
struct ak4642_priv *priv = snd_soc_component_get_drvdata(component);
if (priv->mcko)
ak4642_set_mcko(component, clk_get_rate(priv->mcko));
return 0;
}
static const struct snd_soc_component_driver soc_component_dev_ak4642 = {
.probe = ak4642_probe,
.suspend = ak4642_suspend,
.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),
.idle_bias_on = 1,
.endianness = 1,
};
static const struct regmap_config ak4642_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = FIL1_3,
.reg_defaults = ak4642_reg,
.num_reg_defaults = NUM_AK4642_REG_DEFAULTS,
.cache_type = REGCACHE_RBTREE,
};
static const struct regmap_config ak4643_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = SPK_MS,
.reg_defaults = ak4643_reg,
.num_reg_defaults = ARRAY_SIZE(ak4643_reg),
.cache_type = REGCACHE_RBTREE,
};
static const struct regmap_config ak4648_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = EQ_FBEQE,
.reg_defaults = ak4648_reg,
.num_reg_defaults = ARRAY_SIZE(ak4648_reg),
.cache_type = REGCACHE_RBTREE,
};
static const struct ak4642_drvdata ak4642_drvdata = {
.regmap_config = &ak4642_regmap,
};
static const struct ak4642_drvdata ak4643_drvdata = {
.regmap_config = &ak4643_regmap,
};
static const struct ak4642_drvdata ak4648_drvdata = {
.regmap_config = &ak4648_regmap,
.extended_frequencies = 1,
};
#ifdef CONFIG_COMMON_CLK
static struct clk *ak4642_of_parse_mcko(struct device *dev)
{
struct device_node *np = dev->of_node;
struct clk *clk;
const char *clk_name = np->name;
const char *parent_clk_name = NULL;
u32 rate;
if (of_property_read_u32(np, "clock-frequency", &rate))
return NULL;
if (of_property_read_bool(np, "clocks"))
parent_clk_name = of_clk_get_parent_name(np, 0);
of_property_read_string(np, "clock-output-names", &clk_name);
clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, 0, rate);
if (!IS_ERR(clk))
of_clk_add_provider(np, of_clk_src_simple_get, clk);
return clk;
}
#else
#define ak4642_of_parse_mcko(d) 0
#endif
static int ak4642_i2c_probe(struct i2c_client *i2c)
{
struct device *dev = &i2c->dev;
const struct ak4642_drvdata *drvdata;
struct regmap *regmap;
struct ak4642_priv *priv;
struct clk *mcko = NULL;
if (dev_fwnode(dev)) {
mcko = ak4642_of_parse_mcko(dev);
if (IS_ERR(mcko))
mcko = NULL;
}
drvdata = i2c_get_match_data(i2c);
if (!drvdata)
return dev_err_probe(dev, -EINVAL, "Unknown device type\n");
priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
priv->drvdata = drvdata;
priv->mcko = mcko;
i2c_set_clientdata(i2c, priv);
regmap = devm_regmap_init_i2c(i2c, drvdata->regmap_config);
if (IS_ERR(regmap))
return PTR_ERR(regmap);
return devm_snd_soc_register_component(dev,
&soc_component_dev_ak4642, &ak4642_dai, 1);
}
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",
.of_match_table = ak4642_of_match,
},
.probe = ak4642_i2c_probe,
.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 v2");