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linux_dsm_epyc7002/sound/soc/codecs/es8316.c
Arvind Yadav eb59d73cb5 ASoC: codecs: constify snd_soc_dai_ops structures 
snd_soc_dai_ops are not supposed to change at runtime. All functions
working with snd_soc_dai_ops provided by <sound/soc-dai.h> work with
const snd_soc_dai_ops. So mark the non-const structs as const.

Signed-off-by: Arvind Yadav <arvind.yadav.cs@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-18 13:17:48 +01:00

638 lines
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/*
* es8316.c -- es8316 ALSA SoC audio driver
* Copyright Everest Semiconductor Co.,Ltd
*
* Authors: David Yang <yangxiaohua@everest-semi.com>,
* Daniel Drake <drake@endlessm.com>
*
* 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/acpi.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/mod_devicetable.h>
#include <linux/regmap.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
#include "es8316.h"
/* In slave mode at single speed, the codec is documented as accepting 5
* MCLK/LRCK ratios, but we also add ratio 400, which is commonly used on
* Intel Cherry Trail platforms (19.2MHz MCLK, 48kHz LRCK).
*/
#define NR_SUPPORTED_MCLK_LRCK_RATIOS 6
static const unsigned int supported_mclk_lrck_ratios[] = {
256, 384, 400, 512, 768, 1024
};
struct es8316_priv {
unsigned int sysclk;
unsigned int allowed_rates[NR_SUPPORTED_MCLK_LRCK_RATIOS];
struct snd_pcm_hw_constraint_list sysclk_constraints;
};
/*
* ES8316 controls
*/
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9600, 50, 1);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv,
0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0),
1, 1, TLV_DB_SCALE_ITEM(0, 0, 0),
2, 2, TLV_DB_SCALE_ITEM(250, 0, 0),
3, 3, TLV_DB_SCALE_ITEM(450, 0, 0),
4, 4, TLV_DB_SCALE_ITEM(700, 0, 0),
5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0),
6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0),
7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0),
8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0),
9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0),
10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0),
);
static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv,
0, 0, TLV_DB_SCALE_ITEM(-4800, 0, 0),
1, 3, TLV_DB_SCALE_ITEM(-2400, 1200, 0),
);
static const char * const ng_type_txt[] =
{ "Constant PGA Gain", "Mute ADC Output" };
static const struct soc_enum ng_type =
SOC_ENUM_SINGLE(ES8316_ADC_ALC_NG, 6, 2, ng_type_txt);
static const char * const adcpol_txt[] = { "Normal", "Invert" };
static const struct soc_enum adcpol =
SOC_ENUM_SINGLE(ES8316_ADC_MUTE, 1, 2, adcpol_txt);
static const char *const dacpol_txt[] =
{ "Normal", "R Invert", "L Invert", "L + R Invert" };
static const struct soc_enum dacpol =
SOC_ENUM_SINGLE(ES8316_DAC_SET1, 0, 4, dacpol_txt);
static const struct snd_kcontrol_new es8316_snd_controls[] = {
SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL,
4, 0, 3, 1, hpout_vol_tlv),
SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL,
0, 4, 7, 0, hpmixer_gain_tlv),
SOC_ENUM("Playback Polarity", dacpol),
SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL,
ES8316_DAC_VOLR, 0, 0xc0, 1, dac_vol_tlv),
SOC_SINGLE("DAC Soft Ramp Switch", ES8316_DAC_SET1, 4, 1, 1),
SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1, 2, 4, 0),
SOC_SINGLE("DAC Notch Filter Switch", ES8316_DAC_SET2, 6, 1, 0),
SOC_SINGLE("DAC Double Fs Switch", ES8316_DAC_SET2, 7, 1, 0),
SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3, 0, 7, 0),
SOC_ENUM("Capture Polarity", adcpol),
SOC_SINGLE("Mic Boost Switch", ES8316_ADC_D2SEPGA, 0, 1, 0),
SOC_SINGLE_TLV("ADC Capture Volume", ES8316_ADC_VOLUME,
0, 0xc0, 1, adc_vol_tlv),
SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8316_ADC_PGAGAIN,
4, 10, 0, adc_pga_gain_tlv),
SOC_SINGLE("ADC Soft Ramp Switch", ES8316_ADC_MUTE, 4, 1, 0),
SOC_SINGLE("ADC Double Fs Switch", ES8316_ADC_DMIC, 4, 1, 0),
SOC_SINGLE("ALC Capture Switch", ES8316_ADC_ALC1, 6, 1, 0),
SOC_SINGLE_TLV("ALC Capture Max Volume", ES8316_ADC_ALC1, 0, 28, 0,
alc_max_gain_tlv),
SOC_SINGLE_TLV("ALC Capture Min Volume", ES8316_ADC_ALC2, 0, 28, 0,
alc_min_gain_tlv),
SOC_SINGLE_TLV("ALC Capture Target Volume", ES8316_ADC_ALC3, 4, 10, 0,
alc_target_tlv),
SOC_SINGLE("ALC Capture Hold Time", ES8316_ADC_ALC3, 0, 10, 0),
SOC_SINGLE("ALC Capture Decay Time", ES8316_ADC_ALC4, 4, 10, 0),
SOC_SINGLE("ALC Capture Attack Time", ES8316_ADC_ALC4, 0, 10, 0),
SOC_SINGLE("ALC Capture Noise Gate Switch", ES8316_ADC_ALC_NG,
5, 1, 0),
SOC_SINGLE("ALC Capture Noise Gate Threshold", ES8316_ADC_ALC_NG,
0, 31, 0),
SOC_ENUM("ALC Capture Noise Gate Type", ng_type),
};
/* Analog Input Mux */
static const char * const es8316_analog_in_txt[] = {
"lin1-rin1",
"lin2-rin2",
"lin1-rin1 with 20db Boost",
"lin2-rin2 with 20db Boost"
};
static const unsigned int es8316_analog_in_values[] = { 0, 1, 2, 3 };
static const struct soc_enum es8316_analog_input_enum =
SOC_VALUE_ENUM_SINGLE(ES8316_ADC_PDN_LINSEL, 4, 3,
ARRAY_SIZE(es8316_analog_in_txt),
es8316_analog_in_txt,
es8316_analog_in_values);
static const struct snd_kcontrol_new es8316_analog_in_mux_controls =
SOC_DAPM_ENUM("Route", es8316_analog_input_enum);
static const char * const es8316_dmic_txt[] = {
"dmic disable",
"dmic data at high level",
"dmic data at low level",
};
static const unsigned int es8316_dmic_values[] = { 0, 1, 2 };
static const struct soc_enum es8316_dmic_src_enum =
SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3,
ARRAY_SIZE(es8316_dmic_txt),
es8316_dmic_txt,
es8316_dmic_values);
static const struct snd_kcontrol_new es8316_dmic_src_controls =
SOC_DAPM_ENUM("Route", es8316_dmic_src_enum);
/* hp mixer mux */
static const char * const es8316_hpmux_texts[] = {
"lin1-rin1",
"lin2-rin2",
"lin-rin with Boost",
"lin-rin with Boost and PGA"
};
static const unsigned int es8316_hpmux_values[] = { 0, 1, 2, 3 };
static SOC_ENUM_SINGLE_DECL(es8316_left_hpmux_enum, ES8316_HPMIX_SEL,
4, es8316_hpmux_texts);
static const struct snd_kcontrol_new es8316_left_hpmux_controls =
SOC_DAPM_ENUM("Route", es8316_left_hpmux_enum);
static SOC_ENUM_SINGLE_DECL(es8316_right_hpmux_enum, ES8316_HPMIX_SEL,
0, es8316_hpmux_texts);
static const struct snd_kcontrol_new es8316_right_hpmux_controls =
SOC_DAPM_ENUM("Route", es8316_right_hpmux_enum);
/* headphone Output Mixer */
static const struct snd_kcontrol_new es8316_out_left_mix[] = {
SOC_DAPM_SINGLE("LLIN Switch", ES8316_HPMIX_SWITCH, 6, 1, 0),
SOC_DAPM_SINGLE("Left DAC Switch", ES8316_HPMIX_SWITCH, 7, 1, 0),
};
static const struct snd_kcontrol_new es8316_out_right_mix[] = {
SOC_DAPM_SINGLE("RLIN Switch", ES8316_HPMIX_SWITCH, 2, 1, 0),
SOC_DAPM_SINGLE("Right DAC Switch", ES8316_HPMIX_SWITCH, 3, 1, 0),
};
/* DAC data source mux */
static const char * const es8316_dacsrc_texts[] = {
"LDATA TO LDAC, RDATA TO RDAC",
"LDATA TO LDAC, LDATA TO RDAC",
"RDATA TO LDAC, RDATA TO RDAC",
"RDATA TO LDAC, LDATA TO RDAC",
};
static const unsigned int es8316_dacsrc_values[] = { 0, 1, 2, 3 };
static SOC_ENUM_SINGLE_DECL(es8316_dacsrc_mux_enum, ES8316_DAC_SET1,
6, es8316_dacsrc_texts);
static const struct snd_kcontrol_new es8316_dacsrc_mux_controls =
SOC_DAPM_ENUM("Route", es8316_dacsrc_mux_enum);
static const struct snd_soc_dapm_widget es8316_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("Bias", ES8316_SYS_PDN, 3, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("Analog power", ES8316_SYS_PDN, 4, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("Mic Bias", ES8316_SYS_PDN, 5, 1, NULL, 0),
SND_SOC_DAPM_INPUT("DMIC"),
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
/* Input Mux */
SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
&es8316_analog_in_mux_controls),
SND_SOC_DAPM_SUPPLY("ADC Vref", ES8316_SYS_PDN, 1, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("ADC bias", ES8316_SYS_PDN, 2, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("ADC Clock", ES8316_CLKMGR_CLKSW, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("Line input PGA", ES8316_ADC_PDN_LINSEL,
7, 1, NULL, 0),
SND_SOC_DAPM_ADC("Mono ADC", NULL, ES8316_ADC_PDN_LINSEL, 6, 1),
SND_SOC_DAPM_MUX("Digital Mic Mux", SND_SOC_NOPM, 0, 0,
&es8316_dmic_src_controls),
/* Digital Interface */
SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 1,
ES8316_SERDATA_ADC, 6, 1),
SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0,
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("DAC Source Mux", SND_SOC_NOPM, 0, 0,
&es8316_dacsrc_mux_controls),
SND_SOC_DAPM_SUPPLY("DAC Vref", ES8316_SYS_PDN, 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("DAC Clock", ES8316_CLKMGR_CLKSW, 2, 0, NULL, 0),
SND_SOC_DAPM_DAC("Right DAC", NULL, ES8316_DAC_PDN, 0, 1),
SND_SOC_DAPM_DAC("Left DAC", NULL, ES8316_DAC_PDN, 4, 1),
/* Headphone Output Side */
SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
&es8316_left_hpmux_controls),
SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
&es8316_right_hpmux_controls),
SND_SOC_DAPM_MIXER("Left Headphone Mixer", ES8316_HPMIX_PDN,
5, 1, &es8316_out_left_mix[0],
ARRAY_SIZE(es8316_out_left_mix)),
SND_SOC_DAPM_MIXER("Right Headphone Mixer", ES8316_HPMIX_PDN,
1, 1, &es8316_out_right_mix[0],
ARRAY_SIZE(es8316_out_right_mix)),
SND_SOC_DAPM_PGA("Left Headphone Mixer Out", ES8316_HPMIX_PDN,
4, 1, NULL, 0),
SND_SOC_DAPM_PGA("Right Headphone Mixer Out", ES8316_HPMIX_PDN,
0, 1, NULL, 0),
SND_SOC_DAPM_OUT_DRV("Left Headphone Charge Pump", ES8316_CPHP_OUTEN,
6, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV("Right Headphone Charge Pump", ES8316_CPHP_OUTEN,
2, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8316_CPHP_PDN2,
5, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("Headphone Charge Pump Clock", ES8316_CLKMGR_CLKSW,
4, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV("Left Headphone Driver", ES8316_CPHP_OUTEN,
5, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV("Right Headphone Driver", ES8316_CPHP_OUTEN,
1, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Headphone Out", ES8316_CPHP_PDN1, 2, 1, NULL, 0),
/* pdn_Lical and pdn_Rical bits are documented as Reserved, but must
* be explicitly unset in order to enable HP output
*/
SND_SOC_DAPM_SUPPLY("Left Headphone ical", ES8316_CPHP_ICAL_VOL,
7, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("Right Headphone ical", ES8316_CPHP_ICAL_VOL,
3, 1, NULL, 0),
SND_SOC_DAPM_OUTPUT("HPOL"),
SND_SOC_DAPM_OUTPUT("HPOR"),
};
static const struct snd_soc_dapm_route es8316_dapm_routes[] = {
/* Recording */
{"MIC1", NULL, "Mic Bias"},
{"MIC2", NULL, "Mic Bias"},
{"MIC1", NULL, "Bias"},
{"MIC2", NULL, "Bias"},
{"MIC1", NULL, "Analog power"},
{"MIC2", NULL, "Analog power"},
{"Differential Mux", "lin1-rin1", "MIC1"},
{"Differential Mux", "lin2-rin2", "MIC2"},
{"Line input PGA", NULL, "Differential Mux"},
{"Mono ADC", NULL, "ADC Clock"},
{"Mono ADC", NULL, "ADC Vref"},
{"Mono ADC", NULL, "ADC bias"},
{"Mono ADC", NULL, "Line input PGA"},
/* It's not clear why, but to avoid recording only silence,
* the DAC clock must be running for the ADC to work.
*/
{"Mono ADC", NULL, "DAC Clock"},
{"Digital Mic Mux", "dmic disable", "Mono ADC"},
{"I2S OUT", NULL, "Digital Mic Mux"},
/* Playback */
{"DAC Source Mux", "LDATA TO LDAC, RDATA TO RDAC", "I2S IN"},
{"Left DAC", NULL, "DAC Clock"},
{"Right DAC", NULL, "DAC Clock"},
{"Left DAC", NULL, "DAC Vref"},
{"Right DAC", NULL, "DAC Vref"},
{"Left DAC", NULL, "DAC Source Mux"},
{"Right DAC", NULL, "DAC Source Mux"},
{"Left Headphone Mux", "lin-rin with Boost and PGA", "Line input PGA"},
{"Right Headphone Mux", "lin-rin with Boost and PGA", "Line input PGA"},
{"Left Headphone Mixer", "LLIN Switch", "Left Headphone Mux"},
{"Left Headphone Mixer", "Left DAC Switch", "Left DAC"},
{"Right Headphone Mixer", "RLIN Switch", "Right Headphone Mux"},
{"Right Headphone Mixer", "Right DAC Switch", "Right DAC"},
{"Left Headphone Mixer Out", NULL, "Left Headphone Mixer"},
{"Right Headphone Mixer Out", NULL, "Right Headphone Mixer"},
{"Left Headphone Charge Pump", NULL, "Left Headphone Mixer Out"},
{"Right Headphone Charge Pump", NULL, "Right Headphone Mixer Out"},
{"Left Headphone Charge Pump", NULL, "Headphone Charge Pump"},
{"Right Headphone Charge Pump", NULL, "Headphone Charge Pump"},
{"Left Headphone Charge Pump", NULL, "Headphone Charge Pump Clock"},
{"Right Headphone Charge Pump", NULL, "Headphone Charge Pump Clock"},
{"Left Headphone Driver", NULL, "Left Headphone Charge Pump"},
{"Right Headphone Driver", NULL, "Right Headphone Charge Pump"},
{"HPOL", NULL, "Left Headphone Driver"},
{"HPOR", NULL, "Right Headphone Driver"},
{"HPOL", NULL, "Left Headphone ical"},
{"HPOR", NULL, "Right Headphone ical"},
{"Headphone Out", NULL, "Bias"},
{"Headphone Out", NULL, "Analog power"},
{"HPOL", NULL, "Headphone Out"},
{"HPOR", NULL, "Headphone Out"},
};
static int es8316_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 es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec);
int i;
int count = 0;
es8316->sysclk = freq;
if (freq == 0)
return 0;
/* Limit supported sample rates to ones that can be autodetected
* by the codec running in slave mode.
*/
for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) {
const unsigned int ratio = supported_mclk_lrck_ratios[i];
if (freq % ratio == 0)
es8316->allowed_rates[count++] = freq / ratio;
}
es8316->sysclk_constraints.list = es8316->allowed_rates;
es8316->sysclk_constraints.count = count;
return 0;
}
static int es8316_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
u8 serdata1 = 0;
u8 serdata2 = 0;
u8 clksw;
u8 mask;
if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
dev_err(codec->dev, "Codec driver only supports slave mode\n");
return -EINVAL;
}
if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) {
dev_err(codec->dev, "Codec driver only supports I2S format\n");
return -EINVAL;
}
/* Clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_IF:
serdata1 |= ES8316_SERDATA1_BCLK_INV;
serdata2 |= ES8316_SERDATA2_ADCLRP;
break;
case SND_SOC_DAIFMT_IB_NF:
serdata1 |= ES8316_SERDATA1_BCLK_INV;
break;
case SND_SOC_DAIFMT_NB_IF:
serdata2 |= ES8316_SERDATA2_ADCLRP;
break;
default:
return -EINVAL;
}
mask = ES8316_SERDATA1_MASTER | ES8316_SERDATA1_BCLK_INV;
snd_soc_update_bits(codec, ES8316_SERDATA1, mask, serdata1);
mask = ES8316_SERDATA2_FMT_MASK | ES8316_SERDATA2_ADCLRP;
snd_soc_update_bits(codec, ES8316_SERDATA_ADC, mask, serdata2);
snd_soc_update_bits(codec, ES8316_SERDATA_DAC, mask, serdata2);
/* Enable BCLK and MCLK inputs in slave mode */
clksw = ES8316_CLKMGR_CLKSW_MCLK_ON | ES8316_CLKMGR_CLKSW_BCLK_ON;
snd_soc_update_bits(codec, ES8316_CLKMGR_CLKSW, clksw, clksw);
return 0;
}
static int es8316_pcm_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec);
if (es8316->sysclk == 0) {
dev_err(codec->dev, "No sysclk provided\n");
return -EINVAL;
}
/* The set of sample rates that can be supported depends on the
* MCLK supplied to the CODEC.
*/
snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&es8316->sysclk_constraints);
return 0;
}
static int es8316_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct es8316_priv *es8316 = snd_soc_codec_get_drvdata(codec);
u8 wordlen = 0;
if (!es8316->sysclk) {
dev_err(codec->dev, "No MCLK configured\n");
return -EINVAL;
}
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
wordlen = ES8316_SERDATA2_LEN_16;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
wordlen = ES8316_SERDATA2_LEN_20;
break;
case SNDRV_PCM_FORMAT_S24_LE:
wordlen = ES8316_SERDATA2_LEN_24;
break;
case SNDRV_PCM_FORMAT_S32_LE:
wordlen = ES8316_SERDATA2_LEN_32;
break;
default:
return -EINVAL;
}
snd_soc_update_bits(codec, ES8316_SERDATA_DAC,
ES8316_SERDATA2_LEN_MASK, wordlen);
snd_soc_update_bits(codec, ES8316_SERDATA_ADC,
ES8316_SERDATA2_LEN_MASK, wordlen);
return 0;
}
static int es8316_mute(struct snd_soc_dai *dai, int mute)
{
snd_soc_update_bits(dai->codec, ES8316_DAC_SET1, 0x20,
mute ? 0x20 : 0);
return 0;
}
#define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_LE)
static const struct snd_soc_dai_ops es8316_ops = {
.startup = es8316_pcm_startup,
.hw_params = es8316_pcm_hw_params,
.set_fmt = es8316_set_dai_fmt,
.set_sysclk = es8316_set_dai_sysclk,
.digital_mute = es8316_mute,
};
static struct snd_soc_dai_driver es8316_dai = {
.name = "ES8316 HiFi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = ES8316_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = ES8316_FORMATS,
},
.ops = &es8316_ops,
.symmetric_rates = 1,
};
static int es8316_probe(struct snd_soc_codec *codec)
{
/* Reset codec and enable current state machine */
snd_soc_write(codec, ES8316_RESET, 0x3f);
usleep_range(5000, 5500);
snd_soc_write(codec, ES8316_RESET, ES8316_RESET_CSM_ON);
msleep(30);
/*
* Documentation is unclear, but this value from the vendor driver is
* needed otherwise audio output is silent.
*/
snd_soc_write(codec, ES8316_SYS_VMIDSEL, 0xff);
/*
* Documentation for this register is unclear and incomplete,
* but here is a vendor-provided value that improves volume
* and quality for Intel CHT platforms.
*/
snd_soc_write(codec, ES8316_CLKMGR_ADCOSR, 0x32);
return 0;
}
static const struct snd_soc_codec_driver soc_codec_dev_es8316 = {
.probe = es8316_probe,
.idle_bias_off = true,
.component_driver = {
.controls = es8316_snd_controls,
.num_controls = ARRAY_SIZE(es8316_snd_controls),
.dapm_widgets = es8316_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(es8316_dapm_widgets),
.dapm_routes = es8316_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(es8316_dapm_routes),
},
};
static const struct regmap_config es8316_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = 0x53,
.cache_type = REGCACHE_RBTREE,
};
static int es8316_i2c_probe(struct i2c_client *i2c_client,
const struct i2c_device_id *id)
{
struct es8316_priv *es8316;
struct regmap *regmap;
es8316 = devm_kzalloc(&i2c_client->dev, sizeof(struct es8316_priv),
GFP_KERNEL);
if (es8316 == NULL)
return -ENOMEM;
i2c_set_clientdata(i2c_client, es8316);
regmap = devm_regmap_init_i2c(i2c_client, &es8316_regmap);
if (IS_ERR(regmap))
return PTR_ERR(regmap);
return snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_es8316,
&es8316_dai, 1);
}
static int es8316_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
return 0;
}
static const struct i2c_device_id es8316_i2c_id[] = {
{"es8316", 0 },
{}
};
MODULE_DEVICE_TABLE(i2c, es8316_i2c_id);
static const struct of_device_id es8316_of_match[] = {
{ .compatible = "everest,es8316", },
{},
};
MODULE_DEVICE_TABLE(of, es8316_of_match);
static const struct acpi_device_id es8316_acpi_match[] = {
{"ESSX8316", 0},
{},
};
MODULE_DEVICE_TABLE(acpi, es8316_acpi_match);
static struct i2c_driver es8316_i2c_driver = {
.driver = {
.name = "es8316",
.acpi_match_table = ACPI_PTR(es8316_acpi_match),
.of_match_table = of_match_ptr(es8316_of_match),
},
.probe = es8316_i2c_probe,
.remove = es8316_i2c_remove,
.id_table = es8316_i2c_id,
};
module_i2c_driver(es8316_i2c_driver);
MODULE_DESCRIPTION("Everest Semi ES8316 ALSA SoC Codec Driver");
MODULE_AUTHOR("David Yang <yangxiaohua@everest-semi.com>");
MODULE_LICENSE("GPL v2");
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