/*
* 88pm860x-codec.c -- 88PM860x ALSA SoC Audio Driver
*
* Copyright 2010 Marvell International Ltd.
* Author: Haojian Zhuang <haojian.zhuang@marvell.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/kernel.h>
#include <linux/module.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/mfd/88pm860x.h>
#include <linux/slab.h>
#include <linux/delay.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#include <sound/initval.h>
#include <sound/jack.h>
#include <trace/events/asoc.h>
#include "88pm860x-codec.h"
#define MAX_NAME_LEN 20
#define REG_CACHE_SIZE 0x40
#define REG_CACHE_BASE 0xb0
/* Status Register 1 (0x01) */
#define REG_STATUS_1 0x01
#define MIC_STATUS (1 << 7)
#define HOOK_STATUS (1 << 6)
#define HEADSET_STATUS (1 << 5)
/* Mic Detection Register (0x37) */
#define REG_MIC_DET 0x37
#define CONTINUOUS_POLLING (3 << 1)
#define EN_MIC_DET (1 << 0)
#define MICDET_MASK 0x07
/* Headset Detection Register (0x38) */
#define REG_HS_DET 0x38
#define EN_HS_DET (1 << 0)
/* Misc2 Register (0x42) */
#define REG_MISC2 0x42
#define AUDIO_PLL (1 << 5)
#define AUDIO_SECTION_RESET (1 << 4)
#define AUDIO_SECTION_ON (1 << 3)
/* PCM Interface Register 2 (0xb1) */
#define PCM_INF2_BCLK (1 << 6) /* Bit clock polarity */
#define PCM_INF2_FS (1 << 5) /* Frame Sync polarity */
#define PCM_INF2_MASTER (1 << 4) /* Master / Slave */
#define PCM_INF2_18WL (1 << 3) /* 18 / 16 bits */
#define PCM_GENERAL_I2S 0
#define PCM_EXACT_I2S 1
#define PCM_LEFT_I2S 2
#define PCM_RIGHT_I2S 3
#define PCM_SHORT_FS 4
#define PCM_LONG_FS 5
#define PCM_MODE_MASK 7
/* I2S Interface Register 4 (0xbe) */
#define I2S_EQU_BYP (1 << 6)
/* DAC Offset Register (0xcb) */
#define DAC_MUTE (1 << 7)
#define MUTE_LEFT (1 << 6)
#define MUTE_RIGHT (1 << 2)
/* ADC Analog Register 1 (0xd0) */
#define REG_ADC_ANA_1 0xd0
#define MIC1BIAS_MASK 0x60
/* Earpiece/Speaker Control Register 2 (0xda) */
#define REG_EAR2 0xda
#define RSYNC_CHANGE (1 << 2)
/* Audio Supplies Register 2 (0xdc) */
#define REG_SUPPLIES2 0xdc
#define LDO15_READY (1 << 4)
#define LDO15_EN (1 << 3)
#define CPUMP_READY (1 << 2)
#define CPUMP_EN (1 << 1)
#define AUDIO_EN (1 << 0)
#define SUPPLY_MASK (LDO15_EN | CPUMP_EN | AUDIO_EN)
/* Audio Enable Register 1 (0xdd) */
#define ADC_MOD_RIGHT (1 << 1)
#define ADC_MOD_LEFT (1 << 0)
/* Audio Enable Register 2 (0xde) */
#define ADC_LEFT (1 << 5)
#define ADC_RIGHT (1 << 4)
/* DAC Enable Register 2 (0xe1) */
#define DAC_LEFT (1 << 5)
#define DAC_RIGHT (1 << 4)
#define MODULATOR (1 << 3)
/* Shorts Register (0xeb) */
#define REG_SHORTS 0xeb
#define CLR_SHORT_LO2 (1 << 7)
#define SHORT_LO2 (1 << 6)
#define CLR_SHORT_LO1 (1 << 5)
#define SHORT_LO1 (1 << 4)
#define CLR_SHORT_HS2 (1 << 3)
#define SHORT_HS2 (1 << 2)
#define CLR_SHORT_HS1 (1 << 1)
#define SHORT_HS1 (1 << 0)
/*
* This widget should be just after DAC & PGA in DAPM power-on sequence and
* before DAC & PGA in DAPM power-off sequence.
*/
#define PM860X_DAPM_OUTPUT(wname, wevent) \
{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \
.shift = 0, .invert = 0, .kcontrol_news = NULL, \
.num_kcontrols = 0, .event = wevent, \
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD, }
struct pm860x_det {
struct snd_soc_jack *hp_jack;
struct snd_soc_jack *mic_jack;
int hp_det;
int mic_det;
int hook_det;
int hs_shrt;
int lo_shrt;
};
struct pm860x_priv {
unsigned int sysclk;
unsigned int pcmclk;
unsigned int dir;
unsigned int filter;
struct snd_soc_codec *codec;
struct i2c_client *i2c;
struct pm860x_chip *chip;
struct pm860x_det det;
int irq[4];
unsigned char name[4][MAX_NAME_LEN];
};
/* -9450dB to 0dB in 150dB steps ( mute instead of -9450dB) */
static const DECLARE_TLV_DB_SCALE(dpga_tlv, -9450, 150, 1);
/* -9dB to 0db in 3dB steps */
static const DECLARE_TLV_DB_SCALE(adc_tlv, -900, 300, 0);
/* {-23, -17, -13.5, -11, -9, -6, -3, 0}dB */
static const unsigned int mic_tlv[] = {
TLV_DB_RANGE_HEAD(5),
0, 0, TLV_DB_SCALE_ITEM(-2300, 0, 0),
1, 1, TLV_DB_SCALE_ITEM(-1700, 0, 0),
2, 2, TLV_DB_SCALE_ITEM(-1350, 0, 0),
3, 3, TLV_DB_SCALE_ITEM(-1100, 0, 0),
4, 7, TLV_DB_SCALE_ITEM(-900, 300, 0),
};
/* {0, 0, 0, -6, 0, 6, 12, 18}dB */
static const unsigned int aux_tlv[] = {
TLV_DB_RANGE_HEAD(2),
0, 2, TLV_DB_SCALE_ITEM(0, 0, 0),
3, 7, TLV_DB_SCALE_ITEM(-600, 600, 0),
};
/* {-16, -13, -10, -7, -5.2, -3,3, -2.2, 0}dB, mute instead of -16dB */
static const unsigned int out_tlv[] = {
TLV_DB_RANGE_HEAD(4),
0, 3, TLV_DB_SCALE_ITEM(-1600, 300, 1),
4, 4, TLV_DB_SCALE_ITEM(-520, 0, 0),
5, 5, TLV_DB_SCALE_ITEM(-330, 0, 0),
6, 7, TLV_DB_SCALE_ITEM(-220, 220, 0),
};
static const unsigned int st_tlv[] = {
TLV_DB_RANGE_HEAD(8),
0, 1, TLV_DB_SCALE_ITEM(-12041, 602, 0),
2, 3, TLV_DB_SCALE_ITEM(-11087, 250, 0),
4, 5, TLV_DB_SCALE_ITEM(-10643, 158, 0),
6, 7, TLV_DB_SCALE_ITEM(-10351, 116, 0),
8, 9, TLV_DB_SCALE_ITEM(-10133, 92, 0),
10, 13, TLV_DB_SCALE_ITEM(-9958, 70, 0),
14, 17, TLV_DB_SCALE_ITEM(-9689, 53, 0),
18, 271, TLV_DB_SCALE_ITEM(-9484, 37, 0),
};
/* Sidetone Gain = M * 2^(-5-N) */
struct st_gain {
unsigned int db;
unsigned int m;
unsigned int n;
};
static struct st_gain st_table[] = {
{-12041, 1, 15}, {-11439, 1, 14}, {-11087, 3, 15}, {-10837, 1, 13},
{-10643, 5, 15}, {-10485, 3, 14}, {-10351, 7, 15}, {-10235, 1, 12},
{-10133, 9, 15}, {-10041, 5, 14}, { -9958, 11, 15}, { -9883, 3, 13},
{ -9813, 13, 15}, { -9749, 7, 14}, { -9689, 15, 15}, { -9633, 1, 11},
{ -9580, 17, 15}, { -9531, 9, 14}, { -9484, 19, 15}, { -9439, 5, 13},
{ -9397, 21, 15}, { -9356, 11, 14}, { -9318, 23, 15}, { -9281, 3, 12},
{ -9245, 25, 15}, { -9211, 13, 14}, { -9178, 27, 15}, { -9147, 7, 13},
{ -9116, 29, 15}, { -9087, 15, 14}, { -9058, 31, 15}, { -9031, 1, 10},
{ -8978, 17, 14}, { -8929, 9, 13}, { -8882, 19, 14}, { -8837, 5, 12},
{ -8795, 21, 14}, { -8754, 11, 13}, { -8716, 23, 14}, { -8679, 3, 11},
{ -8643, 25, 14}, { -8609, 13, 13}, { -8576, 27, 14}, { -8545, 7, 12},
{ -8514, 29, 14}, { -8485, 15, 13}, { -8456, 31, 14}, { -8429, 1, 9},
{ -8376, 17, 13}, { -8327, 9, 12}, { -8280, 19, 13}, { -8235, 5, 11},
{ -8193, 21, 13}, { -8152, 11, 12}, { -8114, 23, 13}, { -8077, 3, 10},
{ -8041, 25, 13}, { -8007, 13, 12}, { -7974, 27, 13}, { -7943, 7, 11},
{ -7912, 29, 13}, { -7883, 15, 12}, { -7854, 31, 13}, { -7827, 1, 8},
{ -7774, 17, 12}, { -7724, 9, 11}, { -7678, 19, 12}, { -7633, 5, 10},
{ -7591, 21, 12}, { -7550, 11, 11}, { -7512, 23, 12}, { -7475, 3, 9},
{ -7439, 25, 12}, { -7405, 13, 11}, { -7372, 27, 12}, { -7341, 7, 10},
{ -7310, 29, 12}, { -7281, 15, 11}, { -7252, 31, 12}, { -7225, 1, 7},
{ -7172, 17, 11}, { -7122, 9, 10}, { -7075, 19, 11}, { -7031, 5, 9},
{ -6989, 21, 11}, { -6948, 11, 10}, { -6910, 23, 11}, { -6873, 3, 8},
{ -6837, 25, 11}, { -6803, 13, 10}, { -6770, 27, 11}, { -6739, 7, 9},
{ -6708, 29, 11}, { -6679, 15, 10}, { -6650, 31, 11}, { -6623, 1, 6},
{ -6570, 17, 10}, { -6520, 9, 9}, { -6473, 19, 10}, { -6429, 5, 8},
{ -6386, 21, 10}, { -6346, 11, 9}, { -6307, 23, 10}, { -6270, 3, 7},
{ -6235, 25, 10}, { -6201, 13, 9}, { -6168, 27, 10}, { -6137, 7, 8},
{ -6106, 29, 10}, { -6077, 15, 9}, { -6048, 31, 10}, { -6021, 1, 5},
{ -5968, 17, 9}, { -5918, 9, 8}, { -5871, 19, 9}, { -5827, 5, 7},
{ -5784, 21, 9}, { -5744, 11, 8}, { -5705, 23, 9}, { -5668, 3, 6},
{ -5633, 25, 9}, { -5599, 13, 8}, { -5566, 27, 9}, { -5535, 7, 7},
{ -5504, 29, 9}, { -5475, 15, 8}, { -5446, 31, 9}, { -5419, 1, 4},
{ -5366, 17, 8}, { -5316, 9, 7}, { -5269, 19, 8}, { -5225, 5, 6},
{ -5182, 21, 8}, { -5142, 11, 7}, { -5103, 23, 8}, { -5066, 3, 5},
{ -5031, 25, 8}, { -4997, 13, 7}, { -4964, 27, 8}, { -4932, 7, 6},
{ -4902, 29, 8}, { -4873, 15, 7}, { -4844, 31, 8}, { -4816, 1, 3},
{ -4764, 17, 7}, { -4714, 9, 6}, { -4667, 19, 7}, { -4623, 5, 5},
{ -4580, 21, 7}, { -4540, 11, 6}, { -4501, 23, 7}, { -4464, 3, 4},
{ -4429, 25, 7}, { -4395, 13, 6}, { -4362, 27, 7}, { -4330, 7, 5},
{ -4300, 29, 7}, { -4270, 15, 6}, { -4242, 31, 7}, { -4214, 1, 2},
{ -4162, 17, 6}, { -4112, 9, 5}, { -4065, 19, 6}, { -4021, 5, 4},
{ -3978, 21, 6}, { -3938, 11, 5}, { -3899, 23, 6}, { -3862, 3, 3},
{ -3827, 25, 6}, { -3793, 13, 5}, { -3760, 27, 6}, { -3728, 7, 4},
{ -3698, 29, 6}, { -3668, 15, 5}, { -3640, 31, 6}, { -3612, 1, 1},
{ -3560, 17, 5}, { -3510, 9, 4}, { -3463, 19, 5}, { -3419, 5, 3},
{ -3376, 21, 5}, { -3336, 11, 4}, { -3297, 23, 5}, { -3260, 3, 2},
{ -3225, 25, 5}, { -3191, 13, 4}, { -3158, 27, 5}, { -3126, 7, 3},
{ -3096, 29, 5}, { -3066, 15, 4}, { -3038, 31, 5}, { -3010, 1, 0},
{ -2958, 17, 4}, { -2908, 9, 3}, { -2861, 19, 4}, { -2816, 5, 2},
{ -2774, 21, 4}, { -2734, 11, 3}, { -2695, 23, 4}, { -2658, 3, 1},
{ -2623, 25, 4}, { -2589, 13, 3}, { -2556, 27, 4}, { -2524, 7, 2},
{ -2494, 29, 4}, { -2464, 15, 3}, { -2436, 31, 4}, { -2408, 2, 0},
{ -2356, 17, 3}, { -2306, 9, 2}, { -2259, 19, 3}, { -2214, 5, 1},
{ -2172, 21, 3}, { -2132, 11, 2}, { -2093, 23, 3}, { -2056, 3, 0},
{ -2021, 25, 3}, { -1987, 13, 2}, { -1954, 27, 3}, { -1922, 7, 1},
{ -1892, 29, 3}, { -1862, 15, 2}, { -1834, 31, 3}, { -1806, 4, 0},
{ -1754, 17, 2}, { -1704, 9, 1}, { -1657, 19, 2}, { -1612, 5, 0},
{ -1570, 21, 2}, { -1530, 11, 1}, { -1491, 23, 2}, { -1454, 6, 0},
{ -1419, 25, 2}, { -1384, 13, 1}, { -1352, 27, 2}, { -1320, 7, 0},
{ -1290, 29, 2}, { -1260, 15, 1}, { -1232, 31, 2}, { -1204, 8, 0},
{ -1151, 17, 1}, { -1102, 9, 0}, { -1055, 19, 1}, { -1010, 10, 0},
{ -968, 21, 1}, { -928, 11, 0}, { -889, 23, 1}, { -852, 12, 0},
{ -816, 25, 1}, { -782, 13, 0}, { -750, 27, 1}, { -718, 14, 0},
{ -688, 29, 1}, { -658, 15, 0}, { -630, 31, 1}, { -602, 16, 0},
{ -549, 17, 0}, { -500, 18, 0}, { -453, 19, 0}, { -408, 20, 0},
{ -366, 21, 0}, { -325, 22, 0}, { -287, 23, 0}, { -250, 24, 0},
{ -214, 25, 0}, { -180, 26, 0}, { -148, 27, 0}, { -116, 28, 0},
{ -86, 29, 0}, { -56, 30, 0}, { -28, 31, 0}, { 0, 0, 0},
};
static int pm860x_volatile(unsigned int reg)
{
BUG_ON(reg >= REG_CACHE_SIZE);
switch (reg) {
case PM860X_AUDIO_SUPPLIES_2:
return 1;
}
return 0;
}
static unsigned int pm860x_read_reg_cache(struct snd_soc_codec *codec,
unsigned int reg)
{
unsigned char *cache = codec->reg_cache;
BUG_ON(reg >= REG_CACHE_SIZE);
if (pm860x_volatile(reg))
return cache[reg];
reg += REG_CACHE_BASE;
return pm860x_reg_read(codec->control_data, reg);
}
static int pm860x_write_reg_cache(struct snd_soc_codec *codec,
unsigned int reg, unsigned int value)
{
unsigned char *cache = codec->reg_cache;
BUG_ON(reg >= REG_CACHE_SIZE);
if (!pm860x_volatile(reg))
cache[reg] = (unsigned char)value;
reg += REG_CACHE_BASE;
return pm860x_reg_write(codec->control_data, reg, value);
}
static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
int val[2], val2[2], i;
val[0] = snd_soc_read(codec, reg) & 0x3f;
val[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT) >> 4) & 0xf;
val2[0] = snd_soc_read(codec, reg2) & 0x3f;
val2[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT)) & 0xf;
for (i = 0; i < ARRAY_SIZE(st_table); i++) {
if ((st_table[i].m == val[0]) && (st_table[i].n == val[1]))
ucontrol->value.integer.value[0] = i;
if ((st_table[i].m == val2[0]) && (st_table[i].n == val2[1]))
ucontrol->value.integer.value[1] = i;
}
return 0;
}
static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
int err;
unsigned int val, val2;
val = ucontrol->value.integer.value[0];
val2 = ucontrol->value.integer.value[1];
err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m);
if (err < 0)
return err;
err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0xf0,
st_table[val].n << 4);
if (err < 0)
return err;
err = snd_soc_update_bits(codec, reg2, 0x3f, st_table[val2].m);
if (err < 0)
return err;
err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0x0f,
st_table[val2].n);
return err;
}
static int snd_soc_get_volsw_2r_out(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
unsigned int shift = mc->shift;
int max = mc->max, val, val2;
unsigned int mask = (1 << fls(max)) - 1;
val = snd_soc_read(codec, reg) >> shift;
val2 = snd_soc_read(codec, reg2) >> shift;
ucontrol->value.integer.value[0] = (max - val) & mask;
ucontrol->value.integer.value[1] = (max - val2) & mask;
return 0;
}
static int snd_soc_put_volsw_2r_out(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
unsigned int shift = mc->shift;
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
int err;
unsigned int val, val2, val_mask;
val_mask = mask << shift;
val = ((max - ucontrol->value.integer.value[0]) & mask);
val2 = ((max - ucontrol->value.integer.value[1]) & mask);
val = val << shift;
val2 = val2 << shift;
err = snd_soc_update_bits(codec, reg, val_mask, val);
if (err < 0)
return err;
err = snd_soc_update_bits(codec, reg2, val_mask, val2);
return err;
}
/* DAPM Widget Events */
/*
* A lot registers are belong to RSYNC domain. It requires enabling RSYNC bit
* after updating these registers. Otherwise, these updated registers won't
* be effective.
*/
static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
/*
* In order to avoid current on the load, mute power-on and power-off
* should be transients.
* Unmute by DAC_MUTE. It should be unmuted when DAPM sequence is
* finished.
*/
snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0);
snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
RSYNC_CHANGE, RSYNC_CHANGE);
return 0;
}
static int pm860x_dac_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
unsigned int dac = 0;
int data;
if (!strcmp(w->name, "Left DAC"))
dac = DAC_LEFT;
if (!strcmp(w->name, "Right DAC"))
dac = DAC_RIGHT;
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (dac) {
/* Auto mute in power-on sequence. */
dac |= MODULATOR;
snd_soc_update_bits(codec, PM860X_DAC_OFFSET,
DAC_MUTE, DAC_MUTE);
snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
RSYNC_CHANGE, RSYNC_CHANGE);
/* update dac */
snd_soc_update_bits(codec, PM860X_DAC_EN_2,
dac, dac);
}
break;
case SND_SOC_DAPM_PRE_PMD:
if (dac) {
/* Auto mute in power-off sequence. */
snd_soc_update_bits(codec, PM860X_DAC_OFFSET,
DAC_MUTE, DAC_MUTE);
snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
RSYNC_CHANGE, RSYNC_CHANGE);
/* update dac */
data = snd_soc_read(codec, PM860X_DAC_EN_2);
data &= ~dac;
if (!(data & (DAC_LEFT | DAC_RIGHT)))
data &= ~MODULATOR;
snd_soc_write(codec, PM860X_DAC_EN_2, data);
}
break;
}
return 0;
}
static const char *pm860x_opamp_texts[] = {"-50%", "-25%", "0%", "75%"};
static const char *pm860x_pa_texts[] = {"-33%", "0%", "33%", "66%"};
static const struct soc_enum pm860x_hs1_opamp_enum =
SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts);
static const struct soc_enum pm860x_hs2_opamp_enum =
SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts);
static const struct soc_enum pm860x_hs1_pa_enum =
SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts);
static const struct soc_enum pm860x_hs2_pa_enum =
SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts);
static const struct soc_enum pm860x_lo1_opamp_enum =
SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts);
static const struct soc_enum pm860x_lo2_opamp_enum =
SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts);
static const struct soc_enum pm860x_lo1_pa_enum =
SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts);
static const struct soc_enum pm860x_lo2_pa_enum =
SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts);
static const struct soc_enum pm860x_spk_pa_enum =
SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts);
static const struct soc_enum pm860x_ear_pa_enum =
SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_texts);
static const struct soc_enum pm860x_spk_ear_opamp_enum =
SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts);
static const struct snd_kcontrol_new pm860x_snd_controls[] = {
SOC_DOUBLE_R_TLV("ADC Capture Volume", PM860X_ADC_ANA_2,
PM860X_ADC_ANA_3, 6, 3, 0, adc_tlv),
SOC_DOUBLE_TLV("AUX Capture Volume", PM860X_ADC_ANA_3, 0, 3, 7, 0,
aux_tlv),
SOC_SINGLE_TLV("MIC1 Capture Volume", PM860X_ADC_ANA_2, 0, 7, 0,
mic_tlv),
SOC_SINGLE_TLV("MIC3 Capture Volume", PM860X_ADC_ANA_2, 3, 7, 0,
mic_tlv),
SOC_DOUBLE_R_EXT_TLV("Sidetone Volume", PM860X_SIDETONE_L_GAIN,
PM860X_SIDETONE_R_GAIN, 0, ARRAY_SIZE(st_table)-1,
0, snd_soc_get_volsw_2r_st,
snd_soc_put_volsw_2r_st, st_tlv),
SOC_SINGLE_TLV("Speaker Playback Volume", PM860X_EAR_CTRL_1,
0, 7, 0, out_tlv),
SOC_DOUBLE_R_TLV("Line Playback Volume", PM860X_LO1_CTRL,
PM860X_LO2_CTRL, 0, 7, 0, out_tlv),
SOC_DOUBLE_R_TLV("Headset Playback Volume", PM860X_HS1_CTRL,
PM860X_HS2_CTRL, 0, 7, 0, out_tlv),
SOC_DOUBLE_R_EXT_TLV("Hifi Left Playback Volume",
PM860X_HIFIL_GAIN_LEFT,
PM860X_HIFIL_GAIN_RIGHT, 0, 63, 0,
snd_soc_get_volsw_2r_out,
snd_soc_put_volsw_2r_out, dpga_tlv),
SOC_DOUBLE_R_EXT_TLV("Hifi Right Playback Volume",
PM860X_HIFIR_GAIN_LEFT,
PM860X_HIFIR_GAIN_RIGHT, 0, 63, 0,
snd_soc_get_volsw_2r_out,
snd_soc_put_volsw_2r_out, dpga_tlv),
SOC_DOUBLE_R_EXT_TLV("Lofi Playback Volume", PM860X_LOFI_GAIN_LEFT,
PM860X_LOFI_GAIN_RIGHT, 0, 63, 0,
snd_soc_get_volsw_2r_out,
snd_soc_put_volsw_2r_out, dpga_tlv),
SOC_ENUM("Headset1 Operational Amplifier Current",
pm860x_hs1_opamp_enum),
SOC_ENUM("Headset2 Operational Amplifier Current",
pm860x_hs2_opamp_enum),
SOC_ENUM("Headset1 Amplifier Current", pm860x_hs1_pa_enum),
SOC_ENUM("Headset2 Amplifier Current", pm860x_hs2_pa_enum),
SOC_ENUM("Lineout1 Operational Amplifier Current",
pm860x_lo1_opamp_enum),
SOC_ENUM("Lineout2 Operational Amplifier Current",
pm860x_lo2_opamp_enum),
SOC_ENUM("Lineout1 Amplifier Current", pm860x_lo1_pa_enum),
SOC_ENUM("Lineout2 Amplifier Current", pm860x_lo2_pa_enum),
SOC_ENUM("Speaker Operational Amplifier Current",
pm860x_spk_ear_opamp_enum),
SOC_ENUM("Speaker Amplifier Current", pm860x_spk_pa_enum),
SOC_ENUM("Earpiece Amplifier Current", pm860x_ear_pa_enum),
};
/*
* DAPM Controls
*/
/* PCM Switch / PCM Interface */
static const struct snd_kcontrol_new pcm_switch_controls =
SOC_DAPM_SINGLE("Switch", PM860X_ADC_EN_2, 0, 1, 0);
/* AUX1 Switch */
static const struct snd_kcontrol_new aux1_switch_controls =
SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 4, 1, 0);
/* AUX2 Switch */
static const struct snd_kcontrol_new aux2_switch_controls =
SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 5, 1, 0);
/* Left Ex. PA Switch */
static const struct snd_kcontrol_new lepa_switch_controls =
SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 2, 1, 0);
/* Right Ex. PA Switch */
static const struct snd_kcontrol_new repa_switch_controls =
SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 1, 1, 0);
/* PCM Mux / Mux7 */
static const char *aif1_text[] = {
"PCM L", "PCM R",
};
static const struct soc_enum aif1_enum =
SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 6, 2, aif1_text);
static const struct snd_kcontrol_new aif1_mux =
SOC_DAPM_ENUM("PCM Mux", aif1_enum);
/* I2S Mux / Mux9 */
static const char *i2s_din_text[] = {
"DIN", "DIN1",
};
static const struct soc_enum i2s_din_enum =
SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 1, 2, i2s_din_text);
static const struct snd_kcontrol_new i2s_din_mux =
SOC_DAPM_ENUM("I2S DIN Mux", i2s_din_enum);
/* I2S Mic Mux / Mux8 */
static const char *i2s_mic_text[] = {
"Ex PA", "ADC",
};
static const struct soc_enum i2s_mic_enum =
SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 4, 2, i2s_mic_text);
static const struct snd_kcontrol_new i2s_mic_mux =
SOC_DAPM_ENUM("I2S Mic Mux", i2s_mic_enum);
/* ADCL Mux / Mux2 */
static const char *adcl_text[] = {
"ADCR", "ADCL",
};
static const struct soc_enum adcl_enum =
SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 4, 2, adcl_text);
static const struct snd_kcontrol_new adcl_mux =
SOC_DAPM_ENUM("ADC Left Mux", adcl_enum);
/* ADCR Mux / Mux3 */
static const char *adcr_text[] = {
"ADCL", "ADCR",
};
static const struct soc_enum adcr_enum =
SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 2, 2, adcr_text);
static const struct snd_kcontrol_new adcr_mux =
SOC_DAPM_ENUM("ADC Right Mux", adcr_enum);
/* ADCR EC Mux / Mux6 */
static const char *adcr_ec_text[] = {
"ADCR", "EC",
};
static const struct soc_enum adcr_ec_enum =
SOC_ENUM_SINGLE(PM860X_ADC_EN_2, 3, 2, adcr_ec_text);
static const struct snd_kcontrol_new adcr_ec_mux =
SOC_DAPM_ENUM("ADCR EC Mux", adcr_ec_enum);
/* EC Mux / Mux4 */
static const char *ec_text[] = {
"Left", "Right", "Left + Right",
};
static const struct soc_enum ec_enum =
SOC_ENUM_SINGLE(PM860X_EC_PATH, 1, 3, ec_text);
static const struct snd_kcontrol_new ec_mux =
SOC_DAPM_ENUM("EC Mux", ec_enum);
static const char *dac_text[] = {
"No input", "Right", "Left", "No input",
};
/* DAC Headset 1 Mux / Mux10 */
static const struct soc_enum dac_hs1_enum =
SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 0, 4, dac_text);
static const struct snd_kcontrol_new dac_hs1_mux =
SOC_DAPM_ENUM("DAC HS1 Mux", dac_hs1_enum);
/* DAC Headset 2 Mux / Mux11 */
static const struct soc_enum dac_hs2_enum =
SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 2, 4, dac_text);
static const struct snd_kcontrol_new dac_hs2_mux =
SOC_DAPM_ENUM("DAC HS2 Mux", dac_hs2_enum);
/* DAC Lineout 1 Mux / Mux12 */
static const struct soc_enum dac_lo1_enum =
SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 4, 4, dac_text);
static const struct snd_kcontrol_new dac_lo1_mux =
SOC_DAPM_ENUM("DAC LO1 Mux", dac_lo1_enum);
/* DAC Lineout 2 Mux / Mux13 */
static const struct soc_enum dac_lo2_enum =
SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 6, 4, dac_text);
static const struct snd_kcontrol_new dac_lo2_mux =
SOC_DAPM_ENUM("DAC LO2 Mux", dac_lo2_enum);
/* DAC Spearker Earphone Mux / Mux14 */
static const struct soc_enum dac_spk_ear_enum =
SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_2, 0, 4, dac_text);
static const struct snd_kcontrol_new dac_spk_ear_mux =
SOC_DAPM_ENUM("DAC SP Mux", dac_spk_ear_enum);
/* Headset 1 Mux / Mux15 */
static const char *in_text[] = {
"Digital", "Analog",
};
static const struct soc_enum hs1_enum =
SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 0, 2, in_text);
static const struct snd_kcontrol_new hs1_mux =
SOC_DAPM_ENUM("Headset1 Mux", hs1_enum);
/* Headset 2 Mux / Mux16 */
static const struct soc_enum hs2_enum =
SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 1, 2, in_text);
static const struct snd_kcontrol_new hs2_mux =
SOC_DAPM_ENUM("Headset2 Mux", hs2_enum);
/* Lineout 1 Mux / Mux17 */
static const struct soc_enum lo1_enum =
SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 2, 2, in_text);
static const struct snd_kcontrol_new lo1_mux =
SOC_DAPM_ENUM("Lineout1 Mux", lo1_enum);
/* Lineout 2 Mux / Mux18 */
static const struct soc_enum lo2_enum =
SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 3, 2, in_text);
static const struct snd_kcontrol_new lo2_mux =
SOC_DAPM_ENUM("Lineout2 Mux", lo2_enum);
/* Speaker Earpiece Demux */
static const char *spk_text[] = {
"Earpiece", "Speaker",
};
static const struct soc_enum spk_enum =
SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 6, 2, spk_text);
static const struct snd_kcontrol_new spk_demux =
SOC_DAPM_ENUM("Speaker Earpiece Demux", spk_enum);
/* MIC Mux / Mux1 */
static const char *mic_text[] = {
"Mic 1", "Mic 2",
};
static const struct soc_enum mic_enum =
SOC_ENUM_SINGLE(PM860X_ADC_ANA_4, 4, 2, mic_text);
static const struct snd_kcontrol_new mic_mux =
SOC_DAPM_ENUM("MIC Mux", mic_enum);
static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = {
SND_SOC_DAPM_AIF_IN("PCM SDI", "PCM Playback", 0,
PM860X_ADC_EN_2, 0, 0),
SND_SOC_DAPM_AIF_OUT("PCM SDO", "PCM Capture", 0,
PM860X_PCM_IFACE_3, 1, 1),
SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0,
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0,
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0,
PM860X_I2S_IFACE_3, 5, 1),
SND_SOC_DAPM_SUPPLY("I2S CLK", PM860X_DAC_EN_2, 0, 0, NULL, 0),
SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux),
SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux),
SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux),
SND_SOC_DAPM_MUX("EC Mux", SND_SOC_NOPM, 0, 0, &ec_mux),
SND_SOC_DAPM_MUX("ADCR EC Mux", SND_SOC_NOPM, 0, 0, &adcr_ec_mux),
SND_SOC_DAPM_SWITCH("Left EPA", SND_SOC_NOPM, 0, 0,
&lepa_switch_controls),
SND_SOC_DAPM_SWITCH("Right EPA", SND_SOC_NOPM, 0, 0,
&repa_switch_controls),
SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Left ADC MOD", PM860X_ADC_EN_1,
0, 1, 1, 0),
SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Right ADC MOD", PM860X_ADC_EN_1,
1, 1, 1, 0),
SND_SOC_DAPM_ADC("Left ADC", NULL, PM860X_ADC_EN_2, 5, 0),
SND_SOC_DAPM_ADC("Right ADC", NULL, PM860X_ADC_EN_2, 4, 0),
SND_SOC_DAPM_SWITCH("AUX1 Switch", SND_SOC_NOPM, 0, 0,
&aux1_switch_controls),
SND_SOC_DAPM_SWITCH("AUX2 Switch", SND_SOC_NOPM, 0, 0,
&aux2_switch_controls),
SND_SOC_DAPM_MUX("MIC Mux", SND_SOC_NOPM, 0, 0, &mic_mux),
SND_SOC_DAPM_MICBIAS("Mic1 Bias", PM860X_ADC_ANA_1, 2, 0),
SND_SOC_DAPM_MICBIAS("Mic3 Bias", PM860X_ADC_ANA_1, 7, 0),
SND_SOC_DAPM_PGA("MIC1 Volume", PM860X_ADC_EN_1, 2, 0, NULL, 0),
SND_SOC_DAPM_PGA("MIC3 Volume", PM860X_ADC_EN_1, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("AUX1 Volume", PM860X_ADC_EN_1, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("AUX2 Volume", PM860X_ADC_EN_1, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("Sidetone PGA", PM860X_ADC_EN_2, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("Lofi PGA", PM860X_ADC_EN_2, 2, 0, NULL, 0),
SND_SOC_DAPM_INPUT("AUX1"),
SND_SOC_DAPM_INPUT("AUX2"),
SND_SOC_DAPM_INPUT("MIC1P"),
SND_SOC_DAPM_INPUT("MIC1N"),
SND_SOC_DAPM_INPUT("MIC2P"),
SND_SOC_DAPM_INPUT("MIC2N"),
SND_SOC_DAPM_INPUT("MIC3P"),
SND_SOC_DAPM_INPUT("MIC3N"),
SND_SOC_DAPM_DAC_E("Left DAC", NULL, SND_SOC_NOPM, 0, 0,
pm860x_dac_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_DAC_E("Right DAC", NULL, SND_SOC_NOPM, 0, 0,
pm860x_dac_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_MUX("I2S DIN Mux", SND_SOC_NOPM, 0, 0, &i2s_din_mux),
SND_SOC_DAPM_MUX("DAC HS1 Mux", SND_SOC_NOPM, 0, 0, &dac_hs1_mux),
SND_SOC_DAPM_MUX("DAC HS2 Mux", SND_SOC_NOPM, 0, 0, &dac_hs2_mux),
SND_SOC_DAPM_MUX("DAC LO1 Mux", SND_SOC_NOPM, 0, 0, &dac_lo1_mux),
SND_SOC_DAPM_MUX("DAC LO2 Mux", SND_SOC_NOPM, 0, 0, &dac_lo2_mux),
SND_SOC_DAPM_MUX("DAC SP Mux", SND_SOC_NOPM, 0, 0, &dac_spk_ear_mux),
SND_SOC_DAPM_MUX("Headset1 Mux", SND_SOC_NOPM, 0, 0, &hs1_mux),
SND_SOC_DAPM_MUX("Headset2 Mux", SND_SOC_NOPM, 0, 0, &hs2_mux),
SND_SOC_DAPM_MUX("Lineout1 Mux", SND_SOC_NOPM, 0, 0, &lo1_mux),
SND_SOC_DAPM_MUX("Lineout2 Mux", SND_SOC_NOPM, 0, 0, &lo2_mux),
SND_SOC_DAPM_MUX("Speaker Earpiece Demux", SND_SOC_NOPM, 0, 0,
&spk_demux),
SND_SOC_DAPM_PGA("Headset1 PGA", PM860X_DAC_EN_1, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Headset2 PGA", PM860X_DAC_EN_1, 1, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("HS1"),
SND_SOC_DAPM_OUTPUT("HS2"),
SND_SOC_DAPM_PGA("Lineout1 PGA", PM860X_DAC_EN_1, 2, 0, NULL, 0),
SND_SOC_DAPM_PGA("Lineout2 PGA", PM860X_DAC_EN_1, 3, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LINEOUT1"),
SND_SOC_DAPM_OUTPUT("LINEOUT2"),
SND_SOC_DAPM_PGA("Earpiece PGA", PM860X_DAC_EN_1, 4, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("EARP"),
SND_SOC_DAPM_OUTPUT("EARN"),
SND_SOC_DAPM_PGA("Speaker PGA", PM860X_DAC_EN_1, 5, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LSP"),
SND_SOC_DAPM_OUTPUT("LSN"),
SND_SOC_DAPM_REG(snd_soc_dapm_supply, "VCODEC", PM860X_AUDIO_SUPPLIES_2,
0, SUPPLY_MASK, SUPPLY_MASK, 0),
PM860X_DAPM_OUTPUT("RSYNC", pm860x_rsync_event),
};
static const struct snd_soc_dapm_route 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)
{
int data;
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Enable Audio PLL & Audio section */
data = AUDIO_PLL | AUDIO_SECTION_ON;
pm860x_reg_write(codec->control_data, REG_MISC2, data);
udelay(300);
data = AUDIO_PLL | AUDIO_SECTION_RESET
| AUDIO_SECTION_ON;
pm860x_reg_write(codec->control_data, REG_MISC2, data);
}
break;
case SND_SOC_BIAS_OFF:
data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON;
pm860x_set_bits(codec->control_data, REG_MISC2, data, 0);
break;
}
codec->dapm.bias_level = level;
return 0;
}
static const struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
.digital_mute = pm860x_digital_mute,
.hw_params = pm860x_pcm_hw_params,
.set_fmt = pm860x_pcm_set_dai_fmt,
.set_sysclk = pm860x_set_dai_sysclk,
};
static const struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
.digital_mute = pm860x_digital_mute,
.hw_params = pm860x_i2s_hw_params,
.set_fmt = pm860x_i2s_set_dai_fmt,
.set_sysclk = pm860x_set_dai_sysclk,
};
#define PM860X_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \
SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000)
static struct snd_soc_dai_driver pm860x_dai[] = {
{
/* DAI PCM */
.name = "88pm860x-pcm",
.id = 1,
.playback = {
.stream_name = "PCM Playback",
.channels_min = 2,
.channels_max = 2,
.rates = PM860X_RATES,
.formats = SNDRV_PCM_FORMAT_S16_LE | \
SNDRV_PCM_FORMAT_S18_3LE,
},
.capture = {
.stream_name = "PCM Capture",
.channels_min = 2,
.channels_max = 2,
.rates = PM860X_RATES,
.formats = SNDRV_PCM_FORMAT_S16_LE | \
SNDRV_PCM_FORMAT_S18_3LE,
},
.ops = &pm860x_pcm_dai_ops,
}, {
/* DAI I2S */
.name = "88pm860x-i2s",
.id = 2,
.playback = {
.stream_name = "I2S Playback",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FORMAT_S16_LE | \
SNDRV_PCM_FORMAT_S18_3LE,
},
.capture = {
.stream_name = "I2S Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FORMAT_S16_LE | \
SNDRV_PCM_FORMAT_S18_3LE,
},
.ops = &pm860x_i2s_dai_ops,
},
};
static irqreturn_t pm860x_codec_handler(int irq, void *data)
{
struct pm860x_priv *pm860x = data;
int status, shrt, report = 0, mic_report = 0;
int mask;
status = pm860x_reg_read(pm860x->i2c, REG_STATUS_1);
shrt = pm860x_reg_read(pm860x->i2c, REG_SHORTS);
mask = pm860x->det.hs_shrt | pm860x->det.hook_det | pm860x->det.lo_shrt
| pm860x->det.hp_det;
#ifndef CONFIG_SND_SOC_88PM860X_MODULE
if (status & (HEADSET_STATUS | MIC_STATUS | SHORT_HS1 | SHORT_HS2 |
SHORT_LO1 | SHORT_LO2))
trace_snd_soc_jack_irq(dev_name(pm860x->codec->dev));
#endif
if ((pm860x->det.hp_det & SND_JACK_HEADPHONE)
&& (status & HEADSET_STATUS))
report |= SND_JACK_HEADPHONE;
if ((pm860x->det.mic_det & SND_JACK_MICROPHONE)
&& (status & MIC_STATUS))
mic_report |= SND_JACK_MICROPHONE;
if (pm860x->det.hs_shrt && (shrt & (SHORT_HS1 | SHORT_HS2)))
report |= pm860x->det.hs_shrt;
if (pm860x->det.hook_det && (status & HOOK_STATUS))
report |= pm860x->det.hook_det;
if (pm860x->det.lo_shrt && (shrt & (SHORT_LO1 | SHORT_LO2)))
report |= pm860x->det.lo_shrt;
if (report)
snd_soc_jack_report(pm860x->det.hp_jack, report, mask);
if (mic_report)
snd_soc_jack_report(pm860x->det.mic_jack, SND_JACK_MICROPHONE,
SND_JACK_MICROPHONE);
dev_dbg(pm860x->codec->dev, "headphone report:0x%x, mask:%x\n",
report, mask);
dev_dbg(pm860x->codec->dev, "microphone report:0x%x\n", mic_report);
return IRQ_HANDLED;
}
int pm860x_hs_jack_detect(struct snd_soc_codec *codec,
struct snd_soc_jack *jack,
int det, int hook, int hs_shrt, int lo_shrt)
{
struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
int data;
pm860x->det.hp_jack = jack;
pm860x->det.hp_det = det;
pm860x->det.hook_det = hook;
pm860x->det.hs_shrt = hs_shrt;
pm860x->det.lo_shrt = lo_shrt;
if (det & SND_JACK_HEADPHONE)
pm860x_set_bits(codec->control_data, REG_HS_DET,
EN_HS_DET, EN_HS_DET);
/* headset short detect */
if (hs_shrt) {
data = CLR_SHORT_HS2 | CLR_SHORT_HS1;
pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
}
/* Lineout short detect */
if (lo_shrt) {
data = CLR_SHORT_LO2 | CLR_SHORT_LO1;
pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
}
/* sync status */
pm860x_codec_handler(0, pm860x);
return 0;
}
EXPORT_SYMBOL_GPL(pm860x_hs_jack_detect);
int pm860x_mic_jack_detect(struct snd_soc_codec *codec,
struct snd_soc_jack *jack, int det)
{
struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
pm860x->det.mic_jack = jack;
pm860x->det.mic_det = det;
if (det & SND_JACK_MICROPHONE)
pm860x_set_bits(codec->control_data, REG_MIC_DET,
MICDET_MASK, MICDET_MASK);
/* sync status */
pm860x_codec_handler(0, pm860x);
return 0;
}
EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect);
static int pm860x_probe(struct snd_soc_codec *codec)
{
struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
int i, ret;
pm860x->codec = codec;
codec->control_data = pm860x->i2c;
for (i = 0; i < 4; i++) {
ret = request_threaded_irq(pm860x->irq[i], NULL,
pm860x_codec_handler, IRQF_ONESHOT,
pm860x->name[i], pm860x);
if (ret < 0) {
dev_err(codec->dev, "Failed to request IRQ!\n");
goto out;
}
}
pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE,
REG_CACHE_SIZE, codec->reg_cache);
if (ret < 0) {
dev_err(codec->dev, "Failed to fill register cache: %d\n",
ret);
goto out;
}
return 0;
out:
while (--i >= 0)
free_irq(pm860x->irq[i], pm860x);
return ret;
}
static int pm860x_remove(struct snd_soc_codec *codec)
{
struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
int i;
for (i = 3; i >= 0; i--)
free_irq(pm860x->irq[i], pm860x);
pm860x_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_pm860x = {
.probe = pm860x_probe,
.remove = pm860x_remove,
.read = pm860x_read_reg_cache,
.write = pm860x_write_reg_cache,
.reg_cache_size = REG_CACHE_SIZE,
.reg_word_size = sizeof(u8),
.set_bias_level = pm860x_set_bias_level,
.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 __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 = 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;
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);
return -EINVAL;
}
static int __devexit pm860x_codec_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
platform_set_drvdata(pdev, NULL);
return 0;
}
static struct platform_driver pm860x_codec_driver = {
.driver = {
.name = "88pm860x-codec",
.owner = THIS_MODULE,
},
.probe = pm860x_codec_probe,
.remove = __devexit_p(pm860x_codec_remove),
};
module_platform_driver(pm860x_codec_driver);
MODULE_DESCRIPTION("ASoC 88PM860x driver");
MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:88pm860x-codec");
