/* * DA7210 ALSA Soc codec driver * * Copyright (c) 2009 Dialog Semiconductor * Written by David Chen * * Copyright (C) 2009 Renesas Solutions Corp. * Cleanups by Kuninori Morimoto * * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. */ #include #include #include #include #include #include #include #include #include #include /* DA7210 register space */ #define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 #define DA7210_STARTUP2 0x04 #define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 #define DA7210_AUX1_L 0x09 #define DA7210_AUX1_R 0x0A #define DA7210_AUX2 0x0B #define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F #define DA7210_ADC 0x10 #define DA7210_ADC_EQ1_2 0X11 #define DA7210_ADC_EQ3_4 0x12 #define DA7210_ADC_EQ5 0x13 #define DA7210_DAC_HPF 0x14 #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 #define DA7210_SOFTMUTE 0x18 #define DA7210_DAC_EQ1_2 0x19 #define DA7210_DAC_EQ3_4 0x1A #define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D #define DA7210_OUT1_L 0x1E #define DA7210_OUT1_R 0x1F #define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 #define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 #define DA7210_PLL_DIV1 0x29 #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C #define DA7210_ALC_MAX 0x83 #define DA7210_ALC_MIN 0x84 #define DA7210_ALC_NOIS 0x85 #define DA7210_ALC_ATT 0x86 #define DA7210_ALC_REL 0x87 #define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 #define DA7210_A_CP_MODE 0xA7 /* STARTUP1 bit fields */ #define DA7210_SC_MST_EN (1 << 0) /* MIC_L bit fields */ #define DA7210_MICBIAS_EN (1 << 6) #define DA7210_MIC_L_EN (1 << 7) /* MIC_R bit fields */ #define DA7210_MIC_R_EN (1 << 7) /* INMIX_L bit fields */ #define DA7210_IN_L_EN (1 << 7) /* INMIX_R bit fields */ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ #define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) /* DAC/ADC HPF fields */ #define DA7210_VOICE_F0_MASK (0x7 << 4) #define DA7210_VOICE_F0_25 (1 << 4) #define DA7210_VOICE_EN (1 << 7) /* DAC_SEL bit fields */ #define DA7210_DAC_L_SRC_DAI_L (4 << 0) #define DA7210_DAC_L_EN (1 << 3) #define DA7210_DAC_R_SRC_DAI_R (5 << 4) #define DA7210_DAC_R_EN (1 << 7) /* OUTMIX_L bit fields */ #define DA7210_OUT_L_EN (1 << 7) /* OUTMIX_R bit fields */ #define DA7210_OUT_R_EN (1 << 7) /* HP_CFG bit fields */ #define DA7210_HP_2CAP_MODE (1 << 1) #define DA7210_HP_SENSE_EN (1 << 2) #define DA7210_HP_L_EN (1 << 3) #define DA7210_HP_MODE (1 << 6) #define DA7210_HP_R_EN (1 << 7) /* DAI_SRC_SEL bit fields */ #define DA7210_DAI_OUT_L_SRC (6 << 0) #define DA7210_DAI_OUT_R_SRC (7 << 4) /* DAI_CFG1 bit fields */ #define DA7210_DAI_WORD_S16_LE (0 << 0) #define DA7210_DAI_WORD_S20_3LE (1 << 0) #define DA7210_DAI_WORD_S24_LE (2 << 0) #define DA7210_DAI_WORD_S32_LE (3 << 0) #define DA7210_DAI_FLEN_64BIT (1 << 2) #define DA7210_DAI_MODE_SLAVE (0 << 7) #define DA7210_DAI_MODE_MASTER (1 << 7) /* DAI_CFG3 bit fields */ #define DA7210_DAI_FORMAT_I2SMODE (0 << 0) #define DA7210_DAI_FORMAT_LEFT_J (1 << 0) #define DA7210_DAI_FORMAT_RIGHT_J (2 << 0) #define DA7210_DAI_OE (1 << 3) #define DA7210_DAI_EN (1 << 7) /*PLL_DIV3 bit fields */ #define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) #define DA7210_PLL_BYP (1 << 6) /* PLL bit fields */ #define DA7210_PLL_FS_MASK (0xF << 0) #define DA7210_PLL_FS_8000 (0x1 << 0) #define DA7210_PLL_FS_11025 (0x2 << 0) #define DA7210_PLL_FS_12000 (0x3 << 0) #define DA7210_PLL_FS_16000 (0x5 << 0) #define DA7210_PLL_FS_22050 (0x6 << 0) #define DA7210_PLL_FS_24000 (0x7 << 0) #define DA7210_PLL_FS_32000 (0x9 << 0) #define DA7210_PLL_FS_44100 (0xA << 0) #define DA7210_PLL_FS_48000 (0xB << 0) #define DA7210_PLL_FS_88200 (0xE << 0) #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) /* CONTROL bit fields */ #define DA7210_NOISE_SUP_EN (1 << 3) /* IN_GAIN bit fields */ #define DA7210_INPGA_L_VOL (0x0F << 0) #define DA7210_INPGA_R_VOL (0xF0 << 0) /* ZERO_CROSS bit fields */ #define DA7210_AUX1_L_ZC (1 << 0) #define DA7210_AUX1_R_ZC (1 << 1) #define DA7210_HP_L_ZC (1 << 6) #define DA7210_HP_R_ZC (1 << 7) /* AUX1_L bit fields */ #define DA7210_AUX1_L_VOL (0x3F << 0) /* AUX1_R bit fields */ #define DA7210_AUX1_R_VOL (0x3F << 0) /* Minimum INPGA and AUX1 volume to enable noise suppression */ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ #define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ /* OUT1_L bit fields */ #define DA7210_OUT1_L_EN (1 << 7) /* OUT1_R bit fields */ #define DA7210_OUT1_R_EN (1 << 7) /* OUT2 bit fields */ #define DA7210_OUT2_OUTMIX_R (1 << 5) #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) #define DA7210_VERSION "0.0.1" /* * Playback Volume * * max : 0x3F (+15.0 dB) * (1.5 dB step) * min : 0x11 (-54.0 dB) * mute : 0x10 * reserved : 0x00 - 0x0F * * Reserved area are considered as "mute". */ static const unsigned int hp_out_tlv[] = { TLV_DB_RANGE_HEAD(2), 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), /* -54 dB to +15 dB */ 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), }; static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), /* -54dB to 15dB */ 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) }; static const unsigned int mono_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), /* -18dB to 6dB */ 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); /* ADC and DAC high pass filter f0 value */ static const char const *da7210_hpf_cutoff_txt[] = { "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" }; static const struct soc_enum da7210_dac_hpf_cutoff = SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt); static const struct soc_enum da7210_adc_hpf_cutoff = SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); /* ADC and DAC voice (8kHz) high pass cutoff value */ static const char const *da7210_vf_cutoff_txt[] = { "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" }; static const struct soc_enum da7210_dac_vf_cutoff = SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt); static const struct soc_enum da7210_adc_vf_cutoff = SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); static const char *da7210_hp_mode_txt[] = { "Class H", "Class G" }; static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); /* ALC can be enabled only if noise suppression is disabled */ static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); if (ucontrol->value.integer.value[0]) { /* Check if noise suppression is enabled */ if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) { dev_dbg(codec->dev, "Disable noise suppression to enable ALC\n"); return -EINVAL; } } /* If all conditions are met or we are actually disabling ALC */ return snd_soc_put_volsw(kcontrol, ucontrol); } /* Noise suppression can be enabled only if following conditions are met * ALC disabled * ZC enabled for HP and AUX1 PGA * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB * AUX1_L_VOL and AUX1_R_VOL >= 6 dB */ static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); u8 val; if (ucontrol->value.integer.value[0]) { /* Check if ALC is enabled */ if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN) goto err; /* Check ZC for HP and AUX1 PGA */ if ((snd_soc_read(codec, DA7210_ZERO_CROSS) & (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | DA7210_HP_R_ZC)) != (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | DA7210_HP_R_ZC)) goto err; /* Check INPGA_L_VOL and INPGA_R_VOL */ val = snd_soc_read(codec, DA7210_IN_GAIN); if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || (((val & DA7210_INPGA_R_VOL) >> 4) < DA7210_INPGA_MIN_VOL_NS)) goto err; /* Check AUX1_L_VOL and AUX1_R_VOL */ if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) < DA7210_AUX1_MIN_VOL_NS) || ((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) < DA7210_AUX1_MIN_VOL_NS)) goto err; } /* If all conditions are met or we are actually disabling Noise sup */ return snd_soc_put_volsw(kcontrol, ucontrol); err: return -EINVAL; } static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), SOC_DOUBLE_R_TLV("Digital Playback Volume", DA7210_DAC_L, DA7210_DAC_R, 0, 0x77, 1, dac_gain_tlv), SOC_DOUBLE_R_TLV("Lineout Playback Volume", DA7210_OUT1_L, DA7210_OUT1_R, 0, 0x3f, 0, lineout_vol_tlv), SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, eq_gain_tlv), SOC_SINGLE_TLV("DAC EQ2 Volume", DA7210_DAC_EQ1_2, 4, 0xf, 1, eq_gain_tlv), SOC_SINGLE_TLV("DAC EQ3 Volume", DA7210_DAC_EQ3_4, 0, 0xf, 1, eq_gain_tlv), SOC_SINGLE_TLV("DAC EQ4 Volume", DA7210_DAC_EQ3_4, 4, 0xf, 1, eq_gain_tlv), SOC_SINGLE_TLV("DAC EQ5 Volume", DA7210_DAC_EQ5, 0, 0xf, 1, eq_gain_tlv), /* ADC Equalizer controls */ SOC_SINGLE("ADC EQ Switch", DA7210_ADC_EQ5, 7, 1, 0), SOC_SINGLE_TLV("ADC EQ Master Volume", DA7210_ADC_EQ5, 4, 0x3, 1, adc_eq_master_gain_tlv), SOC_SINGLE_TLV("ADC EQ1 Volume", DA7210_ADC_EQ1_2, 0, 0xf, 1, eq_gain_tlv), SOC_SINGLE_TLV("ADC EQ2 Volume", DA7210_ADC_EQ1_2, 4, 0xf, 1, eq_gain_tlv), SOC_SINGLE_TLV("ADC EQ3 Volume", DA7210_ADC_EQ3_4, 0, 0xf, 1, eq_gain_tlv), SOC_SINGLE_TLV("ADC EQ4 Volume", DA7210_ADC_EQ3_4, 4, 0xf, 1, eq_gain_tlv), SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, eq_gain_tlv), SOC_SINGLE("DAC HPF Switch", DA7210_DAC_HPF, 3, 1, 0), SOC_ENUM("DAC HPF Cutoff", da7210_dac_hpf_cutoff), SOC_SINGLE("DAC Voice Mode Switch", DA7210_DAC_HPF, 7, 1, 0), SOC_ENUM("DAC Voice Cutoff", da7210_dac_vf_cutoff), SOC_SINGLE("ADC HPF Switch", DA7210_ADC_HPF, 3, 1, 0), SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), /* Mute controls */ SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0), SOC_DOUBLE("ADC Capture Switch", DA7210_ADC, 2, 6, 1, 0), SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), /* Zero cross controls */ SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), SOC_ENUM("Headphone Class", da7210_hp_mode_sel), /* ALC controls */ SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, snd_soc_get_volsw, da7210_put_alc_sw), SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0), SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0), SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0), SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; /* * DAPM Controls * * Current DAPM implementation covers almost all codec components e.g. IOs, * mixers, PGAs,ADC and DAC. */ /* In Mixer Left */ static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), }; /* In Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), }; /* Out Mixer Left */ static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), }; /* Out Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), }; /* Mono Mixer */ static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), }; /* DAPM widgets */ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { /* Input Side */ /* Input Lines */ SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), /* Input PGAs */ SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), /* Input Mixers */ SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, &da7210_dapm_inmixl_controls[0], ARRAY_SIZE(da7210_dapm_inmixl_controls)), SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, &da7210_dapm_inmixr_controls[0], ARRAY_SIZE(da7210_dapm_inmixr_controls)), /* ADCs */ SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1), SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1), /* Output Side */ /* DACs */ SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1), SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1), /* Output Mixers */ SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, &da7210_dapm_outmixl_controls[0], ARRAY_SIZE(da7210_dapm_outmixl_controls)), SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, &da7210_dapm_outmixr_controls[0], ARRAY_SIZE(da7210_dapm_outmixr_controls)), SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, &da7210_dapm_monomix_controls[0], ARRAY_SIZE(da7210_dapm_monomix_controls)), /* Output PGAs */ SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("Out1 Left", DA7210_STARTUP2, 0, 1, NULL, 0), SND_SOC_DAPM_PGA("Out1 Right", DA7210_STARTUP2, 1, 1, NULL, 0), SND_SOC_DAPM_PGA("Out2 Mono", DA7210_STARTUP2, 2, 1, NULL, 0), SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), /* Output Lines */ SND_SOC_DAPM_OUTPUT("OUT1L"), SND_SOC_DAPM_OUTPUT("OUT1R"), SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), SND_SOC_DAPM_OUTPUT("OUT2"), }; /* DAPM audio route definition */ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Dest Connecting Widget source */ /* Input path */ {"Mic Left", NULL, "MICL"}, {"Mic Right", NULL, "MICR"}, {"In Mixer Left", "Mic Left Switch", "Mic Left"}, {"In Mixer Left", "Mic Right Switch", "Mic Right"}, {"In Mixer Right", "Mic Right Switch", "Mic Right"}, {"In Mixer Right", "Mic Left Switch", "Mic Left"}, {"INPGA Left", NULL, "In Mixer Left"}, {"ADC Left", NULL, "INPGA Left"}, {"INPGA Right", NULL, "In Mixer Right"}, {"ADC Right", NULL, "INPGA Right"}, /* Output path */ {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, {"Out1 Left", NULL, "OUTPGA Left Enable"}, {"OUT1L", NULL, "Out1 Left"}, {"Out1 Right", NULL, "OUTPGA Right Enable"}, {"OUT1R", NULL, "Out1 Right"}, {"Headphone Left", NULL, "OUTPGA Left Enable"}, {"HPL", NULL, "Headphone Left"}, {"Headphone Right", NULL, "OUTPGA Right Enable"}, {"HPR", NULL, "Headphone Right"}, {"Out2 Mono", NULL, "Mono Mixer"}, {"OUT2", NULL, "Out2 Mono"}, }; /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; }; /* * Register cache */ static const u8 da7210_reg[] = { 0x00, 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R0 - R7 */ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x08, /* R8 - RF */ 0x00, 0x00, 0x00, 0x00, 0x08, 0x10, 0x10, 0x54, /* R10 - R17 */ 0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R18 - R1F */ 0x00, 0x00, 0x00, 0x02, 0x00, 0x76, 0x00, 0x00, /* R20 - R27 */ 0x04, 0x00, 0x00, 0x30, 0x2A, 0x00, 0x40, 0x00, /* R28 - R2F */ 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, /* R30 - R37 */ 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x00, 0x00, /* R38 - R3F */ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R40 - R4F */ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R48 - R4F */ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R50 - R57 */ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R58 - R5F */ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R60 - R67 */ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R68 - R6F */ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R70 - R77 */ 0x00, 0x00, 0x00, 0x00, 0x00, 0x54, 0x54, 0x00, /* R78 - R7F */ 0x00, 0x00, 0x2C, 0x00, 0x00, 0x00, 0x00, 0x00, /* R80 - R87 */ 0x00, /* R88 */ }; static int da7210_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case DA7210_STATUS: return 1; default: return 0; } } /* * Set PCM DAI word length. */ static int da7210_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; u32 dai_cfg1; u32 fs, bypass; /* set DAI source to Left and Right ADC */ snd_soc_write(codec, DA7210_DAI_SRC_SEL, DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC); /* Enable DAI */ snd_soc_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: dai_cfg1 |= DA7210_DAI_WORD_S16_LE; break; case SNDRV_PCM_FORMAT_S20_3LE: dai_cfg1 |= DA7210_DAI_WORD_S20_3LE; break; case SNDRV_PCM_FORMAT_S24_LE: dai_cfg1 |= DA7210_DAI_WORD_S24_LE; break; case SNDRV_PCM_FORMAT_S32_LE: dai_cfg1 |= DA7210_DAI_WORD_S32_LE; break; default: return -EINVAL; } snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; bypass = DA7210_PLL_BYP; break; default: return -EINVAL; } /* Disable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); /* Enable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, DA7210_SC_MST_EN); return 0; } /* * Set DAI mode and Format */ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) { struct snd_soc_codec *codec = codec_dai->codec; u32 dai_cfg1; u32 dai_cfg3; dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: dai_cfg1 |= DA7210_DAI_MODE_SLAVE; break; default: return -EINVAL; } /* FIXME * * It support I2S only now */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE; break; case SND_SOC_DAIFMT_LEFT_J: dai_cfg3 |= DA7210_DAI_FORMAT_LEFT_J; break; case SND_SOC_DAIFMT_RIGHT_J: dai_cfg3 |= DA7210_DAI_FORMAT_RIGHT_J; break; default: return -EINVAL; } /* FIXME * * It support 64bit data transmission only now */ dai_cfg1 |= DA7210_DAI_FLEN_64BIT; snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); snd_soc_write(codec, DA7210_DAI_CFG3, dai_cfg3); return 0; } static int da7210_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB; if (mute) snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4); else snd_soc_write(codec, DA7210_DAC_HPF, mute_reg); return 0; } #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, .digital_mute = da7210_mute, }; static struct snd_soc_dai_driver da7210_dai = { .name = "da7210-hifi", /* playback capabilities */ .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = DA7210_FORMATS, }, /* capture capabilities */ .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = DA7210_FORMATS, }, .ops = &da7210_dai_ops, .symmetric_rates = 1, }; static int da7210_probe(struct snd_soc_codec *codec) { struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); int ret; dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } /* FIXME * * This driver use fixed value here * And below settings expects MCLK = 12.288MHz * * When you select different MCLK, please check... * DA7210_PLL_DIV1 val * DA7210_PLL_DIV2 val * DA7210_PLL_DIV3 val * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx */ /* * make sure that DA7210 use bypass mode before start up */ snd_soc_write(codec, DA7210_STARTUP1, 0); snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); /* * ADC settings */ /* Enable Left & Right MIC PGA and Mic Bias */ snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); /* Enable Left and Right input PGA */ snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); snd_soc_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); /* Enable Left and Right ADC */ snd_soc_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); /* * DAC settings */ /* Enable Left and Right DAC */ snd_soc_write(codec, DA7210_DAC_SEL, DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN | DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN); /* Enable Left and Right out PGA */ snd_soc_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); snd_soc_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); /* Enable Left and Right HeadPhone PGA */ snd_soc_write(codec, DA7210_HP_CFG, DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); /* Enable ramp mode for DAC gain update */ snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); /* * For DA7210 codec, there are two ways to enable/disable analog IOs * and ADC/DAC, * (1) Using "Enable Bit" of register associated with that IO * (or ADC/DAC) * e.g. Mic Left can be enabled using bit 7 of MIC_L(0x7) reg * * (2) Using "Standby Bit" of STARTUP2 or STARTUP3 register * e.g. Mic left can be put to STANDBY using bit 0 of STARTUP3(0x5) * * Out of these two methods, the one using STANDBY bits is preferred * way to enable/disable individual blocks. This is because STANDBY * registers are part of system controller which allows system power * up/down in a controlled, pop-free manner. Also, as per application * note of DA7210, STANDBY register bits are only effective if a * particular IO (or ADC/DAC) is already enabled using enable/disable * register bits. Keeping these things in mind, current DAPM * implementation manipulates only STANDBY bits. * * Overall implementation can be outlined as below, * * - "Enable bit" of an IO or ADC/DAC is used to enable it in probe() * - "STANDBY bit" is controlled by DAPM */ /* Enable Line out amplifiers */ snd_soc_write(codec, DA7210_OUT1_L, DA7210_OUT1_L_EN); snd_soc_write(codec, DA7210_OUT1_R, DA7210_OUT1_R_EN); snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); /* * If 48kHz sound came, it use bypass mode, * and when it is 44.1kHz, it use PLL. * * This time, this driver sets PLL always ON * and controls bypass/PLL mode by switching * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. * see da7210_hw_params */ snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); /* As suggested by Dialog */ snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); snd_soc_write(codec, DA7210_A_PLL1, 0x01); snd_soc_write(codec, DA7210_A_CP_MODE, 0x7C); snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0x00); /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); return 0; } static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .probe = da7210_probe, .reg_cache_size = ARRAY_SIZE(da7210_reg), .reg_word_size = sizeof(u8), .reg_cache_default = da7210_reg, .volatile_register = da7210_volatile_register, .controls = da7210_snd_controls, .num_controls = ARRAY_SIZE(da7210_snd_controls), .dapm_widgets = da7210_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(da7210_dapm_widgets), .dapm_routes = da7210_audio_map, .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int __devinit da7210_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct da7210_priv *da7210; int ret; da7210 = kzalloc(sizeof(struct da7210_priv), GFP_KERNEL); if (!da7210) return -ENOMEM; i2c_set_clientdata(i2c, da7210); da7210->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7210, &da7210_dai, 1); if (ret < 0) kfree(da7210); return ret; } static int __devexit da7210_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); kfree(i2c_get_clientdata(client)); return 0; } static const struct i2c_device_id da7210_i2c_id[] = { { "da7210", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); /* I2C codec control layer */ static struct i2c_driver da7210_i2c_driver = { .driver = { .name = "da7210-codec", .owner = THIS_MODULE, }, .probe = da7210_i2c_probe, .remove = __devexit_p(da7210_i2c_remove), .id_table = da7210_i2c_id, }; #endif static int __init da7210_modinit(void) { int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&da7210_i2c_driver); #endif return ret; } module_init(da7210_modinit); static void __exit da7210_exit(void) { #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&da7210_i2c_driver); #endif } module_exit(da7210_exit); MODULE_DESCRIPTION("ASoC DA7210 driver"); MODULE_AUTHOR("David Chen, Kuninori Morimoto"); MODULE_LICENSE("GPL");