[alsa-devel] [PATCH v2 1/2] ASoC: add es8316 codec driver
Add a codec driver for the Everest ES8316, based on code provided by David Yang from Everest Semi.
I limited the functionality to items where the vendor code was clear, and things that can be tested on the Weibu F3C (Intel Cherry Trail). As a result the initial implementation only supports running in slave mode at single speed (up to 48kHz sample rate) using I2S. HPD is not supported.
Signed-off-by: David Yang yangxiaohua@everest-semi.com [drake@endlessm.com: significant cleanups and simplifications, remove dead/unclear code] Signed-off-by: Daniel Drake drake@endlessm.com ---
Notes: v2: - Add sign-off from David Yang - Only support whole number rates calculated with mclk ratios
sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8316.c | 646 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8316.h | 122 +++++++++ 4 files changed, 774 insertions(+) create mode 100644 sound/soc/codecs/es8316.c create mode 100644 sound/soc/codecs/es8316.h
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 883ed4c8a551..c6286e5ba511 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -72,6 +72,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA9055 if I2C select SND_SOC_DIO2125 select SND_SOC_DMIC + select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_ES7134 @@ -543,6 +544,9 @@ config SND_SOC_HDMI_CODEC config SND_SOC_ES7134 tristate "Everest Semi ES7134 CODEC"
+config SND_SOC_ES8316 + tristate "Everest Semi ES8316 CODEC" + config SND_SOC_ES8328 tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 28a63fdaf982..e878306ce46e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -65,6 +65,7 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o +snd-soc-es8316-objs := es8316.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -300,6 +301,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o +obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c new file mode 100644 index 000000000000..35fe32070285 --- /dev/null +++ b/sound/soc/codecs/es8316.c @@ -0,0 +1,646 @@ +/* + * 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 DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9600, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -9600, 50, 1); +static const DECLARE_TLV_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const DECLARE_TLV_DB_SCALE(mic_bst_tlv, 0, 1200, 0); + +static unsigned int linin_pga_tlv[] = { + TLV_DB_RANGE_HEAD(12), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(300, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(600, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(900, 0, 0), + 4, 4, TLV_DB_SCALE_ITEM(1200, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(1500, 0, 0), + 6, 6, TLV_DB_SCALE_ITEM(1800, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(2100, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(2400, 0, 0), +}; +static unsigned int hpout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 3, TLV_DB_SCALE_ITEM(-4800, 1200, 0), +}; +static const char * const alc_func_txt[] = { "Off", "On" }; +static const struct soc_enum alc_func = + SOC_ENUM_SINGLE(ES8316_ADC_ALC1, 6, 2, alc_func_txt); + +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_ALC6, 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("HP Playback Volume", ES8316_CPHP_ICAL_VOL, + 4, 0, 0, 1, hpout_vol_tlv), + SOC_DOUBLE_TLV("HPMixer Gain", ES8316_HPMIX_VOL, + 0, 4, 7, 0, hpmixer_gain_tlv), + + SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, + ES8316_DAC_VOLR, 0, 0xC0, 1, dac_vol_tlv), + + SOC_SINGLE("Enable DAC Soft Ramp", ES8316_DAC_SET1, 4, 1, 1), + SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1, 2, 4, 0), + + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE("DAC Notch Filter", ES8316_DAC_SET2, 6, 1, 0), + SOC_SINGLE("DAC Double Fs Mode", ES8316_DAC_SET2, 7, 1, 0), + SOC_SINGLE("DAC Volume Control-LeR", ES8316_DAC_SET2, 2, 1, 0), + SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3, 0, 7, 0), + + /* +20dB D2SE PGA Control */ + SOC_SINGLE_TLV("MIC Boost", ES8316_ADC_D2SEPGA, + 0, 1, 0, mic_bst_tlv), + + /* 0-+24dB Lineinput PGA Control */ + SOC_SINGLE_TLV("Input PGA", ES8316_ADC_PGAGAIN, + 4, 8, 0, linin_pga_tlv), + + SOC_SINGLE_TLV("ADC Capture Volume", ES8316_ADC_VOLUME, + 0, 0xc0, 1, adc_vol_tlv), + SOC_SINGLE("ADC Soft Ramp", ES8316_ADC_MUTE, 4, 1, 0), + SOC_ENUM("Capture Polarity", adcpol), + SOC_SINGLE("ADC Double FS Mode", ES8316_ADC_DMIC, 4, 1, 0), + + SOC_SINGLE("ALC Capture Target Volume", + ES8316_ADC_ALC3, 4, 10, 0), + SOC_SINGLE("ALC Capture Max PGA", ES8316_ADC_ALC1, 0, 28, 0), + SOC_SINGLE("ALC Capture Min PGA", ES8316_ADC_ALC2, 0, 28, 0), + SOC_ENUM("ALC Capture Function", alc_func), + 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 NG Threshold", ES8316_ADC_ALC6, 0, 31, 0), + SOC_ENUM("ALC Capture NG Type", ng_type), + SOC_SINGLE("ALC Capture NG Switch", ES8316_ADC_ALC6, 5, 1, 0), +}; + +/* 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_INPUT("DMIC"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_MICBIAS("micbias", SND_SOC_NOPM, 0, 0), + + /* Input Mux */ + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &es8316_analog_in_mux_controls), + + 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 SRC Mux", SND_SOC_NOPM, 0, 0, + &es8316_dacsrc_mux_controls), + + SND_SOC_DAPM_DAC("Right DAC", NULL, SND_SOC_NOPM, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, SND_SOC_NOPM, 4, 1), + + /* Headphone Output Side */ + SND_SOC_DAPM_MUX("Left Hp mux", SND_SOC_NOPM, 0, 0, + &es8316_left_hpmux_controls), + SND_SOC_DAPM_MUX("Right Hp mux", SND_SOC_NOPM, 0, 0, + &es8316_right_hpmux_controls), + SND_SOC_DAPM_MIXER("Left Hp mixer", SND_SOC_NOPM, + 4, 1, &es8316_out_left_mix[0], + ARRAY_SIZE(es8316_out_left_mix)), + SND_SOC_DAPM_MIXER("Right Hp mixer", SND_SOC_NOPM, + 0, 1, &es8316_out_right_mix[0], + ARRAY_SIZE(es8316_out_right_mix)), + + /* Output charge pump */ + SND_SOC_DAPM_PGA("HPCP L", SND_SOC_NOPM, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPCP R", SND_SOC_NOPM, 2, 0, NULL, 0), + + /* Output Driver */ + SND_SOC_DAPM_PGA("HPVOL L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPVOL R", SND_SOC_NOPM, 0, 0, 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, "micbias"}, + {"MIC2", NULL, "micbias"}, + + {"Differential Mux", "lin1-rin1", "MIC1"}, + {"Differential Mux", "lin2-rin2", "MIC2"}, + {"Line input PGA", NULL, "Differential Mux"}, + + {"Mono ADC", NULL, "Line input PGA"}, + + {"Digital Mic Mux", "dmic disable", "Mono ADC"}, + + {"I2S OUT", NULL, "Digital Mic Mux"}, + + /* Playback */ + {"DAC SRC Mux", "LDATA TO LDAC, RDATA TO RDAC", "I2S IN"}, + + {"Left DAC", NULL, "DAC SRC Mux"}, + {"Right DAC", NULL, "DAC SRC Mux"}, + + {"Left Hp mux", "lin-rin with Boost and PGA", "Line input PGA"}, + {"Right Hp mux", "lin-rin with Boost and PGA", "Line input PGA"}, + + {"Left Hp mixer", "LLIN Switch", "Left Hp mux"}, + {"Left Hp mixer", "Left DAC Switch", "Left DAC"}, + + {"Right Hp mixer", "RLIN Switch", "Right Hp mux"}, + {"Right Hp mixer", "Right DAC Switch", "Right DAC"}, + + {"HPCP L", NULL, "Left Hp mixer"}, + {"HPCP R", NULL, "Right Hp mixer"}, + + {"HPVOL L", NULL, "HPCP L"}, + {"HPVOL R", NULL, "HPCP R"}, + + {"HPOL", NULL, "HPVOL L"}, + {"HPOR", NULL, "HPVOL R"}, +}; + +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 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); + + 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; +} + +static int es8316_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_write(codec, ES8316_RESET, 0xC0); + snd_soc_write(codec, ES8316_CPHP_OUTEN, 0x66); + snd_soc_write(codec, ES8316_DAC_PDN, 0x00); + snd_soc_write(codec, ES8316_CPHP_LDOCTL, 0x30); + snd_soc_write(codec, ES8316_CPHP_PDN2, 0x10); + snd_soc_write(codec, ES8316_CPHP_PDN1, 0x03); + snd_soc_write(codec, ES8316_HPMIX_PDN, 0x00); + snd_soc_update_bits(codec, ES8316_ADC_PDN_LINSEL, 0xc0, 0x00); + snd_soc_write(codec, ES8316_SYS_PDN, 0x00); + snd_soc_write(codec, ES8316_SYS_LP1, 0x04); + snd_soc_write(codec, ES8316_SYS_LP2, 0x0c); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, ES8316_CPHP_OUTEN, 0x00); + snd_soc_write(codec, ES8316_DAC_PDN, 0x11); + snd_soc_write(codec, ES8316_CPHP_LDOCTL, 0x03); + snd_soc_write(codec, ES8316_CPHP_PDN2, 0x22); + snd_soc_write(codec, ES8316_CPHP_PDN1, 0x06); + snd_soc_write(codec, ES8316_HPMIX_PDN, 0x33); + snd_soc_update_bits(codec, ES8316_ADC_PDN_LINSEL, 0xc0, 0xc0); + snd_soc_write(codec, ES8316_SYS_PDN, 0x3f); + snd_soc_write(codec, ES8316_SYS_LP1, 0x3f); + snd_soc_write(codec, ES8316_SYS_LP2, 0x1f); + snd_soc_write(codec, ES8316_RESET, 0x00); + break; + } + + return 0; +} + +#define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static 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) +{ + snd_soc_write(codec, ES8316_RESET, 0x3f); + usleep_range(5000, 5500); + snd_soc_write(codec, ES8316_RESET, 0x00); + snd_soc_write(codec, ES8316_SYS_VMIDSEL, 0xFF); + msleep(30); + + snd_soc_write(codec, ES8316_CLKMGR_CLKSEL, 0x08); + snd_soc_write(codec, ES8316_CLKMGR_ADCOSR, 0x32); + snd_soc_write(codec, ES8316_CLKMGR_ADCDIV1, 0x11); + snd_soc_write(codec, ES8316_CLKMGR_ADCDIV2, 0x00); + snd_soc_write(codec, ES8316_CLKMGR_DACDIV1, 0x11); + snd_soc_write(codec, ES8316_CLKMGR_DACDIV2, 0x00); + snd_soc_write(codec, ES8316_CLKMGR_CPDIV, 0x00); + snd_soc_write(codec, ES8316_SERDATA1, 0x04); + snd_soc_write(codec, ES8316_CLKMGR_CLKSW, 0x7f); + snd_soc_write(codec, ES8316_CAL_TYPE, 0x0F); + snd_soc_write(codec, ES8316_CAL_HPLIV, 0x90); + snd_soc_write(codec, ES8316_CAL_HPRIV, 0x90); + snd_soc_write(codec, ES8316_ADC_VOLUME, 0x00); + snd_soc_write(codec, ES8316_ADC_D2SEPGA, 0x00); + snd_soc_write(codec, ES8316_ADC_DMIC, 0x08); + snd_soc_write(codec, ES8316_DAC_SET2, 0x00); + snd_soc_write(codec, ES8316_DAC_SET3, 0x00); + snd_soc_write(codec, ES8316_DAC_VOLL, 0x02); + snd_soc_write(codec, ES8316_DAC_VOLR, 0x02); + snd_soc_write(codec, ES8316_SERDATA_ADC, 0x00); + snd_soc_write(codec, ES8316_SERDATA_DAC, 0x00); + snd_soc_write(codec, ES8316_SYS_VMIDLOW, 0x11); + snd_soc_write(codec, ES8316_SYS_VSEL, 0xfc); + snd_soc_write(codec, ES8316_SYS_REF, 0x28); + snd_soc_write(codec, ES8316_HPMIX_SEL, 0x00); + snd_soc_write(codec, ES8316_HPMIX_SWITCH, 0x88); + snd_soc_write(codec, ES8316_HPMIX_VOL, 0x88); + snd_soc_write(codec, ES8316_CPHP_ICAL_VOL, 0x00); + snd_soc_write(codec, ES8316_GPIO_SEL, 0x00); + snd_soc_write(codec, ES8316_GPIO_DEBUNCE_INT, 0x02); + snd_soc_write(codec, ES8316_TESTMODE, 0xa0); + snd_soc_write(codec, ES8316_TEST1, 0x00); + snd_soc_write(codec, ES8316_TEST2, 0x04); + snd_soc_write(codec, ES8316_RESET, 0xc0); + msleep(50); + snd_soc_write(codec, ES8316_ADC_PGAGAIN, 0x60); + snd_soc_write(codec, ES8316_ADC_D2SEPGA, 0x01); + + /* ADC DS mode, HPF enable */ + snd_soc_write(codec, ES8316_ADC_DMIC, 0x08); + snd_soc_write(codec, ES8316_ADC_ALC1, 0xcd); + snd_soc_write(codec, ES8316_ADC_ALC2, 0x08); + snd_soc_write(codec, ES8316_ADC_ALC3, 0xa0); + snd_soc_write(codec, ES8316_ADC_ALC4, 0x05); + snd_soc_write(codec, ES8316_ADC_ALC5, 0x06); + snd_soc_write(codec, ES8316_ADC_ALC6, 0x61); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_es8316 = { + .probe = es8316_probe, + .set_bias_level = es8316_set_bias_level, + .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[] = { + {"ESSX8316", 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"); diff --git a/sound/soc/codecs/es8316.h b/sound/soc/codecs/es8316.h new file mode 100644 index 000000000000..702bebc69ca8 --- /dev/null +++ b/sound/soc/codecs/es8316.h @@ -0,0 +1,122 @@ +/* + * Copyright Everest Semiconductor Co.,Ltd + * + * Author: David Yang yangxiaohua@everest-semi.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. + * + */ + +#ifndef _ES8316_H +#define _ES8316_H + +/* + * ES8316 register space + */ + +/* Reset Control */ +#define ES8316_RESET 0x00 + +/* Clock Management */ +#define ES8316_CLKMGR_CLKSW 0x01 +#define ES8316_CLKMGR_CLKSEL 0x02 +#define ES8316_CLKMGR_ADCOSR 0x03 +#define ES8316_CLKMGR_ADCDIV1 0x04 +#define ES8316_CLKMGR_ADCDIV2 0x05 +#define ES8316_CLKMGR_DACDIV1 0x06 +#define ES8316_CLKMGR_DACDIV2 0x07 +#define ES8316_CLKMGR_CPDIV 0x08 + +/* Serial Data Port Control */ +#define ES8316_SERDATA1 0x09 +#define ES8316_SERDATA_ADC 0x0a +#define ES8316_SERDATA_DAC 0x0b + +/* System Control */ +#define ES8316_SYS_VMIDSEL 0x0c +#define ES8316_SYS_PDN 0x0d +#define ES8316_SYS_LP1 0x0e +#define ES8316_SYS_LP2 0x0f +#define ES8316_SYS_VMIDLOW 0x10 +#define ES8316_SYS_VSEL 0x11 +#define ES8316_SYS_REF 0x12 + +/* Headphone Mixer */ +#define ES8316_HPMIX_SEL 0x13 +#define ES8316_HPMIX_SWITCH 0x14 +#define ES8316_HPMIX_PDN 0x15 +#define ES8316_HPMIX_VOL 0x16 + +/* Charge Pump Headphone driver */ +#define ES8316_CPHP_OUTEN 0x17 +#define ES8316_CPHP_ICAL_VOL 0x18 +#define ES8316_CPHP_PDN1 0x19 +#define ES8316_CPHP_PDN2 0x1a +#define ES8316_CPHP_LDOCTL 0x1b + +/* Calibration */ +#define ES8316_CAL_TYPE 0x1c +#define ES8316_CAL_SET 0x1d +#define ES8316_CAL_HPLIV 0x1e +#define ES8316_CAL_HPRIV 0x1f +#define ES8316_CAL_HPLMV 0x20 +#define ES8316_CAL_HPRMV 0x21 + +/* ADC Control */ +#define ES8316_ADC_PDN_LINSEL 0x22 +#define ES8316_ADC_PGAGAIN 0x23 +#define ES8316_ADC_D2SEPGA 0x24 +#define ES8316_ADC_DMIC 0x25 +#define ES8316_ADC_MUTE 0x26 +#define ES8316_ADC_VOLUME 0x27 +#define ES8316_ADC_ALC1 0x29 +#define ES8316_ADC_ALC2 0x2a +#define ES8316_ADC_ALC3 0x2b +#define ES8316_ADC_ALC4 0x2c +#define ES8316_ADC_ALC5 0x2d +#define ES8316_ADC_ALC6 0x2e + +/* DAC Control */ +#define ES8316_DAC_PDN 0x2f +#define ES8316_DAC_SET1 0x30 +#define ES8316_DAC_SET2 0x31 +#define ES8316_DAC_SET3 0x32 +#define ES8316_DAC_VOLL 0x33 +#define ES8316_DAC_VOLR 0x34 + +/* GPIO */ +#define ES8316_GPIO_SEL 0x4d +#define ES8316_GPIO_DEBUNCE_INT 0x4e +#define ES8316_GPIO_FLAG 0x4f + +/* Test mode */ +#define ES8316_TESTMODE 0x50 +#define ES8316_TEST1 0x51 +#define ES8316_TEST2 0x52 +#define ES8316_TEST3 0x53 + +/* + * Field definitions + */ + +/* ES8316_SERDATA1 */ +#define ES8316_SERDATA1_MASTER 0x80 +#define ES8316_SERDATA1_BCLK_INV 0x20 + +/* ES8316_SERDATA_ADC and _DAC */ +#define ES8316_SERDATA2_FMT_MASK 0x3 +#define ES8316_SERDATA2_FMT_I2S 0x00 +#define ES8316_SERDATA2_FMT_LEFTJ 0x01 +#define ES8316_SERDATA2_FMT_RIGHTJ 0x02 +#define ES8316_SERDATA2_FMT_PCM 0x03 +#define ES8316_SERDATA2_ADCLRP 0x20 +#define ES8316_SERDATA2_LEN_MASK 0x1c +#define ES8316_SERDATA2_LEN_24 0x00 +#define ES8316_SERDATA2_LEN_20 0x04 +#define ES8316_SERDATA2_LEN_18 0x08 +#define ES8316_SERDATA2_LEN_16 0x0c +#define ES8316_SERDATA2_LEN_32 0x10 + +#endif
Add new machine driver, tested with Weibu F3C MiniPC.
Based heavily on code provided by David Yang @ Everest, and other machine drivers in the same directory.
Signed-off-by: David Yang yangxiaohua@everest-semi.com [drake@endlessm.com: cleanups and modernization] Signed-off-by: Daniel Drake drake@endlessm.com ---
Notes: v2: - add sign-off from David Yang - fix bytbyt typo - mention that BYT-CR/SSP0 is not supported - mention which mic was tested - setup clocks at init time, avoids need for platform clock control
sound/soc/intel/Kconfig | 12 ++ sound/soc/intel/atom/sst/sst_acpi.c | 2 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_es8316.c | 300 +++++++++++++++++++++++++++++++++ 4 files changed, 316 insertions(+) create mode 100644 sound/soc/intel/boards/bytcht_es8316.c
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 67968ef3bbda..013f056c6c12 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -214,6 +214,18 @@ config SND_SOC_INTEL_BYT_CHT_DA7213_MACH platforms with DA7212/7213 audio codec. If unsure select "N".
+config SND_SOC_INTEL_BYT_CHT_ES8316_MACH + tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail with ES8316 codec" + depends on X86_INTEL_LPSS && I2C && ACPI + select SND_SOC_ES8316 + select SND_SST_ATOM_HIFI2_PLATFORM + select SND_SST_IPC_ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI + help + This adds support for ASoC machine driver for Intel(R) Baytrail & + Cherrytrail platforms with ES8316 audio codec. + If unsure select "N". + config SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail platform with no codec (MinnowBoard MAX, Up)" depends on X86_INTEL_LPSS && I2C && ACPI diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index dd250b8b26f2..e0b7346cbb16 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -533,6 +533,8 @@ static struct sst_acpi_mach sst_acpi_chv[] = { &chv_platform_data }, {"DLGS7213", "bytcht_da7213", "intel/fw_sst_22a8.bin", "bytcht_da7213", NULL, &chv_platform_data }, + {"ESSX8316", "bytcht_es8316", "intel/fw_sst_22a8.bin", "cht-bsw", NULL, + &chv_platform_data }, /* some CHT-T platforms rely on RT5640, use Baytrail machine driver */ {"10EC5640", "bytcr_rt5640", "intel/fw_sst_22a8.bin", "bytcr_rt5640", cht_quirk, &chv_platform_data }, diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 56896e09445d..0cca726d249b 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -11,6 +11,7 @@ snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o +snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o @@ -29,6 +30,7 @@ obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o +obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c new file mode 100644 index 000000000000..185d3a233a07 --- /dev/null +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -0,0 +1,300 @@ +/* + * bytcht_es8316.c - ASoc Machine driver for Intel Baytrail/Cherrytrail + * platforms with Everest ES8316 SoC + * + * Copyright (C) 2017 Endless Mobile, Inc. + * 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 as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <asm/platform_sst_audio.h> +#include <linux/clk.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../atom/sst-atom-controls.h" +#include "../common/sst-acpi.h" +#include "../common/sst-dsp.h" + +struct byt_cht_es8316_private { + struct clk *mclk; +}; + +#define CODEC_DAI1 "ES8316 HiFi" + +static inline struct snd_soc_dai *get_codec_dai(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + if (!strncmp(rtd->codec_dai->name, CODEC_DAI1, + strlen(CODEC_DAI1))) + return rtd->codec_dai; + } + return NULL; +} + +static const struct snd_soc_dapm_widget byt_cht_es8316_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + + /* + * The codec supports two analog microphone inputs. I have only + * tested MIC1. A DMIC route could also potentially be added + * if such functionality is found on another platform. + */ + SND_SOC_DAPM_MIC("Microphone 1", NULL), + SND_SOC_DAPM_MIC("Microphone 2", NULL), +}; + +static const struct snd_soc_dapm_route byt_cht_es8316_audio_map[] = { + {"MIC1", NULL, "Microphone 1"}, + {"MIC2", NULL, "Microphone 2"}, + + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + + {"Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx" }, + {"codec_in1", NULL, "ssp2 Rx" }, + {"ssp2 Rx", NULL, "Capture"}, +}; + +static const struct snd_kcontrol_new byt_cht_es8316_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Microphone 1"), + SOC_DAPM_PIN_SWITCH("Microphone 2"), +}; + +static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_card *card = runtime->card; + struct byt_cht_es8316_private *priv = snd_soc_card_get_drvdata(card); + int ret; + + card->dapm.idle_bias_off = true; + + /* + * The firmware might enable the clock at boot (this information + * may or may not be reflected in the enable clock register). + * To change the rate we must disable the clock first to cover these + * cases. Due to common clock framework restrictions that do not allow + * to disable a clock that has not been enabled, we need to enable + * the clock first. + */ + ret = clk_prepare_enable(priv->mclk); + if (!ret) + clk_disable_unprepare(priv->mclk); + + ret = clk_set_rate(priv->mclk, 19200000); + if (ret) + dev_err(card->dev, "unable to set MCLK rate\n"); + + ret = clk_prepare_enable(priv->mclk); + if (!ret) + dev_err(card->dev, "unable to enable MCLK\n"); + + ret = snd_soc_dai_set_sysclk(runtime->codec_dai, 0, 19200000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "can't set codec clock %d\n", ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_pcm_stream byt_cht_es8316_dai_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + int ret; + + /* The DSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + + /* + * Default mode for SSP configuration is TDM 4 slot, override config + * with explicit setting to I2S 2ch 24-bit. The word length is set with + * dai_set_tdm_slot() since there is no other API exposed + */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS + ); + if (ret < 0) { + dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); + return ret; + } + + return 0; +} + +static int byt_cht_es8316_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 48000); +} + +static const struct snd_soc_ops byt_cht_es8316_aif1_ops = { + .startup = byt_cht_es8316_aif1_startup, +}; + +static struct snd_soc_dai_link byt_cht_es8316_dais[] = { + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &byt_cht_es8316_aif1_ops, + }, + + [MERR_DPCM_DEEP_BUFFER] = { + .name = "Deep-Buffer Audio Port", + .stream_name = "Deep-Buffer Audio", + .cpu_dai_name = "deepbuffer-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .ops = &byt_cht_es8316_aif1_ops, + }, + + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + + /* back ends */ + { + /* Only SSP2 has been tested here, so BYT-CR platforms that + * require SSP0 will not work. + */ + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "ES8316 HiFi", + .codec_name = "i2c-ESSX8316:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = byt_cht_es8316_codec_fixup, + .nonatomic = true, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = byt_cht_es8316_init, + }, +}; + + +/* SoC card */ +static struct snd_soc_card byt_cht_es8316_card = { + .name = "bytcht-es8316", + .owner = THIS_MODULE, + .dai_link = byt_cht_es8316_dais, + .num_links = ARRAY_SIZE(byt_cht_es8316_dais), + .dapm_widgets = byt_cht_es8316_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_cht_es8316_widgets), + .dapm_routes = byt_cht_es8316_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_cht_es8316_audio_map), + .controls = byt_cht_es8316_controls, + .num_controls = ARRAY_SIZE(byt_cht_es8316_controls), + .fully_routed = true, +}; + +static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) +{ + int ret = 0; + struct byt_cht_es8316_private *priv; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); + if (!priv) + return -ENOMEM; + + /* register the soc card */ + byt_cht_es8316_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&byt_cht_es8316_card, priv); + + priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (IS_ERR(priv->mclk)) { + ret = PTR_ERR(priv->mclk); + dev_err(&pdev->dev, + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret); + return ret; + } + + ret = devm_snd_soc_register_card(&pdev->dev, &byt_cht_es8316_card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); + return ret; + } + platform_set_drvdata(pdev, &byt_cht_es8316_card); + return ret; +} + +static struct platform_driver snd_byt_cht_es8316_mc_driver = { + .driver = { + .name = "bytcht_es8316", + }, + .probe = snd_byt_cht_es8316_mc_probe, +}; + +module_platform_driver(snd_byt_cht_es8316_mc_driver); +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail Machine driver"); +MODULE_AUTHOR("David Yang yangxiaohua@everest-semi.com"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bytcht_es8316");
On Tue, May 16, 2017 at 02:20:34PM -0600, Daniel Drake wrote:
+static const char * const alc_func_txt[] = { "Off", "On" }; +static const struct soc_enum alc_func =
- SOC_ENUM_SINGLE(ES8316_ADC_ALC1, 6, 2, alc_func_txt);
Use normal boolean controls for simple on/off switches, just a _SINGLE control with a name ending Switch. This applies to many controls in this driver.
+static const struct snd_kcontrol_new es8316_snd_controls[] = {
- SOC_DOUBLE_TLV("HP Playback Volume", ES8316_CPHP_ICAL_VOL,
4, 0, 0, 1, hpout_vol_tlv),- SOC_DOUBLE_TLV("HPMixer Gain", ES8316_HPMIX_VOL,
0, 4, 7, 0, hpmixer_gain_tlv),
All volume controls should end in Volume so that UIs know what to do with them. This is also a repeated problem.
+static int es8316_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)+{
- switch (level) {
- case SND_SOC_BIAS_ON:
snd_soc_write(codec, ES8316_RESET, 0xC0);
Writing to the reset register doens't reset the chip?
snd_soc_write(codec, ES8316_CPHP_OUTEN, 0x66);snd_soc_write(codec, ES8316_DAC_PDN, 0x00);snd_soc_write(codec, ES8316_CPHP_LDOCTL, 0x30);snd_soc_write(codec, ES8316_CPHP_PDN2, 0x10);snd_soc_write(codec, ES8316_CPHP_PDN1, 0x03);snd_soc_write(codec, ES8316_HPMIX_PDN, 0x00);snd_soc_update_bits(codec, ES8316_ADC_PDN_LINSEL, 0xc0, 0x00);snd_soc_write(codec, ES8316_SYS_PDN, 0x00);snd_soc_write(codec, ES8316_SYS_LP1, 0x04);snd_soc_write(codec, ES8316_SYS_LP2, 0x0c);
This all looks like it should be managed by DAPM.
break;- case SND_SOC_BIAS_PREPARE:
break;- case SND_SOC_BIAS_STANDBY:
break;
Transitioning to ON should be undone when we go back down to standby.
+static int es8316_probe(struct snd_soc_codec *codec)
- snd_soc_write(codec, ES8316_CLKMGR_CLKSEL, 0x08);
- snd_soc_write(codec, ES8316_CLKMGR_ADCOSR, 0x32);
- snd_soc_write(codec, ES8316_CLKMGR_ADCDIV1, 0x11);
- snd_soc_write(codec, ES8316_CLKMGR_ADCDIV2, 0x00);
- snd_soc_write(codec, ES8316_CLKMGR_DACDIV1, 0x11);
- snd_soc_write(codec, ES8316_CLKMGR_DACDIV2, 0x00);
- snd_soc_write(codec, ES8316_CLKMGR_CPDIV, 0x00);
- snd_soc_write(codec, ES8316_SERDATA1, 0x04);
- snd_soc_write(codec, ES8316_CLKMGR_CLKSW, 0x7f);
- snd_soc_write(codec, ES8316_CAL_TYPE, 0x0F);
- snd_soc_write(codec, ES8316_CAL_HPLIV, 0x90);
- snd_soc_write(codec, ES8316_CAL_HPRIV, 0x90);
- snd_soc_write(codec, ES8316_ADC_VOLUME, 0x00);
No big undocumented magic register write sequences in probe please, especially not ones that look like they change the default values of user visible controls. The chip should use the hardware defaults where possible to avoid having to decide which use cases to support in the driver. Let userspace pick what works for a given system.
+static const struct i2c_device_id es8316_i2c_id[] = {
- {"ESSX8316", 0 },
- {}
+}; +MODULE_DEVICE_TABLE(i2c, es8316_i2c_id);
Linux I2C device IDs are generally lower case.
participants (2)
-
Daniel Drake -
Mark Brown