linux/sound/soc/codecs/ak4641.c
Lars-Peter Clausen 0b0171e3ad ASoC: ak4641: Cleanup manual bias level transitions
Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner.

Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore either.

The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe()
can also be removed as the core will automatically do this after the CODEC
has been probed.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2014-10-22 11:28:25 +01:00

624 lines
17 KiB
C

/*
* ak4641.c -- AK4641 ALSA Soc Audio driver
*
* Copyright (C) 2008 Harald Welte <laforge@gnufiish.org>
* Copyright (C) 2011 Dmitry Artamonow <mad_soft@inbox.ru>
*
* Based on ak4535.c by Richard Purdie
*
* 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.
*/
#include <linux/module.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/gpio.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include <sound/ak4641.h>
#include "ak4641.h"
/* codec private data */
struct ak4641_priv {
struct regmap *regmap;
unsigned int sysclk;
int deemph;
int playback_fs;
};
/*
* ak4641 register cache
*/
static const struct reg_default ak4641_reg_defaults[] = {
{ 0, 0x00 }, { 1, 0x80 }, { 2, 0x00 }, { 3, 0x80 },
{ 4, 0x02 }, { 5, 0x00 }, { 6, 0x11 }, { 7, 0x05 },
{ 8, 0x00 }, { 9, 0x00 }, { 10, 0x36 }, { 11, 0x10 },
{ 12, 0x00 }, { 13, 0x00 }, { 14, 0x57 }, { 15, 0x00 },
{ 16, 0x88 }, { 17, 0x88 }, { 18, 0x08 }, { 19, 0x08 }
};
static const int deemph_settings[] = {44100, 0, 48000, 32000};
static int ak4641_set_deemph(struct snd_soc_codec *codec)
{
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
int i, best = 0;
for (i = 0 ; i < ARRAY_SIZE(deemph_settings); i++) {
/* if deemphasis is on, select the nearest available rate */
if (ak4641->deemph && deemph_settings[i] != 0 &&
abs(deemph_settings[i] - ak4641->playback_fs) <
abs(deemph_settings[best] - ak4641->playback_fs))
best = i;
if (!ak4641->deemph && deemph_settings[i] == 0)
best = i;
}
dev_dbg(codec->dev, "Set deemphasis %d\n", best);
return snd_soc_update_bits(codec, AK4641_DAC, 0x3, best);
}
static int ak4641_put_deemph(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
int deemph = ucontrol->value.enumerated.item[0];
if (deemph > 1)
return -EINVAL;
ak4641->deemph = deemph;
return ak4641_set_deemph(codec);
}
static int ak4641_get_deemph(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
ucontrol->value.enumerated.item[0] = ak4641->deemph;
return 0;
};
static const char *ak4641_mono_out[] = {"(L + R)/2", "Hi-Z"};
static const char *ak4641_hp_out[] = {"Stereo", "Mono"};
static const char *ak4641_mic_select[] = {"Internal", "External"};
static const char *ak4641_mic_or_dac[] = {"Microphone", "Voice DAC"};
static const DECLARE_TLV_DB_SCALE(mono_gain_tlv, -1700, 2300, 0);
static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 2000, 0);
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1050, 150, 0);
static const DECLARE_TLV_DB_SCALE(master_tlv, -12750, 50, 0);
static const DECLARE_TLV_DB_SCALE(mic_stereo_sidetone_tlv, -2700, 300, 0);
static const DECLARE_TLV_DB_SCALE(mic_mono_sidetone_tlv, -400, 400, 0);
static const DECLARE_TLV_DB_SCALE(capture_tlv, -800, 50, 0);
static const DECLARE_TLV_DB_SCALE(alc_tlv, -800, 50, 0);
static const DECLARE_TLV_DB_SCALE(aux_in_tlv, -2100, 300, 0);
static SOC_ENUM_SINGLE_DECL(ak4641_mono_out_enum,
AK4641_SIG1, 6, ak4641_mono_out);
static SOC_ENUM_SINGLE_DECL(ak4641_hp_out_enum,
AK4641_MODE2, 2, ak4641_hp_out);
static SOC_ENUM_SINGLE_DECL(ak4641_mic_select_enum,
AK4641_MIC, 1, ak4641_mic_select);
static SOC_ENUM_SINGLE_DECL(ak4641_mic_or_dac_enum,
AK4641_BTIF, 4, ak4641_mic_or_dac);
static const struct snd_kcontrol_new ak4641_snd_controls[] = {
SOC_ENUM("Mono 1 Output", ak4641_mono_out_enum),
SOC_SINGLE_TLV("Mono 1 Gain Volume", AK4641_SIG1, 7, 1, 1,
mono_gain_tlv),
SOC_ENUM("Headphone Output", ak4641_hp_out_enum),
SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0,
ak4641_get_deemph, ak4641_put_deemph),
SOC_SINGLE_TLV("Mic Boost Volume", AK4641_MIC, 0, 1, 0, mic_boost_tlv),
SOC_SINGLE("ALC Operation Time", AK4641_TIMER, 0, 3, 0),
SOC_SINGLE("ALC Recovery Time", AK4641_TIMER, 2, 3, 0),
SOC_SINGLE("ALC ZC Time", AK4641_TIMER, 4, 3, 0),
SOC_SINGLE("ALC 1 Switch", AK4641_ALC1, 5, 1, 0),
SOC_SINGLE_TLV("ALC Volume", AK4641_ALC2, 0, 71, 0, alc_tlv),
SOC_SINGLE("Left Out Enable Switch", AK4641_SIG2, 1, 1, 0),
SOC_SINGLE("Right Out Enable Switch", AK4641_SIG2, 0, 1, 0),
SOC_SINGLE_TLV("Capture Volume", AK4641_PGA, 0, 71, 0, capture_tlv),
SOC_DOUBLE_R_TLV("Master Playback Volume", AK4641_LATT,
AK4641_RATT, 0, 255, 1, master_tlv),
SOC_SINGLE_TLV("AUX In Volume", AK4641_VOL, 0, 15, 0, aux_in_tlv),
SOC_SINGLE("Equalizer Switch", AK4641_DAC, 2, 1, 0),
SOC_SINGLE_TLV("EQ1 100 Hz Volume", AK4641_EQLO, 0, 15, 1, eq_tlv),
SOC_SINGLE_TLV("EQ2 250 Hz Volume", AK4641_EQLO, 4, 15, 1, eq_tlv),
SOC_SINGLE_TLV("EQ3 1 kHz Volume", AK4641_EQMID, 0, 15, 1, eq_tlv),
SOC_SINGLE_TLV("EQ4 3.5 kHz Volume", AK4641_EQMID, 4, 15, 1, eq_tlv),
SOC_SINGLE_TLV("EQ5 10 kHz Volume", AK4641_EQHI, 0, 15, 1, eq_tlv),
};
/* Mono 1 Mixer */
static const struct snd_kcontrol_new ak4641_mono1_mixer_controls[] = {
SOC_DAPM_SINGLE_TLV("Mic Mono Sidetone Volume", AK4641_VOL, 7, 1, 0,
mic_mono_sidetone_tlv),
SOC_DAPM_SINGLE("Mic Mono Sidetone Switch", AK4641_SIG1, 4, 1, 0),
SOC_DAPM_SINGLE("Mono Playback Switch", AK4641_SIG1, 5, 1, 0),
};
/* Stereo Mixer */
static const struct snd_kcontrol_new ak4641_stereo_mixer_controls[] = {
SOC_DAPM_SINGLE_TLV("Mic Sidetone Volume", AK4641_VOL, 4, 7, 0,
mic_stereo_sidetone_tlv),
SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4641_SIG2, 4, 1, 0),
SOC_DAPM_SINGLE("Playback Switch", AK4641_SIG2, 7, 1, 0),
SOC_DAPM_SINGLE("Aux Bypass Switch", AK4641_SIG2, 5, 1, 0),
};
/* Input Mixer */
static const struct snd_kcontrol_new ak4641_input_mixer_controls[] = {
SOC_DAPM_SINGLE("Mic Capture Switch", AK4641_MIC, 2, 1, 0),
SOC_DAPM_SINGLE("Aux Capture Switch", AK4641_MIC, 5, 1, 0),
};
/* Mic mux */
static const struct snd_kcontrol_new ak4641_mic_mux_control =
SOC_DAPM_ENUM("Mic Select", ak4641_mic_select_enum);
/* Input mux */
static const struct snd_kcontrol_new ak4641_input_mux_control =
SOC_DAPM_ENUM("Input Select", ak4641_mic_or_dac_enum);
/* mono 2 switch */
static const struct snd_kcontrol_new ak4641_mono2_control =
SOC_DAPM_SINGLE("Switch", AK4641_SIG1, 0, 1, 0);
/* ak4641 dapm widgets */
static const struct snd_soc_dapm_widget ak4641_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0,
&ak4641_stereo_mixer_controls[0],
ARRAY_SIZE(ak4641_stereo_mixer_controls)),
SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0,
&ak4641_mono1_mixer_controls[0],
ARRAY_SIZE(ak4641_mono1_mixer_controls)),
SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0,
&ak4641_input_mixer_controls[0],
ARRAY_SIZE(ak4641_input_mixer_controls)),
SND_SOC_DAPM_MUX("Mic Mux", SND_SOC_NOPM, 0, 0,
&ak4641_mic_mux_control),
SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
&ak4641_input_mux_control),
SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0,
&ak4641_mono2_control),
SND_SOC_DAPM_OUTPUT("LOUT"),
SND_SOC_DAPM_OUTPUT("ROUT"),
SND_SOC_DAPM_OUTPUT("MOUT1"),
SND_SOC_DAPM_OUTPUT("MOUT2"),
SND_SOC_DAPM_OUTPUT("MICOUT"),
SND_SOC_DAPM_ADC("ADC", "HiFi Capture", AK4641_PM1, 0, 0),
SND_SOC_DAPM_PGA("Mic", AK4641_PM1, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("AUX In", AK4641_PM1, 2, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mono Out", AK4641_PM1, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("Line Out", AK4641_PM1, 4, 0, NULL, 0),
SND_SOC_DAPM_DAC("DAC", "HiFi Playback", AK4641_PM2, 0, 0),
SND_SOC_DAPM_PGA("Mono Out 2", AK4641_PM2, 3, 0, NULL, 0),
SND_SOC_DAPM_ADC("Voice ADC", "Voice Capture", AK4641_BTIF, 0, 0),
SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0),
SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4641_MIC, 3, 0),
SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4641_MIC, 4, 0),
SND_SOC_DAPM_INPUT("MICIN"),
SND_SOC_DAPM_INPUT("MICEXT"),
SND_SOC_DAPM_INPUT("AUX"),
SND_SOC_DAPM_INPUT("AIN"),
};
static const struct snd_soc_dapm_route ak4641_audio_map[] = {
/* Stereo Mixer */
{"Stereo Mixer", "Playback Switch", "DAC"},
{"Stereo Mixer", "Mic Sidetone Switch", "Input Mux"},
{"Stereo Mixer", "Aux Bypass Switch", "AUX In"},
/* Mono 1 Mixer */
{"Mono1 Mixer", "Mic Mono Sidetone Switch", "Input Mux"},
{"Mono1 Mixer", "Mono Playback Switch", "DAC"},
/* Mic */
{"Mic", NULL, "AIN"},
{"Mic Mux", "Internal", "Mic Int Bias"},
{"Mic Mux", "External", "Mic Ext Bias"},
{"Mic Int Bias", NULL, "MICIN"},
{"Mic Ext Bias", NULL, "MICEXT"},
{"MICOUT", NULL, "Mic Mux"},
/* Input Mux */
{"Input Mux", "Microphone", "Mic"},
{"Input Mux", "Voice DAC", "Voice DAC"},
/* Line Out */
{"LOUT", NULL, "Line Out"},
{"ROUT", NULL, "Line Out"},
{"Line Out", NULL, "Stereo Mixer"},
/* Mono 1 Out */
{"MOUT1", NULL, "Mono Out"},
{"Mono Out", NULL, "Mono1 Mixer"},
/* Mono 2 Out */
{"MOUT2", NULL, "Mono 2 Enable"},
{"Mono 2 Enable", "Switch", "Mono Out 2"},
{"Mono Out 2", NULL, "Stereo Mixer"},
{"Voice ADC", NULL, "Mono 2 Enable"},
/* Aux In */
{"AUX In", NULL, "AUX"},
/* ADC */
{"ADC", NULL, "Input Mixer"},
{"Input Mixer", "Mic Capture Switch", "Mic"},
{"Input Mixer", "Aux Capture Switch", "AUX In"},
};
static int ak4641_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
ak4641->sysclk = freq;
return 0;
}
static int ak4641_i2s_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;
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
int rate = params_rate(params), fs = 256;
u8 mode2;
if (rate)
fs = ak4641->sysclk / rate;
else
return -EINVAL;
/* set fs */
switch (fs) {
case 1024:
mode2 = (0x2 << 5);
break;
case 512:
mode2 = (0x1 << 5);
break;
case 256:
mode2 = (0x0 << 5);
break;
default:
dev_err(codec->dev, "Error: unsupported fs=%d\n", fs);
return -EINVAL;
}
snd_soc_update_bits(codec, AK4641_MODE2, (0x3 << 5), mode2);
/* Update de-emphasis filter for the new rate */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ak4641->playback_fs = rate;
ak4641_set_deemph(codec);
}
return 0;
}
static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
u8 btif;
int ret;
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
btif = (0x3 << 5);
break;
case SND_SOC_DAIFMT_LEFT_J:
btif = (0x2 << 5);
break;
case SND_SOC_DAIFMT_DSP_A: /* MSB after FRM */
btif = (0x0 << 5);
break;
case SND_SOC_DAIFMT_DSP_B: /* MSB during FRM */
btif = (0x1 << 5);
break;
default:
return -EINVAL;
}
ret = snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif);
if (ret < 0)
return ret;
return 0;
}
static int ak4641_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
u8 mode1 = 0;
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
mode1 = 0x02;
break;
case SND_SOC_DAIFMT_LEFT_J:
mode1 = 0x01;
break;
default:
return -EINVAL;
}
return snd_soc_write(codec, AK4641_MODE1, mode1);
}
static int ak4641_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
return snd_soc_update_bits(codec, AK4641_DAC, 0x20, mute ? 0x20 : 0);
}
static int ak4641_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
struct ak4641_platform_data *pdata = codec->dev->platform_data;
int ret;
switch (level) {
case SND_SOC_BIAS_ON:
/* unmute */
snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0);
break;
case SND_SOC_BIAS_PREPARE:
/* mute */
snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0x20);
break;
case SND_SOC_BIAS_STANDBY:
if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
if (pdata && gpio_is_valid(pdata->gpio_power))
gpio_set_value(pdata->gpio_power, 1);
mdelay(1);
if (pdata && gpio_is_valid(pdata->gpio_npdn))
gpio_set_value(pdata->gpio_npdn, 1);
mdelay(1);
ret = regcache_sync(ak4641->regmap);
if (ret) {
dev_err(codec->dev,
"Failed to sync cache: %d\n", ret);
return ret;
}
}
snd_soc_update_bits(codec, AK4641_PM1, 0x80, 0x80);
snd_soc_update_bits(codec, AK4641_PM2, 0x80, 0);
break;
case SND_SOC_BIAS_OFF:
snd_soc_update_bits(codec, AK4641_PM1, 0x80, 0);
if (pdata && gpio_is_valid(pdata->gpio_npdn))
gpio_set_value(pdata->gpio_npdn, 0);
if (pdata && gpio_is_valid(pdata->gpio_power))
gpio_set_value(pdata->gpio_power, 0);
regcache_mark_dirty(ak4641->regmap);
break;
}
codec->dapm.bias_level = level;
return 0;
}
#define AK4641_RATES (SNDRV_PCM_RATE_8000_48000)
#define AK4641_RATES_BT (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
SNDRV_PCM_RATE_16000)
#define AK4641_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
static const struct snd_soc_dai_ops ak4641_i2s_dai_ops = {
.hw_params = ak4641_i2s_hw_params,
.set_fmt = ak4641_i2s_set_dai_fmt,
.digital_mute = ak4641_mute,
.set_sysclk = ak4641_set_dai_sysclk,
};
static const struct snd_soc_dai_ops ak4641_pcm_dai_ops = {
.hw_params = NULL, /* rates are controlled by BT chip */
.set_fmt = ak4641_pcm_set_dai_fmt,
.digital_mute = ak4641_mute,
.set_sysclk = ak4641_set_dai_sysclk,
};
static struct snd_soc_dai_driver ak4641_dai[] = {
{
.name = "ak4641-hifi",
.id = 1,
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
.channels_max = 2,
.rates = AK4641_RATES,
.formats = AK4641_FORMATS,
},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = AK4641_RATES,
.formats = AK4641_FORMATS,
},
.ops = &ak4641_i2s_dai_ops,
.symmetric_rates = 1,
},
{
.name = "ak4641-voice",
.id = 1,
.playback = {
.stream_name = "Voice Playback",
.channels_min = 1,
.channels_max = 1,
.rates = AK4641_RATES_BT,
.formats = AK4641_FORMATS,
},
.capture = {
.stream_name = "Voice Capture",
.channels_min = 1,
.channels_max = 1,
.rates = AK4641_RATES_BT,
.formats = AK4641_FORMATS,
},
.ops = &ak4641_pcm_dai_ops,
.symmetric_rates = 1,
},
};
static struct snd_soc_codec_driver soc_codec_dev_ak4641 = {
.controls = ak4641_snd_controls,
.num_controls = ARRAY_SIZE(ak4641_snd_controls),
.dapm_widgets = ak4641_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ak4641_dapm_widgets),
.dapm_routes = ak4641_audio_map,
.num_dapm_routes = ARRAY_SIZE(ak4641_audio_map),
.set_bias_level = ak4641_set_bias_level,
.suspend_bias_off = true,
};
static const struct regmap_config ak4641_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = AK4641_BTIF,
.reg_defaults = ak4641_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(ak4641_reg_defaults),
.cache_type = REGCACHE_RBTREE,
};
static int ak4641_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct ak4641_platform_data *pdata = i2c->dev.platform_data;
struct ak4641_priv *ak4641;
int ret;
ak4641 = devm_kzalloc(&i2c->dev, sizeof(struct ak4641_priv),
GFP_KERNEL);
if (!ak4641)
return -ENOMEM;
ak4641->regmap = devm_regmap_init_i2c(i2c, &ak4641_regmap);
if (IS_ERR(ak4641->regmap))
return PTR_ERR(ak4641->regmap);
if (pdata) {
if (gpio_is_valid(pdata->gpio_power)) {
ret = gpio_request_one(pdata->gpio_power,
GPIOF_OUT_INIT_LOW, "ak4641 power");
if (ret)
goto err_out;
}
if (gpio_is_valid(pdata->gpio_npdn)) {
ret = gpio_request_one(pdata->gpio_npdn,
GPIOF_OUT_INIT_LOW, "ak4641 npdn");
if (ret)
goto err_gpio;
udelay(1); /* > 150 ns */
gpio_set_value(pdata->gpio_npdn, 1);
}
}
i2c_set_clientdata(i2c, ak4641);
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641,
ak4641_dai, ARRAY_SIZE(ak4641_dai));
if (ret != 0)
goto err_gpio2;
return 0;
err_gpio2:
if (pdata) {
if (gpio_is_valid(pdata->gpio_power))
gpio_set_value(pdata->gpio_power, 0);
if (gpio_is_valid(pdata->gpio_npdn))
gpio_free(pdata->gpio_npdn);
}
err_gpio:
if (pdata && gpio_is_valid(pdata->gpio_power))
gpio_free(pdata->gpio_power);
err_out:
return ret;
}
static int ak4641_i2c_remove(struct i2c_client *i2c)
{
struct ak4641_platform_data *pdata = i2c->dev.platform_data;
snd_soc_unregister_codec(&i2c->dev);
if (pdata) {
if (gpio_is_valid(pdata->gpio_power)) {
gpio_set_value(pdata->gpio_power, 0);
gpio_free(pdata->gpio_power);
}
if (gpio_is_valid(pdata->gpio_npdn))
gpio_free(pdata->gpio_npdn);
}
return 0;
}
static const struct i2c_device_id ak4641_i2c_id[] = {
{ "ak4641", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, ak4641_i2c_id);
static struct i2c_driver ak4641_i2c_driver = {
.driver = {
.name = "ak4641",
.owner = THIS_MODULE,
},
.probe = ak4641_i2c_probe,
.remove = ak4641_i2c_remove,
.id_table = ak4641_i2c_id,
};
module_i2c_driver(ak4641_i2c_driver);
MODULE_DESCRIPTION("SoC AK4641 driver");
MODULE_AUTHOR("Harald Welte <laforge@gnufiish.org>");
MODULE_LICENSE("GPL");