sound/soc/codecs/88pm860x-codec.c

/*
 * 88pm860x-codec.c -- 88PM860x ALSA SoC Audio Driver
 *
 * Copyright 2010 Marvell International Ltd.
 * Author: Haojian Zhuang <haojian.zhuang@marvell.com>
 *
 * 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/kernel.h>
#include <linux/module.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/mfd/88pm860x.h>
#include <linux/slab.h>
#include <linux/delay.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#include <sound/initval.h>
#include <sound/jack.h>
#include <trace/events/asoc.h>

#include "88pm860x-codec.h"

#define MAX_NAME_LEN		20
#define REG_CACHE_SIZE		0x40
#define REG_CACHE_BASE		0xb0

/* Status Register 1 (0x01) */
#define REG_STATUS_1		0x01
#define MIC_STATUS		(1 << 7)
#define HOOK_STATUS		(1 << 6)
#define HEADSET_STATUS		(1 << 5)

/* Mic Detection Register (0x37) */
#define REG_MIC_DET		0x37
#define CONTINUOUS_POLLING	(3 << 1)
#define EN_MIC_DET		(1 << 0)
#define MICDET_MASK		0x07

/* Headset Detection Register (0x38) */
#define REG_HS_DET		0x38
#define EN_HS_DET		(1 << 0)

/* Misc2 Register (0x42) */
#define REG_MISC2		0x42
#define AUDIO_PLL		(1 << 5)
#define AUDIO_SECTION_RESET	(1 << 4)
#define AUDIO_SECTION_ON	(1 << 3)

/* PCM Interface Register 2 (0xb1) */
#define PCM_INF2_BCLK		(1 << 6)	/* Bit clock polarity */
#define PCM_INF2_FS		(1 << 5)	/* Frame Sync polarity */
#define PCM_INF2_MASTER		(1 << 4)	/* Master / Slave */
#define PCM_INF2_18WL		(1 << 3)	/* 18 / 16 bits */
#define PCM_GENERAL_I2S		0
#define PCM_EXACT_I2S		1
#define PCM_LEFT_I2S		2
#define PCM_RIGHT_I2S		3
#define PCM_SHORT_FS		4
#define PCM_LONG_FS		5
#define PCM_MODE_MASK		7

/* I2S Interface Register 4 (0xbe) */
#define I2S_EQU_BYP		(1 << 6)

/* DAC Offset Register (0xcb) */
#define DAC_MUTE		(1 << 7)
#define MUTE_LEFT		(1 << 6)
#define MUTE_RIGHT		(1 << 2)

/* ADC Analog Register 1 (0xd0) */
#define REG_ADC_ANA_1		0xd0
#define MIC1BIAS_MASK		0x60

/* Earpiece/Speaker Control Register 2 (0xda) */
#define REG_EAR2		0xda
#define RSYNC_CHANGE		(1 << 2)

/* Audio Supplies Register 2 (0xdc) */
#define REG_SUPPLIES2		0xdc
#define LDO15_READY		(1 << 4)
#define LDO15_EN		(1 << 3)
#define CPUMP_READY		(1 << 2)
#define CPUMP_EN		(1 << 1)
#define AUDIO_EN		(1 << 0)
#define SUPPLY_MASK		(LDO15_EN | CPUMP_EN | AUDIO_EN)

/* Audio Enable Register 1 (0xdd) */
#define ADC_MOD_RIGHT		(1 << 1)
#define ADC_MOD_LEFT		(1 << 0)

/* Audio Enable Register 2 (0xde) */
#define ADC_LEFT		(1 << 5)
#define ADC_RIGHT		(1 << 4)

/* DAC Enable Register 2 (0xe1) */
#define DAC_LEFT		(1 << 5)
#define DAC_RIGHT		(1 << 4)
#define MODULATOR		(1 << 3)

/* Shorts Register (0xeb) */
#define REG_SHORTS		0xeb
#define CLR_SHORT_LO2		(1 << 7)
#define SHORT_LO2		(1 << 6)
#define CLR_SHORT_LO1		(1 << 5)
#define SHORT_LO1		(1 << 4)
#define CLR_SHORT_HS2		(1 << 3)
#define SHORT_HS2		(1 << 2)
#define CLR_SHORT_HS1		(1 << 1)
#define SHORT_HS1		(1 << 0)

/*
 * This widget should be just after DAC & PGA in DAPM power-on sequence and
 * before DAC & PGA in DAPM power-off sequence.
 */
#define PM860X_DAPM_OUTPUT(wname, wevent)	\
{	.id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \
	.shift = 0, .invert = 0, .kcontrol_news = NULL, \
	.num_kcontrols = 0, .event = wevent, \
	.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD, }

struct pm860x_det {
	struct snd_soc_jack	*hp_jack;
	struct snd_soc_jack	*mic_jack;
	int			hp_det;
	int			mic_det;
	int			hook_det;
	int			hs_shrt;
	int			lo_shrt;
};

struct pm860x_priv {
	unsigned int		sysclk;
	unsigned int		pcmclk;
	unsigned int		dir;
	unsigned int		filter;
	struct snd_soc_codec	*codec;
	struct i2c_client	*i2c;
	struct pm860x_chip	*chip;
	struct pm860x_det	det;

	int			irq[4];
	unsigned char		name[4][MAX_NAME_LEN];
};

/* -9450dB to 0dB in 150dB steps ( mute instead of -9450dB) */
static const DECLARE_TLV_DB_SCALE(dpga_tlv, -9450, 150, 1);

/* -9dB to 0db in 3dB steps */
static const DECLARE_TLV_DB_SCALE(adc_tlv, -900, 300, 0);

/* {-23, -17, -13.5, -11, -9, -6, -3, 0}dB */
static const unsigned int mic_tlv[] = {
	TLV_DB_RANGE_HEAD(5),
	0, 0, TLV_DB_SCALE_ITEM(-2300, 0, 0),
	1, 1, TLV_DB_SCALE_ITEM(-1700, 0, 0),
	2, 2, TLV_DB_SCALE_ITEM(-1350, 0, 0),
	3, 3, TLV_DB_SCALE_ITEM(-1100, 0, 0),
	4, 7, TLV_DB_SCALE_ITEM(-900, 300, 0),
};

/* {0, 0, 0, -6, 0, 6, 12, 18}dB */
static const unsigned int aux_tlv[] = {
	TLV_DB_RANGE_HEAD(2),
	0, 2, TLV_DB_SCALE_ITEM(0, 0, 0),
	3, 7, TLV_DB_SCALE_ITEM(-600, 600, 0),
};

/* {-16, -13, -10, -7, -5.2, -3,3, -2.2, 0}dB, mute instead of -16dB */
static const unsigned int out_tlv[] = {
	TLV_DB_RANGE_HEAD(4),
	0, 3, TLV_DB_SCALE_ITEM(-1600, 300, 1),
	4, 4, TLV_DB_SCALE_ITEM(-520, 0, 0),
	5, 5, TLV_DB_SCALE_ITEM(-330, 0, 0),
	6, 7, TLV_DB_SCALE_ITEM(-220, 220, 0),
};

static const unsigned int st_tlv[] = {
	TLV_DB_RANGE_HEAD(8),
	0, 1, TLV_DB_SCALE_ITEM(-12041, 602, 0),
	2, 3, TLV_DB_SCALE_ITEM(-11087, 250, 0),
	4, 5, TLV_DB_SCALE_ITEM(-10643, 158, 0),
	6, 7, TLV_DB_SCALE_ITEM(-10351, 116, 0),
	8, 9, TLV_DB_SCALE_ITEM(-10133, 92, 0),
	10, 13, TLV_DB_SCALE_ITEM(-9958, 70, 0),
	14, 17, TLV_DB_SCALE_ITEM(-9689, 53, 0),
	18, 271, TLV_DB_SCALE_ITEM(-9484, 37, 0),
};

/* Sidetone Gain = M * 2^(-5-N) */
struct st_gain {
	unsigned int	db;
	unsigned int	m;
	unsigned int	n;
};

static struct st_gain st_table[] = {
	{-12041,  1, 15}, {-11439,  1, 14}, {-11087,  3, 15}, {-10837,  1, 13},
	{-10643,  5, 15}, {-10485,  3, 14}, {-10351,  7, 15}, {-10235,  1, 12},
	{-10133,  9, 15}, {-10041,  5, 14}, { -9958, 11, 15}, { -9883,  3, 13},
	{ -9813, 13, 15}, { -9749,  7, 14}, { -9689, 15, 15}, { -9633,  1, 11},
	{ -9580, 17, 15}, { -9531,  9, 14}, { -9484, 19, 15}, { -9439,  5, 13},
	{ -9397, 21, 15}, { -9356, 11, 14}, { -9318, 23, 15}, { -9281,  3, 12},
	{ -9245, 25, 15}, { -9211, 13, 14}, { -9178, 27, 15}, { -9147,  7, 13},
	{ -9116, 29, 15}, { -9087, 15, 14}, { -9058, 31, 15}, { -9031,  1, 10},
	{ -8978, 17, 14}, { -8929,  9, 13}, { -8882, 19, 14}, { -8837,  5, 12},
	{ -8795, 21, 14}, { -8754, 11, 13}, { -8716, 23, 14}, { -8679,  3, 11},
	{ -8643, 25, 14}, { -8609, 13, 13}, { -8576, 27, 14}, { -8545,  7, 12},
	{ -8514, 29, 14}, { -8485, 15, 13}, { -8456, 31, 14}, { -8429,  1,  9},
	{ -8376, 17, 13}, { -8327,  9, 12}, { -8280, 19, 13}, { -8235,  5, 11},
	{ -8193, 21, 13}, { -8152, 11, 12}, { -8114, 23, 13}, { -8077,  3, 10},
	{ -8041, 25, 13}, { -8007, 13, 12}, { -7974, 27, 13}, { -7943,  7, 11},
	{ -7912, 29, 13}, { -7883, 15, 12}, { -7854, 31, 13}, { -7827,  1,  8},
	{ -7774, 17, 12}, { -7724,  9, 11}, { -7678, 19, 12}, { -7633,  5, 10},
	{ -7591, 21, 12}, { -7550, 11, 11}, { -7512, 23, 12}, { -7475,  3,  9},
	{ -7439, 25, 12}, { -7405, 13, 11}, { -7372, 27, 12}, { -7341,  7, 10},
	{ -7310, 29, 12}, { -7281, 15, 11}, { -7252, 31, 12}, { -7225,  1,  7},
	{ -7172, 17, 11}, { -7122,  9, 10}, { -7075, 19, 11}, { -7031,  5,  9},
	{ -6989, 21, 11}, { -6948, 11, 10}, { -6910, 23, 11}, { -6873,  3,  8},
	{ -6837, 25, 11}, { -6803, 13, 10}, { -6770, 27, 11}, { -6739,  7,  9},
	{ -6708, 29, 11}, { -6679, 15, 10}, { -6650, 31, 11}, { -6623,  1,  6},
	{ -6570, 17, 10}, { -6520,  9,  9}, { -6473, 19, 10}, { -6429,  5,  8},
	{ -6386, 21, 10}, { -6346, 11,  9}, { -6307, 23, 10}, { -6270,  3,  7},
	{ -6235, 25, 10}, { -6201, 13,  9}, { -6168, 27, 10}, { -6137,  7,  8},
	{ -6106, 29, 10}, { -6077, 15,  9}, { -6048, 31, 10}, { -6021,  1,  5},
	{ -5968, 17,  9}, { -5918,  9,  8}, { -5871, 19,  9}, { -5827,  5,  7},
	{ -5784, 21,  9}, { -5744, 11,  8}, { -5705, 23,  9}, { -5668,  3,  6},
	{ -5633, 25,  9}, { -5599, 13,  8}, { -5566, 27,  9}, { -5535,  7,  7},
	{ -5504, 29,  9}, { -5475, 15,  8}, { -5446, 31,  9}, { -5419,  1,  4},
	{ -5366, 17,  8}, { -5316,  9,  7}, { -5269, 19,  8}, { -5225,  5,  6},
	{ -5182, 21,  8}, { -5142, 11,  7}, { -5103, 23,  8}, { -5066,  3,  5},
	{ -5031, 25,  8}, { -4997, 13,  7}, { -4964, 27,  8}, { -4932,  7,  6},
	{ -4902, 29,  8}, { -4873, 15,  7}, { -4844, 31,  8}, { -4816,  1,  3},
	{ -4764, 17,  7}, { -4714,  9,  6}, { -4667, 19,  7}, { -4623,  5,  5},
	{ -4580, 21,  7}, { -4540, 11,  6}, { -4501, 23,  7}, { -4464,  3,  4},
	{ -4429, 25,  7}, { -4395, 13,  6}, { -4362, 27,  7}, { -4330,  7,  5},
	{ -4300, 29,  7}, { -4270, 15,  6}, { -4242, 31,  7}, { -4214,  1,  2},
	{ -4162, 17,  6}, { -4112,  9,  5}, { -4065, 19,  6}, { -4021,  5,  4},
	{ -3978, 21,  6}, { -3938, 11,  5}, { -3899, 23,  6}, { -3862,  3,  3},
	{ -3827, 25,  6}, { -3793, 13,  5}, { -3760, 27,  6}, { -3728,  7,  4},
	{ -3698, 29,  6}, { -3668, 15,  5}, { -3640, 31,  6}, { -3612,  1,  1},
	{ -3560, 17,  5}, { -3510,  9,  4}, { -3463, 19,  5}, { -3419,  5,  3},
	{ -3376, 21,  5}, { -3336, 11,  4}, { -3297, 23,  5}, { -3260,  3,  2},
	{ -3225, 25,  5}, { -3191, 13,  4}, { -3158, 27,  5}, { -3126,  7,  3},
	{ -3096, 29,  5}, { -3066, 15,  4}, { -3038, 31,  5}, { -3010,  1,  0},
	{ -2958, 17,  4}, { -2908,  9,  3}, { -2861, 19,  4}, { -2816,  5,  2},
	{ -2774, 21,  4}, { -2734, 11,  3}, { -2695, 23,  4}, { -2658,  3,  1},
	{ -2623, 25,  4}, { -2589, 13,  3}, { -2556, 27,  4}, { -2524,  7,  2},
	{ -2494, 29,  4}, { -2464, 15,  3}, { -2436, 31,  4}, { -2408,  2,  0},
	{ -2356, 17,  3}, { -2306,  9,  2}, { -2259, 19,  3}, { -2214,  5,  1},
	{ -2172, 21,  3}, { -2132, 11,  2}, { -2093, 23,  3}, { -2056,  3,  0},
	{ -2021, 25,  3}, { -1987, 13,  2}, { -1954, 27,  3}, { -1922,  7,  1},
	{ -1892, 29,  3}, { -1862, 15,  2}, { -1834, 31,  3}, { -1806,  4,  0},
	{ -1754, 17,  2}, { -1704,  9,  1}, { -1657, 19,  2}, { -1612,  5,  0},
	{ -1570, 21,  2}, { -1530, 11,  1}, { -1491, 23,  2}, { -1454,  6,  0},
	{ -1419, 25,  2}, { -1384, 13,  1}, { -1352, 27,  2}, { -1320,  7,  0},
	{ -1290, 29,  2}, { -1260, 15,  1}, { -1232, 31,  2}, { -1204,  8,  0},
	{ -1151, 17,  1}, { -1102,  9,  0}, { -1055, 19,  1}, { -1010, 10,  0},
	{  -968, 21,  1}, {  -928, 11,  0}, {  -889, 23,  1}, {  -852, 12,  0},
	{  -816, 25,  1}, {  -782, 13,  0}, {  -750, 27,  1}, {  -718, 14,  0},
	{  -688, 29,  1}, {  -658, 15,  0}, {  -630, 31,  1}, {  -602, 16,  0},
	{  -549, 17,  0}, {  -500, 18,  0}, {  -453, 19,  0}, {  -408, 20,  0},
	{  -366, 21,  0}, {  -325, 22,  0}, {  -287, 23,  0}, {  -250, 24,  0},
	{  -214, 25,  0}, {  -180, 26,  0}, {  -148, 27,  0}, {  -116, 28,  0},
	{   -86, 29,  0}, {   -56, 30,  0}, {   -28, 31,  0}, {     0,  0,  0},
};

static int pm860x_volatile(unsigned int reg)
{
	BUG_ON(reg >= REG_CACHE_SIZE);

	switch (reg) {
	case PM860X_AUDIO_SUPPLIES_2:
		return 1;
	}

	return 0;
}

static unsigned int pm860x_read_reg_cache(struct snd_soc_codec *codec,
					  unsigned int reg)
{
	unsigned char *cache = codec->reg_cache;

	BUG_ON(reg >= REG_CACHE_SIZE);

	if (pm860x_volatile(reg))
		return cache[reg];

	reg += REG_CACHE_BASE;

	return pm860x_reg_read(codec->control_data, reg);
}

static int pm860x_write_reg_cache(struct snd_soc_codec *codec,
				  unsigned int reg, unsigned int value)
{
	unsigned char *cache = codec->reg_cache;

	BUG_ON(reg >= REG_CACHE_SIZE);

	if (!pm860x_volatile(reg))
		cache[reg] = (unsigned char)value;

	reg += REG_CACHE_BASE;

	return pm860x_reg_write(codec->control_data, reg, value);
}

static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol,
				   struct snd_ctl_elem_value *ucontrol)
{
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	int val[2], val2[2], i;

	val[0] = snd_soc_read(codec, reg) & 0x3f;
	val[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT) >> 4) & 0xf;
	val2[0] = snd_soc_read(codec, reg2) & 0x3f;
	val2[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT)) & 0xf;

	for (i = 0; i < ARRAY_SIZE(st_table); i++) {
		if ((st_table[i].m == val[0]) && (st_table[i].n == val[1]))
			ucontrol->value.integer.value[0] = i;
		if ((st_table[i].m == val2[0]) && (st_table[i].n == val2[1]))
			ucontrol->value.integer.value[1] = i;
	}
	return 0;
}

static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol,
				   struct snd_ctl_elem_value *ucontrol)
{
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	int err;
	unsigned int val, val2;

	val = ucontrol->value.integer.value[0];
	val2 = ucontrol->value.integer.value[1];

	err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m);
	if (err < 0)
		return err;
	err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0xf0,
				  st_table[val].n << 4);
	if (err < 0)
		return err;

	err = snd_soc_update_bits(codec, reg2, 0x3f, st_table[val2].m);
	if (err < 0)
		return err;
	err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0x0f,
				  st_table[val2].n);
	return err;
}

static int snd_soc_get_volsw_2r_out(struct snd_kcontrol *kcontrol,
				    struct snd_ctl_elem_value *ucontrol)
{
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	unsigned int shift = mc->shift;
	int max = mc->max, val, val2;
	unsigned int mask = (1 << fls(max)) - 1;

	val = snd_soc_read(codec, reg) >> shift;
	val2 = snd_soc_read(codec, reg2) >> shift;
	ucontrol->value.integer.value[0] = (max - val) & mask;
	ucontrol->value.integer.value[1] = (max - val2) & mask;

	return 0;
}

static int snd_soc_put_volsw_2r_out(struct snd_kcontrol *kcontrol,
				    struct snd_ctl_elem_value *ucontrol)
{
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	unsigned int shift = mc->shift;
	int max = mc->max;
	unsigned int mask = (1 << fls(max)) - 1;
	int err;
	unsigned int val, val2, val_mask;

	val_mask = mask << shift;
	val = ((max - ucontrol->value.integer.value[0]) & mask);
	val2 = ((max - ucontrol->value.integer.value[1]) & mask);

	val = val << shift;
	val2 = val2 << shift;

	err = snd_soc_update_bits(codec, reg, val_mask, val);
	if (err < 0)
		return err;

	err = snd_soc_update_bits(codec, reg2, val_mask, val2);
	return err;
}

/* DAPM Widget Events */
/*
 * A lot registers are belong to RSYNC domain. It requires enabling RSYNC bit
 * after updating these registers. Otherwise, these updated registers won't
 * be effective.
 */
static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
			      struct snd_kcontrol *kcontrol, int event)
{
	struct snd_soc_codec *codec = w->codec;

	/*
	 * In order to avoid current on the load, mute power-on and power-off
	 * should be transients.
	 * Unmute by DAC_MUTE. It should be unmuted when DAPM sequence is
	 * finished.
	 */
	snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0);
	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
			    RSYNC_CHANGE, RSYNC_CHANGE);
	return 0;
}

static int pm860x_dac_event(struct snd_soc_dapm_widget *w,
			    struct snd_kcontrol *kcontrol, int event)
{
	struct snd_soc_codec *codec = w->codec;
	unsigned int dac = 0;
	int data;

	if (!strcmp(w->name, "Left DAC"))
		dac = DAC_LEFT;
	if (!strcmp(w->name, "Right DAC"))
		dac = DAC_RIGHT;
	switch (event) {
	case SND_SOC_DAPM_PRE_PMU:
		if (dac) {
			/* Auto mute in power-on sequence. */
			dac |= MODULATOR;
			snd_soc_update_bits(codec, PM860X_DAC_OFFSET,
					    DAC_MUTE, DAC_MUTE);
			snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
					    RSYNC_CHANGE, RSYNC_CHANGE);
			/* update dac */
			snd_soc_update_bits(codec, PM860X_DAC_EN_2,
					    dac, dac);
		}
		break;
	case SND_SOC_DAPM_PRE_PMD:
		if (dac) {
			/* Auto mute in power-off sequence. */
			snd_soc_update_bits(codec, PM860X_DAC_OFFSET,
					    DAC_MUTE, DAC_MUTE);
			snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
					    RSYNC_CHANGE, RSYNC_CHANGE);
			/* update dac */
			data = snd_soc_read(codec, PM860X_DAC_EN_2);
			data &= ~dac;
			if (!(data & (DAC_LEFT | DAC_RIGHT)))
				data &= ~MODULATOR;
			snd_soc_write(codec, PM860X_DAC_EN_2, data);
		}
		break;
	}
	return 0;
}

static const char *pm860x_opamp_texts[] = {"-50%", "-25%", "0%", "75%"};

static const char *pm860x_pa_texts[] = {"-33%", "0%", "33%", "66%"};

static const struct soc_enum pm860x_hs1_opamp_enum =
	SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts);

static const struct soc_enum pm860x_hs2_opamp_enum =
	SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts);

static const struct soc_enum pm860x_hs1_pa_enum =
	SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts);

static const struct soc_enum pm860x_hs2_pa_enum =
	SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts);

static const struct soc_enum pm860x_lo1_opamp_enum =
	SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts);

static const struct soc_enum pm860x_lo2_opamp_enum =
	SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts);

static const struct soc_enum pm860x_lo1_pa_enum =
	SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts);

static const struct soc_enum pm860x_lo2_pa_enum =
	SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts);

static const struct soc_enum pm860x_spk_pa_enum =
	SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts);

static const struct soc_enum pm860x_ear_pa_enum =
	SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_texts);

static const struct soc_enum pm860x_spk_ear_opamp_enum =
	SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts);

static const struct snd_kcontrol_new pm860x_snd_controls[] = {
	SOC_DOUBLE_R_TLV("ADC Capture Volume", PM860X_ADC_ANA_2,
			PM860X_ADC_ANA_3, 6, 3, 0, adc_tlv),
	SOC_DOUBLE_TLV("AUX Capture Volume", PM860X_ADC_ANA_3, 0, 3, 7, 0,
			aux_tlv),
	SOC_SINGLE_TLV("MIC1 Capture Volume", PM860X_ADC_ANA_2, 0, 7, 0,
			mic_tlv),
	SOC_SINGLE_TLV("MIC3 Capture Volume", PM860X_ADC_ANA_2, 3, 7, 0,
			mic_tlv),
	SOC_DOUBLE_R_EXT_TLV("Sidetone Volume", PM860X_SIDETONE_L_GAIN,
			     PM860X_SIDETONE_R_GAIN, 0, ARRAY_SIZE(st_table)-1,
			     0, snd_soc_get_volsw_2r_st,
			     snd_soc_put_volsw_2r_st, st_tlv),
	SOC_SINGLE_TLV("Speaker Playback Volume", PM860X_EAR_CTRL_1,
			0, 7, 0, out_tlv),
	SOC_DOUBLE_R_TLV("Line Playback Volume", PM860X_LO1_CTRL,
			 PM860X_LO2_CTRL, 0, 7, 0, out_tlv),
	SOC_DOUBLE_R_TLV("Headset Playback Volume", PM860X_HS1_CTRL,
			 PM860X_HS2_CTRL, 0, 7, 0, out_tlv),
	SOC_DOUBLE_R_EXT_TLV("Hifi Left Playback Volume",
			     PM860X_HIFIL_GAIN_LEFT,
			     PM860X_HIFIL_GAIN_RIGHT, 0, 63, 0,
			     snd_soc_get_volsw_2r_out,
			     snd_soc_put_volsw_2r_out, dpga_tlv),
	SOC_DOUBLE_R_EXT_TLV("Hifi Right Playback Volume",
			     PM860X_HIFIR_GAIN_LEFT,
			     PM860X_HIFIR_GAIN_RIGHT, 0, 63, 0,
			     snd_soc_get_volsw_2r_out,
			     snd_soc_put_volsw_2r_out, dpga_tlv),
	SOC_DOUBLE_R_EXT_TLV("Lofi Playback Volume", PM860X_LOFI_GAIN_LEFT,
			     PM860X_LOFI_GAIN_RIGHT, 0, 63, 0,
			     snd_soc_get_volsw_2r_out,
			     snd_soc_put_volsw_2r_out, dpga_tlv),
	SOC_ENUM("Headset1 Operational Amplifier Current",
		 pm860x_hs1_opamp_enum),
	SOC_ENUM("Headset2 Operational Amplifier Current",
		 pm860x_hs2_opamp_enum),
	SOC_ENUM("Headset1 Amplifier Current", pm860x_hs1_pa_enum),
	SOC_ENUM("Headset2 Amplifier Current", pm860x_hs2_pa_enum),
	SOC_ENUM("Lineout1 Operational Amplifier Current",
		 pm860x_lo1_opamp_enum),
	SOC_ENUM("Lineout2 Operational Amplifier Current",
		 pm860x_lo2_opamp_enum),
	SOC_ENUM("Lineout1 Amplifier Current", pm860x_lo1_pa_enum),
	SOC_ENUM("Lineout2 Amplifier Current", pm860x_lo2_pa_enum),
	SOC_ENUM("Speaker Operational Amplifier Current",
		 pm860x_spk_ear_opamp_enum),
	SOC_ENUM("Speaker Amplifier Current", pm860x_spk_pa_enum),
	SOC_ENUM("Earpiece Amplifier Current", pm860x_ear_pa_enum),
};

/*
 * DAPM Controls
 */

/* PCM Switch / PCM Interface */
static const struct snd_kcontrol_new pcm_switch_controls =
	SOC_DAPM_SINGLE("Switch", PM860X_ADC_EN_2, 0, 1, 0);

/* AUX1 Switch */
static const struct snd_kcontrol_new aux1_switch_controls =
	SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 4, 1, 0);

/* AUX2 Switch */
static const struct snd_kcontrol_new aux2_switch_controls =
	SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 5, 1, 0);

/* Left Ex. PA Switch */
static const struct snd_kcontrol_new lepa_switch_controls =
	SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 2, 1, 0);

/* Right Ex. PA Switch */
static const struct snd_kcontrol_new repa_switch_controls =
	SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 1, 1, 0);

/* PCM Mux / Mux7 */
static const char *aif1_text[] = {
	"PCM L", "PCM R",
};

static const struct soc_enum aif1_enum =
	SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 6, 2, aif1_text);

static const struct snd_kcontrol_new aif1_mux =
	SOC_DAPM_ENUM("PCM Mux", aif1_enum);

/* I2S Mux / Mux9 */
static const char *i2s_din_text[] = {
	"DIN", "DIN1",
};

static const struct soc_enum i2s_din_enum =
	SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 1, 2, i2s_din_text);

static const struct snd_kcontrol_new i2s_din_mux =
	SOC_DAPM_ENUM("I2S DIN Mux", i2s_din_enum);

/* I2S Mic Mux / Mux8 */
static const char *i2s_mic_text[] = {
	"Ex PA", "ADC",
};

static const struct soc_enum i2s_mic_enum =
	SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 4, 2, i2s_mic_text);

static const struct snd_kcontrol_new i2s_mic_mux =
	SOC_DAPM_ENUM("I2S Mic Mux", i2s_mic_enum);

/* ADCL Mux / Mux2 */
static const char *adcl_text[] = {
	"ADCR", "ADCL",
};

static const struct soc_enum adcl_enum =
	SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 4, 2, adcl_text);

static const struct snd_kcontrol_new adcl_mux =
	SOC_DAPM_ENUM("ADC Left Mux", adcl_enum);

/* ADCR Mux / Mux3 */
static const char *adcr_text[] = {
	"ADCL", "ADCR",
};

static const struct soc_enum adcr_enum =
	SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 2, 2, adcr_text);

static const struct snd_kcontrol_new adcr_mux =
	SOC_DAPM_ENUM("ADC Right Mux", adcr_enum);

/* ADCR EC Mux / Mux6 */
static const char *adcr_ec_text[] = {
	"ADCR", "EC",
};

static const struct soc_enum adcr_ec_enum =
	SOC_ENUM_SINGLE(PM860X_ADC_EN_2, 3, 2, adcr_ec_text);

static const struct snd_kcontrol_new adcr_ec_mux =
	SOC_DAPM_ENUM("ADCR EC Mux", adcr_ec_enum);

/* EC Mux / Mux4 */
static const char *ec_text[] = {
	"Left", "Right", "Left + Right",
};

static const struct soc_enum ec_enum =
	SOC_ENUM_SINGLE(PM860X_EC_PATH, 1, 3, ec_text);

static const struct snd_kcontrol_new ec_mux =
	SOC_DAPM_ENUM("EC Mux", ec_enum);

static const char *dac_text[] = {
	"No input", "Right", "Left", "No input",
};

/* DAC Headset 1 Mux / Mux10 */
static const struct soc_enum dac_hs1_enum =
	SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 0, 4, dac_text);

static const struct snd_kcontrol_new dac_hs1_mux =
	SOC_DAPM_ENUM("DAC HS1 Mux", dac_hs1_enum);

/* DAC Headset 2 Mux / Mux11 */
static const struct soc_enum dac_hs2_enum =
	SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 2, 4, dac_text);

static const struct snd_kcontrol_new dac_hs2_mux =
	SOC_DAPM_ENUM("DAC HS2 Mux", dac_hs2_enum);

/* DAC Lineout 1 Mux / Mux12 */
static const struct soc_enum dac_lo1_enum =
	SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 4, 4, dac_text);

static const struct snd_kcontrol_new dac_lo1_mux =
	SOC_DAPM_ENUM("DAC LO1 Mux", dac_lo1_enum);

/* DAC Lineout 2 Mux / Mux13 */
static const struct soc_enum dac_lo2_enum =
	SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 6, 4, dac_text);

static const struct snd_kcontrol_new dac_lo2_mux =
	SOC_DAPM_ENUM("DAC LO2 Mux", dac_lo2_enum);

/* DAC Spearker Earphone Mux / Mux14 */
static const struct soc_enum dac_spk_ear_enum =
	SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_2, 0, 4, dac_text);

static const struct snd_kcontrol_new dac_spk_ear_mux =
	SOC_DAPM_ENUM("DAC SP Mux", dac_spk_ear_enum);

/* Headset 1 Mux / Mux15 */
static const char *in_text[] = {
	"Digital", "Analog",
};

static const struct soc_enum hs1_enum =
	SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 0, 2, in_text);

static const struct snd_kcontrol_new hs1_mux =
	SOC_DAPM_ENUM("Headset1 Mux", hs1_enum);

/* Headset 2 Mux / Mux16 */
static const struct soc_enum hs2_enum =
	SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 1, 2, in_text);

static const struct snd_kcontrol_new hs2_mux =
	SOC_DAPM_ENUM("Headset2 Mux", hs2_enum);

/* Lineout 1 Mux / Mux17 */
static const struct soc_enum lo1_enum =
	SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 2, 2, in_text);

static const struct snd_kcontrol_new lo1_mux =
	SOC_DAPM_ENUM("Lineout1 Mux", lo1_enum);

/* Lineout 2 Mux / Mux18 */
static const struct soc_enum lo2_enum =
	SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 3, 2, in_text);

static const struct snd_kcontrol_new lo2_mux =
	SOC_DAPM_ENUM("Lineout2 Mux", lo2_enum);

/* Speaker Earpiece Demux */
static const char *spk_text[] = {
	"Earpiece", "Speaker",
};

static const struct soc_enum spk_enum =
	SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 6, 2, spk_text);

static const struct snd_kcontrol_new spk_demux =
	SOC_DAPM_ENUM("Speaker Earpiece Demux", spk_enum);

/* MIC Mux / Mux1 */
static const char *mic_text[] = {
	"Mic 1", "Mic 2",
};

static const struct soc_enum mic_enum =
	SOC_ENUM_SINGLE(PM860X_ADC_ANA_4, 4, 2, mic_text);

static const struct snd_kcontrol_new mic_mux =
	SOC_DAPM_ENUM("MIC Mux", mic_enum);

static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = {
	SND_SOC_DAPM_AIF_IN("PCM SDI", "PCM Playback", 0,
			    PM860X_ADC_EN_2, 0, 0),
	SND_SOC_DAPM_AIF_OUT("PCM SDO", "PCM Capture", 0,
			     PM860X_PCM_IFACE_3, 1, 1),


	SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0,
			    SND_SOC_NOPM, 0, 0),
	SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0,
			    SND_SOC_NOPM, 0, 0),
	SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0,
			     PM860X_I2S_IFACE_3, 5, 1),
	SND_SOC_DAPM_SUPPLY("I2S CLK", PM860X_DAC_EN_2, 0, 0, NULL, 0),
	SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux),
	SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux),
	SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux),
	SND_SOC_DAPM_MUX("EC Mux", SND_SOC_NOPM, 0, 0, &ec_mux),
	SND_SOC_DAPM_MUX("ADCR EC Mux", SND_SOC_NOPM, 0, 0, &adcr_ec_mux),
	SND_SOC_DAPM_SWITCH("Left EPA", SND_SOC_NOPM, 0, 0,
			    &lepa_switch_controls),
	SND_SOC_DAPM_SWITCH("Right EPA", SND_SOC_NOPM, 0, 0,
			    &repa_switch_controls),

	SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Left ADC MOD", PM860X_ADC_EN_1,
			 0, 1, 1, 0),
	SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Right ADC MOD", PM860X_ADC_EN_1,
			 1, 1, 1, 0),
	SND_SOC_DAPM_ADC("Left ADC", NULL, PM860X_ADC_EN_2, 5, 0),
	SND_SOC_DAPM_ADC("Right ADC", NULL, PM860X_ADC_EN_2, 4, 0),

	SND_SOC_DAPM_SWITCH("AUX1 Switch", SND_SOC_NOPM, 0, 0,
			    &aux1_switch_controls),
	SND_SOC_DAPM_SWITCH("AUX2 Switch", SND_SOC_NOPM, 0, 0,
			    &aux2_switch_controls),

	SND_SOC_DAPM_MUX("MIC Mux", SND_SOC_NOPM, 0, 0, &mic_mux),
	SND_SOC_DAPM_MICBIAS("Mic1 Bias", PM860X_ADC_ANA_1, 2, 0),
	SND_SOC_DAPM_MICBIAS("Mic3 Bias", PM860X_ADC_ANA_1, 7, 0),
	SND_SOC_DAPM_PGA("MIC1 Volume", PM860X_ADC_EN_1, 2, 0, NULL, 0),
	SND_SOC_DAPM_PGA("MIC3 Volume", PM860X_ADC_EN_1, 3, 0, NULL, 0),
	SND_SOC_DAPM_PGA("AUX1 Volume", PM860X_ADC_EN_1, 4, 0, NULL, 0),
	SND_SOC_DAPM_PGA("AUX2 Volume", PM860X_ADC_EN_1, 5, 0, NULL, 0),
	SND_SOC_DAPM_PGA("Sidetone PGA", PM860X_ADC_EN_2, 1, 0, NULL, 0),
	SND_SOC_DAPM_PGA("Lofi PGA", PM860X_ADC_EN_2, 2, 0, NULL, 0),

	SND_SOC_DAPM_INPUT("AUX1"),
	SND_SOC_DAPM_INPUT("AUX2"),
	SND_SOC_DAPM_INPUT("MIC1P"),
	SND_SOC_DAPM_INPUT("MIC1N"),
	SND_SOC_DAPM_INPUT("MIC2P"),
	SND_SOC_DAPM_INPUT("MIC2N"),
	SND_SOC_DAPM_INPUT("MIC3P"),
	SND_SOC_DAPM_INPUT("MIC3N"),

	SND_SOC_DAPM_DAC_E("Left DAC", NULL, SND_SOC_NOPM, 0, 0,
			   pm860x_dac_event,
			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
	SND_SOC_DAPM_DAC_E("Right DAC", NULL, SND_SOC_NOPM, 0, 0,
			   pm860x_dac_event,
			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),

	SND_SOC_DAPM_MUX("I2S DIN Mux", SND_SOC_NOPM, 0, 0, &i2s_din_mux),
	SND_SOC_DAPM_MUX("DAC HS1 Mux", SND_SOC_NOPM, 0, 0, &dac_hs1_mux),
	SND_SOC_DAPM_MUX("DAC HS2 Mux", SND_SOC_NOPM, 0, 0, &dac_hs2_mux),
	SND_SOC_DAPM_MUX("DAC LO1 Mux", SND_SOC_NOPM, 0, 0, &dac_lo1_mux),
	SND_SOC_DAPM_MUX("DAC LO2 Mux", SND_SOC_NOPM, 0, 0, &dac_lo2_mux),
	SND_SOC_DAPM_MUX("DAC SP Mux", SND_SOC_NOPM, 0, 0, &dac_spk_ear_mux),
	SND_SOC_DAPM_MUX("Headset1 Mux", SND_SOC_NOPM, 0, 0, &hs1_mux),
	SND_SOC_DAPM_MUX("Headset2 Mux", SND_SOC_NOPM, 0, 0, &hs2_mux),
	SND_SOC_DAPM_MUX("Lineout1 Mux", SND_SOC_NOPM, 0, 0, &lo1_mux),
	SND_SOC_DAPM_MUX("Lineout2 Mux", SND_SOC_NOPM, 0, 0, &lo2_mux),
	SND_SOC_DAPM_MUX("Speaker Earpiece Demux", SND_SOC_NOPM, 0, 0,
			 &spk_demux),


	SND_SOC_DAPM_PGA("Headset1 PGA", PM860X_DAC_EN_1, 0, 0, NULL, 0),
	SND_SOC_DAPM_PGA("Headset2 PGA", PM860X_DAC_EN_1, 1, 0, NULL, 0),
	SND_SOC_DAPM_OUTPUT("HS1"),
	SND_SOC_DAPM_OUTPUT("HS2"),
	SND_SOC_DAPM_PGA("Lineout1 PGA", PM860X_DAC_EN_1, 2, 0, NULL, 0),
	SND_SOC_DAPM_PGA("Lineout2 PGA", PM860X_DAC_EN_1, 3, 0, NULL, 0),
	SND_SOC_DAPM_OUTPUT("LINEOUT1"),
	SND_SOC_DAPM_OUTPUT("LINEOUT2"),
	SND_SOC_DAPM_PGA("Earpiece PGA", PM860X_DAC_EN_1, 4, 0, NULL, 0),
	SND_SOC_DAPM_OUTPUT("EARP"),
	SND_SOC_DAPM_OUTPUT("EARN"),
	SND_SOC_DAPM_PGA("Speaker PGA", PM860X_DAC_EN_1, 5, 0, NULL, 0),
	SND_SOC_DAPM_OUTPUT("LSP"),
	SND_SOC_DAPM_OUTPUT("LSN"),
	SND_SOC_DAPM_REG(snd_soc_dapm_supply, "VCODEC", PM860X_AUDIO_SUPPLIES_2,
			 0, SUPPLY_MASK, SUPPLY_MASK, 0),

	PM860X_DAPM_OUTPUT("RSYNC", pm860x_rsync_event),
};

static const struct snd_soc_dapm_route audio_map[] = {
	/* supply */
	{"Left DAC", NULL, "VCODEC"},
	{"Right DAC", NULL, "VCODEC"},
	{"Left ADC", NULL, "VCODEC"},
	{"Right ADC", NULL, "VCODEC"},
	{"Left ADC", NULL, "Left ADC MOD"},
	{"Right ADC", NULL, "Right ADC MOD"},

	/* I2S Clock */
	{"I2S DIN", NULL, "I2S CLK"},
	{"I2S DIN1", NULL, "I2S CLK"},
	{"I2S DOUT", NULL, "I2S CLK"},

	/* PCM/AIF1 Inputs */
	{"PCM SDO", NULL, "ADC Left Mux"},
	{"PCM SDO", NULL, "ADCR EC Mux"},

	/* PCM/AFI2 Outputs */
	{"Lofi PGA", NULL, "PCM SDI"},
	{"Lofi PGA", NULL, "Sidetone PGA"},
	{"Left DAC", NULL, "Lofi PGA"},
	{"Right DAC", NULL, "Lofi PGA"},

	/* I2S/AIF2 Inputs */
	{"MIC Mux", "Mic 1", "MIC1P"},
	{"MIC Mux", "Mic 1", "MIC1N"},
	{"MIC Mux", "Mic 2", "MIC2P"},
	{"MIC Mux", "Mic 2", "MIC2N"},
	{"MIC1 Volume", NULL, "MIC Mux"},
	{"MIC3 Volume", NULL, "MIC3P"},
	{"MIC3 Volume", NULL, "MIC3N"},
	{"Left ADC", NULL, "MIC1 Volume"},
	{"Right ADC", NULL, "MIC3 Volume"},
	{"ADC Left Mux", "ADCR", "Right ADC"},
	{"ADC Left Mux", "ADCL", "Left ADC"},
	{"ADC Right Mux", "ADCL", "Left ADC"},
	{"ADC Right Mux", "ADCR", "Right ADC"},
	{"Left EPA", "Switch", "Left DAC"},
	{"Right EPA", "Switch", "Right DAC"},
	{"EC Mux", "Left", "Left DAC"},
	{"EC Mux", "Right", "Right DAC"},
	{"EC Mux", "Left + Right", "Left DAC"},
	{"EC Mux", "Left + Right", "Right DAC"},
	{"ADCR EC Mux", "ADCR", "ADC Right Mux"},
	{"ADCR EC Mux", "EC", "EC Mux"},
	{"I2S Mic Mux", "Ex PA", "Left EPA"},
	{"I2S Mic Mux", "Ex PA", "Right EPA"},
	{"I2S Mic Mux", "ADC", "ADC Left Mux"},
	{"I2S Mic Mux", "ADC", "ADCR EC Mux"},
	{"I2S DOUT", NULL, "I2S Mic Mux"},

	/* I2S/AIF2 Outputs */
	{"I2S DIN Mux", "DIN", "I2S DIN"},
	{"I2S DIN Mux", "DIN1", "I2S DIN1"},
	{"Left DAC", NULL, "I2S DIN Mux"},
	{"Right DAC", NULL, "I2S DIN Mux"},
	{"DAC HS1 Mux", "Left", "Left DAC"},
	{"DAC HS1 Mux", "Right", "Right DAC"},
	{"DAC HS2 Mux", "Left", "Left DAC"},
	{"DAC HS2 Mux", "Right", "Right DAC"},
	{"DAC LO1 Mux", "Left", "Left DAC"},
	{"DAC LO1 Mux", "Right", "Right DAC"},
	{"DAC LO2 Mux", "Left", "Left DAC"},
	{"DAC LO2 Mux", "Right", "Right DAC"},
	{"Headset1 Mux", "Digital", "DAC HS1 Mux"},
	{"Headset2 Mux", "Digital", "DAC HS2 Mux"},
	{"Lineout1 Mux", "Digital", "DAC LO1 Mux"},
	{"Lineout2 Mux", "Digital", "DAC LO2 Mux"},
	{"Headset1 PGA", NULL, "Headset1 Mux"},
	{"Headset2 PGA", NULL, "Headset2 Mux"},
	{"Lineout1 PGA", NULL, "Lineout1 Mux"},
	{"Lineout2 PGA", NULL, "Lineout2 Mux"},
	{"DAC SP Mux", "Left", "Left DAC"},
	{"DAC SP Mux", "Right", "Right DAC"},
	{"Speaker Earpiece Demux", "Speaker", "DAC SP Mux"},
	{"Speaker PGA", NULL, "Speaker Earpiece Demux"},
	{"Earpiece PGA", NULL, "Speaker Earpiece Demux"},

	{"RSYNC", NULL, "Headset1 PGA"},
	{"RSYNC", NULL, "Headset2 PGA"},
	{"RSYNC", NULL, "Lineout1 PGA"},
	{"RSYNC", NULL, "Lineout2 PGA"},
	{"RSYNC", NULL, "Speaker PGA"},
	{"RSYNC", NULL, "Speaker PGA"},
	{"RSYNC", NULL, "Earpiece PGA"},
	{"RSYNC", NULL, "Earpiece PGA"},

	{"HS1", NULL, "RSYNC"},
	{"HS2", NULL, "RSYNC"},
	{"LINEOUT1", NULL, "RSYNC"},
	{"LINEOUT2", NULL, "RSYNC"},
	{"LSP", NULL, "RSYNC"},
	{"LSN", NULL, "RSYNC"},
	{"EARP", NULL, "RSYNC"},
	{"EARN", NULL, "RSYNC"},
};

/*
 * Use MUTE_LEFT & MUTE_RIGHT to implement digital mute.
 * These bits can also be used to mute.
 */
static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute)
{
	struct snd_soc_codec *codec = codec_dai->codec;
	int data = 0, mask = MUTE_LEFT | MUTE_RIGHT;

	if (mute)
		data = mask;
	snd_soc_update_bits(codec, PM860X_DAC_OFFSET, mask, data);
	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
			    RSYNC_CHANGE, RSYNC_CHANGE);
	return 0;
}

static int pm860x_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;
	unsigned char inf = 0, mask = 0;

	/* bit size */
	switch (params_format(params)) {
	case SNDRV_PCM_FORMAT_S16_LE:
		inf &= ~PCM_INF2_18WL;
		break;
	case SNDRV_PCM_FORMAT_S18_3LE:
		inf |= PCM_INF2_18WL;
		break;
	default:
		return -EINVAL;
	}
	mask |= PCM_INF2_18WL;
	snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf);

	/* sample rate */
	switch (params_rate(params)) {
	case 8000:
		inf = 0;
		break;
	case 16000:
		inf = 3;
		break;
	case 32000:
		inf = 6;
		break;
	case 48000:
		inf = 8;
		break;
	default:
		return -EINVAL;
	}
	snd_soc_update_bits(codec, PM860X_PCM_RATE, 0x0f, inf);

	return 0;
}

static int pm860x_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
				  unsigned int fmt)
{
	struct snd_soc_codec *codec = codec_dai->codec;
	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
	unsigned char inf = 0, mask = 0;
	int ret = -EINVAL;

	mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER;

	/* set master/slave audio interface */
	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
	case SND_SOC_DAIFMT_CBM_CFM:
	case SND_SOC_DAIFMT_CBM_CFS:
		if (pm860x->dir == PM860X_CLK_DIR_OUT) {
			inf |= PCM_INF2_MASTER;
			ret = 0;
		}
		break;
	case SND_SOC_DAIFMT_CBS_CFS:
		if (pm860x->dir == PM860X_CLK_DIR_IN) {
			inf &= ~PCM_INF2_MASTER;
			ret = 0;
		}
		break;
	}

	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
	case SND_SOC_DAIFMT_I2S:
		inf |= PCM_EXACT_I2S;
		ret = 0;
		break;
	}
	mask |= PCM_MODE_MASK;
	if (ret)
		return ret;
	snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf);
	return 0;
}

static int pm860x_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 pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);

	if (dir == PM860X_CLK_DIR_OUT)
		pm860x->dir = PM860X_CLK_DIR_OUT;
	else {
		pm860x->dir = PM860X_CLK_DIR_IN;
		return -EINVAL;
	}

	return 0;
}

static int pm860x_i2s_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;
	unsigned char inf;

	/* bit size */
	switch (params_format(params)) {
	case SNDRV_PCM_FORMAT_S16_LE:
		inf = 0;
		break;
	case SNDRV_PCM_FORMAT_S18_3LE:
		inf = PCM_INF2_18WL;
		break;
	default:
		return -EINVAL;
	}
	snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, PCM_INF2_18WL, inf);

	/* sample rate */
	switch (params_rate(params)) {
	case 8000:
		inf = 0;
		break;
	case 11025:
		inf = 1;
		break;
	case 16000:
		inf = 3;
		break;
	case 22050:
		inf = 4;
		break;
	case 32000:
		inf = 6;
		break;
	case 44100:
		inf = 7;
		break;
	case 48000:
		inf = 8;
		break;
	default:
		return -EINVAL;
	}
	snd_soc_update_bits(codec, PM860X_I2S_IFACE_4, 0xf, inf);

	return 0;
}

static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
				  unsigned int fmt)
{
	struct snd_soc_codec *codec = codec_dai->codec;
	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
	unsigned char inf = 0, mask = 0;

	mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER;

	/* set master/slave audio interface */
	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
	case SND_SOC_DAIFMT_CBM_CFM:
		if (pm860x->dir == PM860X_CLK_DIR_OUT)
			inf |= PCM_INF2_MASTER;
		else
			return -EINVAL;
		break;
	case SND_SOC_DAIFMT_CBS_CFS:
		if (pm860x->dir == PM860X_CLK_DIR_IN)
			inf &= ~PCM_INF2_MASTER;
		else
			return -EINVAL;
		break;
	default:
		return -EINVAL;
	}

	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
	case SND_SOC_DAIFMT_I2S:
		inf |= PCM_EXACT_I2S;
		break;
	default:
		return -EINVAL;
	}
	mask |= PCM_MODE_MASK;
	snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, mask, inf);
	return 0;
}

static int pm860x_set_bias_level(struct snd_soc_codec *codec,
				 enum snd_soc_bias_level level)
{
	int data;

	switch (level) {
	case SND_SOC_BIAS_ON:
		break;

	case SND_SOC_BIAS_PREPARE:
		break;

	case SND_SOC_BIAS_STANDBY:
		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
			/* Enable Audio PLL & Audio section */
			data = AUDIO_PLL | AUDIO_SECTION_ON;
			pm860x_reg_write(codec->control_data, REG_MISC2, data);
			udelay(300);
			data = AUDIO_PLL | AUDIO_SECTION_RESET
				| AUDIO_SECTION_ON;
			pm860x_reg_write(codec->control_data, REG_MISC2, data);
		}
		break;

	case SND_SOC_BIAS_OFF:
		data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON;
		pm860x_set_bits(codec->control_data, REG_MISC2, data, 0);
		break;
	}
	codec->dapm.bias_level = level;
	return 0;
}

static const struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
	.digital_mute	= pm860x_digital_mute,
	.hw_params	= pm860x_pcm_hw_params,
	.set_fmt	= pm860x_pcm_set_dai_fmt,
	.set_sysclk	= pm860x_set_dai_sysclk,
};

static const struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
	.digital_mute	= pm860x_digital_mute,
	.hw_params	= pm860x_i2s_hw_params,
	.set_fmt	= pm860x_i2s_set_dai_fmt,
	.set_sysclk	= pm860x_set_dai_sysclk,
};

#define PM860X_RATES	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |	\
			 SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000)

static struct snd_soc_dai_driver pm860x_dai[] = {
	{
		/* DAI PCM */
		.name	= "88pm860x-pcm",
		.id	= 1,
		.playback = {
			.stream_name	= "PCM Playback",
			.channels_min	= 2,
			.channels_max	= 2,
			.rates		= PM860X_RATES,
			.formats	= SNDRV_PCM_FORMAT_S16_LE | \
					  SNDRV_PCM_FORMAT_S18_3LE,
		},
		.capture = {
			.stream_name	= "PCM Capture",
			.channels_min	= 2,
			.channels_max	= 2,
			.rates		= PM860X_RATES,
			.formats	= SNDRV_PCM_FORMAT_S16_LE | \
					  SNDRV_PCM_FORMAT_S18_3LE,
		},
		.ops	= &pm860x_pcm_dai_ops,
	}, {
		/* DAI I2S */
		.name	= "88pm860x-i2s",
		.id	= 2,
		.playback = {
			.stream_name	= "I2S Playback",
			.channels_min	= 2,
			.channels_max	= 2,
			.rates		= SNDRV_PCM_RATE_8000_48000,
			.formats	= SNDRV_PCM_FORMAT_S16_LE | \
					  SNDRV_PCM_FORMAT_S18_3LE,
		},
		.capture = {
			.stream_name	= "I2S Capture",
			.channels_min	= 2,
			.channels_max	= 2,
			.rates		= SNDRV_PCM_RATE_8000_48000,
			.formats	= SNDRV_PCM_FORMAT_S16_LE | \
					  SNDRV_PCM_FORMAT_S18_3LE,
		},
		.ops	= &pm860x_i2s_dai_ops,
	},
};

static irqreturn_t pm860x_codec_handler(int irq, void *data)
{
	struct pm860x_priv *pm860x = data;
	int status, shrt, report = 0, mic_report = 0;
	int mask;

	status = pm860x_reg_read(pm860x->i2c, REG_STATUS_1);
	shrt = pm860x_reg_read(pm860x->i2c, REG_SHORTS);
	mask = pm860x->det.hs_shrt | pm860x->det.hook_det | pm860x->det.lo_shrt
		| pm860x->det.hp_det;

#ifndef CONFIG_SND_SOC_88PM860X_MODULE
	if (status & (HEADSET_STATUS | MIC_STATUS | SHORT_HS1 | SHORT_HS2 |
		      SHORT_LO1 | SHORT_LO2))
		trace_snd_soc_jack_irq(dev_name(pm860x->codec->dev));
#endif

	if ((pm860x->det.hp_det & SND_JACK_HEADPHONE)
		&& (status & HEADSET_STATUS))
		report |= SND_JACK_HEADPHONE;

	if ((pm860x->det.mic_det & SND_JACK_MICROPHONE)
		&& (status & MIC_STATUS))
		mic_report |= SND_JACK_MICROPHONE;

	if (pm860x->det.hs_shrt && (shrt & (SHORT_HS1 | SHORT_HS2)))
		report |= pm860x->det.hs_shrt;

	if (pm860x->det.hook_det && (status & HOOK_STATUS))
		report |= pm860x->det.hook_det;

	if (pm860x->det.lo_shrt && (shrt & (SHORT_LO1 | SHORT_LO2)))
		report |= pm860x->det.lo_shrt;

	if (report)
		snd_soc_jack_report(pm860x->det.hp_jack, report, mask);
	if (mic_report)
		snd_soc_jack_report(pm860x->det.mic_jack, SND_JACK_MICROPHONE,
				    SND_JACK_MICROPHONE);

	dev_dbg(pm860x->codec->dev, "headphone report:0x%x, mask:%x\n",
		report, mask);
	dev_dbg(pm860x->codec->dev, "microphone report:0x%x\n", mic_report);
	return IRQ_HANDLED;
}

int pm860x_hs_jack_detect(struct snd_soc_codec *codec,
			  struct snd_soc_jack *jack,
			  int det, int hook, int hs_shrt, int lo_shrt)
{
	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
	int data;

	pm860x->det.hp_jack = jack;
	pm860x->det.hp_det = det;
	pm860x->det.hook_det = hook;
	pm860x->det.hs_shrt = hs_shrt;
	pm860x->det.lo_shrt = lo_shrt;

	if (det & SND_JACK_HEADPHONE)
		pm860x_set_bits(codec->control_data, REG_HS_DET,
				EN_HS_DET, EN_HS_DET);
	/* headset short detect */
	if (hs_shrt) {
		data = CLR_SHORT_HS2 | CLR_SHORT_HS1;
		pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
	}
	/* Lineout short detect */
	if (lo_shrt) {
		data = CLR_SHORT_LO2 | CLR_SHORT_LO1;
		pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
	}

	/* sync status */
	pm860x_codec_handler(0, pm860x);
	return 0;
}
EXPORT_SYMBOL_GPL(pm860x_hs_jack_detect);

int pm860x_mic_jack_detect(struct snd_soc_codec *codec,
			   struct snd_soc_jack *jack, int det)
{
	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);

	pm860x->det.mic_jack = jack;
	pm860x->det.mic_det = det;

	if (det & SND_JACK_MICROPHONE)
		pm860x_set_bits(codec->control_data, REG_MIC_DET,
				MICDET_MASK, MICDET_MASK);

	/* sync status */
	pm860x_codec_handler(0, pm860x);
	return 0;
}
EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect);

static int pm860x_probe(struct snd_soc_codec *codec)
{
	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
	struct snd_soc_dapm_context *dapm = &codec->dapm;
	int i, ret;

	pm860x->codec = codec;

	codec->control_data = pm860x->i2c;

	for (i = 0; i < 4; i++) {
		ret = request_threaded_irq(pm860x->irq[i], NULL,
					   pm860x_codec_handler, IRQF_ONESHOT,
					   pm860x->name[i], pm860x);
		if (ret < 0) {
			dev_err(codec->dev, "Failed to request IRQ!\n");
			goto out;
		}
	}

	pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

	ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE,
			       REG_CACHE_SIZE, codec->reg_cache);
	if (ret < 0) {
		dev_err(codec->dev, "Failed to fill register cache: %d\n",
			ret);
		goto out;
	}

	snd_soc_add_controls(codec, pm860x_snd_controls,
			     ARRAY_SIZE(pm860x_snd_controls));
	snd_soc_dapm_new_controls(dapm, pm860x_dapm_widgets,
				  ARRAY_SIZE(pm860x_dapm_widgets));
	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
	return 0;

out:
	while (--i >= 0)
		free_irq(pm860x->irq[i], pm860x);
	return ret;
}

static int pm860x_remove(struct snd_soc_codec *codec)
{
	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
	int i;

	for (i = 3; i >= 0; i--)
		free_irq(pm860x->irq[i], pm860x);
	pm860x_set_bias_level(codec, SND_SOC_BIAS_OFF);
	return 0;
}

static struct snd_soc_codec_driver soc_codec_dev_pm860x = {
	.probe		= pm860x_probe,
	.remove		= pm860x_remove,
	.read		= pm860x_read_reg_cache,
	.write		= pm860x_write_reg_cache,
	.reg_cache_size	= REG_CACHE_SIZE,
	.reg_word_size	= sizeof(u8),
	.set_bias_level	= pm860x_set_bias_level,
};

static int __devinit pm860x_codec_probe(struct platform_device *pdev)
{
	struct pm860x_chip *chip = dev_get_drvdata(pdev->dev.parent);
	struct pm860x_priv *pm860x;
	struct resource *res;
	int i, ret;

	pm860x = kzalloc(sizeof(struct pm860x_priv), GFP_KERNEL);
	if (pm860x == NULL)
		return -ENOMEM;

	pm860x->chip = chip;
	pm860x->i2c = (chip->id == CHIP_PM8607) ? chip->client
			: chip->companion;
	platform_set_drvdata(pdev, pm860x);

	for (i = 0; i < 4; i++) {
		res = platform_get_resource(pdev, IORESOURCE_IRQ, i);
		if (!res) {
			dev_err(&pdev->dev, "Failed to get IRQ resources\n");
			goto out;
		}
		pm860x->irq[i] = res->start + chip->irq_base;
		strncpy(pm860x->name[i], res->name, MAX_NAME_LEN);
	}

	ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pm860x,
				     pm860x_dai, ARRAY_SIZE(pm860x_dai));
	if (ret) {
		dev_err(&pdev->dev, "Failed to register codec\n");
		goto out;
	}
	return ret;

out:
	platform_set_drvdata(pdev, NULL);
	kfree(pm860x);
	return -EINVAL;
}

static int __devexit pm860x_codec_remove(struct platform_device *pdev)
{
	struct pm860x_priv *pm860x = platform_get_drvdata(pdev);

	snd_soc_unregister_codec(&pdev->dev);
	platform_set_drvdata(pdev, NULL);
	kfree(pm860x);
	return 0;
}

static struct platform_driver pm860x_codec_driver = {
	.driver	= {
		.name	= "88pm860x-codec",
		.owner	= THIS_MODULE,
	},
	.probe	= pm860x_codec_probe,
	.remove	= __devexit_p(pm860x_codec_remove),
};

module_platform_driver(pm860x_codec_driver);

MODULE_DESCRIPTION("ASoC 88PM860x driver");
MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:88pm860x-codec");