/* * alc5632.c -- ALC5632 ALSA SoC Audio Codec * * Copyright (C) 2011 The AC100 Kernel Team * * Authors: Leon Romanovsky * Andrey Danin * Ilya Petrov * Marc Dietrich * * Based on alc5623.c by Arnaud Patard * * 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 "alc5632.h" /* * ALC5632 register cache */ static struct reg_default alc5632_reg_defaults[] = { { 2, 0x8080 }, /* R2 - Speaker Output Volume */ { 4, 0x8080 }, /* R4 - Headphone Output Volume */ { 6, 0x8080 }, /* R6 - AUXOUT Volume */ { 8, 0xC800 }, /* R8 - Phone Input */ { 10, 0xE808 }, /* R10 - LINE_IN Volume */ { 12, 0x1010 }, /* R12 - STEREO DAC Input Volume */ { 14, 0x0808 }, /* R14 - MIC Input Volume */ { 16, 0xEE0F }, /* R16 - Stereo DAC and MIC Routing Control */ { 18, 0xCBCB }, /* R18 - ADC Record Gain */ { 20, 0x7F7F }, /* R20 - ADC Record Mixer Control */ { 24, 0xE010 }, /* R24 - Voice DAC Volume */ { 28, 0x8008 }, /* R28 - Output Mixer Control */ { 34, 0x0000 }, /* R34 - Microphone Control */ { 36, 0x00C0 }, /* R36 - Codec Digital MIC/Digital Boost Control */ { 46, 0x0000 }, /* R46 - Stereo DAC/Voice DAC/Stereo ADC Function Select */ { 52, 0x8000 }, /* R52 - Main Serial Data Port Control (Stereo I2S) */ { 54, 0x0000 }, /* R54 - Extend Serial Data Port Control (VoDAC_I2S/PCM) */ { 58, 0x0000 }, /* R58 - Power Management Addition 1 */ { 60, 0x0000 }, /* R60 - Power Management Addition 2 */ { 62, 0x8000 }, /* R62 - Power Management Addition 3 */ { 64, 0x0C0A }, /* R64 - General Purpose Control Register 1 */ { 66, 0x0000 }, /* R66 - General Purpose Control Register 2 */ { 68, 0x0000 }, /* R68 - PLL1 Control */ { 70, 0x0000 }, /* R70 - PLL2 Control */ { 76, 0xBE3E }, /* R76 - GPIO Pin Configuration */ { 78, 0xBE3E }, /* R78 - GPIO Pin Polarity */ { 80, 0x0000 }, /* R80 - GPIO Pin Sticky */ { 82, 0x0000 }, /* R82 - GPIO Pin Wake Up */ { 86, 0x0000 }, /* R86 - Pin Sharing */ { 90, 0x0009 }, /* R90 - Soft Volume Control Setting */ { 92, 0x0000 }, /* R92 - GPIO_Output Pin Control */ { 94, 0x3000 }, /* R94 - MISC Control */ { 96, 0x3075 }, /* R96 - Stereo DAC Clock Control_1 */ { 98, 0x1010 }, /* R98 - Stereo DAC Clock Control_2 */ { 100, 0x3110 }, /* R100 - VoDAC_PCM Clock Control_1 */ { 104, 0x0553 }, /* R104 - Pseudo Stereo and Spatial Effect Block Control */ { 106, 0x0000 }, /* R106 - Private Register Address */ }; /* codec private data */ struct alc5632_priv { struct regmap *regmap; u8 id; unsigned int sysclk; }; static bool alc5632_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case ALC5632_RESET: case ALC5632_PWR_DOWN_CTRL_STATUS: case ALC5632_GPIO_PIN_STATUS: case ALC5632_OVER_CURR_STATUS: case ALC5632_HID_CTRL_DATA: case ALC5632_EQ_CTRL: case ALC5632_VENDOR_ID1: case ALC5632_VENDOR_ID2: return true; default: break; } return false; } static inline int alc5632_reset(struct regmap *map) { return regmap_write(map, ALC5632_RESET, 0x59B4); } 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, ALC5632_HID_CTRL_INDEX, 0x46); switch (event) { case SND_SOC_DAPM_PRE_PMU: snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0xFFFF); break; case SND_SOC_DAPM_POST_PMD: snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0); break; } return 0; } /* * ALC5632 Controls */ /* -34.5db min scale, 1.5db steps, no mute */ static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); /* -46.5db min scale, 1.5db steps, no mute */ static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); /* -16.5db min scale, 1.5db steps, no mute */ 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), }; /* 0db min scale, 6 db steps, no mute */ static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); /* 0db min scalem 0.75db steps, no mute */ static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0); static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = { /* left starts at bit 8, right at bit 0 */ /* 31 steps (5 bit), -46.5db scale */ SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), /* bit 15 mutes left, bit 7 right */ SOC_DOUBLE("Speaker Playback Switch", ALC5632_SPK_OUT_VOL, 15, 7, 1, 1), SOC_DOUBLE_TLV("Headphone Playback Volume", ALC5632_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Headphone Playback Switch", ALC5632_HP_OUT_VOL, 15, 7, 1, 1), }; static const struct snd_kcontrol_new alc5632_snd_controls[] = { SOC_DOUBLE_TLV("Auxout Playback Volume", ALC5632_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Auxout Playback Switch", ALC5632_AUX_OUT_VOL, 15, 7, 1, 1), SOC_SINGLE_TLV("Voice DAC Playback Volume", ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv), SOC_SINGLE_TLV("Phone Capture Volume", ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), SOC_DOUBLE_TLV("LineIn Capture Volume", ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), SOC_DOUBLE_TLV("Master Playback Volume", ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), SOC_DOUBLE("Master Playback Switch", ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1), SOC_SINGLE_TLV("Mic1 Capture Volume", ALC5632_MIC_VOL, 8, 31, 1, vol_tlv), SOC_SINGLE_TLV("Mic2 Capture Volume", ALC5632_MIC_VOL, 0, 31, 1, vol_tlv), SOC_DOUBLE_TLV("Rec Capture Volume", ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv), SOC_SINGLE_TLV("Mic 1 Boost Volume", ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv), SOC_SINGLE_TLV("Mic 2 Boost Volume", ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv), SOC_SINGLE_TLV("Digital Boost Volume", ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), }; /* * DAPM Controls */ static const struct snd_kcontrol_new alc5632_hp_mixer_controls[] = { SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5632_LINE_IN_VOL, 15, 1, 1), SOC_DAPM_SINGLE("PHONE2HP Playback Switch", ALC5632_PHONE_IN_VOL, 15, 1, 1), SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 15, 1, 1), SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 11, 1, 1), SOC_DAPM_SINGLE("VOICE2HP Playback Switch", ALC5632_VOICE_DAC_VOL, 15, 1, 1), }; static const struct snd_kcontrol_new alc5632_hpl_mixer_controls[] = { SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5632_ADC_REC_GAIN, 15, 1, 1), SOC_DAPM_SINGLE("DACL2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 3, 1, 1), }; static const struct snd_kcontrol_new alc5632_hpr_mixer_controls[] = { SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5632_ADC_REC_GAIN, 7, 1, 1), SOC_DAPM_SINGLE("DACR2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 2, 1, 1), }; static const struct snd_kcontrol_new alc5632_mono_mixer_controls[] = { SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5632_ADC_REC_GAIN, 14, 1, 1), SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5632_ADC_REC_GAIN, 6, 1, 1), SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5632_LINE_IN_VOL, 13, 1, 1), SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 13, 1, 1), SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 9, 1, 1), SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 0, 1, 1), SOC_DAPM_SINGLE("VOICE2MONO Playback Switch", ALC5632_VOICE_DAC_VOL, 13, 1, 1), }; static const struct snd_kcontrol_new alc5632_speaker_mixer_controls[] = { SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5632_LINE_IN_VOL, 14, 1, 1), SOC_DAPM_SINGLE("PHONE2SPK Playback Switch", ALC5632_PHONE_IN_VOL, 14, 1, 1), SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 14, 1, 1), SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 10, 1, 1), SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 1, 1, 1), SOC_DAPM_SINGLE("VOICE2SPK Playback Switch", ALC5632_VOICE_DAC_VOL, 14, 1, 1), }; /* Left Record Mixer */ static const struct snd_kcontrol_new alc5632_captureL_mixer_controls[] = { SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1), SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1), SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1), SOC_DAPM_SINGLE("Left Phone Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1), SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1), SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1), SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1), }; /* Right Record Mixer */ static const struct snd_kcontrol_new alc5632_captureR_mixer_controls[] = { SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1), SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1), SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1), SOC_DAPM_SINGLE("Right Phone Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1), SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1), SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), }; static const char *alc5632_spk_n_sour_sel[] = { "RN/-R", "RP/+R", "LN/-R", "Mute"}; static const char *alc5632_hpl_out_input_sel[] = { "Vmid", "HP Left Mix"}; static const char *alc5632_hpr_out_input_sel[] = { "Vmid", "HP Right Mix"}; static const char *alc5632_spkout_input_sel[] = { "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; static const char *alc5632_aux_out_input_sel[] = { "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; /* auxout output mux */ static const struct soc_enum alc5632_aux_out_input_enum = SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel); static const struct snd_kcontrol_new alc5632_auxout_mux_controls = SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum); /* speaker output mux */ static const struct soc_enum alc5632_spkout_input_enum = SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel); static const struct snd_kcontrol_new alc5632_spkout_mux_controls = SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum); /* headphone left output mux */ static const struct soc_enum alc5632_hpl_out_input_enum = SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel); static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls = SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum); /* headphone right output mux */ static const struct soc_enum alc5632_hpr_out_input_enum = SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel); static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls = SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum); /* speaker output N select */ static const struct soc_enum alc5632_spk_n_sour_enum = SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel); static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls = SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum); /* speaker amplifier */ static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"}; static const struct soc_enum alc5632_amp_enum = SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names); static const struct snd_kcontrol_new alc5632_amp_mux_controls = SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum); static const struct snd_soc_dapm_widget alc5632_dapm_widgets[] = { /* Muxes */ SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, &alc5632_auxout_mux_controls), SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, &alc5632_spkout_mux_controls), SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &alc5632_hpl_out_mux_controls), SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &alc5632_hpr_out_mux_controls), SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, &alc5632_spkoutn_mux_controls), /* output mixers */ SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, &alc5632_hp_mixer_controls[0], ARRAY_SIZE(alc5632_hp_mixer_controls)), SND_SOC_DAPM_MIXER("HPR Mix", ALC5632_PWR_MANAG_ADD2, 4, 0, &alc5632_hpr_mixer_controls[0], ARRAY_SIZE(alc5632_hpr_mixer_controls)), SND_SOC_DAPM_MIXER("HPL Mix", ALC5632_PWR_MANAG_ADD2, 5, 0, &alc5632_hpl_mixer_controls[0], ARRAY_SIZE(alc5632_hpl_mixer_controls)), SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("Mono Mix", ALC5632_PWR_MANAG_ADD2, 2, 0, &alc5632_mono_mixer_controls[0], ARRAY_SIZE(alc5632_mono_mixer_controls)), SND_SOC_DAPM_MIXER("Speaker Mix", ALC5632_PWR_MANAG_ADD2, 3, 0, &alc5632_speaker_mixer_controls[0], ARRAY_SIZE(alc5632_speaker_mixer_controls)), /* input mixers */ SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5632_PWR_MANAG_ADD2, 1, 0, &alc5632_captureL_mixer_controls[0], ARRAY_SIZE(alc5632_captureL_mixer_controls)), SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5632_PWR_MANAG_ADD2, 0, 0, &alc5632_captureR_mixer_controls[0], ARRAY_SIZE(alc5632_captureR_mixer_controls)), SND_SOC_DAPM_DAC("Left DAC", "HiFi Playback", ALC5632_PWR_MANAG_ADD2, 9, 0), SND_SOC_DAPM_DAC("Right DAC", "HiFi Playback", ALC5632_PWR_MANAG_ADD2, 8, 0), SND_SOC_DAPM_MIXER("DAC Left Channel", ALC5632_PWR_MANAG_ADD1, 15, 0, NULL, 0), SND_SOC_DAPM_MIXER("DAC Right Channel", ALC5632_PWR_MANAG_ADD1, 14, 0, NULL, 0), SND_SOC_DAPM_MIXER("I2S Mix", ALC5632_PWR_MANAG_ADD1, 11, 0, NULL, 0), SND_SOC_DAPM_MIXER("Phone 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", "HiFi Capture", ALC5632_PWR_MANAG_ADD2, 7, 0), SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", ALC5632_PWR_MANAG_ADD2, 6, 0), SND_SOC_DAPM_PGA("Left Headphone", ALC5632_PWR_MANAG_ADD3, 11, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Headphone", ALC5632_PWR_MANAG_ADD3, 10, 0, NULL, 0), SND_SOC_DAPM_PGA("Left Speaker", ALC5632_PWR_MANAG_ADD3, 13, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Speaker", ALC5632_PWR_MANAG_ADD3, 12, 0, NULL, 0), SND_SOC_DAPM_PGA("Aux Out", ALC5632_PWR_MANAG_ADD3, 14, 0, NULL, 0), SND_SOC_DAPM_PGA("Left LineIn", ALC5632_PWR_MANAG_ADD3, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("Right LineIn", ALC5632_PWR_MANAG_ADD3, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("Phone", ALC5632_PWR_MANAG_ADD3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("Phone ADMix", ALC5632_PWR_MANAG_ADD3, 4, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC1 PGA", ALC5632_PWR_MANAG_ADD3, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC2 PGA", ALC5632_PWR_MANAG_ADD3, 2, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5632_PWR_MANAG_ADD3, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5632_PWR_MANAG_ADD3, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Bias1", ALC5632_PWR_MANAG_ADD1, 3, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Bias2", ALC5632_PWR_MANAG_ADD1, 2, 0, NULL, 0), SND_SOC_DAPM_PGA_E("D Amp", ALC5632_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", ALC5632_PWR_MANAG_ADD2, 15, 0, NULL, 0), SND_SOC_DAPM_MUX("AB-D Amp Mux", ALC5632_PWR_MANAG_ADD1, 10, 0, &alc5632_amp_mux_controls), SND_SOC_DAPM_OUTPUT("AUXOUT"), 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("PHONEP"), SND_SOC_DAPM_INPUT("PHONEN"), SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_VMID("Vmid"), }; static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { /* 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"}, {"Phone Mix", NULL, "Phone"}, {"Phone Mix", NULL, "Phone ADMix"}, {"AUXOUT", NULL, "Aux Out"}, /* DAC */ {"DAC Right Channel", NULL, "I2S Mix"}, {"DAC Left Channel", NULL, "I2S Mix"}, /* 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", "PHONE2HP Playback Switch", "Phone Mix"}, {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, {"HPR Mix", "DACR2HP Playback Switch", "DAC Right Channel"}, {"HPL Mix", "DACL2HP Playback Switch", "DAC Left Channel"}, /* speaker mixer */ {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, {"Speaker Mix", "PHONE2SPK Playback Switch", "Phone Mix"}, {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, {"Speaker Mix", "DAC2SPK Playback Switch", "DAC Left Channel"}, /* 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", "VOICE2MONO Playback Switch", "Phone Mix"}, {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"}, /* Left record mixer */ {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, {"Left Capture Mix", "Left Phone Capture Switch", "PHONEN"}, {"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 Phone Capture Switch", "PHONEP"}, {"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"}, {"Aux Out", NULL, "AuxOut Mux"}, /* input pga */ {"Left LineIn", NULL, "LINEINL"}, {"Right LineIn", NULL, "LINEINR"}, {"Phone", NULL, "PHONEP"}, {"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", "Left Speaker"}, {"SpeakerOut N Mux", "RP/+R", "Left Speaker"}, {"SpeakerOut N Mux", "LN/-R", "Left Speaker"}, {"SpeakerOut N Mux", "Mute", "Vmid"}, {"SpeakerOut N Mux", "RN/-R", "Right Speaker"}, {"SpeakerOut N Mux", "RP/+R", "Right Speaker"}, {"SpeakerOut N Mux", "LN/-R", "Right Speaker"}, {"SpeakerOut N Mux", "Mute", "Vmid"}, {"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"}, {"Left Speaker", NULL, "AB-D Amp Mux"}, {"Right Speaker", NULL, "AB-D Amp Mux"}, {"SPKOUT", NULL, "Left Speaker"}, {"SPKOUT", NULL, "Right Speaker"}, {"SPKOUTN", NULL, "SpeakerOut N Mux"}, }; /* PLL divisors */ struct _pll_div { u32 pll_in; u32 pll_out; u16 regvalue; }; /* Note : pll code from original alc5632 driver. Not sure of how good it is */ /* usefull 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}, }; /* FOUT = MCLK*(N+2)/((M+2)*(K+2)) N: bit 15:8 (div 2 .. div 257) K: bit 6:4 typical 2 M: bit 3:0 (div 2 .. div 17) same as for 5623 - thanks! */ 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 alc5632_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 < ALC5632_PLL_FR_MCLK || pll_id > ALC5632_PLL_FR_VBCLK) return -EINVAL; /* Disable PLL power */ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, ALC5632_PWR_ADD2_PLL1, 0); snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, ALC5632_PWR_ADD2_PLL2, 0); /* pll is not used in slave mode */ reg = snd_soc_read(codec, ALC5632_DAI_CONTROL); if (reg & ALC5632_DAI_SDP_SLAVE_MODE) return 0; if (!freq_in || !freq_out) return 0; switch (pll_id) { case ALC5632_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 ALC5632_PLL_FR_BCLK: 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 = ALC5632_PLL_FR_BCLK; pll_div = codec_slave_pll_div[i].regvalue; break; } } break; case ALC5632_PLL_FR_VBCLK: 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 voice clock */ gbl_clk = ALC5632_PLL_FR_VBCLK; pll_div = codec_slave_pll_div[i].regvalue; break; } } break; default: return -EINVAL; } if (!pll_div) return -EINVAL; /* choose MCLK/BCLK/VBCLK */ snd_soc_write(codec, ALC5632_GPCR2, gbl_clk); /* choose PLL1 clock rate */ snd_soc_write(codec, ALC5632_PLL1_CTRL, pll_div); /* enable PLL1 */ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, ALC5632_PWR_ADD2_PLL1, ALC5632_PWR_ADD2_PLL1); /* enable PLL2 */ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, ALC5632_PWR_ADD2_PLL2, ALC5632_PWR_ADD2_PLL2); /* use PLL1 as main SYSCLK */ snd_soc_update_bits(codec, ALC5632_GPCR1, ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1, ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1); 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[] = { {512*1, 0x3075}, }; static int get_coeff(struct snd_soc_codec *codec, int rate) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); int i; for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { if (coeff_div[i].fs * rate == alc5632->sysclk) return i; } return -EINVAL; } /* * Clock after PLL and dividers */ static int alc5632_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 alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); switch (freq) { case 8192000: case 11289600: case 12288000: case 16384000: case 16934400: case 18432000: case 22579200: case 24576000: alc5632->sysclk = freq; return 0; } return -EINVAL; } static int alc5632_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 = ALC5632_DAI_SDP_MASTER_MODE; break; case SND_SOC_DAIFMT_CBS_CFS: iface = ALC5632_DAI_SDP_SLAVE_MODE; break; default: return -EINVAL; } /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: iface |= ALC5632_DAI_I2S_DF_I2S; break; case SND_SOC_DAIFMT_LEFT_J: iface |= ALC5632_DAI_I2S_DF_LEFT; break; case SND_SOC_DAIFMT_DSP_A: iface |= ALC5632_DAI_I2S_DF_PCM_A; break; case SND_SOC_DAIFMT_DSP_B: iface |= ALC5632_DAI_I2S_DF_PCM_B; 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 |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; break; case SND_SOC_DAIFMT_IB_NF: iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; break; case SND_SOC_DAIFMT_NB_IF: break; default: return -EINVAL; } return snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); } static int alc5632_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; int coeff, rate; u16 iface; iface = snd_soc_read(codec, ALC5632_DAI_CONTROL); iface &= ~ALC5632_DAI_I2S_DL_MASK; /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: iface |= ALC5632_DAI_I2S_DL_16; break; case SNDRV_PCM_FORMAT_S20_3LE: iface |= ALC5632_DAI_I2S_DL_20; break; case SNDRV_PCM_FORMAT_S24_LE: iface |= ALC5632_DAI_I2S_DL_24; break; default: return -EINVAL; } /* set iface & srate */ snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); rate = params_rate(params); coeff = get_coeff(codec, rate); if (coeff < 0) return -EINVAL; coeff = coeff_div[coeff].regvalue; snd_soc_write(codec, ALC5632_DAC_CLK_CTRL1, coeff); return 0; } static int alc5632_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 hp_mute = ALC5632_MISC_HP_DEPOP_MUTE_L |ALC5632_MISC_HP_DEPOP_MUTE_R; u16 mute_reg = snd_soc_read(codec, ALC5632_MISC_CTRL) & ~hp_mute; if (mute) mute_reg |= hp_mute; return snd_soc_write(codec, ALC5632_MISC_CTRL, mute_reg); } #define ALC5632_ADD2_POWER_EN (ALC5632_PWR_ADD2_VREF) #define ALC5632_ADD3_POWER_EN (ALC5632_PWR_ADD3_MIC1_BOOST_AD) #define ALC5632_ADD1_POWER_EN \ (ALC5632_PWR_ADD1_DAC_REF \ | ALC5632_PWR_ADD1_SOFTGEN_EN \ | ALC5632_PWR_ADD1_HP_OUT_AMP \ | ALC5632_PWR_ADD1_HP_OUT_ENH_AMP \ | ALC5632_PWR_ADD1_MAIN_BIAS) static void enable_power_depop(struct snd_soc_codec *codec) { snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, ALC5632_PWR_ADD1_SOFTGEN_EN, ALC5632_PWR_ADD1_SOFTGEN_EN); snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3, ALC5632_ADD3_POWER_EN, ALC5632_ADD3_POWER_EN); snd_soc_update_bits(codec, ALC5632_MISC_CTRL, ALC5632_MISC_HP_DEPOP_MODE2_EN, ALC5632_MISC_HP_DEPOP_MODE2_EN); /* "normal" mode: 0 @ 26 */ /* set all PR0-7 mixers to 0 */ snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS, ALC5632_PWR_DOWN_CTRL_STATUS_MASK, 0); msleep(500); snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, ALC5632_ADD2_POWER_EN, ALC5632_ADD2_POWER_EN); snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, ALC5632_ADD1_POWER_EN, ALC5632_ADD1_POWER_EN); /* disable HP Depop2 */ snd_soc_update_bits(codec, ALC5632_MISC_CTRL, ALC5632_MISC_HP_DEPOP_MODE2_EN, 0); } static int alc5632_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_update_bits(codec, ALC5632_PWR_MANAG_ADD1, ALC5632_PWR_MANAG_ADD1_MASK, ALC5632_PWR_ADD1_MAIN_BIAS); snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, ALC5632_PWR_MANAG_ADD2_MASK, ALC5632_PWR_ADD2_VREF); /* "normal" mode: 0 @ 26 */ snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS, ALC5632_PWR_DOWN_CTRL_STATUS_MASK, 0xffff ^ (ALC5632_PWR_VREF_PR3 | ALC5632_PWR_VREF_PR2)); break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, ALC5632_PWR_MANAG_ADD2_MASK, 0); snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3, ALC5632_PWR_MANAG_ADD3_MASK, 0); snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, ALC5632_PWR_MANAG_ADD1_MASK, 0); break; } codec->dapm.bias_level = level; return 0; } #define ALC5632_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ | SNDRV_PCM_FMTBIT_S24_LE \ | SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops alc5632_dai_ops = { .hw_params = alc5632_pcm_hw_params, .digital_mute = alc5632_mute, .set_fmt = alc5632_set_dai_fmt, .set_sysclk = alc5632_set_dai_sysclk, .set_pll = alc5632_set_dai_pll, }; static struct snd_soc_dai_driver alc5632_dai = { .name = "alc5632-hifi", .playback = { .stream_name = "HiFi Playback", .channels_min = 1, .channels_max = 2, .rate_min = 8000, .rate_max = 48000, .rates = SNDRV_PCM_RATE_8000_48000, .formats = ALC5632_FORMATS,}, .capture = { .stream_name = "HiFi Capture", .channels_min = 1, .channels_max = 2, .rate_min = 8000, .rate_max = 48000, .rates = SNDRV_PCM_RATE_8000_48000, .formats = ALC5632_FORMATS,}, .ops = &alc5632_dai_ops, .symmetric_rates = 1, }; #ifdef CONFIG_PM static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg) { alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static int alc5632_resume(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); regcache_sync(alc5632->regmap); alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } #else #define alc5632_suspend NULL #define alc5632_resume NULL #endif static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); int ret; codec->control_data = alc5632->regmap; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } /* power on device */ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); switch (alc5632->id) { case 0x5c: snd_soc_add_controls(codec, alc5632_vol_snd_controls, ARRAY_SIZE(alc5632_vol_snd_controls)); break; default: return -EINVAL; } return ret; } /* power down chip */ static int alc5632_remove(struct snd_soc_codec *codec) { alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static struct snd_soc_codec_driver soc_codec_device_alc5632 = { .probe = alc5632_probe, .remove = alc5632_remove, .suspend = alc5632_suspend, .resume = alc5632_resume, .set_bias_level = alc5632_set_bias_level, .controls = alc5632_snd_controls, .num_controls = ARRAY_SIZE(alc5632_snd_controls), .dapm_widgets = alc5632_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(alc5632_dapm_widgets), .dapm_routes = alc5632_dapm_routes, .num_dapm_routes = ARRAY_SIZE(alc5632_dapm_routes), }; static struct regmap_config alc5632_regmap = { .reg_bits = 8, .val_bits = 16, .max_register = ALC5632_MAX_REGISTER, .reg_defaults = alc5632_reg_defaults, .num_reg_defaults = ARRAY_SIZE(alc5632_reg_defaults), .volatile_reg = alc5632_volatile_register, .cache_type = REGCACHE_RBTREE, }; /* * alc5632 2 wire address is determined by A1 pin * state during powerup. * low = 0x1a * high = 0x1b */ static __devinit int alc5632_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct alc5632_priv *alc5632; int ret, ret1, ret2; unsigned int vid1, vid2; alc5632 = devm_kzalloc(&client->dev, sizeof(struct alc5632_priv), GFP_KERNEL); if (alc5632 == NULL) return -ENOMEM; i2c_set_clientdata(client, alc5632); alc5632->regmap = regmap_init_i2c(client, &alc5632_regmap); if (IS_ERR(alc5632->regmap)) { ret = PTR_ERR(alc5632->regmap); dev_err(&client->dev, "regmap_init() failed: %d\n", ret); return ret; } ret1 = regmap_read(alc5632->regmap, ALC5632_VENDOR_ID1, &vid1); ret2 = regmap_read(alc5632->regmap, ALC5632_VENDOR_ID2, &vid2); if (ret1 != 0 || ret2 != 0) { dev_err(&client->dev, "Failed to read chip ID: ret1=%d, ret2=%d\n", ret1, ret2); regmap_exit(alc5632->regmap); return -EIO; } vid2 >>= 8; if ((vid1 != 0x10EC) || (vid2 != id->driver_data)) { dev_err(&client->dev, "Device is not a ALC5632: VID1=0x%x, VID2=0x%x\n", vid1, vid2); regmap_exit(alc5632->regmap); return -EINVAL; } ret = alc5632_reset(alc5632->regmap); if (ret < 0) { dev_err(&client->dev, "Failed to issue reset\n"); regmap_exit(alc5632->regmap); return ret; } alc5632->id = vid2; switch (alc5632->id) { case 0x5c: alc5632_dai.name = "alc5632-hifi"; break; default: return -EINVAL; } ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5632, &alc5632_dai, 1); if (ret < 0) { dev_err(&client->dev, "Failed to register codec: %d\n", ret); regmap_exit(alc5632->regmap); return ret; } return ret; } static int alc5632_i2c_remove(struct i2c_client *client) { struct alc5632_priv *alc5632 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); regmap_exit(alc5632->regmap); return 0; } static const struct i2c_device_id alc5632_i2c_table[] = { {"alc5632", 0x5c}, {} }; MODULE_DEVICE_TABLE(i2c, alc5632_i2c_table); /* i2c codec control layer */ static struct i2c_driver alc5632_i2c_driver = { .driver = { .name = "alc5632", .owner = THIS_MODULE, }, .probe = alc5632_i2c_probe, .remove = __devexit_p(alc5632_i2c_remove), .id_table = alc5632_i2c_table, }; static int __init alc5632_modinit(void) { int ret; ret = i2c_add_driver(&alc5632_i2c_driver); if (ret != 0) { printk(KERN_ERR "%s: can't add i2c driver", __func__); return ret; } return ret; } module_init(alc5632_modinit); static void __exit alc5632_modexit(void) { i2c_del_driver(&alc5632_i2c_driver); } module_exit(alc5632_modexit); MODULE_DESCRIPTION("ASoC ALC5632 driver"); MODULE_AUTHOR("Leon Romanovsky "); MODULE_LICENSE("GPL");