/*
   * 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 <linux/regmap.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)	\
  	SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, 0, 0, NULL, 0, wevent, \
  			    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 regmap		*regmap;
  	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 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];
  
  	if (val >= ARRAY_SIZE(st_table) || val2 >= ARRAY_SIZE(st_table))
  		return -EINVAL;
  
  	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 pm860x_dapm_routes[] = {
  	/* 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)
  {
  	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
  	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(pm860x->i2c, REG_MISC2, data);
  			udelay(300);
  			data = AUDIO_PLL | AUDIO_SECTION_RESET
  				| AUDIO_SECTION_ON;
  			pm860x_reg_write(pm860x->i2c, REG_MISC2, data);
  		}
  		break;
  
  	case SND_SOC_BIAS_OFF:
  		data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON;
  		pm860x_set_bits(pm860x->i2c, 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
  ",
  		report, mask);
  	dev_dbg(pm860x->codec->dev, "microphone report:0x%x
  ", 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(pm860x->i2c, 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(pm860x->i2c, REG_SHORTS, data, data);
  	}
  	/* Lineout short detect */
  	if (lo_shrt) {
  		data = CLR_SHORT_LO2 | CLR_SHORT_LO1;
  		pm860x_set_bits(pm860x->i2c, 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(pm860x->i2c, 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);
  	int i, ret;
  
  	pm860x->codec = codec;
  
  	codec->control_data = pm860x->regmap;
  	ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
  	if (ret)
  		return ret;
  
  	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!
  ");
  			goto out;
  		}
  	}
  
  	pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
  
  	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,
  	.set_bias_level	= pm860x_set_bias_level,
  
  	.controls = pm860x_snd_controls,
  	.num_controls = ARRAY_SIZE(pm860x_snd_controls),
  	.dapm_widgets = pm860x_dapm_widgets,
  	.num_dapm_widgets = ARRAY_SIZE(pm860x_dapm_widgets),
  	.dapm_routes = pm860x_dapm_routes,
  	.num_dapm_routes = ARRAY_SIZE(pm860x_dapm_routes),
  };
  
  static int 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 = devm_kzalloc(&pdev->dev, sizeof(struct pm860x_priv),
  			      GFP_KERNEL);
  	if (pm860x == NULL)
  		return -ENOMEM;
  
  	pm860x->chip = chip;
  	pm860x->i2c = (chip->id == CHIP_PM8607) ? chip->client
  			: chip->companion;
  	pm860x->regmap = (chip->id == CHIP_PM8607) ? chip->regmap
  			: chip->regmap_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
  ");
  			return -EINVAL;
  		}
  		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
  ");
  		return -EINVAL;
  	}
  	return ret;
  }
  
  static int pm860x_codec_remove(struct platform_device *pdev)
  {
  	snd_soc_unregister_codec(&pdev->dev);
  	return 0;
  }
  
  static struct platform_driver pm860x_codec_driver = {
  	.driver	= {
  		.name	= "88pm860x-codec",
  		.owner	= THIS_MODULE,
  	},
  	.probe	= pm860x_codec_probe,
  	.remove	= 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");