| Author | Tokens | Token Proportion | Commits | Commit Proportion |
|---|---|---|---|---|
| Daniel Drake | 3089 | 97.85% | 1 | 25.00% |
| Kuninori Morimoto | 66 | 2.09% | 1 | 25.00% |
| Arvind Yadav | 1 | 0.03% | 1 | 25.00% |
| Bhumika Goyal | 1 | 0.03% | 1 | 25.00% |
| Total | 3157 | 4 |
/* * 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_component *component = codec_dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); 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_component *component = codec_dai->component; u8 serdata1 = 0; u8 serdata2 = 0; u8 clksw; u8 mask; if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { dev_err(component->dev, "Codec driver only supports slave mode\n"); return -EINVAL; } if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) { dev_err(component->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_component_update_bits(component, ES8316_SERDATA1, mask, serdata1); mask = ES8316_SERDATA2_FMT_MASK | ES8316_SERDATA2_ADCLRP; snd_soc_component_update_bits(component, ES8316_SERDATA_ADC, mask, serdata2); snd_soc_component_update_bits(component, 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_component_update_bits(component, 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_component *component = dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); if (es8316->sysclk == 0) { dev_err(component->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_component *component = dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); u8 wordlen = 0; if (!es8316->sysclk) { dev_err(component->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_component_update_bits(component, ES8316_SERDATA_DAC, ES8316_SERDATA2_LEN_MASK, wordlen); snd_soc_component_update_bits(component, ES8316_SERDATA_ADC, ES8316_SERDATA2_LEN_MASK, wordlen); return 0; } static int es8316_mute(struct snd_soc_dai *dai, int mute) { snd_soc_component_update_bits(dai->component, 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_component *component) { /* Reset codec and enable current state machine */ snd_soc_component_write(component, ES8316_RESET, 0x3f); usleep_range(5000, 5500); snd_soc_component_write(component, 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_component_write(component, 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_component_write(component, ES8316_CLKMGR_ADCOSR, 0x32); return 0; } static const struct snd_soc_component_driver soc_component_dev_es8316 = { .probe = es8316_probe, .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), .use_pmdown_time = 1, .endianness = 1, .non_legacy_dai_naming = 1, }; 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 devm_snd_soc_register_component(&i2c_client->dev, &soc_component_dev_es8316, &es8316_dai, 1); } 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, .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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