diff options
author | Leon Romanovsky <leon@leon.nu> | 2011-11-05 12:38:02 +0200 |
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committer | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2011-11-10 12:00:35 +0000 |
commit | 94d5f7c0255bd712d68732a0180558d45fe6eac5 (patch) | |
tree | 61663a6dd14f9d9c83e2be395abd5a7a0b0a9533 /sound/soc/codecs/alc5632.c | |
parent | d66a327ddad647fd1678fd24d9070846737c6834 (diff) | |
download | linux-94d5f7c0255bd712d68732a0180558d45fe6eac5.tar.bz2 |
ASoC: Add new Realtek ALC5632 CODEC driver
This driver implements basic functionality, using I²C for the control
channel.
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Diffstat (limited to 'sound/soc/codecs/alc5632.c')
-rw-r--r-- | sound/soc/codecs/alc5632.c | 1153 |
1 files changed, 1153 insertions, 0 deletions
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c new file mode 100644 index 000000000000..ee6a497b5e71 --- /dev/null +++ b/sound/soc/codecs/alc5632.c @@ -0,0 +1,1153 @@ +/* +* alc5632.c -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon@leon.nu> +* Andrey Danin <danindrey@mail.ru> +* Ilya Petrov <ilya.muromec@gmail.com> +* Marc Dietrich <marvin24@gmx.de> +* +* 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 <linux/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/initval.h> + +#include "alc5632.h" + +/* + * ALC5632 register cache + */ +static const u16 alc5632_reg_defaults[] = { + 0x59B4, 0x0000, 0x8080, 0x0000, /* 0 */ + 0x8080, 0x0000, 0x8080, 0x0000, /* 4 */ + 0xC800, 0x0000, 0xE808, 0x0000, /* 8 */ + 0x1010, 0x0000, 0x0808, 0x0000, /* 12 */ + 0xEE0F, 0x0000, 0xCBCB, 0x0000, /* 16 */ + 0x7F7F, 0x0000, 0x0000, 0x0000, /* 20 */ + 0xE010, 0x0000, 0x0000, 0x0000, /* 24 */ + 0x8008, 0x0000, 0x0000, 0x0000, /* 28 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 32 */ + 0x00C0, 0x0000, 0xEF00, 0x0000, /* 36 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 40 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 44 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 48 */ + 0x8000, 0x0000, 0x0000, 0x0000, /* 52 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x0000, 0x0000, 0x8000, 0x0000, /* 60 */ + 0x0C0A, 0x0000, 0x0000, 0x0000, /* 64 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 68 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 72 */ + 0xBE3E, 0x0000, 0xBE3E, 0x0000, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 80 */ + 0x803A, 0x0000, 0x0000, 0x0000, /* 84 */ + 0x0000, 0x0000, 0x0009, 0x0000, /* 88 */ + 0x0000, 0x0000, 0x3000, 0x0000, /* 92 */ + 0x3075, 0x0000, 0x1010, 0x0000, /* 96 */ + 0x3110, 0x0000, 0x0000, 0x0000, /* 100 */ + 0x0553, 0x0000, 0x0000, 0x0000, /* 104 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 108 */ +}; + +/* codec private data */ +struct alc5632_priv { + enum snd_soc_control_type control_type; + void *control_data; + struct mutex mutex; + u8 id; + unsigned int sysclk; +}; + +static int alc5632_volatile_register(struct snd_soc_codec *codec, + 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: + return 1; + + default: + break; + } + + return 0; +} + +static inline int alc5632_reset(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, ALC5632_RESET, 0); + return snd_soc_read(codec, ALC5632_RESET); +} + +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("Line Playback Volume", + ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + /* bit 15 mutes left, bit 7 right */ + SOC_DOUBLE("Line 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("Stereo DAC Playback Volume", + ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), + SOC_DOUBLE("Stereo DAC 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, +}; + +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) +{ + int ret; + + /* mark cache as needed to sync */ + codec->cache_sync = 1; + + ret = snd_soc_cache_sync(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +#define ALC5632_REC_UNMUTE (ALC5632_ADC_REC_MIC2 \ + | ALC5632_ADC_REC_LINE_IN | ALC5632_ADC_REC_AUX \ + | ALC5632_ADC_REC_HP | ALC5632_ADC_REC_SPK \ + | ALC5632_ADC_REC_MONOMIX) + +#define ALC5632_MIC_ROUTE (ALC5632_MIC_ROUTE_HP \ + | ALC5632_MIC_ROUTE_SPK \ + | ALC5632_MIC_ROUTE_MONOMIX) + +#define ALC5632_PWR_DEFAULT (ALC5632_PWR_ADC_STATUS \ + | ALC5632_PWR_DAC_STATUS \ + | ALC5632_PWR_AMIX_STATUS \ + | ALC5632_PWR_VREF_STATUS) + +#define ALC5632_ADC_REC_GAIN_COMP(x) (int)((x - ALC5632_ADC_REC_GAIN_BASE) \ + / ALC5632_ADC_REC_GAIN_STEP) + +#define ALC5632_MIC_BOOST_COMP(x) (int)(x / ALC5632_MIC_BOOST_STEP) + +#define ALC5632_SPK_OUT_VOL_COMP(x) (int)(x / ALC5632_SPK_OUT_VOL_STEP) + +static int alc5632_probe(struct snd_soc_codec *codec) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5632->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + alc5632_reset(codec); + + /* 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, + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = alc5632_reg_defaults, + .reg_cache_size = ARRAY_SIZE(alc5632_reg_defaults), + .volatile_register = alc5632_volatile_register, + .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), +}; + +/* + * alc5632 2 wire address is determined by A1 pin + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int alc5632_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct alc5632_priv *alc5632; + int ret, vid1, vid2; + + vid1 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID1); + if (vid1 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_info(&client->dev, "got vid1: %x\n", vid1); + } + vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); + + vid2 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID2); + if (vid2 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_info(&client->dev, "got vid2: %x\n", vid2); + } + vid2 = (vid2 & 0xff); + + 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; + } + + alc5632 = devm_kzalloc(&client->dev, + sizeof(struct alc5632_priv), GFP_KERNEL); + if (alc5632 == NULL) + return -ENOMEM; + + alc5632->id = vid2; + switch (alc5632->id) { + case 0x5c: + alc5632_dai.name = "alc5632-hifi"; + break; + default: + return -EINVAL; + } + + i2c_set_clientdata(client, alc5632); + alc5632->control_data = client; + alc5632->control_type = SND_SOC_I2C; + mutex_init(&alc5632->mutex); + + 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); + + return ret; +} + +static int alc5632_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + 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 <leon@leon.nu>"); +MODULE_LICENSE("GPL"); |