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TERES/SOFTWARE/A64-TERES/linux-a64/sound/soc/sunxi/sun8iw11_codec.c
Dimitar Gamishev f9b0e7a283 linux
2017-10-13 14:07:04 +03:00

1130 lines
35 KiB
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/*
* sound\soc\sunxi\sun8iw10_codec.c
* (C) Copyright 2014-2017
* Reuuimlla Technology Co., Ltd. <www.allwinnertech.com>
* huangxin <huangxin@allwinnertech.com>
*
* some simple description for this code
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License as
* published by the Free Software Foundation; either version 2 of
* the License, or (at your option) any later version.
*
*/
#include <linux/module.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/clk.h>
#include <linux/gpio.h>
#include <linux/io.h>
#include <linux/regulator/consumer.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include <linux/of.h>
#include <linux/of_address.h>
#include <linux/of_device.h>
#include <linux/pm.h>
#include <linux/of_gpio.h>
#include <linux/sys_config.h>
#include "sunxi_cpudai.h"
#define DRV_NAME "sunxi-internal-codec"
void __iomem *codec_digitaladress;
void __iomem *codec_analogadress;
struct spk_gpio_ spk_gpio;
static const DECLARE_TLV_DB_SCALE(dig_vol_tlv, -7424, 0, 0);
static const DECLARE_TLV_DB_SCALE(headphone_vol_tlv, -6300, 100, 0);
static const DECLARE_TLV_DB_SCALE(lineinl_to_routp_mix_vol_tlv, -450, 150, 0);
static const DECLARE_TLV_DB_SCALE(lineinr_to_loutp_mix_vol_tlv, -450, 150, 0);
static const DECLARE_TLV_DB_SCALE(fmlr_to_lroutp_mix_vol_tlv, -450, 150, 0);
static const DECLARE_TLV_DB_SCALE(lineinln_to_lroutp_mix_vol_tlv, -450, 150, 0);
static const DECLARE_TLV_DB_SCALE(mic1_to_lroutp_mix_vol_tlv, -450, 150, 0);
static const DECLARE_TLV_DB_SCALE(mic2_to_lroutp_mix_vol_tlv, -450, 150, 0);
static const DECLARE_TLV_DB_SCALE(mic1_boost_vol_tlv, 0, 300, 0);
static const DECLARE_TLV_DB_SCALE(mic2_boost_vol_tlv, 0, 300, 0);
static const DECLARE_TLV_DB_SCALE(adc_input_gain_tlv, -450, 150, 0);
struct codec_sr {
unsigned int samplerate;
int srbit;
};
static const struct codec_sr codec_sr_s[] = {
{44100, 0},
{48000, 0},
{8000, 5},
{11025, 4},
{12000, 4},
{16000, 3},
{22050, 2},
{24000, 2},
{32000, 1},
{96000, 7},
{192000, 6},
};
static struct label reg_labels[]={
LABEL(AC_DAC_DPC),
LABEL(AC_DAC_FIFOC),
LABEL(AC_DAC_FIFOS),
LABEL(AC_ADC_FIFOC),
LABEL(AC_ADC_FIFOS),
LABEL(AC_ADC_RXDATA),
LABEL(AC_ADC_TXDATA),
LABEL(AC_DAC_CNT),
LABEL(AC_ADC_CNT),
LABEL(AC_DAC_DG),
LABEL(AC_ADC_DG),
LABEL(AC_HMIC_CTRL),
LABEL(AC_HMIC_DATA),
LABEL(HP_VOLC),
LABEL(LOMIXSC),
LABEL(ROMIXSC),
LABEL(DAC_PA_SRC),
LABEL(LINEIN_GCTRL),
LABEL(FM_GCTRL),
LABEL(MICIN_GCTRL),
LABEL(PAEN_HP_CTRL),
LABEL(PHONEOUT_CTRL),
LABEL(MIC2G_LINEEN_CTRL),
LABEL(MIC1G_MICBIAS_CTRL),
LABEL(LADCMIXSC),
LABEL(RADCMIXSC),
LABEL(PA_POP_CTR),
LABEL(ADC_AP_EN),
LABEL(ADDA_APTO),
LABEL(ADDA_APT1),
LABEL(ADDA_APT2),
LABEL(DA16_CAL_CTRL),
LABEL(BIAS_DA16_CAL_CTR),
LABEL(DA16_CALI_DATA),
LABEL(BIAS_CALI_DATA),
LABEL(BIAS_CALI_SET),
LABEL_END,
};
/*
*enable the codec function which should be enable during system init.
*/
static int codec_init(struct sunxi_codec *sunxi_internal_codec)
{
struct snd_soc_codec *codec = sunxi_internal_codec->codec;
if (sunxi_internal_codec->hp_dirused) {
snd_soc_update_bits(codec, PAEN_HP_CTRL, (0x3<<HPCOM_FC), (0x3<<HPCOM_FC));
snd_soc_update_bits(codec, PAEN_HP_CTRL, (0x1<<HPCOM_PT), (0x1<<HPCOM_PT));
} else {
snd_soc_update_bits(codec, PAEN_HP_CTRL, (0x3<<HPCOM_FC), (0x0<<HPCOM_FC));
snd_soc_update_bits(codec, PAEN_HP_CTRL, (0x1<<HPCOM_PT), (0x0<<HPCOM_PT));
}
snd_soc_update_bits(codec, DAC_PA_SRC, (0x1<<LHPPAMUTE), (0x0<<LHPPAMUTE));
snd_soc_update_bits(codec, DAC_PA_SRC, (0x1<<RHPPAMUTE), (0x0<<RHPPAMUTE));
/*when TX FIFO available room less than or equal N,
* DRQ Requeest will be de-asserted.
*/
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x3<<DAC_DRQ_CLR_CNT), (0x3<<DAC_DRQ_CLR_CNT));
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x1<<FIFO_FLUSH), (0x1<<FIFO_FLUSH));
/*
* 0:64-Tap FIR
* 1:32-Tap FIR
*/
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x1<<FIR_VER), (0x0<<FIR_VER));
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x1<<ADC_FIFO_FLUSH), (0x1<<ADC_FIFO_FLUSH));
return 0;
}
static int late_enable_dac(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct sunxi_codec *sunxi_internal_codec = snd_soc_codec_get_drvdata(codec);
mutex_lock(&sunxi_internal_codec->dac_mutex);
pr_debug("..dac power state change.\n");
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (sunxi_internal_codec->dac_enable == 0) {
snd_soc_update_bits(codec, AC_DAC_DPC, (0x1<<EN_DAC), (0x1<<EN_DAC));
}
sunxi_internal_codec->dac_enable++;
break;
case SND_SOC_DAPM_POST_PMD:
if (sunxi_internal_codec->dac_enable > 0) {
sunxi_internal_codec->dac_enable--;
if (sunxi_internal_codec->dac_enable == 0) {
snd_soc_update_bits(codec, AC_DAC_DPC, (0x1<<EN_DAC), (0x0<<EN_DAC));
}
}
break;
}
mutex_unlock(&sunxi_internal_codec->dac_mutex);
return 0;
}
static int late_enable_adc(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct sunxi_codec *sunxi_internal_codec = snd_soc_codec_get_drvdata(codec);
mutex_lock(&sunxi_internal_codec->adc_mutex);
pr_debug("..adc power state change.\n");
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (sunxi_internal_codec->adc_enable == 0) {
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x1<<EN_AD), (0x1<<EN_AD));
}
sunxi_internal_codec->adc_enable++;
break;
case SND_SOC_DAPM_POST_PMD:
if (sunxi_internal_codec->adc_enable > 0) {
sunxi_internal_codec->adc_enable--;
if (sunxi_internal_codec->adc_enable == 0) {
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x1<<EN_AD), (0x0<<EN_AD));
}
}
break;
}
mutex_unlock(&sunxi_internal_codec->adc_mutex);
return 0;
}
static int ac_headphone_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
struct snd_soc_codec *codec = w->codec;
pr_debug("..headphone power state change.\n");
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/*open*/
snd_soc_update_bits(codec, PAEN_HP_CTRL, (0x1<<HPPAEN), (0x1<<HPPAEN));
msleep(10);
snd_soc_update_bits(codec, DAC_PA_SRC, (0x1<<LHPPAMUTE), (0x1<<LHPPAMUTE));
snd_soc_update_bits(codec, DAC_PA_SRC, (0x1<<RHPPAMUTE), (0x1<<RHPPAMUTE));
break;
case SND_SOC_DAPM_PRE_PMD:
/*close*/
snd_soc_update_bits(codec, PAEN_HP_CTRL, (0x1<<HPPAEN), (0x0<<HPPAEN));
snd_soc_update_bits(codec, DAC_PA_SRC, (0x1<<LHPPAMUTE), (0x0<<LHPPAMUTE));
snd_soc_update_bits(codec, DAC_PA_SRC, (0x1<<RHPPAMUTE), (0x0<<RHPPAMUTE));
break;
}
return 0;
}
static const struct snd_kcontrol_new sunxi_codec_controls[] = {
SOC_SINGLE_TLV("digital volume", AC_DAC_DPC, DVOL, 0x3f, 0, dig_vol_tlv),
/*analog control*/
SOC_SINGLE_TLV("headphone volume", HP_VOLC, HPVOL, 0x3f, 0, headphone_vol_tlv),
SOC_SINGLE_TLV("mic1 to lr output mixer control", MICIN_GCTRL, MIC1_GAIN, 0x7, 0, mic1_to_lroutp_mix_vol_tlv),
SOC_SINGLE_TLV("mic2 to lr output mixer control", MICIN_GCTRL, MIC2_GAIN, 0x7, 0, mic2_to_lroutp_mix_vol_tlv),
SOC_SINGLE_TLV("MIC1 boost AMP gain control", MIC1G_MICBIAS_CTRL, MIC1_BOOST, 0x7, 0, mic1_boost_vol_tlv),
SOC_SINGLE_TLV("MIC2 boost AMP gain control", MIC2G_LINEEN_CTRL, MIC2BOOST, 0x7, 0, mic2_boost_vol_tlv),
/*ADC*/
SOC_SINGLE_TLV("ADC input gain control", ADC_AP_EN, ADCG, 0x7, 0, adc_input_gain_tlv),
SOC_SINGLE_TLV("lineinl to r_output mixer gain", LINEIN_GCTRL, LINEINRG, 0x7, 0, lineinl_to_routp_mix_vol_tlv),
SOC_SINGLE_TLV("lineinr to l_output mixer gain", LINEIN_GCTRL, LINEINLG, 0x7, 0, lineinr_to_loutp_mix_vol_tlv),
SOC_SINGLE_TLV("lineinln to lr_output mixer gain", FM_GCTRL, LINEING, 0x7, 0, lineinln_to_lroutp_mix_vol_tlv),
SOC_SINGLE_TLV("FMlr to lr output mixer gain", FM_GCTRL, FMG, 0x7, 0, fmlr_to_lroutp_mix_vol_tlv),
};
/*
* output mixer source select
* analog:0x01:defined left output mixer
*/
static const struct snd_kcontrol_new ac_loutmix_controls[] = {
SOC_DAPM_SINGLE("DACR Switch", LOMIXSC, LMIX_RDAC, 1, 0),
SOC_DAPM_SINGLE("DACL Switch", LOMIXSC, LMIX_LDAC, 1, 0),
SOC_DAPM_SINGLE("FML Switch", LOMIXSC, LMIX_FML, 1, 0),
SOC_DAPM_SINGLE("LINEINL Switch", LOMIXSC, LMIX_LINEINL, 1, 0),
SOC_DAPM_SINGLE("LINEINLR Switch", LOMIXSC, LMIX_LINEINLR, 1, 0),
SOC_DAPM_SINGLE("MIC2Booststage Switch", LOMIXSC, LMIX_MIC2_BST, 1, 0),
SOC_DAPM_SINGLE("MIC1Booststage Switch", LOMIXSC, LMIX_MIC1_BST, 1, 0),
};
/*
* analog:0x02:defined right output mixer
*/
static const struct snd_kcontrol_new ac_routmix_controls[] = {
SOC_DAPM_SINGLE("DACL Switch", ROMIXSC, RMIX_LDAC, 1, 0),
SOC_DAPM_SINGLE("DACR Switch", ROMIXSC, RMIX_RDAC, 1, 0),
SOC_DAPM_SINGLE("FMR Switch", ROMIXSC, RMIX_FMR, 1, 0),
SOC_DAPM_SINGLE("LINEINR Switch", ROMIXSC, RMIX_LINEINR, 1, 0),
SOC_DAPM_SINGLE("LINEINLR Switch", ROMIXSC, RMIX_LINEINLR, 1, 0),
SOC_DAPM_SINGLE("MIC2Booststage Switch", ROMIXSC, RMIX_MIC2_BST, 1, 0),
SOC_DAPM_SINGLE("MIC1Booststage Switch", ROMIXSC, RMIX_MIC1_BST, 1, 0),
};
/*
* hp source select
* 0x0a:headphone input source
*/
static const char *ac_hp_r_func_sel[] = {
"DACR HPR Switch", "Right Analog Mixer HPR Switch"};
static const struct soc_enum ac_hp_r_func_enum =
SOC_ENUM_SINGLE(DAC_PA_SRC, RHPIS, 2, ac_hp_r_func_sel);
static const struct snd_kcontrol_new ac_hp_r_func_controls =
SOC_DAPM_ENUM("HP_R Mux", ac_hp_r_func_enum);
static const char *ac_hp_l_func_sel[] = {
"DACL HPL Switch", "Left Analog Mixer HPL Switch"};
static const struct soc_enum ac_hp_l_func_enum =
SOC_ENUM_SINGLE(DAC_PA_SRC, LHPIS, 2, ac_hp_l_func_sel);
static const struct snd_kcontrol_new ac_hp_l_func_controls =
SOC_DAPM_ENUM("HP_L Mux", ac_hp_l_func_enum);
/*
* LADC SOURCE SELECT
* 0x0c:defined left input adc mixer
*/
static const struct snd_kcontrol_new ac_ladcmix_controls[] = {
SOC_DAPM_SINGLE("r_output mixer Switch", LADCMIXSC, LADC_ROUT_MIX, 1, 0),
SOC_DAPM_SINGLE("l_output mixer Switch", LADCMIXSC, LADC_LOUT_MIX, 1, 0),
SOC_DAPM_SINGLE("fml mixer Switch", LADCMIXSC, LADC_FML, 1, 0),
SOC_DAPM_SINGLE("lineinl Switch", LADCMIXSC, LADC_LINEINL, 1, 0),
SOC_DAPM_SINGLE("lineinlr Switch", LADCMIXSC, LADC_LINEINLR, 1, 0),
SOC_DAPM_SINGLE("MIC2 boost Switch", LADCMIXSC, LADC_MIC2_BST, 1, 0),
SOC_DAPM_SINGLE("MIC1 boost Switch", LADCMIXSC, LADC_MIC1_BST, 1, 0),
};
/*
* RADC SOURCE SELECT
* 0x0d:defined right input adc mixer
*/
static const struct snd_kcontrol_new ac_radcmix_controls[] = {
SOC_DAPM_SINGLE("l_output mixer Switch", RADCMIXSC, RADC_LOUT_MIX, 1, 0),
SOC_DAPM_SINGLE("r_output mixer Switch", RADCMIXSC, RADC_ROUT_MIX, 1, 0),
SOC_DAPM_SINGLE("fmr mixer Switch", RADCMIXSC, RADC_FMR, 1, 0),
SOC_DAPM_SINGLE("lineinr Switch", RADCMIXSC, RADC_LINER, 1, 0),
SOC_DAPM_SINGLE("lineinlr Switch", RADCMIXSC, RADC_LINEINLR, 1, 0),
SOC_DAPM_SINGLE("MIC2 boost Switch", RADCMIXSC, RADC_MIC2_BST, 1, 0),
SOC_DAPM_SINGLE("MIC1 boost Switch", RADCMIXSC, RADC_MIC1_BST, 1, 0),
};
/*
* PHONEOUT SOURCE SELECT
* 0x0d:defined phoneout mixer
*/
static const struct snd_kcontrol_new ac_phoneoutmix_controls[] = {
SOC_DAPM_SINGLE("l_output mixer Switch", PHONEOUT_CTRL, PHONEOUTS0, 1, 0),
SOC_DAPM_SINGLE("r_output mixer Switch", PHONEOUT_CTRL, PHONEOUTS1, 1, 0),
SOC_DAPM_SINGLE("mic2 boost Switch", PHONEOUT_CTRL, PHONEOUTS2, 1, 0),
SOC_DAPM_SINGLE("mic1 boost Switch", PHONEOUT_CTRL, PHONEOUTS3, 1, 0),
};
/*0x0b:mic2 source select*/
static const char *mic2src_text[] = {"MIC3","MIC2"};
static const struct soc_enum mic2src_enum =
SOC_ENUM_SINGLE(MIC1G_MICBIAS_CTRL, MIC2_SS, 2, mic2src_text);
static const struct snd_kcontrol_new mic2src_mux =
SOC_DAPM_ENUM("MIC2 SRC", mic2src_enum);
/*built widget*/
static const struct snd_soc_dapm_widget ac_dapm_widgets[] = {
SND_SOC_DAPM_AIF_IN("DAC_L", "Playback", 0, DAC_PA_SRC, DACALEN, 0),
SND_SOC_DAPM_AIF_IN("DAC_R", "Playback", 0, DAC_PA_SRC, DACAREN, 0),
/*0x0a*/
SND_SOC_DAPM_MIXER_E("Left Output Mixer", DAC_PA_SRC, LMIXEN, 0,
ac_loutmix_controls, ARRAY_SIZE(ac_loutmix_controls), late_enable_dac, SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MIXER_E("Right Output Mixer", DAC_PA_SRC, RMIXEN, 0,
ac_routmix_controls, ARRAY_SIZE(ac_routmix_controls), late_enable_dac, SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MUX("HP_R Mux", SND_SOC_NOPM, 0, 0, &ac_hp_r_func_controls),
SND_SOC_DAPM_MUX("HP_L Mux", SND_SOC_NOPM, 0, 0, &ac_hp_l_func_controls),
/*output widget*/
SND_SOC_DAPM_OUTPUT("HPOUTL"),
SND_SOC_DAPM_OUTPUT("HPOUTR"),
/*headphone*/
SND_SOC_DAPM_HP("Headphone", ac_headphone_event),
/*
* INPUT widget
* 0x0e Headset Microphone Bias Control Register
*/
/*Microphone Bias Control Register*/
SND_SOC_DAPM_MICBIAS("MainMic Bias", MIC1G_MICBIAS_CTRL, MMICBIASEN, 0),
SND_SOC_DAPM_INPUT("MIC1P"),
SND_SOC_DAPM_INPUT("MIC1N"),
SND_SOC_DAPM_INPUT("MIC2"),
SND_SOC_DAPM_INPUT("MIC3"),
SND_SOC_DAPM_INPUT("LINEINR"),
SND_SOC_DAPM_INPUT("LINEINL"),
SND_SOC_DAPM_INPUT("LINEINLR"),
SND_SOC_DAPM_INPUT("FML"),
SND_SOC_DAPM_INPUT("FMR"),
/*mic1 reference*/
SND_SOC_DAPM_PGA("MIC1 PGA", MIC1G_MICBIAS_CTRL, MIC1_AMPEN, 0, NULL, 0),
/*0x0a mic2 reference*/
SND_SOC_DAPM_PGA("MIC2 PGA", MIC2G_LINEEN_CTRL, MIC2AMPEN, 0, NULL, 0),
/*0x0b: mic2 source select*/
SND_SOC_DAPM_MUX("MIC2 SRC", SND_SOC_NOPM, 0, 0, &mic2src_mux),
SND_SOC_DAPM_MIXER_E("LADC input Mixer", ADC_AP_EN, ADCLEN, 0,
ac_ladcmix_controls, ARRAY_SIZE(ac_ladcmix_controls),late_enable_adc, SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MIXER_E("RADC input Mixer", ADC_AP_EN, ADCREN, 0,
ac_radcmix_controls, ARRAY_SIZE(ac_radcmix_controls),late_enable_adc, SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_OUT("ADC_L", "Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("ADC_R", "Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MIXER("phone Output Mixer", PHONEOUT_CTRL, PHONEOUTEN, 0,
ac_phoneoutmix_controls, ARRAY_SIZE(ac_phoneoutmix_controls)),
SND_SOC_DAPM_OUTPUT("PHONEOUTP"),
SND_SOC_DAPM_OUTPUT("PHONEOUTN"),
};
static const struct snd_soc_dapm_route ac_dapm_routes[] = {
/*PLAYBACK*/
{"Left Output Mixer", "DACL Switch", "DAC_L"},
{"Left Output Mixer", "DACR Switch", "DAC_R"},
{"Left Output Mixer", "FML Switch", "FML"},
{"Left Output Mixer", "LINEINL Switch", "LINEINL"},
{"Left Output Mixer", "LINEINLR Switch", "LINEINLR"},
{"Left Output Mixer", "MIC2Booststage Switch", "MIC2 PGA"},
{"Left Output Mixer", "MIC1Booststage Switch", "MIC1 PGA"},
{"Right Output Mixer", "DACR Switch", "DAC_R"},
{"Right Output Mixer", "DACL Switch", "DAC_L"},
{"Right Output Mixer", "FMR Switch", "FMR"},
{"Right Output Mixer", "LINEINR Switch", "LINEINR"},
{"Right Output Mixer", "LINEINLR Switch", "LINEINLR"},
{"Right Output Mixer", "MIC2Booststage Switch", "MIC2 PGA"},
{"Right Output Mixer", "MIC1Booststage Switch", "MIC1 PGA"},
/*hp mux*/
{"HP_R Mux", "DACR HPR Switch", "DAC_R"},
{"HP_R Mux", "Right Analog Mixer HPR Switch", "Right Output Mixer"},
{"HP_L Mux", "DACL HPL Switch", "DAC_L"},
{"HP_L Mux", "Left Analog Mixer HPL Switch", "Left Output Mixer"},
/*hp endpoint*/
{"HPOUTR", NULL, "HP_R Mux"},
{"HPOUTL", NULL, "HP_L Mux"},
{"Headphone", NULL, "HPOUTR"},
{"Headphone", NULL, "HPOUTL"},
/*CAPTURE*/
{"MIC1 PGA", NULL, "MIC1P"},
{"MIC1 PGA", NULL, "MIC1N"},
{"MIC2 SRC", "MIC2", "MIC2"},
{"MIC2 SRC", "MIC3", "MIC3"},
{"MIC2 PGA", NULL, "MIC2 SRC"},
/*LADC SOURCE mixer*/
{"LADC input Mixer", "MIC1 boost Switch", "MIC1 PGA"},
{"LADC input Mixer", "MIC2 boost Switch", "MIC2 PGA"},
{"LADC input Mixer", "lineinlr Switch", "LINEINLR"},
{"LADC input Mixer", "lineinl Switch", "LINEINL"},
{"LADC input Mixer", "fml mixer Switch", "FML"},
{"LADC input Mixer", "l_output mixer Switch", "Left Output Mixer"},
{"LADC input Mixer", "r_output mixer Switch", "Right Output Mixer"},
/*RADC SOURCE mixer*/
{"RADC input Mixer", "MIC1 boost Switch", "MIC1 PGA"},
{"RADC input Mixer", "MIC2 boost Switch", "MIC2 PGA"},
{"RADC input Mixer", "lineinlr Switch", "LINEINLR"},
{"RADC input Mixer", "lineinr Switch", "LINEINR"},
{"RADC input Mixer", "fmr mixer Switch", "FMR"},
{"RADC input Mixer", "r_output mixer Switch", "Right Output Mixer"},
{"RADC input Mixer", "l_output mixer Switch", "Left Output Mixer"},
/*ADC--ADCMUX*/
{"ADC_L", NULL, "LADC input Mixer"},
{"ADC_R", NULL, "RADC input Mixer"},
/*phoneout*/
{"phone Output Mixer", "MIC1 boost Switch","MIC1 PGA"},
{"phone Output Mixer", "mic2 boost Switch","MIC2 PGA"},
{"phone Output Mixer", "r_output mixer Switch","Right Output Mixer"},
{"phone Output Mixer", "l_output mixer Switch","Left Output Mixer"},
{"PHONEOUTP", NULL,"phone Output Mixer"},
{"PHONEOUTN", NULL,"phone Output Mixer"},
};
static int codec_start(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai)
{
return 0;
}
static int codec_mute(struct snd_soc_dai *codec_dai, int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct sunxi_codec *sunxi_internal_codec = snd_soc_codec_get_drvdata(codec);
if(sunxi_internal_codec->spkenable == true)
msleep(sunxi_internal_codec->pa_sleep_time);
return 0;
}
static void codec_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai)
{
}
static int codec_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
/*enable dac drq*/
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x1<<DAC_DRQ_EN), (0x1<<DAC_DRQ_EN));
return 0;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x1<<DAC_DRQ_EN), (0x0<<DAC_DRQ_EN));
return 0;
default:
return -EINVAL;
}
} else {
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x1<<ADC_DRQ_EN), (0x1<<ADC_DRQ_EN));
return 0;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x1<<ADC_DRQ_EN), (0x0<<ADC_DRQ_EN));
return 0;
default:
pr_err("error:%s,%d\n", __func__, __LINE__);
return -EINVAL;
}
}
return 0;
}
static int codec_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *codec_dai)
{
int i = 0;
struct snd_soc_codec *codec = codec_dai->codec;
for (i = 0; i < ARRAY_SIZE(codec_sr_s); i++) {
if (codec_sr_s[i].samplerate == params_rate(params)) {
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x7<<DAC_FS), (codec_sr_s[i].srbit<<DAC_FS));
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x7<<DAC_FS), (codec_sr_s[i].srbit<<DAC_FS));
} else {
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x7<<ADFS), (codec_sr_s[i].srbit<<ADFS));
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x7<<ADFS), (codec_sr_s[i].srbit<<ADFS));
}
break;
}
}
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S24_LE:
case SNDRV_PCM_FORMAT_S32_LE:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/*set TX FIFO MODE:24bit*/
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x3<<FIFO_MODE), (0x2<<FIFO_MODE));
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x1<<TX_SAMPLE_BITS), (0x1<<TX_SAMPLE_BITS));
} else {
/*set RX FIFO MODE:24bit*/
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x1<<RX_FIFO_MODE), (0x0<<RX_FIFO_MODE));
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x1<<RX_SAMPLE_BITS), (0x1<<RX_SAMPLE_BITS));
}
break;
case SNDRV_PCM_FORMAT_S16_LE:
default:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/*set TX FIFO MODE:16bit*/
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x3<<FIFO_MODE), (0x3<<FIFO_MODE));
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x1<<TX_SAMPLE_BITS), (0x0<<TX_SAMPLE_BITS));
} else {
/*set RX FIFO MODE:16bit*/
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x1<<RX_FIFO_MODE), (0x1<<RX_FIFO_MODE));
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x1<<RX_SAMPLE_BITS), (0x0<<RX_SAMPLE_BITS));
}
break;
}
if (params_channels(params)==1) {
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x1<<DAC_MONO_EN), (0x1<<DAC_MONO_EN));
} else {
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x1<<ADC_MONO_EN), (0x1<<ADC_MONO_EN));
}
} else {
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
snd_soc_update_bits(codec, AC_DAC_FIFOC, (0x1<<DAC_MONO_EN), (0x0<<DAC_MONO_EN));
} else {
snd_soc_update_bits(codec, AC_ADC_FIFOC, (0x1<<ADC_MONO_EN), (0x0<<ADC_MONO_EN));
}
}
return 0;
}
static int codec_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
return 0;
}
static int codec_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 sunxi_codec *sunxi_internal_codec = snd_soc_codec_get_drvdata(codec);
if (clk_set_rate(sunxi_internal_codec->pllclk, freq)) {
pr_err("[audio-cpudai]try to set the pll clk rate failed!\n");
}
return 0;
}
static int codec_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
codec->dapm.bias_level = level;
return 0;
}
static const struct snd_soc_dai_ops codec_dai_ops = {
.startup = codec_start,
.set_fmt = codec_set_dai_fmt,
.hw_params = codec_hw_params,
.shutdown = codec_shutdown,
.digital_mute = codec_mute,
.set_sysclk = codec_set_dai_sysclk,
.trigger = codec_trigger,
};
static struct snd_soc_dai_driver codec_dai[] = {
{
.name = "sun8iw11codec",
.id = 1,
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
},
.ops = &codec_dai_ops,
},
};
static const struct snd_soc_component_driver sunxi_i2s_component = {
.name = DRV_NAME,
};
static int codec_soc_probe(struct snd_soc_codec *codec)
{
int ret = 0;
struct snd_soc_dapm_context *dapm = &codec->dapm;
struct sunxi_codec *sunxi_internal_codec = snd_soc_codec_get_drvdata(codec);
sunxi_internal_codec->codec = codec;
sunxi_internal_codec->dac_enable = 0;
sunxi_internal_codec->adc_enable = 0;
mutex_init(&sunxi_internal_codec->dac_mutex);
mutex_init(&sunxi_internal_codec->adc_mutex);
/* Add virtual switch */
ret = snd_soc_add_codec_controls(codec, sunxi_codec_controls,
ARRAY_SIZE(sunxi_codec_controls));
if (ret) {
pr_err("[audio-codec] Failed to register audio mode control, "
"will continue without it.\n");
}
snd_soc_dapm_new_controls(dapm, ac_dapm_widgets, ARRAY_SIZE(ac_dapm_widgets));
snd_soc_dapm_add_routes(dapm, ac_dapm_routes, ARRAY_SIZE(ac_dapm_routes));
codec_init(sunxi_internal_codec);
return 0;
}
int audio_gpio_iodisable(u32 gpio)
{
char pin_name[8];
u32 config,ret;
sunxi_gpio_to_name(gpio, pin_name);
config = (((7) << 16) | (0 & 0xFFFF));
ret = pin_config_set(SUNXI_PINCTRL, pin_name, config);
return ret;
}
static int codec_suspend(struct snd_soc_codec *codec)
{
struct sunxi_codec *sunxi_internal_codec = snd_soc_codec_get_drvdata(codec);
pr_debug("[audio codec]:suspend start.\n");
if (spk_gpio.cfg) {
audio_gpio_iodisable(spk_gpio.gpio);
}
if (sunxi_internal_codec->moduleclk != NULL) {
clk_disable(sunxi_internal_codec->moduleclk);
}
if (sunxi_internal_codec->pllclk != NULL) {
clk_disable(sunxi_internal_codec->pllclk);
}
if (sunxi_internal_codec->vol_supply.cpvdd) {
regulator_disable(sunxi_internal_codec->vol_supply.cpvdd);
}
if (sunxi_internal_codec->vol_supply.avcc) {
regulator_disable(sunxi_internal_codec->vol_supply.avcc);
}
pr_debug("[audio codec]:suspend end..\n");
return 0;
}
static int codec_resume(struct snd_soc_codec *codec)
{
int ret ;
struct sunxi_codec *sunxi_internal_codec = snd_soc_codec_get_drvdata(codec);
pr_debug("[audio codec]:resume start\n");
if (sunxi_internal_codec->vol_supply.cpvdd) {
ret = regulator_enable(sunxi_internal_codec->vol_supply.cpvdd);
if (ret) {
pr_err("[%s]: cpvdd:regulator_enable() failed!\n",__func__);
}
}
if (sunxi_internal_codec->vol_supply.avcc) {
ret = regulator_enable(sunxi_internal_codec->vol_supply.avcc);
if (ret) {
pr_err("[%s]: avcc:regulator_enable() failed!\n",__func__);
}
}
if (sunxi_internal_codec->pllclk != NULL) {
if (clk_prepare_enable(sunxi_internal_codec->pllclk)) {
pr_err("open sunxi_internal_codec->pllclk failed! line = %d\n", __LINE__);
}
}
if (sunxi_internal_codec->moduleclk != NULL) {
if (clk_prepare_enable(sunxi_internal_codec->moduleclk)) {
pr_err("open sunxi_internal_codec->moduleclk failed! line = %d\n", __LINE__);
}
}
codec_init(sunxi_internal_codec);
if (spk_gpio.cfg) {
gpio_direction_output(spk_gpio.gpio, 1);
gpio_set_value(spk_gpio.gpio, 0);
}
pr_debug("[audio codec]:resume end..\n");
return 0;
}
/* power down chip */
static int codec_soc_remove(struct snd_soc_codec *codec)
{
return 0;
}
static unsigned int codec_read(struct snd_soc_codec *codec,
unsigned int reg)
{
struct sunxi_codec *sunxi_internal_codec = snd_soc_codec_get_drvdata(codec);
if (reg <= 0x1e) {
/*analog reg*/
return read_prcm_wvalue(reg,sunxi_internal_codec->codec_abase);
} else {
/*digital reg*/
return codec_rdreg(sunxi_internal_codec->codec_dbase + reg);
}
}
static int codec_write(struct snd_soc_codec *codec,
unsigned int reg, unsigned int value)
{
struct sunxi_codec *sunxi_internal_codec = snd_soc_codec_get_drvdata(codec);
if (reg <= 0x1e) {
/*analog reg*/
write_prcm_wvalue(reg, value,sunxi_internal_codec->codec_abase);
} else {
/*digital reg*/
codec_wrreg(sunxi_internal_codec->codec_dbase + reg, value);
}
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_codec = {
.probe = codec_soc_probe,
.remove = codec_soc_remove,
.suspend = codec_suspend,
.resume = codec_resume,
.set_bias_level = codec_set_bias_level,
.read = codec_read,
.write = codec_write,
.ignore_pmdown_time = 1,
};
static ssize_t show_audio_reg(struct device *dev, struct device_attribute *attr,
char *buf)
{
int count = 0;
int i = 0;
int reg_group =0;
count += sprintf(buf, "dump audio reg:\n");
while (reg_labels[i].name != NULL) {
if (reg_labels[i].value == 0) {
reg_group++;
}
if (reg_group == 1) {
count +=sprintf(buf + count, "%s 0x%p: 0x%x\n", reg_labels[i].name,
(codec_digitaladress + reg_labels[i].value),
readl(codec_digitaladress + reg_labels[i].value) );
} else if (reg_group == 2) {
count +=sprintf(buf + count, "%s 0x%x: 0x%x\n", reg_labels[i].name,
(reg_labels[i].value),
read_prcm_wvalue(reg_labels[i].value,codec_analogadress) );
}
i++;
}
return count;
}
/* ex:
*param 1: 0 read;1 write
*param 2: 1 digital reg; 2 analog reg
*param 3: reg value;
*param 4: write value;
read:
echo 0,1,0x00> audio_reg
echo 0,2,0x00> audio_reg
write:
echo 1,1,0x00,0xa > audio_reg
echo 1,2,0x00,0xff > audio_reg
*/
static ssize_t store_audio_reg(struct device *dev, struct device_attribute *attr,
const char *buf, size_t count)
{
int ret;
int rw_flag;
int reg_val_read;
int input_reg_val =0;
int input_reg_group =0;
int input_reg_offset =0;
ret = sscanf(buf, "%d,%d,0x%x,0x%x", &rw_flag,&input_reg_group, &input_reg_offset, &input_reg_val);
printk("ret:%d, reg_group:%d, reg_offset:%d, reg_val:0x%x\n", ret, input_reg_group, input_reg_offset, input_reg_val);
if (!(input_reg_group ==1 || input_reg_group ==2)) {
pr_err("not exist reg group\n");
ret = count;
goto out;
}
if (!(rw_flag ==1 || rw_flag ==0)) {
pr_err("not rw_flag\n");
ret = count;
goto out;
}
if (input_reg_group == 1) {
if (rw_flag) {
writel(input_reg_val, codec_digitaladress + input_reg_offset);
} else {
reg_val_read = readl(codec_digitaladress + input_reg_offset);
printk("\n\n Reg[0x%x] : 0x%x\n\n",input_reg_offset,reg_val_read);
}
} else if (input_reg_group == 2) {
if (rw_flag) {
write_prcm_wvalue(input_reg_offset, input_reg_val & 0xff,codec_analogadress);
} else {
reg_val_read = read_prcm_wvalue(input_reg_offset,codec_analogadress);
printk("\n\n Reg[0x%x] : 0x%x\n\n",input_reg_offset,reg_val_read);
}
}
ret = count;
out:
return ret;
}
static DEVICE_ATTR(audio_reg, 0644, show_audio_reg, store_audio_reg);
static struct attribute *audio_debug_attrs[] = {
&dev_attr_audio_reg.attr,
NULL,
};
static struct attribute_group audio_debug_attr_group = {
.name = "audio_reg_debug",
.attrs = audio_debug_attrs,
};
static const struct of_device_id sunxi_codec_of_match[] = {
{ .compatible = "allwinner,sunxi-internal-codec", },
{},
};
static int __init sunxi_internal_codec_probe(struct platform_device *pdev)
{
s32 ret = 0;
u32 temp_val = 0;
struct resource res;
struct gpio_config config;
const struct of_device_id *device;
struct sunxi_codec *sunxi_internal_codec;
struct device_node *node = pdev->dev.of_node;
if (!node) {
dev_err(&pdev->dev,
"can not get dt node for this device.\n");
ret = -EINVAL;
goto err0;
}
sunxi_internal_codec = devm_kzalloc(&pdev->dev, sizeof(struct sunxi_codec), GFP_KERNEL);
if (!sunxi_internal_codec) {
dev_err(&pdev->dev, "Can't allocate sunxi_codec\n");
ret = -ENOMEM;
goto err0;
}
dev_set_drvdata(&pdev->dev, sunxi_internal_codec);
device = of_match_device(sunxi_codec_of_match, &pdev->dev);
if (!device) {
ret = -ENODEV;
goto err1;
}
ret = of_address_to_resource(node, 0, &res);
if (ret) {
dev_err(&pdev->dev, "Can't parse device node resource\n");
return -ENODEV;
}
sunxi_internal_codec->pllclk = of_clk_get(node, 0);
sunxi_internal_codec->moduleclk= of_clk_get(node, 1);
if (IS_ERR(sunxi_internal_codec->pllclk) || IS_ERR(sunxi_internal_codec->moduleclk)){
dev_err(&pdev->dev, "[audio-cpudai]Can't get cpudai clocks\n");
if (IS_ERR(sunxi_internal_codec->pllclk))
ret = PTR_ERR(sunxi_internal_codec->pllclk);
else
ret = PTR_ERR(sunxi_internal_codec->moduleclk);
goto err1;
} else {
if (clk_set_parent(sunxi_internal_codec->moduleclk, sunxi_internal_codec->pllclk)) {
pr_err("try to set parent of sunxi_spdif->moduleclk to sunxi_spdif->pllclk failed! line = %d\n",__LINE__);
}
clk_prepare_enable(sunxi_internal_codec->pllclk);
clk_prepare_enable(sunxi_internal_codec->moduleclk);
}
#if 0
/*voltage*/
sunxi_internal_codec->vol_supply.cpvdd = regulator_get(NULL, "vcc-cpvdd");/*HPVCC*/
if (!sunxi_internal_codec->vol_supply.cpvdd) {
pr_err("get audio cpvdd failed\n");
ret = -EFAULT;
goto err1;
} else {
ret = regulator_enable(sunxi_internal_codec->vol_supply.cpvdd);
if (ret) {
pr_err("[%s]: cpvdd:regulator_enable() failed!\n",__func__);
goto err1;
}
}
sunxi_internal_codec->vol_supply.avcc = regulator_get(NULL, "vcc-avcc");
if (!sunxi_internal_codec->vol_supply.avcc) {
pr_err("[%s]:get audio avcc failed\n",__func__);
ret = -EFAULT;
goto err1;
} else {
ret = regulator_enable(sunxi_internal_codec->vol_supply.avcc);
if (ret) {
pr_err("[%s]: avcc:regulator_enable() failed!\n",__func__);
goto err1;
}
}
#endif
sunxi_internal_codec->codec_abase = NULL;
sunxi_internal_codec->codec_dbase = NULL;
sunxi_internal_codec->codec_dbase = of_iomap(node, 0);
if (NULL == sunxi_internal_codec->codec_dbase) {
pr_err("[audio-codec]Can't map codec digital registers\n");
} else {
codec_digitaladress = sunxi_internal_codec->codec_dbase;
}
sunxi_internal_codec->codec_abase = of_iomap(node, 1);
if (NULL == sunxi_internal_codec->codec_abase) {
pr_err("[audio-codec]Can't map codec analog registers\n");
} else {
codec_analogadress = sunxi_internal_codec->codec_abase;
}
/*initial speaker gpio */
spk_gpio.gpio = of_get_named_gpio_flags(node, "gpio-spk", 0, (enum of_gpio_flags *)&config);
if (!gpio_is_valid(spk_gpio.gpio)) {
pr_err("failed to get gpio-spk gpio from dts,spk_gpio:%d\n",spk_gpio.gpio);
spk_gpio.cfg = 0;
} else {
ret = devm_gpio_request(&pdev->dev, spk_gpio.gpio, "SPK");
if (ret) {
spk_gpio.cfg = 0;
pr_err("failed to request gpio-spk gpio\n");
goto err1;
} else {
spk_gpio.cfg = 1;
gpio_direction_output(spk_gpio.gpio, 1);
gpio_set_value(spk_gpio.gpio, 0);
}
}
ret = of_property_read_u32(node, "headphonevol",&temp_val);
if (ret < 0) {
pr_err("[audio-codec]headphonevol configurations missing or invalid.\n");
ret = -EINVAL;
goto err1;
} else {
sunxi_internal_codec->gain_config.headphonevol = temp_val;
}
ret = of_property_read_u32(node, "spkervol",&temp_val);
if (ret < 0) {
pr_err("[audio-codec]spkervol configurations missing or invalid.\n");
ret = -EINVAL;
goto err1;
} else {
sunxi_internal_codec->gain_config.spkervol = temp_val;
}
ret = of_property_read_u32(node, "maingain",&temp_val);
if (ret < 0) {
pr_err("[audio-codec]maingain configurations missing or invalid.\n");
ret = -EINVAL;
goto err1;
} else {
sunxi_internal_codec->gain_config.maingain = temp_val;
}
ret = of_property_read_u32(node, "headsetmicgain",&temp_val);
if (ret < 0) {
pr_err("[audio-codec]headsetmicgain configurations missing or invalid.\n");
ret = -EINVAL;
goto err1;
} else {
sunxi_internal_codec->gain_config.headsetmicgain = temp_val;
}
ret = of_property_read_u32(node, "hp_dirused",&temp_val);
if (ret < 0) {
pr_err("[audio-codec]hp_dirused configurations missing or invalid.\n");
ret = -EINVAL;
goto err1;
} else {
sunxi_internal_codec->hp_dirused = temp_val;
}
ret = of_property_read_u32(node, "pa_sleep_time",&temp_val);
if (ret < 0) {
pr_err("[audio-codec]pa_sleep_time configurations missing or invalid.\n");
ret = -EINVAL;
goto err1;
} else {
sunxi_internal_codec->pa_sleep_time = temp_val;
}
pr_debug("headphonevol:%d, spkervol:%d, maingain:%d, pa_sleep_time:%d\n",
sunxi_internal_codec->gain_config.headphonevol,
sunxi_internal_codec->gain_config.spkervol,
sunxi_internal_codec->gain_config.maingain,
sunxi_internal_codec->pa_sleep_time
);
snd_soc_register_codec(&pdev->dev, &soc_codec_dev_codec, codec_dai, ARRAY_SIZE(codec_dai));
ret = sysfs_create_group(&pdev->dev.kobj, &audio_debug_attr_group);
if (ret){
pr_err("[audio-codec]failed to create attr group\n");
}
return 0;
err1:
devm_kfree(&pdev->dev, sunxi_internal_codec);
err0:
return ret;
}
static int __exit sunxi_internal_codec_remove(struct platform_device *pdev)
{
sysfs_remove_group(&pdev->dev.kobj, &audio_debug_attr_group);
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static void sunxi_internal_codec_shutdown(struct platform_device *pdev)
{
struct sunxi_codec * sunxi_internal_codec = dev_get_drvdata(&pdev->dev);
snd_soc_update_bits(sunxi_internal_codec->codec, PAEN_HP_CTRL, (0x1<<HPPAEN), (0x0<<HPPAEN));
if (spk_gpio.cfg)
gpio_set_value(spk_gpio.gpio, 0);
}
static struct platform_driver sunxi_internal_codec_driver = {
.driver = {
.name = DRV_NAME,
.owner = THIS_MODULE,
.of_match_table = sunxi_codec_of_match,
},
.probe = sunxi_internal_codec_probe,
.remove = __exit_p(sunxi_internal_codec_remove),
.shutdown = sunxi_internal_codec_shutdown,
};
module_platform_driver(sunxi_internal_codec_driver);
MODULE_DESCRIPTION("codec ALSA soc codec driver");
MODULE_AUTHOR("huanxin<huanxin@allwinnertech.com>");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:sunxi-pcm-codec");
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