[PATCH 4/5] ASoC: add 88pm860x codec driver

[PATCH 4/5] ASoC: add 88pm860x codec driver

[Haojian Zhuang]

Add 88PM860x codec driver. 88PM860x codec supports two interfaces. And it
also supports headset/mic/hook/short detection.

Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com>
---
 sound/soc/codecs/88pm860x-codec.c | 1567 +++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/88pm860x-codec.h |   97 +++
 sound/soc/codecs/Kconfig          |    4 +
 sound/soc/codecs/Makefile         |    2 +
 4 files changed, 1670 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/codecs/88pm860x-codec.c
 create mode 100644 sound/soc/codecs/88pm860x-codec.h

diff --git a/sound/soc/codecs/88pm860x-codec.c
b/sound/soc/codecs/88pm860x-codec.c
new file mode 100644
index 0000000..0e5b445
--- /dev/null
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -0,0 +1,1567 @@
+/*
+ * 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 <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.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)
+
+/* 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)
+
+/* 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 SHORT_LO2		(1 << 6)
+#define SHORT_LO1		(1 << 4)
+#define SHORT_HS2		(1 << 2)
+#define SHORT_HS1		(1 << 0)
+
+enum {
+	FILTER_BYPASS = 0,
+	FILTER_LOW_PASS_1,
+	FILTER_LOW_PASS_2,
+	FILTER_HIGH_PASS_3,
+	FILTER_MAX,
+};
+
+struct pm860x_hsdet {
+	struct snd_soc_jack	*jack;
+	int			det;
+};
+
+struct pm860x_priv {
+	unsigned int		sysclk;
+	unsigned int		pcmclk;
+	unsigned int		dir;
+	unsigned int		automute;
+	unsigned int		filter;
+	struct snd_soc_codec	*codec;
+	struct i2c_client	*i2c;
+	struct pm860x_chip	*chip;
+
+	int			irq[4];
+	unsigned char		name[4][MAX_NAME_LEN];
+	struct pm860x_hsdet	hsdet;
+	unsigned char		reg_cache[REG_CACHE_SIZE];
+};
+
+/* -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;
+}
+
+static int snd_soc_get_equalizer(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+
+	if (pm860x->filter >= FILTER_MAX)
+		return -EINVAL;
+
+	ucontrol->value.enumerated.item[0] = pm860x->filter;
+	return 0;
+}
+
+static int snd_soc_put_equalizer(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+
+	if (ucontrol->value.integer.value[0] >= FILTER_MAX)
+		return -EINVAL;
+	pm860x->filter = ucontrol->value.integer.value[0];
+	switch (pm860x->filter) {
+	case FILTER_BYPASS:
+		snd_soc_update_bits(codec, PM860X_I2S_IFACE_4, I2S_EQU_BYP,
+				    I2S_EQU_BYP);
+		snd_soc_write(codec, PM860X_EQUALIZER_N0_1, 0);
+		snd_soc_write(codec, PM860X_EQUALIZER_N0_2, 0);
+		snd_soc_write(codec, PM860X_EQUALIZER_N1_1, 0);
+		snd_soc_write(codec, PM860X_EQUALIZER_N1_2, 0);
+		snd_soc_write(codec, PM860X_EQUALIZER_D1_1, 0);
+		snd_soc_write(codec, PM860X_EQUALIZER_D1_2, 0);
+		snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, RSYNC_CHANGE,
+				    RSYNC_CHANGE);
+		return 0;
+	case FILTER_LOW_PASS_1:
+		snd_soc_write(codec, PM860X_EQUALIZER_N0_1, 0xf3);
+		snd_soc_write(codec, PM860X_EQUALIZER_N0_2, 0x3a);
+		snd_soc_write(codec, PM860X_EQUALIZER_N1_1, 0xc3);
+		snd_soc_write(codec, PM860X_EQUALIZER_N1_2, 0xf4);
+		snd_soc_write(codec, PM860X_EQUALIZER_D1_1, 0xb5);
+		snd_soc_write(codec, PM860X_EQUALIZER_D1_2, 0xaf);
+		break;
+	case FILTER_LOW_PASS_2:
+		snd_soc_write(codec, PM860X_EQUALIZER_N0_1, 0x36);
+		snd_soc_write(codec, PM860X_EQUALIZER_N0_2, 0x42);
+		snd_soc_write(codec, PM860X_EQUALIZER_N1_1, 0x3c);
+		snd_soc_write(codec, PM860X_EQUALIZER_N1_2, 0xfb);
+		snd_soc_write(codec, PM860X_EQUALIZER_D1_1, 0x73);
+		snd_soc_write(codec, PM860X_EQUALIZER_D1_2, 0xbd);
+		break;
+	case FILTER_HIGH_PASS_3:
+		snd_soc_write(codec, PM860X_EQUALIZER_N0_1, 0x55);
+		snd_soc_write(codec, PM860X_EQUALIZER_N0_2, 0x39);
+		snd_soc_write(codec, PM860X_EQUALIZER_N1_1, 0x5a);
+		snd_soc_write(codec, PM860X_EQUALIZER_N1_2, 0xf3);
+		snd_soc_write(codec, PM860X_EQUALIZER_D1_1, 0x7e);
+		snd_soc_write(codec, PM860X_EQUALIZER_D1_2, 0x53);
+		break;
+	}
+	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, RSYNC_CHANGE,
+			    RSYNC_CHANGE);
+	snd_soc_update_bits(codec, PM860X_I2S_IFACE_4, I2S_EQU_BYP, 0);
+	return 0;
+}
+
+/* DAPM Widget Events */
+static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
+			      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+
+	/* unmute DAC */
+	if (pm860x->automute) {
+		snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0);
+		pm860x->automute = 0;
+	}
+	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+			    RSYNC_CHANGE, RSYNC_CHANGE);
+	return 0;
+}
+
+static int pm860x_adc_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	unsigned int en1 = 0, en2 = 0;
+
+	if (!strcmp(w->name, "Left ADC")) {
+		en1 = ADC_MOD_LEFT;
+		en2 = ADC_LEFT;
+	}
+	if (!strcmp(w->name, "Right ADC")) {
+		en1 = ADC_MOD_RIGHT;
+		en2 = ADC_RIGHT;
+	}
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_update_bits(codec, PM860X_ADC_EN_1, en1, en1);
+		snd_soc_update_bits(codec, PM860X_ADC_EN_2, en2, en2);
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		snd_soc_update_bits(codec, PM860X_ADC_EN_1, en1, 0);
+		snd_soc_update_bits(codec, PM860X_ADC_EN_2, en2, 0);
+		break;
+	}
+	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;
+	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+	unsigned int data, dac = 0;
+
+	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) {
+			pm860x->automute = 1;
+			dac |= MODULATOR;
+			/* mute */
+			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) {
+			pm860x->automute = 1;
+			/* mute */
+			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 int pm860x_spk_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+
+	dev_dbg(codec->dev, "event:%x\n", event);
+	return 0;
+}
+
+static const char *pm860x_equalizer_texts[] = {"Filter Bypass",
+					       "Low Pass Filter1",
+					       "Low Pass Filter 2",
+					       "High Pass Filter 3"};
+
+static const struct soc_enum pm860x_equalizer_enum =
+	SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, ARRAY_SIZE(pm860x_equalizer_texts),
+			pm860x_equalizer_texts);
+;
+
+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_enum[] = {
+	SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts),
+	SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts),
+	SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts),
+	SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts),
+	SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts),
+	SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts),
+	SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts),
+	SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts),
+	SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts),
+	SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts),
+	SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_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 Capture 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_EXT("Equalizer", pm860x_equalizer_enum, snd_soc_get_equalizer,
+		     snd_soc_put_equalizer),
+	SOC_ENUM("Headset1 Operational Amplifier Current", pm860x_enum[0]),
+	SOC_ENUM("Headset2 Operational Amplifier Current", pm860x_enum[1]),
+	SOC_ENUM("Headset1 Amplifier Current", pm860x_enum[2]),
+	SOC_ENUM("Headset2 Amplifier Current", pm860x_enum[3]),
+	SOC_ENUM("Lineout1 Operational Amplifier Current", pm860x_enum[4]),
+	SOC_ENUM("Lineout2 Operational Amplifier Current", pm860x_enum[5]),
+	SOC_ENUM("Lineout1 Amplifier Current", pm860x_enum[6]),
+	SOC_ENUM("Lineout2 Amplifier Current", pm860x_enum[7]),
+	SOC_ENUM("Speaker Operational Amplifier Current", pm860x_enum[8]),
+	SOC_ENUM("Speaker Amplifier Current", pm860x_enum[9]),
+	SOC_ENUM("Earpiece Amplifier Current", pm860x_enum[10]),
+};
+
+/*
+ * 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, SND_SOC_NOPM, 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_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_ADC_E("Left ADC", NULL, SND_SOC_NOPM, 0, 0,
+			   pm860x_adc_event,
+			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_ADC_E("Right ADC", NULL, SND_SOC_NOPM, 0, 0,
+			   pm860x_adc_event,
+			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+
+	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_E("Speaker Earpiece Demux", SND_SOC_NOPM, 0, 0,
+			   &spk_demux, pm860x_spk_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	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_POST("RSYNC", pm860x_rsync_event),
+
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* 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"},
+
+	{"HS1", NULL, "Headset1 PGA"},
+	{"HS2", NULL, "Headset2 PGA"},
+	{"LINEOUT1", NULL, "Lineout1 PGA"},
+	{"LINEOUT2", NULL, "Lineout2 PGA"},
+	{"LSP", NULL, "Speaker PGA"},
+	{"LSN", NULL, "Speaker PGA"},
+	{"EARP", NULL, "Earpiece PGA"},
+	{"EARN", NULL, "Earpiece PGA"},
+};
+
+static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int data = 0;
+
+	if (mute)
+		data = DAC_MUTE;
+	snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, data);
+	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+			    RSYNC_CHANGE, RSYNC_CHANGE);
+	return 0;
+}
+
+static int set_dai_fmt(struct pm860x_priv *pm860x, unsigned int fmt,
+		       unsigned char *inf, unsigned char *mask)
+{
+	int ret = 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:
+	case SND_SOC_DAIFMT_CBM_CFS:
+		if (pm860x->dir == PM860X_CLK_DIR_OUT)
+			*inf |= PCM_INF2_MASTER;
+		else
+			ret = -EINVAL;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		if (pm860x->dir == PM860X_CLK_DIR_IN)
+			*inf &= ~PCM_INF2_MASTER;
+		else
+			ret = -EINVAL;
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+	if (ret)
+		return ret;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		*inf |= PCM_EXACT_I2S;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		*inf |= PCM_LEFT_I2S;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		*inf |= PCM_RIGHT_I2S;
+		break;
+	default:
+		return -EINVAL;
+	}
+	*mask |= PCM_MODE_MASK;
+	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;
+	int data;
+
+	/* 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;
+	data = snd_soc_read(codec, PM860X_PCM_IFACE_2) & ~mask;
+	data |= inf;
+	snd_soc_write(codec, PM860X_PCM_IFACE_2, data);
+
+	/* 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);
+
+	/* enable PCM interface */
+	snd_soc_update_bits(codec, PM860X_ADC_EN_2, 1, 1);
+	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+			    RSYNC_CHANGE, RSYNC_CHANGE);
+	return 0;
+}
+
+static int pm860x_pcm_hw_free(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	/* disable PCM interface */
+	snd_soc_update_bits(codec, PM860X_ADC_EN_2, 1, 0);
+	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+			    RSYNC_CHANGE, RSYNC_CHANGE);
+	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 data, ret;
+
+	ret = set_dai_fmt(pm860x, fmt, &inf, &mask);
+	if (!ret) {
+		data = snd_soc_read(codec, PM860X_PCM_IFACE_2) & ~mask;
+		data |= inf;
+		snd_soc_write(codec, PM860X_PCM_IFACE_2, data);
+	}
+	return ret;
+}
+
+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 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;
+	int data;
+
+	/*
+	 * Enable LDO15 for VDD_CODEC, audio charger pump for VDDSTP/VDDSTN
+	 * and reset audio module.
+	 */
+	data = LDO15_EN | CPUMP_EN | AUDIO_EN;
+	snd_soc_update_bits(codec, PM860X_AUDIO_SUPPLIES_2, data, data);
+
+	/* 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);
+
+	/* enable I2S interface */
+	snd_soc_update_bits(codec, PM860X_DAC_EN_2, 1, 1);
+	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+			    RSYNC_CHANGE, RSYNC_CHANGE);
+	return 0;
+}
+
+static int pm860x_i2s_hw_free(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	int data;
+
+	/* disable I2S interface */
+	snd_soc_update_bits(codec, PM860X_DAC_EN_2, 1, 0);
+	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+			    RSYNC_CHANGE, RSYNC_CHANGE);
+
+	/*
+	 * Disable LDO15 for VDD_CODEC, audio charger pump for VDDSTP/VDDSTN
+	 * and reset audio module.
+	 */
+	data = LDO15_EN | CPUMP_EN | AUDIO_EN;
+	snd_soc_update_bits(codec, PM860X_AUDIO_SUPPLIES_2, data, 0);
+	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;
+	int data, ret;
+
+	ret = set_dai_fmt(pm860x, fmt, &inf, &mask);
+	if (!ret) {
+		data = snd_soc_read(codec, PM860X_I2S_IFACE_2) & ~mask;
+		data |= inf;
+		snd_soc_write(codec, PM860X_I2S_IFACE_2, data);
+	}
+	return ret;
+}
+
+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->bias_level == SND_SOC_BIAS_OFF) {
+			/* Enable Audio PLL & Audio section */
+			data = AUDIO_PLL | AUDIO_SECTION_RESET
+				| AUDIO_SECTION_ON;
+			pm860x_reg_write(codec->control_data, REG_MISC2, data);
+
+			/* Enable Mic1 Bias & MICDET, HSDET */
+			pm860x_set_bits(codec->control_data, REG_ADC_ANA_1,
+					MIC1BIAS_MASK, MIC1BIAS_MASK);
+			pm860x_set_bits(codec->control_data, REG_MIC_DET,
+					MICDET_MASK, MICDET_MASK);
+			pm860x_set_bits(codec->control_data, REG_HS_DET,
+					EN_HS_DET, EN_HS_DET);
+		}
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		/* disable Mic1 Bias & MICDET, HSDET */
+		pm860x_set_bits(codec->control_data, REG_MIC_DET,
+				MICDET_MASK, 0);
+		pm860x_set_bits(codec->control_data, REG_HS_DET,
+				EN_HS_DET, 0);
+		pm860x_set_bits(codec->control_data, REG_ADC_ANA_1,
+				MIC1BIAS_MASK, 0);
+		data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON;
+		pm860x_set_bits(codec->control_data, REG_MISC2, data, 0);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+static struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
+	.digital_mute	= pm860x_digital_mute,
+	.hw_params	= pm860x_pcm_hw_params,
+	.hw_free	= pm860x_pcm_hw_free,
+	.set_fmt	= pm860x_pcm_set_dai_fmt,
+	.set_sysclk	= pm860x_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
+	.digital_mute	= pm860x_digital_mute,
+	.hw_params	= pm860x_i2s_hw_params,
+	.hw_free	= pm860x_i2s_hw_free,
+	.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)
+
+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,
+	},
+};
+EXPORT_SYMBOL_GPL(pm860x_dai);
+
+static irqreturn_t pm860x_codec_handler(int irq, void *data)
+{
+	struct pm860x_priv *pm860x = data;
+	int status, shrt, report = 0;
+
+	status = pm860x_reg_read(pm860x->i2c, REG_STATUS_1);
+	shrt = pm860x_reg_read(pm860x->i2c, REG_SHORTS);
+
+	if (pm860x->hsdet.det & PM860X_DET_HEADSET) {
+		if (status & HEADSET_STATUS)
+			report |= PM860X_DET_HEADSET;
+	}
+	if (pm860x->hsdet.det & PM860X_DET_MIC) {
+		if (status & MIC_STATUS)
+			report |= PM860X_DET_MIC;
+	}
+	if (pm860x->hsdet.det & PM860X_DET_HOOK) {
+		if (status & HOOK_STATUS)
+			report |= PM860X_DET_HOOK;
+	}
+	if (pm860x->hsdet.det & PM860X_SHORT_LINEOUT) {
+		if (shrt & (SHORT_LO1 | SHORT_LO2))
+			report |= PM860X_SHORT_LINEOUT;
+	}
+	if (pm860x->hsdet.det & PM860X_SHORT_HEADSET) {
+		if (shrt & (SHORT_HS1 | SHORT_HS2))
+			report |= PM860X_SHORT_HEADSET;
+	}
+	snd_soc_jack_report(pm860x->hsdet.jack, report, PM860X_DET_MASK);
+	dev_dbg(pm860x->codec->dev, "report:0x%x\n", report);
+	return IRQ_HANDLED;
+}
+
+/*
+ * Enable headset detection via 88PM860x IRQ
+ *
+ * Enable headset detection via IRQ on 88PM860x. If GPIOs are being used to
+ * bring out signals to the processor then processor GPIOs should be
+ * configured using snd_soc_jack_add_gpios() instead.
+ */
+int pm860x_hs_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
+		      int det)
+{
+	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+
+	if (!jack) {
+		dev_err(codec->dev, "Wrong jack is specified\n");
+		return -EINVAL;
+	}
+	pm860x->hsdet.jack = jack;
+	pm860x->hsdet.det = det;
+
+	pm860x_codec_handler(0, pm860x);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(pm860x_hs_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->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_irq;
+		}
+	}
+
+	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_codec;
+	}
+
+	snd_soc_add_controls(codec, pm860x_snd_controls,
+			     ARRAY_SIZE(pm860x_snd_controls));
+	snd_soc_dapm_new_controls(codec, pm860x_dapm_widgets,
+				  ARRAY_SIZE(pm860x_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	return 0;
+
+out_codec:
+	i = 3;
+out_irq:
+	for (; i >= 0; i--)
+		free_irq(pm860x->irq[i], pm860x);
+	return -EINVAL;
+}
+
+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),
+};
+
+static __init int pm860x_init(void)
+{
+	return platform_driver_register(&pm860x_codec_driver);
+}
+module_init(pm860x_init);
+
+static __exit void pm860x_exit(void)
+{
+	platform_driver_unregister(&pm860x_codec_driver);
+}
+module_exit(pm860x_exit);
+
+MODULE_DESCRIPTION("ASoC 88PM860x driver");
+MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:88pm860x-codec");
+
diff --git a/sound/soc/codecs/88pm860x-codec.h
b/sound/soc/codecs/88pm860x-codec.h
new file mode 100644
index 0000000..8094669
--- /dev/null
+++ b/sound/soc/codecs/88pm860x-codec.h
@@ -0,0 +1,97 @@
+/*
+ * 88pm860x-codec.h -- 88PM860x ALSA SoC Audio Driver
+ *
+ * Copyright 2010 Marvell International Ltd.
+ *	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.
+ */
+
+#ifndef __88PM860X_H
+#define __88PM860X_H
+
+/* The offset of these registers are 0xb0 */
+#define PM860X_PCM_IFACE_1		0x00
+#define PM860X_PCM_IFACE_2		0x01
+#define PM860X_PCM_IFACE_3		0x02
+#define PM860X_PCM_RATE			0x03
+#define PM860X_EC_PATH			0x04
+#define PM860X_SIDETONE_L_GAIN		0x05
+#define PM860X_SIDETONE_R_GAIN		0x06
+#define PM860X_SIDETONE_SHIFT		0x07
+#define PM860X_ADC_OFFSET_1		0x08
+#define PM860X_ADC_OFFSET_2		0x09
+#define PM860X_DMIC_DELAY		0x0a
+
+#define PM860X_I2S_IFACE_1		0x0b
+#define PM860X_I2S_IFACE_2		0x0c
+#define PM860X_I2S_IFACE_3		0x0d
+#define PM860X_I2S_IFACE_4		0x0e
+#define PM860X_EQUALIZER_N0_1		0x0f
+#define PM860X_EQUALIZER_N0_2		0x10
+#define PM860X_EQUALIZER_N1_1		0x11
+#define PM860X_EQUALIZER_N1_2		0x12
+#define PM860X_EQUALIZER_D1_1		0x13
+#define PM860X_EQUALIZER_D1_2		0x14
+#define PM860X_LOFI_GAIN_LEFT		0x15
+#define PM860X_LOFI_GAIN_RIGHT		0x16
+#define PM860X_HIFIL_GAIN_LEFT		0x17
+#define PM860X_HIFIL_GAIN_RIGHT		0x18
+#define PM860X_HIFIR_GAIN_LEFT		0x19
+#define PM860X_HIFIR_GAIN_RIGHT		0x1a
+#define PM860X_DAC_OFFSET		0x1b
+#define PM860X_OFFSET_LEFT_1		0x1c
+#define PM860X_OFFSET_LEFT_2		0x1d
+#define PM860X_OFFSET_RIGHT_1		0x1e
+#define PM860X_OFFSET_RIGHT_2		0x1f
+#define PM860X_ADC_ANA_1		0x20
+#define PM860X_ADC_ANA_2		0x21
+#define PM860X_ADC_ANA_3		0x22
+#define PM860X_ADC_ANA_4		0x23
+#define PM860X_ANA_TO_ANA		0x24
+#define PM860X_HS1_CTRL			0x25
+#define PM860X_HS2_CTRL			0x26
+#define PM860X_LO1_CTRL			0x27
+#define PM860X_LO2_CTRL			0x28
+#define PM860X_EAR_CTRL_1		0x29
+#define PM860X_EAR_CTRL_2		0x2a
+#define PM860X_AUDIO_SUPPLIES_1		0x2b
+#define PM860X_AUDIO_SUPPLIES_2		0x2c
+#define PM860X_ADC_EN_1			0x2d
+#define PM860X_ADC_EN_2			0x2e
+#define PM860X_DAC_EN_1			0x2f
+#define PM860X_DAC_EN_2			0x31
+#define PM860X_AUDIO_CAL_1		0x32
+#define PM860X_AUDIO_CAL_2		0x33
+#define PM860X_AUDIO_CAL_3		0x34
+#define PM860X_AUDIO_CAL_4		0x35
+#define PM860X_AUDIO_CAL_5		0x36
+#define PM860X_ANA_INPUT_SEL_1		0x37
+#define PM860X_ANA_INPUT_SEL_2		0x38
+
+#define PM860X_PCM_IFACE_4		0x39
+#define PM860X_I2S_IFACE_5		0x3a
+
+#define PM860X_SHORTS			0x3b
+#define PM860X_PLL_ADJ_1		0x3c
+#define PM860X_PLL_ADJ_2		0x3d
+
+/* bits definition */
+#define MUTE_ALL			(1 << 7)
+
+#define PM860X_CLK_DIR_IN		0
+#define PM860X_CLK_DIR_OUT		1
+
+#define PM860X_DET_HEADSET	(1 << 0)
+#define PM860X_DET_MIC		(1 << 1)
+#define PM860X_DET_HOOK		(1 << 2)
+#define PM860X_SHORT_HEADSET	(1 << 3)
+#define PM860X_SHORT_LINEOUT	(1 << 4)
+#define PM860X_DET_MASK		0x1F
+
+extern int pm860x_hs_detect(struct snd_soc_codec *, struct snd_soc_jack *,
+			    int);
+
+#endif	/* __88PM860X_H */
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index bfdd92b..a3cfc18 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI

 config SND_SOC_ALL_CODECS
 	tristate "Build all ASoC CODEC drivers"
+	select SND_SOC_88PM860X if MFD_88PM860X
 	select SND_SOC_L3
 	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
 	select SND_SOC_AD1836 if SPI_MASTER
@@ -85,6 +86,9 @@ config SND_SOC_ALL_CODECS

           If unsure select "N".

+config SND_SOC_88PM860X
+	tristate
+
 config SND_SOC_WM_HUBS
 	tristate
 	default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 9c3c39f..b9c4358 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,3 +1,4 @@
+snd-soc-88pm860x-objs := 88pm860x-codec.o
 snd-soc-ac97-objs := ac97.o
 snd-soc-ad1836-objs := ad1836.o
 snd-soc-ad193x-objs := ad193x.o
@@ -67,6 +68,7 @@ snd-soc-tpa6130a2-objs := tpa6130a2.o
 snd-soc-wm2000-objs := wm2000.o
 snd-soc-wm9090-objs := wm9090.o

+obj-$(CONFIG_SND_SOC_88PM860X)	+= snd-soc-88pm860x.o
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1836)	+= snd-soc-ad1836.o
 obj-$(CONFIG_SND_SOC_AD193X)	+= snd-soc-ad193x.o
-- 
1.5.6.5

[PATCH 4/5] ASoC: add 88pm860x codec driver

[Haojian Zhuang]

[PATCH 4/5] ASoC: add 88pm860x codec driver

[Mark Brown]

On Fri, Aug 13, 2010 at 09:55:44PM +0800, Haojian Zhuang wrote:

> +static int snd_soc_put_equalizer(struct snd_kcontrol *kcontrol,
> +				 struct snd_ctl_elem_value *ucontrol)
> +{
> +	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
> +	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
> +
> +	if (ucontrol->value.integer.value[0] >= FILTER_MAX)
> +		return -EINVAL;
> +	pm860x->filter = ucontrol->value.integer.value[0];
> +	switch (pm860x->filter) {
> +	case FILTER_BYPASS:
> +		snd_soc_update_bits(codec, PM860X_I2S_IFACE_4, I2S_EQU_BYP,
> +				    I2S_EQU_BYP);
> +		snd_soc_write(codec, PM860X_EQUALIZER_N0_1, 0);
> +		snd_soc_write(codec, PM860X_EQUALIZER_N0_2, 0);
> +		snd_soc_write(codec, PM860X_EQUALIZER_N1_1, 0);
> +		snd_soc_write(codec, PM860X_EQUALIZER_N1_2, 0);
> +		snd_soc_write(codec, PM860X_EQUALIZER_D1_1, 0);
> +		snd_soc_write(codec, PM860X_EQUALIZER_D1_2, 0);
> +		snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, RSYNC_CHANGE,
> +				    RSYNC_CHANGE);
> +		return 0;

Rather than hard coding every possible set of EQ configurations into the
driver it would be much better to allow the user to supply these via
platform data.  That way if users want to supply their own EQ
coefficients they will be able to.  Several Wolfson CODEC drivers have
examples of doing this.

It's fine to provide a default set of configurations in case the user
doesn't specify anything in platform data.

> +	case FILTER_HIGH_PASS_3:
> +		snd_soc_write(codec, PM860X_EQUALIZER_N0_1, 0x55);
> +		snd_soc_write(codec, PM860X_EQUALIZER_N0_2, 0x39);
> +		snd_soc_write(codec, PM860X_EQUALIZER_N1_1, 0x5a);
> +		snd_soc_write(codec, PM860X_EQUALIZER_N1_2, 0xf3);
> +		snd_soc_write(codec, PM860X_EQUALIZER_D1_1, 0x7e);
> +		snd_soc_write(codec, PM860X_EQUALIZER_D1_2, 0x53);
> +		break;
> +	}

You're also missing a default.

> +/* DAPM Widget Events */
> +static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
> +			      struct snd_kcontrol *kcontrol, int event)
> +{
> +	struct snd_soc_codec *codec = w->codec;
> +	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
> +
> +	/* unmute DAC */
> +	if (pm860x->automute) {
> +		snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0);
> +		pm860x->automute = 0;
> +	}
> +	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
> +			    RSYNC_CHANGE, RSYNC_CHANGE);
> +	return 0;
> +}

What's this doing?  There's also some similar code in the DAC widget
event - I think some comments explaining what's being done here are
required at the very least.

> +static int pm860x_adc_event(struct snd_soc_dapm_widget *w,
> +			    struct snd_kcontrol *kcontrol, int event)
> +{
> +	struct snd_soc_codec *codec = w->codec;
> +	unsigned int en1 = 0, en2 = 0;
> +
> +	if (!strcmp(w->name, "Left ADC")) {
> +		en1 = ADC_MOD_LEFT;
> +		en2 = ADC_LEFT;
> +	}
> +	if (!strcmp(w->name, "Right ADC")) {
> +		en1 = ADC_MOD_RIGHT;
> +		en2 = ADC_RIGHT;
> +	}
> +	switch (event) {
> +	case SND_SOC_DAPM_PRE_PMU:
> +		snd_soc_update_bits(codec, PM860X_ADC_EN_1, en1, en1);
> +		snd_soc_update_bits(codec, PM860X_ADC_EN_2, en2, en2);

Can you do this more simply by using a supply widget for the en1
register bits?

> +static int pm860x_spk_event(struct snd_soc_dapm_widget *w,
> +			    struct snd_kcontrol *kcontrol, int event)
> +{
> +	struct snd_soc_codec *codec = w->codec;
> +
> +	dev_dbg(codec->dev, "event:%x\n", event);
> +	return 0;
> +}

Remove this, it's purely for debug.

> +static const struct soc_enum pm860x_enum[] = {
> +	SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts),

Don't put your enums in an array, it is error prone and hard to read.
Just define variables for each enum.

> +	SOC_DOUBLE_R_EXT_TLV("Sidetone Capture 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),

The use of Capture in the name looks wrong - Sidetone paths usually go
to the output.  I'd just say Sidetone by itself.

> +/* Headset 1 Mux / Mux15 */
> +static const char *in_text[] = {
> +	"DIGITAL", "ANALOG",

Why ALL CAPS?

> +static int set_dai_fmt(struct pm860x_priv *pm860x, unsigned int fmt,
> +		       unsigned char *inf, unsigned char *mask)
> +{
> +	int ret = 0;

> +	default:
> +		ret = -EINVAL;
> +		break;
> +	}
> +	if (ret)
> +		return ret;

It'd be as easy to just return immediately rather than deferring the
return - probably more legible too.

> +	case SND_SOC_DAIFMT_I2S:
> +		*inf |= PCM_EXACT_I2S;
> +		break;
> +	case SND_SOC_DAIFMT_LEFT_J:
> +		*inf |= PCM_LEFT_I2S;
> +		break;
> +	case SND_SOC_DAIFMT_RIGHT_J:
> +		*inf |= PCM_RIGHT_I2S;
> +		break;

This is wrong - left and right justfied formats are not I2S.  Either the
device doesn't support them or there's some missing configuration.

> +	data = snd_soc_read(codec, PM860X_PCM_IFACE_2) & ~mask;
> +	data |= inf;
> +	snd_soc_write(codec, PM860X_PCM_IFACE_2, data);

snd_soc_update_bits().

> +static int pm860x_pcm_hw_free(struct snd_pcm_substream *substream,
> +			      struct snd_soc_dai *dai)
> +{
> +	struct snd_soc_codec *codec = dai->codec;
> +
> +	/* disable PCM interface */
> +	snd_soc_update_bits(codec, PM860X_ADC_EN_2, 1, 0);
> +	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
> +			    RSYNC_CHANGE, RSYNC_CHANGE);
> +	return 0;
> +}

This looks like it should be done by DAPM.

> +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;
> +	int data;
> +
> +	/*
> +	 * Enable LDO15 for VDD_CODEC, audio charger pump for VDDSTP/VDDSTN
> +	 * and reset audio module.
> +	 */
> +	data = LDO15_EN | CPUMP_EN | AUDIO_EN;
> +	snd_soc_update_bits(codec, PM860X_AUDIO_SUPPLIES_2, data, data);

These look like they should all be handled via DAPM, probably supply
widgets.

> +static int pm860x_i2s_hw_free(struct snd_pcm_substream *substream,
> +			      struct snd_soc_dai *dai)
> +{

This also looks like DAPM stuff.

> +			/* Enable Mic1 Bias & MICDET, HSDET */
> +			pm860x_set_bits(codec->control_data, REG_ADC_ANA_1,
> +					MIC1BIAS_MASK, MIC1BIAS_MASK);
> +			pm860x_set_bits(codec->control_data, REG_MIC_DET,
> +					MICDET_MASK, MICDET_MASK);
> +			pm860x_set_bits(codec->control_data, REG_HS_DET,
> +					EN_HS_DET, EN_HS_DET);

The microphone bias should be handled via DAPM.

> +EXPORT_SYMBOL_GPL(pm860x_dai);

No need to export in multi-component.

> +static irqreturn_t pm860x_codec_handler(int irq, void *data)
> +{
> +	struct pm860x_priv *pm860x = data;
> +	int status, shrt, report = 0;
> +
> +	status = pm860x_reg_read(pm860x->i2c, REG_STATUS_1);
> +	shrt = pm860x_reg_read(pm860x->i2c, REG_SHORTS);
> +
> +	if (pm860x->hsdet.det & PM860X_DET_HEADSET) {
> +		if (status & HEADSET_STATUS)
> +			report |= PM860X_DET_HEADSET;
> +	}
> +	if (pm860x->hsdet.det & PM860X_DET_MIC) {
> +		if (status & MIC_STATUS)
> +			report |= PM860X_DET_MIC;
> +	}
> +	if (pm860x->hsdet.det & PM860X_DET_HOOK) {
> +		if (status & HOOK_STATUS)
> +			report |= PM860X_DET_HOOK;
> +	}
> +	if (pm860x->hsdet.det & PM860X_SHORT_LINEOUT) {
> +		if (shrt & (SHORT_LO1 | SHORT_LO2))
> +			report |= PM860X_SHORT_LINEOUT;
> +	}
> +	if (pm860x->hsdet.det & PM860X_SHORT_HEADSET) {
> +		if (shrt & (SHORT_HS1 | SHORT_HS2))
> +			report |= PM860X_SHORT_HEADSET;
> +	}
> +	snd_soc_jack_report(pm860x->hsdet.jack, report, PM860X_DET_MASK);

This looks very wrong - you're injecting PM860X specific report codes
into the jack subsystem.  You should be reporting codes supplied by the
user, or standard codes at the very least.

> +/*
> + * Enable headset detection via 88PM860x IRQ
> + *
> + * Enable headset detection via IRQ on 88PM860x. If GPIOs are being used to
> + * bring out signals to the processor then processor GPIOs should be
> + * configured using snd_soc_jack_add_gpios() instead.

Is direct export of signals supported by this device, the comment looks
like cut'n'paste from one of my drivers.

> +/* bits definition */
> +#define MUTE_ALL			(1 << 7)

Namespacing.

[PATCH 4/5] ASoC: add 88pm860x codec driver

[Haojian Zhuang]

On Fri, Aug 13, 2010 at 10:23 PM, Mark Brown
<broonie@opensource.wolfsonmicro.com> wrote:
> On Fri, Aug 13, 2010 at 09:55:44PM +0800, Haojian Zhuang wrote:
>
>> +static int snd_soc_put_equalizer(struct snd_kcontrol *kcontrol,
>> + ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?struct snd_ctl_elem_value *ucontrol)
>> +{
>> + ? ? struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
>> + ? ? struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
>> +
>> + ? ? if (ucontrol->value.integer.value[0] >= FILTER_MAX)
>> + ? ? ? ? ? ? return -EINVAL;
>> + ? ? pm860x->filter = ucontrol->value.integer.value[0];
>> + ? ? switch (pm860x->filter) {
>> + ? ? case FILTER_BYPASS:
>> + ? ? ? ? ? ? snd_soc_update_bits(codec, PM860X_I2S_IFACE_4, I2S_EQU_BYP,
>> + ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? I2S_EQU_BYP);
>> + ? ? ? ? ? ? snd_soc_write(codec, PM860X_EQUALIZER_N0_1, 0);
>> + ? ? ? ? ? ? snd_soc_write(codec, PM860X_EQUALIZER_N0_2, 0);
>> + ? ? ? ? ? ? snd_soc_write(codec, PM860X_EQUALIZER_N1_1, 0);
>> + ? ? ? ? ? ? snd_soc_write(codec, PM860X_EQUALIZER_N1_2, 0);
>> + ? ? ? ? ? ? snd_soc_write(codec, PM860X_EQUALIZER_D1_1, 0);
>> + ? ? ? ? ? ? snd_soc_write(codec, PM860X_EQUALIZER_D1_2, 0);
>> + ? ? ? ? ? ? snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, RSYNC_CHANGE,
>> + ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? RSYNC_CHANGE);
>> + ? ? ? ? ? ? return 0;
>
> Rather than hard coding every possible set of EQ configurations into the
> driver it would be much better to allow the user to supply these via
> platform data. ?That way if users want to supply their own EQ
> coefficients they will be able to. ?Several Wolfson CODEC drivers have
> examples of doing this.
>
> It's fine to provide a default set of configurations in case the user
> doesn't specify anything in platform data.
>
OK.

>> + ? ? case FILTER_HIGH_PASS_3:
>> + ? ? ? ? ? ? snd_soc_write(codec, PM860X_EQUALIZER_N0_1, 0x55);
>> + ? ? ? ? ? ? snd_soc_write(codec, PM860X_EQUALIZER_N0_2, 0x39);
>> + ? ? ? ? ? ? snd_soc_write(codec, PM860X_EQUALIZER_N1_1, 0x5a);
>> + ? ? ? ? ? ? snd_soc_write(codec, PM860X_EQUALIZER_N1_2, 0xf3);
>> + ? ? ? ? ? ? snd_soc_write(codec, PM860X_EQUALIZER_D1_1, 0x7e);
>> + ? ? ? ? ? ? snd_soc_write(codec, PM860X_EQUALIZER_D1_2, 0x53);
>> + ? ? ? ? ? ? break;
>> + ? ? }
>
> You're also missing a default.
>
Bypass is default.  After audio part is reset, EQ is in bypass state.

>> +/* DAPM Widget Events */
>> +static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
>> + ? ? ? ? ? ? ? ? ? ? ? ? ? struct snd_kcontrol *kcontrol, int event)
>> +{
>> + ? ? struct snd_soc_codec *codec = w->codec;
>> + ? ? struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
>> +
>> + ? ? /* unmute DAC */
>> + ? ? if (pm860x->automute) {
>> + ? ? ? ? ? ? snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0);
>> + ? ? ? ? ? ? pm860x->automute = 0;
>> + ? ? }
>> + ? ? snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
>> + ? ? ? ? ? ? ? ? ? ? ? ? RSYNC_CHANGE, RSYNC_CHANGE);
>> + ? ? return 0;
>> +}
>
> What's this doing? ?There's also some similar code in the DAC widget
> event - I think some comments explaining what's being done here are
> required at the very least.
>
A lot of registers belong to RSYNC domain. It means that those
changing won't be valid until RSYNC bit is set. So I'll set RSYNC
again in a lot of areas.

>> +static int pm860x_adc_event(struct snd_soc_dapm_widget *w,
>> + ? ? ? ? ? ? ? ? ? ? ? ? struct snd_kcontrol *kcontrol, int event)
>> +{
>> + ? ? struct snd_soc_codec *codec = w->codec;
>> + ? ? unsigned int en1 = 0, en2 = 0;
>> +
>> + ? ? if (!strcmp(w->name, "Left ADC")) {
>> + ? ? ? ? ? ? en1 = ADC_MOD_LEFT;
>> + ? ? ? ? ? ? en2 = ADC_LEFT;
>> + ? ? }
>> + ? ? if (!strcmp(w->name, "Right ADC")) {
>> + ? ? ? ? ? ? en1 = ADC_MOD_RIGHT;
>> + ? ? ? ? ? ? en2 = ADC_RIGHT;
>> + ? ? }
>> + ? ? switch (event) {
>> + ? ? case SND_SOC_DAPM_PRE_PMU:
>> + ? ? ? ? ? ? snd_soc_update_bits(codec, PM860X_ADC_EN_1, en1, en1);
>> + ? ? ? ? ? ? snd_soc_update_bits(codec, PM860X_ADC_EN_2, en2, en2);
>
> Can you do this more simply by using a supply widget for the en1
> register bits?
>
When I enable this, I need to touch two different registers at the
same time. So I have to implement the adc_event().

>> +static int pm860x_spk_event(struct snd_soc_dapm_widget *w,
>> + ? ? ? ? ? ? ? ? ? ? ? ? struct snd_kcontrol *kcontrol, int event)
>> +{
>> + ? ? struct snd_soc_codec *codec = w->codec;
>> +
>> + ? ? dev_dbg(codec->dev, "event:%x\n", event);
>> + ? ? return 0;
>> +}
>
> Remove this, it's purely for debug.
>
OK.

>> +static const struct soc_enum pm860x_enum[] = {
>> + ? ? SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts),
>
> Don't put your enums in an array, it is error prone and hard to read.
> Just define variables for each enum.
>
OK.


>> +/* Headset 1 Mux / Mux15 */
>> +static const char *in_text[] = {
>> + ? ? "DIGITAL", "ANALOG",
>
> Why ALL CAPS?
>
Sure. I'll change to lowcases.

>> + ? ? data = snd_soc_read(codec, PM860X_PCM_IFACE_2) & ~mask;
>> + ? ? data |= inf;
>> + ? ? snd_soc_write(codec, PM860X_PCM_IFACE_2, data);
>
> snd_soc_update_bits().
>
>> +static int pm860x_pcm_hw_free(struct snd_pcm_substream *substream,
>> + ? ? ? ? ? ? ? ? ? ? ? ? ? struct snd_soc_dai *dai)
>> +{
>> + ? ? struct snd_soc_codec *codec = dai->codec;
>> +
>> + ? ? /* disable PCM interface */
>> + ? ? snd_soc_update_bits(codec, PM860X_ADC_EN_2, 1, 0);
>> + ? ? snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
>> + ? ? ? ? ? ? ? ? ? ? ? ? RSYNC_CHANGE, RSYNC_CHANGE);
>> + ? ? return 0;
>> +}
>
> This looks like it should be done by DAPM.
>
But I don't want to export this to amixer.

>> +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;
>> + ? ? int data;
>> +
>> + ? ? /*
>> + ? ? ?* Enable LDO15 for VDD_CODEC, audio charger pump for VDDSTP/VDDSTN
>> + ? ? ?* and reset audio module.
>> + ? ? ?*/
>> + ? ? data = LDO15_EN | CPUMP_EN | AUDIO_EN;
>> + ? ? snd_soc_update_bits(codec, PM860X_AUDIO_SUPPLIES_2, data, data);
>
> These look like they should all be handled via DAPM, probably supply
> widgets.
>
>> +static int pm860x_i2s_hw_free(struct snd_pcm_substream *substream,
>> + ? ? ? ? ? ? ? ? ? ? ? ? ? struct snd_soc_dai *dai)
>> +{
>
> This also looks like DAPM stuff.
>
>> + ? ? ? ? ? ? ? ? ? ? /* Enable Mic1 Bias & MICDET, HSDET */
>> + ? ? ? ? ? ? ? ? ? ? pm860x_set_bits(codec->control_data, REG_ADC_ANA_1,
>> + ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? MIC1BIAS_MASK, MIC1BIAS_MASK);
>> + ? ? ? ? ? ? ? ? ? ? pm860x_set_bits(codec->control_data, REG_MIC_DET,
>> + ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? MICDET_MASK, MICDET_MASK);
>> + ? ? ? ? ? ? ? ? ? ? pm860x_set_bits(codec->control_data, REG_HS_DET,
>> + ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? EN_HS_DET, EN_HS_DET);
>
> The microphone bias should be handled via DAPM.
>
This one is different. Without this, mic/headset detection can't be
finished. This bit have to be set in initialization.

[PATCH 4/5] ASoC: add 88pm860x codec driver

[Mark Brown]

On Fri, Aug 13, 2010 at 10:53:52PM +0800, Haojian Zhuang wrote:
> On Fri, Aug 13, 2010 at 10:23 PM, Mark Brown

> >> + ? ? case FILTER_HIGH_PASS_3:
> >> + ? ? ? ? ? ? break;
> >> + ? ? }

> > You're also missing a default.

> Bypass is default.  After audio part is reset, EQ is in bypass state.

That's not my point - you're missing a default: in the case statement in
case someone passes in an unsupported value.

> >> +static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
> >> + ? ? ? ? ? ? ? ? ? ? ? ? ? struct snd_kcontrol *kcontrol, int event)
> >> +{
> >> + ? ? struct snd_soc_codec *codec = w->codec;
> >> + ? ? struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
> >> +
> >> + ? ? /* unmute DAC */
> >> + ? ? if (pm860x->automute) {
> >> + ? ? ? ? ? ? snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0);
> >> + ? ? ? ? ? ? pm860x->automute = 0;
> >> + ? ? }
> >> + ? ? snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
> >> + ? ? ? ? ? ? ? ? ? ? ? ? RSYNC_CHANGE, RSYNC_CHANGE);
> >> + ? ? return 0;
> >> +}

> > What's this doing? ?There's also some similar code in the DAC widget
> > event - I think some comments explaining what's being done here are
> > required at the very least.

> A lot of registers belong to RSYNC domain. It means that those
> changing won't be valid until RSYNC bit is set. So I'll set RSYNC
> again in a lot of areas.

That explains the RSYNC bit but not this automute variable - what's that
about?

> >> + ? ? case SND_SOC_DAPM_PRE_PMU:
> >> + ? ? ? ? ? ? snd_soc_update_bits(codec, PM860X_ADC_EN_1, en1, en1);
> >> + ? ? ? ? ? ? snd_soc_update_bits(codec, PM860X_ADC_EN_2, en2, en2);

> > Can you do this more simply by using a supply widget for the en1
> > register bits?

> When I enable this, I need to touch two different registers at the
> same time. So I have to implement the adc_event().

You presumably don't need to touch them at *exactly* the same time since
they're separate registers so there's always going to be some latency.
What is the actual constraint here, and why can't a supply widget handle
it?

> >> +/* Headset 1 Mux / Mux15 */
> >> +static const char *in_text[] = {
> >> + ? ? "DIGITAL", "ANALOG",

> > Why ALL CAPS?

> Sure. I'll change to lowcases.

Probably better mixed case (eg, "Digital") for UI use.

> >> +static int pm860x_pcm_hw_free(struct snd_pcm_substream *substream,
> >> + ? ? ? ? ? ? ? ? ? ? ? ? ? struct snd_soc_dai *dai)
> >> +{
> >> + ? ? struct snd_soc_codec *codec = dai->codec;
> >> +
> >> + ? ? /* disable PCM interface */
> >> + ? ? snd_soc_update_bits(codec, PM860X_ADC_EN_2, 1, 0);
> >> + ? ? snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
> >> + ? ? ? ? ? ? ? ? ? ? ? ? RSYNC_CHANGE, RSYNC_CHANGE);
> >> + ? ? return 0;
> >> +}

> > This looks like it should be done by DAPM.

> But I don't want to export this to amixer.

So don't.  DAPM only exports things to the application layer if there is
actual control.  Pure power control is invisible to userspace.

> >> + ? ? ? ? ? ? ? ? ? ? /* Enable Mic1 Bias & MICDET, HSDET */
> >> + ? ? ? ? ? ? ? ? ? ? pm860x_set_bits(codec->control_data, REG_ADC_ANA_1,
> >> + ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? MIC1BIAS_MASK, MIC1BIAS_MASK);
> >> + ? ? ? ? ? ? ? ? ? ? pm860x_set_bits(codec->control_data, REG_MIC_DET,
> >> + ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? MICDET_MASK, MICDET_MASK);
> >> + ? ? ? ? ? ? ? ? ? ? pm860x_set_bits(codec->control_data, REG_HS_DET,
> >> + ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? EN_HS_DET, EN_HS_DET);

> > The microphone bias should be handled via DAPM.

> This one is different. Without this, mic/headset detection can't be
> finished. This bit have to be set in initialization.

Why do they have to be set during initialisation?  I'd expect machine
drivers that use microphone detection to for enable the micbias with
snd_soc_dapm_force_enable_pin() and for the detection to only be enabled
when detect is in use (so one of the API calls setting up a jack gets
called).  Not every system is going to want this feature (they may have
no microphone or non-removable microphones) but with the code above
every system is going to have to leave both bias and detection enabled
at all times.