/* * alc5623.c -- alc562[123] ALSA Soc Audio driver * * Copyright 2008 Realtek Microelectronics * Author: flove Ethan * * Copyright 2010 Arnaud Patard * * * Based on WM8753.c * * 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 #include #include #include #include #include #include #include #include #include #include #include #include #include #include "alc5623.h" static int caps_charge = 2000; module_param(caps_charge, int, 0); MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); /* codec private data */ struct alc5623_priv { enum snd_soc_control_type control_type; u8 id; unsigned int sysclk; u16 reg_cache[ALC5623_VENDOR_ID2+2]; unsigned int add_ctrl; unsigned int jack_det_ctrl; }; static void alc5623_fill_cache(struct snd_soc_codec *codec) { int i, step = codec->driver->reg_cache_step; u16 *cache = codec->reg_cache; /* not really efficient ... */ codec->cache_bypass = 1; for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) cache[i] = snd_soc_read(codec, i); codec->cache_bypass = 0; } static inline int alc5623_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, ALC5623_RESET, 0); } static int amp_mixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { /* to power-on/off class-d amp generators/speaker */ /* need to write to 'index-46h' register : */ /* so write index num (here 0x46) to reg 0x6a */ /* and then 0xffff/0 to reg 0x6c */ snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46); switch (event) { case SND_SOC_DAPM_PRE_PMU: snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF); break; case SND_SOC_DAPM_POST_PMD: snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0); break; } return 0; } /* * ALC5623 Controls */ static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); static const unsigned int boost_tlv[] = { TLV_DB_RANGE_HEAD(3), 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), }; static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); static const struct snd_kcontrol_new alc5621_vol_snd_controls[] = { SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Speaker Playback Switch", ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), SOC_DOUBLE_TLV("Headphone Playback Volume", ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Headphone Playback Switch", ALC5623_HP_OUT_VOL, 15, 7, 1, 1), }; static const struct snd_kcontrol_new alc5622_vol_snd_controls[] = { SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Speaker Playback Switch", ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), SOC_DOUBLE_TLV("Line Playback Volume", ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Line Playback Switch", ALC5623_HP_OUT_VOL, 15, 7, 1, 1), }; static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = { SOC_DOUBLE_TLV("Line Playback Volume", ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Line Playback Switch", ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), SOC_DOUBLE_TLV("Headphone Playback Volume", ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Headphone Playback Switch", ALC5623_HP_OUT_VOL, 15, 7, 1, 1), }; static const struct snd_kcontrol_new alc5623_snd_controls[] = { SOC_DOUBLE_TLV("Auxout Playback Volume", ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Auxout Playback Switch", ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1), SOC_DOUBLE_TLV("PCM Playback Volume", ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv), SOC_DOUBLE_TLV("AuxI Capture Volume", ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv), SOC_DOUBLE_TLV("LineIn Capture Volume", ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), SOC_SINGLE_TLV("Mic1 Capture Volume", ALC5623_MIC_VOL, 8, 31, 1, vol_tlv), SOC_SINGLE_TLV("Mic2 Capture Volume", ALC5623_MIC_VOL, 0, 31, 1, vol_tlv), SOC_DOUBLE_TLV("Rec Capture Volume", ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv), SOC_SINGLE_TLV("Mic 1 Boost Volume", ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv), SOC_SINGLE_TLV("Mic 2 Boost Volume", ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv), SOC_SINGLE_TLV("Digital Boost Volume", ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv), }; /* * DAPM Controls */ static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = { SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1), SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1), SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1), SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1), SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1), }; static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = { SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1), }; static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = { SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1), }; static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = { SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1), SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1), SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1), SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1), SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1), SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1), SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1), }; static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = { SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1), SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1), SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1), SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1), SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1), }; /* Left Record Mixer */ static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = { SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1), SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1), SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1), SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1), SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1), SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1), SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1), }; /* Right Record Mixer */ static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = { SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1), SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1), SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1), SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1), SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1), SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1), SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1), }; static const char *alc5623_spk_n_sour_sel[] = { "RN/-R", "RP/+R", "LN/-R", "Vmid" }; static const char *alc5623_hpl_out_input_sel[] = { "Vmid", "HP Left Mix"}; static const char *alc5623_hpr_out_input_sel[] = { "Vmid", "HP Right Mix"}; static const char *alc5623_spkout_input_sel[] = { "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; static const char *alc5623_aux_out_input_sel[] = { "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; /* auxout output mux */ static const struct soc_enum alc5623_aux_out_input_enum = SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel); static const struct snd_kcontrol_new alc5623_auxout_mux_controls = SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum); /* speaker output mux */ static const struct soc_enum alc5623_spkout_input_enum = SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel); static const struct snd_kcontrol_new alc5623_spkout_mux_controls = SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum); /* headphone left output mux */ static const struct soc_enum alc5623_hpl_out_input_enum = SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel); static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls = SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum); /* headphone right output mux */ static const struct soc_enum alc5623_hpr_out_input_enum = SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel); static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls = SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum); /* speaker output N select */ static const struct soc_enum alc5623_spk_n_sour_enum = SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel); static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls = SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum); static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = { /* Muxes */ SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, &alc5623_auxout_mux_controls), SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, &alc5623_spkout_mux_controls), SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &alc5623_hpl_out_mux_controls), SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &alc5623_hpr_out_mux_controls), SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, &alc5623_spkoutn_mux_controls), /* output mixers */ SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, &alc5623_hp_mixer_controls[0], ARRAY_SIZE(alc5623_hp_mixer_controls)), SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0, &alc5623_hpr_mixer_controls[0], ARRAY_SIZE(alc5623_hpr_mixer_controls)), SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0, &alc5623_hpl_mixer_controls[0], ARRAY_SIZE(alc5623_hpl_mixer_controls)), SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0, &alc5623_mono_mixer_controls[0], ARRAY_SIZE(alc5623_mono_mixer_controls)), SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0, &alc5623_speaker_mixer_controls[0], ARRAY_SIZE(alc5623_speaker_mixer_controls)), /* input mixers */ SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0, &alc5623_captureL_mixer_controls[0], ARRAY_SIZE(alc5623_captureL_mixer_controls)), SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0, &alc5623_captureR_mixer_controls[0], ARRAY_SIZE(alc5623_captureR_mixer_controls)), SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", ALC5623_PWR_MANAG_ADD2, 9, 0), SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", ALC5623_PWR_MANAG_ADD2, 8, 0), SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0), SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", ALC5623_PWR_MANAG_ADD2, 7, 0), SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", ALC5623_PWR_MANAG_ADD2, 6, 0), SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0), SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0), SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0), SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0), SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0), SND_SOC_DAPM_OUTPUT("AUXOUTL"), SND_SOC_DAPM_OUTPUT("AUXOUTR"), SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), SND_SOC_DAPM_OUTPUT("SPKOUT"), SND_SOC_DAPM_OUTPUT("SPKOUTN"), SND_SOC_DAPM_INPUT("LINEINL"), SND_SOC_DAPM_INPUT("LINEINR"), SND_SOC_DAPM_INPUT("AUXINL"), SND_SOC_DAPM_INPUT("AUXINR"), SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_VMID("Vmid"), }; static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"}; static const struct soc_enum alc5623_amp_enum = SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names); static const struct snd_kcontrol_new alc5623_amp_mux_controls = SOC_DAPM_ENUM("Route", alc5623_amp_enum); static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = { SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0, amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0), SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0, &alc5623_amp_mux_controls), }; static const struct snd_soc_dapm_route intercon[] = { /* virtual mixer - mixes left & right channels */ {"I2S Mix", NULL, "Left DAC"}, {"I2S Mix", NULL, "Right DAC"}, {"Line Mix", NULL, "Right LineIn"}, {"Line Mix", NULL, "Left LineIn"}, {"AuxI Mix", NULL, "Left AuxI"}, {"AuxI Mix", NULL, "Right AuxI"}, {"AUXOUTL", NULL, "Left AuxOut"}, {"AUXOUTR", NULL, "Right AuxOut"}, /* HP mixer */ {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, {"HPL Mix", NULL, "HP Mix"}, {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, {"HPR Mix", NULL, "HP Mix"}, {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"}, {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"}, /* speaker mixer */ {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"}, {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"}, /* mono mixer */ {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"}, {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"}, /* Left record mixer */ {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"}, {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, /*Right record mixer */ {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"}, {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, /* headphone left mux */ {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, {"Left Headphone Mux", "Vmid", "Vmid"}, /* headphone right mux */ {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, {"Right Headphone Mux", "Vmid", "Vmid"}, /* speaker out mux */ {"SpeakerOut Mux", "Vmid", "Vmid"}, {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, /* Mono/Aux Out mux */ {"AuxOut Mux", "Vmid", "Vmid"}, {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, {"AuxOut Mux", "Mono Mix", "Mono Mix"}, /* output pga */ {"HPL", NULL, "Left Headphone"}, {"Left Headphone", NULL, "Left Headphone Mux"}, {"HPR", NULL, "Right Headphone"}, {"Right Headphone", NULL, "Right Headphone Mux"}, {"Left AuxOut", NULL, "AuxOut Mux"}, {"Right AuxOut", NULL, "AuxOut Mux"}, /* input pga */ {"Left LineIn", NULL, "LINEINL"}, {"Right LineIn", NULL, "LINEINR"}, {"Left AuxI", NULL, "AUXINL"}, {"Right AuxI", NULL, "AUXINR"}, {"MIC1 Pre Amp", NULL, "MIC1"}, {"MIC2 Pre Amp", NULL, "MIC2"}, {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, /* left ADC */ {"Left ADC", NULL, "Left Capture Mix"}, /* right ADC */ {"Right ADC", NULL, "Right Capture Mix"}, {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"}, {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"}, {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"}, {"SpeakerOut N Mux", "Vmid", "Vmid"}, {"SPKOUT", NULL, "SpeakerOut"}, {"SPKOUTN", NULL, "SpeakerOut N Mux"}, }; static const struct snd_soc_dapm_route intercon_spk[] = { {"SpeakerOut", NULL, "SpeakerOut Mux"}, }; static const struct snd_soc_dapm_route intercon_amp_spk[] = { {"AB Amp", NULL, "SpeakerOut Mux"}, {"D Amp", NULL, "SpeakerOut Mux"}, {"AB-D Amp Mux", "AB Amp", "AB Amp"}, {"AB-D Amp Mux", "D Amp", "D Amp"}, {"SpeakerOut", NULL, "AB-D Amp Mux"}, }; /* PLL divisors */ struct _pll_div { u32 pll_in; u32 pll_out; u16 regvalue; }; /* Note : pll code from original alc5623 driver. Not sure of how good it is */ /* useful only for master mode */ static const struct _pll_div codec_master_pll_div[] = { { 2048000, 8192000, 0x0ea0}, { 3686400, 8192000, 0x4e27}, { 12000000, 8192000, 0x456b}, { 13000000, 8192000, 0x495f}, { 13100000, 8192000, 0x0320}, { 2048000, 11289600, 0xf637}, { 3686400, 11289600, 0x2f22}, { 12000000, 11289600, 0x3e2f}, { 13000000, 11289600, 0x4d5b}, { 13100000, 11289600, 0x363b}, { 2048000, 16384000, 0x1ea0}, { 3686400, 16384000, 0x9e27}, { 12000000, 16384000, 0x452b}, { 13000000, 16384000, 0x542f}, { 13100000, 16384000, 0x03a0}, { 2048000, 16934400, 0xe625}, { 3686400, 16934400, 0x9126}, { 12000000, 16934400, 0x4d2c}, { 13000000, 16934400, 0x742f}, { 13100000, 16934400, 0x3c27}, { 2048000, 22579200, 0x2aa0}, { 3686400, 22579200, 0x2f20}, { 12000000, 22579200, 0x7e2f}, { 13000000, 22579200, 0x742f}, { 13100000, 22579200, 0x3c27}, { 2048000, 24576000, 0x2ea0}, { 3686400, 24576000, 0xee27}, { 12000000, 24576000, 0x2915}, { 13000000, 24576000, 0x772e}, { 13100000, 24576000, 0x0d20}, }; static const struct _pll_div codec_slave_pll_div[] = { { 1024000, 16384000, 0x3ea0}, { 1411200, 22579200, 0x3ea0}, { 1536000, 24576000, 0x3ea0}, { 2048000, 16384000, 0x1ea0}, { 2822400, 22579200, 0x1ea0}, { 3072000, 24576000, 0x1ea0}, }; static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { int i; struct snd_soc_codec *codec = codec_dai->codec; int gbl_clk = 0, pll_div = 0; u16 reg; if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK) return -ENODEV; /* Disable PLL power */ snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_PWR_ADD2_PLL, 0); /* pll is not used in slave mode */ reg = snd_soc_read(codec, ALC5623_DAI_CONTROL); if (reg & ALC5623_DAI_SDP_SLAVE_MODE) return 0; if (!freq_in || !freq_out) return 0; switch (pll_id) { case ALC5623_PLL_FR_MCLK: for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) { if (codec_master_pll_div[i].pll_in == freq_in && codec_master_pll_div[i].pll_out == freq_out) { /* PLL source from MCLK */ pll_div = codec_master_pll_div[i].regvalue; break; } } break; case ALC5623_PLL_FR_BCK: for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { if (codec_slave_pll_div[i].pll_in == freq_in && codec_slave_pll_div[i].pll_out == freq_out) { /* PLL source from Bitclk */ gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK; pll_div = codec_slave_pll_div[i].regvalue; break; } } break; default: return -EINVAL; } if (!pll_div) return -EINVAL; snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk); snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div); snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_PWR_ADD2_PLL, ALC5623_PWR_ADD2_PLL); gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL; snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk); return 0; } struct _coeff_div { u16 fs; u16 regvalue; }; /* codec hifi mclk (after PLL) clock divider coefficients */ /* values inspired from column BCLK=32Fs of Appendix A table */ static const struct _coeff_div coeff_div[] = { {256*8, 0x3a69}, {384*8, 0x3c6b}, {256*4, 0x2a69}, {384*4, 0x2c6b}, {256*2, 0x1a69}, {384*2, 0x1c6b}, {256*1, 0x0a69}, {384*1, 0x0c6b}, }; static int get_coeff(struct snd_soc_codec *codec, int rate) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); int i; for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { if (coeff_div[i].fs * rate == alc5623->sysclk) return i; } return -EINVAL; } /* * Clock after PLL and dividers */ static int alc5623_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 alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); switch (freq) { case 8192000: case 11289600: case 12288000: case 16384000: case 16934400: case 18432000: case 22579200: case 24576000: alc5623->sysclk = freq; return 0; } return -EINVAL; } static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; u16 iface = 0; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: iface = ALC5623_DAI_SDP_MASTER_MODE; break; case SND_SOC_DAIFMT_CBS_CFS: iface = ALC5623_DAI_SDP_SLAVE_MODE; break; default: return -EINVAL; } /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: iface |= ALC5623_DAI_I2S_DF_I2S; break; case SND_SOC_DAIFMT_RIGHT_J: iface |= ALC5623_DAI_I2S_DF_RIGHT; break; case SND_SOC_DAIFMT_LEFT_J: iface |= ALC5623_DAI_I2S_DF_LEFT; break; case SND_SOC_DAIFMT_DSP_A: iface |= ALC5623_DAI_I2S_DF_PCM; break; case SND_SOC_DAIFMT_DSP_B: iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE; break; default: return -EINVAL; } /* clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_IF: iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL; break; case SND_SOC_DAIFMT_IB_NF: iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL; break; case SND_SOC_DAIFMT_NB_IF: break; default: return -EINVAL; } return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface); } static int alc5623_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 alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); int coeff, rate; u16 iface; iface = snd_soc_read(codec, ALC5623_DAI_CONTROL); iface &= ~ALC5623_DAI_I2S_DL_MASK; /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: iface |= ALC5623_DAI_I2S_DL_16; break; case SNDRV_PCM_FORMAT_S20_3LE: iface |= ALC5623_DAI_I2S_DL_20; break; case SNDRV_PCM_FORMAT_S24_LE: iface |= ALC5623_DAI_I2S_DL_24; break; case SNDRV_PCM_FORMAT_S32_LE: iface |= ALC5623_DAI_I2S_DL_32; break; default: return -EINVAL; } /* set iface & srate */ snd_soc_write(codec, ALC5623_DAI_CONTROL, iface); rate = params_rate(params); coeff = get_coeff(codec, rate); if (coeff < 0) return -EINVAL; coeff = coeff_div[coeff].regvalue; dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n", __func__, alc5623->sysclk, rate, coeff); snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff); return 0; } static int alc5623_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT; u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute; if (mute) mute_reg |= hp_mute; return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg); } #define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \ | ALC5623_PWR_ADD2_DAC_REF_CIR) #define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \ | ALC5623_PWR_ADD3_MIC1_BOOST_AD) #define ALC5623_ADD1_POWER_EN \ (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \ | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \ | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP) #define ALC5623_ADD1_POWER_EN_5622 \ (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \ | ALC5623_PWR_ADD1_HP_OUT_AMP) static void enable_power_depop(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1, ALC5623_PWR_ADD1_SOFTGEN_EN, ALC5623_PWR_ADD1_SOFTGEN_EN); snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN); snd_soc_update_bits(codec, ALC5623_MISC_CTRL, ALC5623_MISC_HP_DEPOP_MODE2_EN, ALC5623_MISC_HP_DEPOP_MODE2_EN); msleep(500); snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN); /* avoid writing '1' into 5622 reserved bits */ if (alc5623->id == 0x22) snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, ALC5623_ADD1_POWER_EN_5622); else snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, ALC5623_ADD1_POWER_EN); /* disable HP Depop2 */ snd_soc_update_bits(codec, ALC5623_MISC_CTRL, ALC5623_MISC_HP_DEPOP_MODE2_EN, 0); } static int alc5623_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { switch (level) { case SND_SOC_BIAS_ON: enable_power_depop(codec); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_PWR_ADD2_VREF); snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_PWR_ADD3_MAIN_BIAS); break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0); snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0); snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0); break; } codec->dapm.bias_level = level; return 0; } #define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ | SNDRV_PCM_FMTBIT_S24_LE \ | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops alc5623_dai_ops = { .hw_params = alc5623_pcm_hw_params, .digital_mute = alc5623_mute, .set_fmt = alc5623_set_dai_fmt, .set_sysclk = alc5623_set_dai_sysclk, .set_pll = alc5623_set_dai_pll, }; static struct snd_soc_dai_driver alc5623_dai = { .name = "alc5623-hifi", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rate_min = 8000, .rate_max = 48000, .rates = SNDRV_PCM_RATE_8000_48000, .formats = ALC5623_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rate_min = 8000, .rate_max = 48000, .rates = SNDRV_PCM_RATE_8000_48000, .formats = ALC5623_FORMATS,}, .ops = &alc5623_dai_ops, }; static int alc5623_suspend(struct snd_soc_codec *codec) { alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static int alc5623_resume(struct snd_soc_codec *codec) { int i, step = codec->driver->reg_cache_step; u16 *cache = codec->reg_cache; /* Sync reg_cache with the hardware */ for (i = 2 ; i < codec->driver->reg_cache_size ; i += step) snd_soc_write(codec, i, cache[i]); alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge alc5623 caps */ if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); codec->dapm.bias_level = SND_SOC_BIAS_ON; alc5623_set_bias_level(codec, codec->dapm.bias_level); } return 0; } static int alc5623_probe(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } alc5623_reset(codec); alc5623_fill_cache(codec); /* power on device */ alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (alc5623->add_ctrl) { snd_soc_write(codec, ALC5623_ADD_CTRL_REG, alc5623->add_ctrl); } if (alc5623->jack_det_ctrl) { snd_soc_write(codec, ALC5623_JACK_DET_CTRL, alc5623->jack_det_ctrl); } switch (alc5623->id) { case 0x21: snd_soc_add_controls(codec, alc5621_vol_snd_controls, ARRAY_SIZE(alc5621_vol_snd_controls)); break; case 0x22: snd_soc_add_controls(codec, alc5622_vol_snd_controls, ARRAY_SIZE(alc5622_vol_snd_controls)); break; case 0x23: snd_soc_add_controls(codec, alc5623_vol_snd_controls, ARRAY_SIZE(alc5623_vol_snd_controls)); break; default: return -EINVAL; } snd_soc_add_controls(codec, alc5623_snd_controls, ARRAY_SIZE(alc5623_snd_controls)); snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets, ARRAY_SIZE(alc5623_dapm_widgets)); /* set up audio path interconnects */ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); switch (alc5623->id) { case 0x21: case 0x22: snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets, ARRAY_SIZE(alc5623_dapm_amp_widgets)); snd_soc_dapm_add_routes(dapm, intercon_amp_spk, ARRAY_SIZE(intercon_amp_spk)); break; case 0x23: snd_soc_dapm_add_routes(dapm, intercon_spk, ARRAY_SIZE(intercon_spk)); break; default: return -EINVAL; } return ret; } /* power down chip */ static int alc5623_remove(struct snd_soc_codec *codec) { alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static struct snd_soc_codec_driver soc_codec_device_alc5623 = { .probe = alc5623_probe, .remove = alc5623_remove, .suspend = alc5623_suspend, .resume = alc5623_resume, .set_bias_level = alc5623_set_bias_level, .reg_cache_size = ALC5623_VENDOR_ID2+2, .reg_word_size = sizeof(u16), .reg_cache_step = 2, }; /* * ALC5623 2 wire address is determined by A1 pin * state during powerup. * low = 0x1a * high = 0x1b */ static __devinit int alc5623_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct alc5623_platform_data *pdata; struct alc5623_priv *alc5623; int ret, vid1, vid2; vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1); if (vid1 < 0) { dev_err(&client->dev, "failed to read I2C\n"); return -EIO; } vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2); if (vid2 < 0) { dev_err(&client->dev, "failed to read I2C\n"); return -EIO; } if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { dev_err(&client->dev, "unknown or wrong codec\n"); dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", 0x10ec, id->driver_data, vid1, vid2); return -ENODEV; } dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), GFP_KERNEL); if (alc5623 == NULL) return -ENOMEM; pdata = client->dev.platform_data; if (pdata) { alc5623->add_ctrl = pdata->add_ctrl; alc5623->jack_det_ctrl = pdata->jack_det_ctrl; } alc5623->id = vid2; switch (alc5623->id) { case 0x21: alc5623_dai.name = "alc5621-hifi"; break; case 0x22: alc5623_dai.name = "alc5622-hifi"; break; case 0x23: alc5623_dai.name = "alc5623-hifi"; break; default: return -EINVAL; } i2c_set_clientdata(client, alc5623); alc5623->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5623, &alc5623_dai, 1); if (ret != 0) dev_err(&client->dev, "Failed to register codec: %d\n", ret); return ret; } static __devexit int alc5623_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); return 0; } static const struct i2c_device_id alc5623_i2c_table[] = { {"alc5621", 0x21}, {"alc5622", 0x22}, {"alc5623", 0x23}, {} }; MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table); /* i2c codec control layer */ static struct i2c_driver alc5623_i2c_driver = { .driver = { .name = "alc562x-codec", .owner = THIS_MODULE, }, .probe = alc5623_i2c_probe, .remove = __devexit_p(alc5623_i2c_remove), .id_table = alc5623_i2c_table, }; static int __init alc5623_modinit(void) { int ret; ret = i2c_add_driver(&alc5623_i2c_driver); if (ret != 0) { printk(KERN_ERR "%s: can't add i2c driver", __func__); return ret; } return ret; } module_init(alc5623_modinit); static void __exit alc5623_modexit(void) { i2c_del_driver(&alc5623_i2c_driver); } module_exit(alc5623_modexit); MODULE_DESCRIPTION("ASoC alc5621/2/3 driver"); MODULE_AUTHOR("Arnaud Patard "); MODULE_LICENSE("GPL");