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linux-stable/sound/soc/codecs/alc5632.c
Leon Romanovsky 88c494b99a ASoC: Remove unnecessary backslash from alc5632 codec 
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-11 08:47:00 +00:00

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/*
* 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;
}
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");
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