[alsa-devel] [PATCH v4] sound/soc/codecs: add LAPIS Semiconductor ML26124
ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high resistance to voltage noise. On chip regulator realizes power supply rejection ratio be over 90dB so more than 50dB is improved than ever. ML26124-01HB/ ML26124-02GD can deliver stable audio performance without being affected by noise from the power supply circuit and peripheral components. The chip also includes a composite video signal output, which can be applied to various portable device requirements. The ML26124 is realized these functions into very small package the size is only 2.56mm x 2.46mm therefore can be construct high quality sound system easily. ML26124-01HB is 25pin WCSP package; ML26124-02GD is 32pin WQFN package.
Signed-off-by: Tomoya MORINAGA tomoya.rohm@gmail.com --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ml26124.c | 595 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ml26124.h | 122 +++++++++ 4 files changed, 723 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ml26124.c create mode 100644 sound/soc/codecs/ml26124.h
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 665d924..ce8e622 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -38,6 +38,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_ML26124 if I2C select SND_SOC_PCM3008 select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -211,6 +212,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate
+config SND_SOC_ML26124 + tristate + config SND_SOC_PCM3008 tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5119a7e..c1353ff 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -24,6 +24,7 @@ snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-ml26124-objs := ml26124.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -122,6 +123,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c new file mode 100644 index 0000000..7753a9c --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,595 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "ml26124.h" + +struct ml26124_priv { + u32 mclk; + u32 channels; +}; + +/* ML26124 configuration */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0); +static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); + +static const char * const ml26124_companding[] = {"16bit PCM", "u-law", + "A-law"}; +static const struct soc_enum ml26124_adc_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); +static const struct soc_enum ml26124_dac_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); + +static const struct snd_kcontrol_new ml26124_snd_controls[] = { + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE_TLV("EQ Band0 Input Volume", ML26124_EQ_GAIN_BRAND0, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band1 Input Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band2 Input Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band3 Input Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band4 Input Volume", ML26124_EQ_GAIN_BRAND4, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, + 0xf, 1, alclvl), + SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, + 7, 0, mingain), + SOC_SINGLE_TLV("ALC MAX Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, + 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Limitter Min Input Volume", + ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain), + SOC_SINGLE_TLV("Playback Limitter Max Input Volume", + ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), + SOC_SINGLE("NOISE High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), + SOC_SINGLE("Zero Cross Comparator Switch", ML26124_PW_ZCCMP_PW_MNG, 1, + 1, 0), + SOC_SINGLE("EQ BAND0 Switch", ML26124_FILTER_EN, 2, 1, 0), + SOC_SINGLE("EQ BAND1 Switch", ML26124_FILTER_EN, 3, 1, 0), + SOC_SINGLE("EQ BAND2 Switch", ML26124_FILTER_EN, 4, 1, 0), + SOC_SINGLE("EQ BAND3 Switch", ML26124_FILTER_EN, 5, 1, 0), + SOC_SINGLE("EQ BAND4 Switch", ML26124_FILTER_EN, 6, 1, 0), + SOC_SINGLE("Play Limitter", ML26124_DVOL_CTL, 0, 1, 0), + SOC_SINGLE("Capture Limitter", ML26124_DVOL_CTL, 1, 1, 0), + SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), + SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0), + SOC_ENUM("DAC Companding", ml26124_dac_companding_enum), + SOC_ENUM("ADC Companding", ml26124_adc_companding_enum), +}; + +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), + SOC_DAPM_SINGLE("Line in", ML26124_SPK_AMP_OUT, 3, 1, 0), + SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), +}; + +/* Input mux */ +static const char * const ml26124_input_select[] = {"Analog MIC in", + "Digital MIC in"}; + +static const struct soc_enum ml26124_insel_enum = + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 1, ml26124_input_select); + +static const struct snd_kcontrol_new ml26124_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); + +static const struct snd_kcontrol_new ml26124_line_control = + SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0); + +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MCLK", ML26124_CLK_EN, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", ML26124_CLK_EN, 1, 0, NULL, 0), + SND_SOC_DAPM_MICBIAS("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 0), + SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0, 0, + &ml26124_output_mixer_controls[0], + ARRAY_SIZE(ml26124_output_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), + SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ml26124_input_mux_controls), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ml26124_line_control), + SND_SOC_DAPM_INPUT("MDIN"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_OUTPUT("SPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("MICBIASO"), +}; + +static const struct snd_soc_dapm_route ml26124_intercon[] = { + /* Supply */ + {"DAC", NULL, "MCLK"}, + {"ADC", NULL, "MCLK"}, + {"DAC", NULL, "PLL"}, + {"ADC", NULL, "PLL"}, + + /* output mixer */ + {"Output Mixer", "PGA Switch", "PGA"}, + {"Output Mixer", "DAC Switch", "DAC"}, + {"Output Mixer", NULL, "Line in"}, + + /* outputs */ + {"SPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + {"MICBIASO", NULL, "MICBIAS"}, + {"LOUT", NULL, "Line Out Enable"}, + + /* input mux */ + {"Input Mux", "Analog MIC in", "MICBIAS"}, + {"Input Mux", "Digital MIC in", "MICBIAS"}, + {"ADC", NULL, "Input Mux"}, + + /* inputs */ + {"DAC", NULL, "MDIN"}, + {"PGA", NULL, "MIN"}, + {"SPOUT", "Line Out Enable", "LIN"}, +}; + +struct clk_coeff { + u32 mclk; + u32 rate; + u8 pllnl; + u8 pllnh; + u8 pllml; + u8 pllmh; + u8 plldiv; +}; + +/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */ +static const struct clk_coeff coeff_div[] = { + {12288800, 16000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288800, 32000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288800, 48000, 0xc, 0x0, 0x30, 0x0, 0x4}, +}; + +/* Get sampling rate value of sampling rate setting register (0x0) */ +static inline int get_srate(int rate) +{ + int srate; + + switch (rate) { + case 16000: + srate = 3; + break; + case 32000: + srate = 6; + break; + case 48000: + srate = 8; + break; + default: + return -EINVAL; + } + return srate; +} + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int ml26124_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + int i = get_coeff(priv->mclk, params_rate(hw_params)); + + priv->channels = params_channels(hw_params); + + switch (params_rate(hw_params)) { + case 16000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 32000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 48000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + default: + pr_err("%s:this rate is no support for ml26124\n", __func__); + return -EINVAL; + } + + return 0; +} + +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ + +static int ml26124_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_ON); + else + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_OFF); + return 0; +} + +#define ML26124_SAI_NO_DELAY BIT(1) +#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */ + +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + unsigned char mode; + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned char mask = ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + mode = 0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + /* SAI configuration */ + if (priv->channels == 1) { + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + } else { + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY); + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY); + } + + return 0; +} + +static int ml26124_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 ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 12288000: + priv->mclk = freq; + break; + default: + return -EINVAL; + } + return 0; +} + +#define ML26134_CACHESIZE 79 +static const u16 ml26124_reg[ML26134_CACHESIZE] = { + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, + 0x0, 0x0, 0x0, /* System Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* Power Management */ + 0x4, /* Analog Reference Control */ + 0x10, 0x0, 0x33, 0x0, 0x0, /* Input/Output Amplifier Control */ + 0x0, 0x0, 0x1, /* Analog Path Contorl */ + 0x0, 0x0, 0x0, /* Audio Interface Control */ + 0x1, 0x0, 0x0, 0xff, 0xff, 0x10, /* DSP Control */ + 0x7, 0x7, 0x7, 0x7, 0x7, /* DSP Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x2, 0x3, 0x0, 0xb, 0x70, 0x0, 0x0, /* ALC Control */ + 0x4, 0x5, 0xd, 0x70, 0x10, 0x0, /* Playback Limiter Control */ + 0x1, 0x1, 0x1 /* Video Amplifer Control */ +}; + +#define ML26124_VMID BIT(0) + +static void ml26124_sync_cache(struct snd_soc_codec *codec) +{ + u8 *reg_cache = codec->reg_cache; + int i; + + if (!codec->cache_sync) + return; + + codec->cache_only = 0; + + for (i = 1; i < codec->driver->reg_cache_size; i++) { + if (reg_cache[i] == ml26124_reg[i]) + continue; + + snd_soc_write(codec, i, reg_cache[i]); + } + + codec->cache_sync = 0; +} + +static int ml26124_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + /* VMID ON */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, ML26124_VMID); + msleep(500); + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ml26124_sync_cache(codec); + break; + case SND_SOC_BIAS_OFF: + /* VMID OFF */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int ml26124_pcm_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (cmd == SNDRV_PCM_TRIGGER_STOP) { + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, 0); + return 0; + } else if (cmd == SNDRV_PCM_TRIGGER_START) { + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, + 1); + else + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, + 2); + } + + return 0; +} + +#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000) +#define ML26124_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static const struct snd_soc_dai_ops ml26124_dai_ops = { + .hw_params = ml26124_hw_params, + .digital_mute = ml26124_mute, + .set_fmt = ml26124_set_dai_fmt, + .set_sysclk = ml26124_set_dai_sysclk, + .trigger = ml26124_pcm_trigger, +}; + +static struct snd_soc_dai_driver ml26124_dai = { + .name = "ml26124-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .ops = &ml26124_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int ml26124_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int ml26124_resume(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define ml26124_suspend NULL +#define ml26124_resume NULL +#endif + +static int ml26124_probe(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Software Reset */ + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); + + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* power down chip */ +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { + .probe = ml26124_probe, + .suspend = ml26124_suspend, + .resume = ml26124_resume, + .set_bias_level = ml26124_set_bias_level, + .reg_cache_size = ML26134_CACHESIZE, + .reg_word_size = sizeof(u8), + .reg_cache_default = ml26124_reg, + .dapm_widgets = ml26124_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), + .dapm_routes = ml26124_intercon, + .num_dapm_routes = ARRAY_SIZE(ml26124_intercon), + .controls = ml26124_snd_controls, + .num_controls = ARRAY_SIZE(ml26124_snd_controls), +}; + +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + return snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ml26124_i2c_id[] = { + { "ml26124", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); + +static struct i2c_driver ml26124_i2c_driver = { + .driver = { + .name = "ml26124", + .owner = THIS_MODULE, + }, + .probe = ml26124_i2c_probe, + .remove = __devexit_p(ml26124_i2c_remove), + .id_table = ml26124_i2c_id, +}; + +static int __init ml26124_modinit(void) +{ + int ret = 0; + + ret = i2c_add_driver(&ml26124_i2c_driver); + if (ret != 0) + pr_err("Failed to register ML26124 I2C driver: %d\n", ret); + + return ret; +} +module_init(ml26124_modinit); + +static void __exit ml26124_exit(void) +{ + i2c_del_driver(&ml26124_i2c_driver); +} +module_exit(ml26124_exit); + +MODULE_AUTHOR("Tomoya MORINAGA tomoya-linux@dsn.lapis-semi.com"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h new file mode 100644 index 0000000..d30b530 --- /dev/null +++ b/sound/soc/codecs/ml26124.h @@ -0,0 +1,122 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML26124_H +#define ML26124_H + +/* Clock Control Register */ +#define ML26124_SMPLING_RATE 0x00 +#define ML26124_PLLNL 0x02 +#define ML26124_PLLNH 0x04 +#define ML26124_PLLML 0x06 +#define ML26124_PLLMH 0x08 +#define ML26124_PLLDIV 0x0a +#define ML26124_CLK_EN 0x0c +#define ML26124_CLK_CTL 0x0e + +/* System Control Register */ +#define ML26124_SW_RST 0x10 +#define ML26124_REC_PLYBAK_RUN 0x12 +#define ML26124_MIC_TIM 0x14 + +/* Power Mnagement Register */ +#define ML26124_PW_REF_PW_MNG 0x20 +#define ML26124_PW_IN_PW_MNG 0x22 +#define ML26124_PW_DAC_PW_MNG 0x24 +#define ML26124_PW_SPAMP_PW_MNG 0x26 +#define ML26124_PW_LOUT_PW_MNG 0x28 +#define ML26124_PW_VOUT_PW_MNG 0x2a +#define ML26124_PW_ZCCMP_PW_MNG 0x2e + +/* Analog Reference Control Register */ +#define ML26124_PW_MICBIAS_VOL 0x30 + +/* Input/Output Amplifier Control Register */ +#define ML26124_PW_MIC_IN_VOL 0x32 +#define ML26124_PW_MIC_BOST_VOL 0x38 +#define ML26124_PW_SPK_AMP_VOL 0x3a +#define ML26124_PW_AMP_VOL_FUNC 0x48 +#define ML26124_PW_AMP_VOL_FADE 0x4a + +/* Analog Path Control Register */ +#define ML26124_SPK_AMP_OUT 0x54 +#define ML26124_MIC_IF_CTL 0x5a +#define ML26124_MIC_SELECT 0xe8 + +/* Audio Interface Control Register */ +#define ML26124_SAI_TRANS_CTL 0x60 +#define ML26124_SAI_RCV_CTL 0x62 +#define ML26124_SAI_MODE_SEL 0x64 + +/* DSP Control Register */ +#define ML26124_FILTER_EN 0x66 +#define ML26124_DVOL_CTL 0x68 +#define ML26124_MIXER_VOL_CTL 0x6a +#define ML26124_RECORD_DIG_VOL 0x6c +#define ML26124_PLBAK_DIG_VOL 0x70 +#define ML26124_DIGI_BOOST_VOL 0x72 +#define ML26124_EQ_GAIN_BRAND0 0x74 +#define ML26124_EQ_GAIN_BRAND1 0x76 +#define ML26124_EQ_GAIN_BRAND2 0x78 +#define ML26124_EQ_GAIN_BRAND3 0x7a +#define ML26124_EQ_GAIN_BRAND4 0x7c +#define ML26124_HPF2_CUTOFF 0x7e +#define ML26124_EQBRAND0_F0L 0x80 +#define ML26124_EQBRAND0_F0H 0x82 +#define ML26124_EQBRAND0_F1L 0x84 +#define ML26124_EQBRAND0_F1H 0x86 +#define ML26124_EQBRAND1_F0L 0x88 +#define ML26124_EQBRAND1_F0H 0x8a +#define ML26124_EQBRAND1_F1L 0x8c +#define ML26124_EQBRAND1_F1H 0x8e +#define ML26124_EQBRAND2_F0L 0x90 +#define ML26124_EQBRAND2_F0H 0x92 +#define ML26124_EQBRAND2_F1L 0x94 +#define ML26124_EQBRAND2_F1H 0x96 +#define ML26124_EQBRAND3_F0L 0x98 +#define ML26124_EQBRAND3_F0H 0x9a +#define ML26124_EQBRAND3_F1L 0x9c +#define ML26124_EQBRAND3_F1H 0x9e +#define ML26124_EQBRAND4_F0L 0xa0 +#define ML26124_EQBRAND4_F0H 0xa2 +#define ML26124_EQBRAND4_F1L 0xa4 +#define ML26124_EQBRAND4_F1H 0xa6 + +/* ALC Control Register */ +#define ML26124_ALC_MODE 0xb0 +#define ML26124_ALC_ATTACK_TIM 0xb2 +#define ML26124_ALC_DECAY_TIM 0xb4 +#define ML26124_ALC_HOLD_TIM 0xb6 +#define ML26124_ALC_TARGET_LEV 0xb8 +#define ML26124_ALC_MAXMIN_GAIN 0xba +#define ML26124_NOIS_GATE_THRSH 0xbc +#define ML26124_ALC_ZERO_TIMOUT 0xbe + +/* Playback Limiter Control Register */ +#define ML26124_PL_ATTACKTIME 0xc0 +#define ML26124_PL_DECAYTIME 0xc2 +#define ML26124_PL_TARGETTIME 0xc4 +#define ML26124_PL_MAXMIN_GAIN 0xc6 +#define ML26124_PLYBAK_BOST_VOL 0xc8 +#define ML26124_PL_0CROSS_TIMOUT 0xca + +/* Video Amplifer Control Register */ +#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0 +#define ML26124_VIDEO_AMP_SETUP1 0xd2 +#define ML26124_VIDEO_AMP_CTL2 0xd4 + +#endif
On Tue, Nov 29, 2011 at 09:28:56PM +0900, Tomoya MORINAGA wrote:
ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high resistance to voltage noise. On chip regulator realizes power supply rejection ratio be over 90dB so more than 50dB is improved than ever. ML26124-01HB/
Given the questions you just asked myself and Lars-Peter it seems fairly clear that the review comments from the previous version so I'm not going to review this, please resend once the review comments have been addressed. Lars-Peter concern seemed especially serious, I'd expect large sections of the audio functionality to not work without being connected to the DAPM map.
In general if resubmitting code that has outstanding review issues you should highlight the known issues and explain why you are resending; usually if there's an active discussion in progress it's better to wait for that to die down.
Hi Mark and Peter,
On Tue, Nov 29, 2011 at 03:38:00PM +0100, Lars-Peter Clausen wrote:>>> Also this doesn't make much sense:>> + {"Input Mux", "Analog MIC in", "MICBIAS"},>> + {"Input Mux", "Digital MIC in", "MICBIAS"},
Let me discuss about intercon settings.
Anticipated routes are like below. Analog MIC(single ended) -> PGA -> ADC -> SAI Analog MIC(differential) -> PGA -> ADC -> SAI Digital MIC -> SAI SAI -> DAC -> SPOUT SAI -> DAC -> LOUT LINEIN -> SPOUT
So, I set like below. Please give us your opinion.
/* ML26124 configuration */ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0); static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0);
static const char * const ml26124_companding[] = {"16bit PCM", "u-law", "A-law"}; static const struct soc_enum ml26124_adc_companding_enum = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); static const struct soc_enum ml26124_dac_companding_enum = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding);
static const struct snd_kcontrol_new ml26124_snd_controls[] = { SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, 0xff, 1, digital_tlv), SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, 0xff, 1, digital_tlv), SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, 0x3f, 0, boost_vol), SOC_SINGLE_TLV("EQ Band0 Input Volume", ML26124_EQ_GAIN_BRAND0, 0, 0xff, 1, digital_tlv), SOC_SINGLE_TLV("EQ Band1 Input Volume", ML26124_EQ_GAIN_BRAND1, 0, 0xff, 1, digital_tlv), SOC_SINGLE_TLV("EQ Band2 Input Volume", ML26124_EQ_GAIN_BRAND2, 0, 0xff, 1, digital_tlv), SOC_SINGLE_TLV("EQ Band3 Input Volume", ML26124_EQ_GAIN_BRAND3, 0, 0xff, 1, digital_tlv), SOC_SINGLE_TLV("EQ Band4 Input Volume", ML26124_EQ_GAIN_BRAND4, 0, 0xff, 1, digital_tlv), SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, 0xf, 1, alclvl), SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, 7, 0, mingain), SOC_SINGLE_TLV("ALC MAX Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, 7, 1, maxgain), SOC_SINGLE_TLV("Playback Limitter Min Input Volume", ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain), SOC_SINGLE_TLV("Playback Limitter Max Input Volume", ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain), SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, 0x3f, 0, boost_vol), SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), SOC_SINGLE("NOISE High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), SOC_SINGLE("Zero Cross Comparator Switch", ML26124_PW_ZCCMP_PW_MNG, 1, 1, 0), SOC_SINGLE("EQ BAND0 Switch", ML26124_FILTER_EN, 2, 1, 0), SOC_SINGLE("EQ BAND1 Switch", ML26124_FILTER_EN, 3, 1, 0), SOC_SINGLE("EQ BAND2 Switch", ML26124_FILTER_EN, 4, 1, 0), SOC_SINGLE("EQ BAND3 Switch", ML26124_FILTER_EN, 5, 1, 0), SOC_SINGLE("EQ BAND4 Switch", ML26124_FILTER_EN, 6, 1, 0), SOC_SINGLE("Play Limitter", ML26124_DVOL_CTL, 0, 1, 0), SOC_SINGLE("Capture Limitter", ML26124_DVOL_CTL, 1, 1, 0), SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0), SOC_ENUM("DAC Companding", ml26124_dac_companding_enum), SOC_ENUM("ADC Companding", ml26124_adc_companding_enum), };
static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), SOC_DAPM_SINGLE("Line in", ML26124_SPK_AMP_OUT, 3, 1, 0), SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), };
/* Input mux */ static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in", "Analog MIC Differential in" "Digital MIC in"};
static const struct soc_enum ml26124_insel_enum = SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 2, ml26124_input_select);
static const struct snd_kcontrol_new ml26124_input_mux_controls = SOC_DAPM_ENUM("Input Select", ml26124_insel_enum);
static const struct snd_kcontrol_new ml26124_line_control = SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0);
static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("MCLK", ML26124_CLK_EN, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("PLL", ML26124_CLK_EN, 1, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 0), SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0, 0, &ml26124_output_mixer_controls[0], ARRAY_SIZE(ml26124_output_mixer_controls)), SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ml26124_input_mux_controls), SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, &ml26124_line_control), SND_SOC_DAPM_INPUT("MDIN"), SND_SOC_DAPM_INPUT("MIN"), SND_SOC_DAPM_INPUT("LIN"), SND_SOC_DAPM_OUTPUT("SPOUT"), SND_SOC_DAPM_OUTPUT("LOUT"), };
static const struct snd_soc_dapm_route ml26124_intercon[] = { /* Supply */ {"DAC", NULL, "MCLK"}, {"ADC", NULL, "MCLK"}, {"DAC", NULL, "PLL"}, {"ADC", NULL, "PLL"},
/* output mixer */ {"Output Mixer", "PGA Switch", "PGA"}, {"Output Mixer", "DAC Switch", "DAC"}, {"Output Mixer", NULL, "Line in"},
/* outputs */ {"LOUT", "Line Out Enable", "Output Mixer"}, {"SPOUT", NULL, "Output Mixer"},
/* input */ {"Input Mux", "Analog MIC SingleEnded in", "MIN"}, {"Input Mux", "Analog MIC Differential in", "MIN"}, {"Input Mux", "Digital MIC in", "MDIN"}, {"ADC", "PGA", "Input Mux"}, };
participants (2)
-
Mark Brown -
Tomoya MORINAGA