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