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linux-stable/sound/soc/codecs/88pm860x-codec.c
Lars-Peter Clausen 85e7652d89 ASoC: Constify snd_soc_dai_ops structs 
Commit 1ee46ebd("ASoC: Make the DAI ops constant in the DAI structure")
introduced the possibility to have constant DAI ops structures, yet this is
barley used in both existing drivers and also new drivers being submitted,
although none of them modifies its DAI ops structure. The later is not
surprising since existing drivers are often used as templates for new drivers.
So this patch just constifies all existing snd_soc_dai_ops structs to eliminate
the issue altogether.

The patch was generated with the following coccinelle semantic patch:
// <smpl>
@@
identifier ops;
@@
-struct snd_soc_dai_ops ops =
+const struct snd_soc_dai_ops ops =
{ ... };
// </smpl>

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-23 10:40:46 +00:00

1500 lines
44 KiB
C
Raw Blame History

/*
* 88pm860x-codec.c -- 88PM860x ALSA SoC Audio Driver
*
* Copyright 2010 Marvell International Ltd.
* Author: Haojian Zhuang <haojian.zhuang@marvell.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/kernel.h>
#include <linux/module.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/mfd/88pm860x.h>
#include <linux/slab.h>
#include <linux/delay.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#include <sound/initval.h>
#include <sound/jack.h>
#include <trace/events/asoc.h>
#include "88pm860x-codec.h"
#define MAX_NAME_LEN 20
#define REG_CACHE_SIZE 0x40
#define REG_CACHE_BASE 0xb0
/* Status Register 1 (0x01) */
#define REG_STATUS_1 0x01
#define MIC_STATUS (1 << 7)
#define HOOK_STATUS (1 << 6)
#define HEADSET_STATUS (1 << 5)
/* Mic Detection Register (0x37) */
#define REG_MIC_DET 0x37
#define CONTINUOUS_POLLING (3 << 1)
#define EN_MIC_DET (1 << 0)
#define MICDET_MASK 0x07
/* Headset Detection Register (0x38) */
#define REG_HS_DET 0x38
#define EN_HS_DET (1 << 0)
/* Misc2 Register (0x42) */
#define REG_MISC2 0x42
#define AUDIO_PLL (1 << 5)
#define AUDIO_SECTION_RESET (1 << 4)
#define AUDIO_SECTION_ON (1 << 3)
/* PCM Interface Register 2 (0xb1) */
#define PCM_INF2_BCLK (1 << 6) /* Bit clock polarity */
#define PCM_INF2_FS (1 << 5) /* Frame Sync polarity */
#define PCM_INF2_MASTER (1 << 4) /* Master / Slave */
#define PCM_INF2_18WL (1 << 3) /* 18 / 16 bits */
#define PCM_GENERAL_I2S 0
#define PCM_EXACT_I2S 1
#define PCM_LEFT_I2S 2
#define PCM_RIGHT_I2S 3
#define PCM_SHORT_FS 4
#define PCM_LONG_FS 5
#define PCM_MODE_MASK 7
/* I2S Interface Register 4 (0xbe) */
#define I2S_EQU_BYP (1 << 6)
/* DAC Offset Register (0xcb) */
#define DAC_MUTE (1 << 7)
#define MUTE_LEFT (1 << 6)
#define MUTE_RIGHT (1 << 2)
/* ADC Analog Register 1 (0xd0) */
#define REG_ADC_ANA_1 0xd0
#define MIC1BIAS_MASK 0x60
/* Earpiece/Speaker Control Register 2 (0xda) */
#define REG_EAR2 0xda
#define RSYNC_CHANGE (1 << 2)
/* Audio Supplies Register 2 (0xdc) */
#define REG_SUPPLIES2 0xdc
#define LDO15_READY (1 << 4)
#define LDO15_EN (1 << 3)
#define CPUMP_READY (1 << 2)
#define CPUMP_EN (1 << 1)
#define AUDIO_EN (1 << 0)
#define SUPPLY_MASK (LDO15_EN | CPUMP_EN | AUDIO_EN)
/* Audio Enable Register 1 (0xdd) */
#define ADC_MOD_RIGHT (1 << 1)
#define ADC_MOD_LEFT (1 << 0)
/* Audio Enable Register 2 (0xde) */
#define ADC_LEFT (1 << 5)
#define ADC_RIGHT (1 << 4)
/* DAC Enable Register 2 (0xe1) */
#define DAC_LEFT (1 << 5)
#define DAC_RIGHT (1 << 4)
#define MODULATOR (1 << 3)
/* Shorts Register (0xeb) */
#define REG_SHORTS 0xeb
#define CLR_SHORT_LO2 (1 << 7)
#define SHORT_LO2 (1 << 6)
#define CLR_SHORT_LO1 (1 << 5)
#define SHORT_LO1 (1 << 4)
#define CLR_SHORT_HS2 (1 << 3)
#define SHORT_HS2 (1 << 2)
#define CLR_SHORT_HS1 (1 << 1)
#define SHORT_HS1 (1 << 0)
/*
* This widget should be just after DAC & PGA in DAPM power-on sequence and
* before DAC & PGA in DAPM power-off sequence.
*/
#define PM860X_DAPM_OUTPUT(wname, wevent) \
{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \
.shift = 0, .invert = 0, .kcontrol_news = NULL, \
.num_kcontrols = 0, .event = wevent, \
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD, }
struct pm860x_det {
struct snd_soc_jack *hp_jack;
struct snd_soc_jack *mic_jack;
int hp_det;
int mic_det;
int hook_det;
int hs_shrt;
int lo_shrt;
};
struct pm860x_priv {
unsigned int sysclk;
unsigned int pcmclk;
unsigned int dir;
unsigned int filter;
struct snd_soc_codec *codec;
struct i2c_client *i2c;
struct pm860x_chip *chip;
struct pm860x_det det;
int irq[4];
unsigned char name[4][MAX_NAME_LEN];
};
/* -9450dB to 0dB in 150dB steps ( mute instead of -9450dB) */
static const DECLARE_TLV_DB_SCALE(dpga_tlv, -9450, 150, 1);
/* -9dB to 0db in 3dB steps */
static const DECLARE_TLV_DB_SCALE(adc_tlv, -900, 300, 0);
/* {-23, -17, -13.5, -11, -9, -6, -3, 0}dB */
static const unsigned int mic_tlv[] = {
TLV_DB_RANGE_HEAD(5),
0, 0, TLV_DB_SCALE_ITEM(-2300, 0, 0),
1, 1, TLV_DB_SCALE_ITEM(-1700, 0, 0),
2, 2, TLV_DB_SCALE_ITEM(-1350, 0, 0),
3, 3, TLV_DB_SCALE_ITEM(-1100, 0, 0),
4, 7, TLV_DB_SCALE_ITEM(-900, 300, 0),
};
/* {0, 0, 0, -6, 0, 6, 12, 18}dB */
static const unsigned int aux_tlv[] = {
TLV_DB_RANGE_HEAD(2),
0, 2, TLV_DB_SCALE_ITEM(0, 0, 0),
3, 7, TLV_DB_SCALE_ITEM(-600, 600, 0),
};
/* {-16, -13, -10, -7, -5.2, -3,3, -2.2, 0}dB, mute instead of -16dB */
static const unsigned int out_tlv[] = {
TLV_DB_RANGE_HEAD(4),
0, 3, TLV_DB_SCALE_ITEM(-1600, 300, 1),
4, 4, TLV_DB_SCALE_ITEM(-520, 0, 0),
5, 5, TLV_DB_SCALE_ITEM(-330, 0, 0),
6, 7, TLV_DB_SCALE_ITEM(-220, 220, 0),
};
static const unsigned int st_tlv[] = {
TLV_DB_RANGE_HEAD(8),
0, 1, TLV_DB_SCALE_ITEM(-12041, 602, 0),
2, 3, TLV_DB_SCALE_ITEM(-11087, 250, 0),
4, 5, TLV_DB_SCALE_ITEM(-10643, 158, 0),
6, 7, TLV_DB_SCALE_ITEM(-10351, 116, 0),
8, 9, TLV_DB_SCALE_ITEM(-10133, 92, 0),
10, 13, TLV_DB_SCALE_ITEM(-9958, 70, 0),
14, 17, TLV_DB_SCALE_ITEM(-9689, 53, 0),
18, 271, TLV_DB_SCALE_ITEM(-9484, 37, 0),
};
/* Sidetone Gain = M * 2^(-5-N) */
struct st_gain {
unsigned int db;
unsigned int m;
unsigned int n;
};
static struct st_gain st_table[] = {
{-12041, 1, 15}, {-11439, 1, 14}, {-11087, 3, 15}, {-10837, 1, 13},
{-10643, 5, 15}, {-10485, 3, 14}, {-10351, 7, 15}, {-10235, 1, 12},
{-10133, 9, 15}, {-10041, 5, 14}, { -9958, 11, 15}, { -9883, 3, 13},
{ -9813, 13, 15}, { -9749, 7, 14}, { -9689, 15, 15}, { -9633, 1, 11},
{ -9580, 17, 15}, { -9531, 9, 14}, { -9484, 19, 15}, { -9439, 5, 13},
{ -9397, 21, 15}, { -9356, 11, 14}, { -9318, 23, 15}, { -9281, 3, 12},
{ -9245, 25, 15}, { -9211, 13, 14}, { -9178, 27, 15}, { -9147, 7, 13},
{ -9116, 29, 15}, { -9087, 15, 14}, { -9058, 31, 15}, { -9031, 1, 10},
{ -8978, 17, 14}, { -8929, 9, 13}, { -8882, 19, 14}, { -8837, 5, 12},
{ -8795, 21, 14}, { -8754, 11, 13}, { -8716, 23, 14}, { -8679, 3, 11},
{ -8643, 25, 14}, { -8609, 13, 13}, { -8576, 27, 14}, { -8545, 7, 12},
{ -8514, 29, 14}, { -8485, 15, 13}, { -8456, 31, 14}, { -8429, 1, 9},
{ -8376, 17, 13}, { -8327, 9, 12}, { -8280, 19, 13}, { -8235, 5, 11},
{ -8193, 21, 13}, { -8152, 11, 12}, { -8114, 23, 13}, { -8077, 3, 10},
{ -8041, 25, 13}, { -8007, 13, 12}, { -7974, 27, 13}, { -7943, 7, 11},
{ -7912, 29, 13}, { -7883, 15, 12}, { -7854, 31, 13}, { -7827, 1, 8},
{ -7774, 17, 12}, { -7724, 9, 11}, { -7678, 19, 12}, { -7633, 5, 10},
{ -7591, 21, 12}, { -7550, 11, 11}, { -7512, 23, 12}, { -7475, 3, 9},
{ -7439, 25, 12}, { -7405, 13, 11}, { -7372, 27, 12}, { -7341, 7, 10},
{ -7310, 29, 12}, { -7281, 15, 11}, { -7252, 31, 12}, { -7225, 1, 7},
{ -7172, 17, 11}, { -7122, 9, 10}, { -7075, 19, 11}, { -7031, 5, 9},
{ -6989, 21, 11}, { -6948, 11, 10}, { -6910, 23, 11}, { -6873, 3, 8},
{ -6837, 25, 11}, { -6803, 13, 10}, { -6770, 27, 11}, { -6739, 7, 9},
{ -6708, 29, 11}, { -6679, 15, 10}, { -6650, 31, 11}, { -6623, 1, 6},
{ -6570, 17, 10}, { -6520, 9, 9}, { -6473, 19, 10}, { -6429, 5, 8},
{ -6386, 21, 10}, { -6346, 11, 9}, { -6307, 23, 10}, { -6270, 3, 7},
{ -6235, 25, 10}, { -6201, 13, 9}, { -6168, 27, 10}, { -6137, 7, 8},
{ -6106, 29, 10}, { -6077, 15, 9}, { -6048, 31, 10}, { -6021, 1, 5},
{ -5968, 17, 9}, { -5918, 9, 8}, { -5871, 19, 9}, { -5827, 5, 7},
{ -5784, 21, 9}, { -5744, 11, 8}, { -5705, 23, 9}, { -5668, 3, 6},
{ -5633, 25, 9}, { -5599, 13, 8}, { -5566, 27, 9}, { -5535, 7, 7},
{ -5504, 29, 9}, { -5475, 15, 8}, { -5446, 31, 9}, { -5419, 1, 4},
{ -5366, 17, 8}, { -5316, 9, 7}, { -5269, 19, 8}, { -5225, 5, 6},
{ -5182, 21, 8}, { -5142, 11, 7}, { -5103, 23, 8}, { -5066, 3, 5},
{ -5031, 25, 8}, { -4997, 13, 7}, { -4964, 27, 8}, { -4932, 7, 6},
{ -4902, 29, 8}, { -4873, 15, 7}, { -4844, 31, 8}, { -4816, 1, 3},
{ -4764, 17, 7}, { -4714, 9, 6}, { -4667, 19, 7}, { -4623, 5, 5},
{ -4580, 21, 7}, { -4540, 11, 6}, { -4501, 23, 7}, { -4464, 3, 4},
{ -4429, 25, 7}, { -4395, 13, 6}, { -4362, 27, 7}, { -4330, 7, 5},
{ -4300, 29, 7}, { -4270, 15, 6}, { -4242, 31, 7}, { -4214, 1, 2},
{ -4162, 17, 6}, { -4112, 9, 5}, { -4065, 19, 6}, { -4021, 5, 4},
{ -3978, 21, 6}, { -3938, 11, 5}, { -3899, 23, 6}, { -3862, 3, 3},
{ -3827, 25, 6}, { -3793, 13, 5}, { -3760, 27, 6}, { -3728, 7, 4},
{ -3698, 29, 6}, { -3668, 15, 5}, { -3640, 31, 6}, { -3612, 1, 1},
{ -3560, 17, 5}, { -3510, 9, 4}, { -3463, 19, 5}, { -3419, 5, 3},
{ -3376, 21, 5}, { -3336, 11, 4}, { -3297, 23, 5}, { -3260, 3, 2},
{ -3225, 25, 5}, { -3191, 13, 4}, { -3158, 27, 5}, { -3126, 7, 3},
{ -3096, 29, 5}, { -3066, 15, 4}, { -3038, 31, 5}, { -3010, 1, 0},
{ -2958, 17, 4}, { -2908, 9, 3}, { -2861, 19, 4}, { -2816, 5, 2},
{ -2774, 21, 4}, { -2734, 11, 3}, { -2695, 23, 4}, { -2658, 3, 1},
{ -2623, 25, 4}, { -2589, 13, 3}, { -2556, 27, 4}, { -2524, 7, 2},
{ -2494, 29, 4}, { -2464, 15, 3}, { -2436, 31, 4}, { -2408, 2, 0},
{ -2356, 17, 3}, { -2306, 9, 2}, { -2259, 19, 3}, { -2214, 5, 1},
{ -2172, 21, 3}, { -2132, 11, 2}, { -2093, 23, 3}, { -2056, 3, 0},
{ -2021, 25, 3}, { -1987, 13, 2}, { -1954, 27, 3}, { -1922, 7, 1},
{ -1892, 29, 3}, { -1862, 15, 2}, { -1834, 31, 3}, { -1806, 4, 0},
{ -1754, 17, 2}, { -1704, 9, 1}, { -1657, 19, 2}, { -1612, 5, 0},
{ -1570, 21, 2}, { -1530, 11, 1}, { -1491, 23, 2}, { -1454, 6, 0},
{ -1419, 25, 2}, { -1384, 13, 1}, { -1352, 27, 2}, { -1320, 7, 0},
{ -1290, 29, 2}, { -1260, 15, 1}, { -1232, 31, 2}, { -1204, 8, 0},
{ -1151, 17, 1}, { -1102, 9, 0}, { -1055, 19, 1}, { -1010, 10, 0},
{ -968, 21, 1}, { -928, 11, 0}, { -889, 23, 1}, { -852, 12, 0},
{ -816, 25, 1}, { -782, 13, 0}, { -750, 27, 1}, { -718, 14, 0},
{ -688, 29, 1}, { -658, 15, 0}, { -630, 31, 1}, { -602, 16, 0},
{ -549, 17, 0}, { -500, 18, 0}, { -453, 19, 0}, { -408, 20, 0},
{ -366, 21, 0}, { -325, 22, 0}, { -287, 23, 0}, { -250, 24, 0},
{ -214, 25, 0}, { -180, 26, 0}, { -148, 27, 0}, { -116, 28, 0},
{ -86, 29, 0}, { -56, 30, 0}, { -28, 31, 0}, { 0, 0, 0},
};
static int pm860x_volatile(unsigned int reg)
{
BUG_ON(reg >= REG_CACHE_SIZE);
switch (reg) {
case PM860X_AUDIO_SUPPLIES_2:
return 1;
}
return 0;
}
static unsigned int pm860x_read_reg_cache(struct snd_soc_codec *codec,
unsigned int reg)
{
unsigned char *cache = codec->reg_cache;
BUG_ON(reg >= REG_CACHE_SIZE);
if (pm860x_volatile(reg))
return cache[reg];
reg += REG_CACHE_BASE;
return pm860x_reg_read(codec->control_data, reg);
}
static int pm860x_write_reg_cache(struct snd_soc_codec *codec,
unsigned int reg, unsigned int value)
{
unsigned char *cache = codec->reg_cache;
BUG_ON(reg >= REG_CACHE_SIZE);
if (!pm860x_volatile(reg))
cache[reg] = (unsigned char)value;
reg += REG_CACHE_BASE;
return pm860x_reg_write(codec->control_data, reg, value);
}
static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
int val[2], val2[2], i;
val[0] = snd_soc_read(codec, reg) & 0x3f;
val[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT) >> 4) & 0xf;
val2[0] = snd_soc_read(codec, reg2) & 0x3f;
val2[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT)) & 0xf;
for (i = 0; i < ARRAY_SIZE(st_table); i++) {
if ((st_table[i].m == val[0]) && (st_table[i].n == val[1]))
ucontrol->value.integer.value[0] = i;
if ((st_table[i].m == val2[0]) && (st_table[i].n == val2[1]))
ucontrol->value.integer.value[1] = i;
}
return 0;
}
static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
int err;
unsigned int val, val2;
val = ucontrol->value.integer.value[0];
val2 = ucontrol->value.integer.value[1];
err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m);
if (err < 0)
return err;
err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0xf0,
st_table[val].n << 4);
if (err < 0)
return err;
err = snd_soc_update_bits(codec, reg2, 0x3f, st_table[val2].m);
if (err < 0)
return err;
err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0x0f,
st_table[val2].n);
return err;
}
static int snd_soc_get_volsw_2r_out(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
unsigned int shift = mc->shift;
int max = mc->max, val, val2;
unsigned int mask = (1 << fls(max)) - 1;
val = snd_soc_read(codec, reg) >> shift;
val2 = snd_soc_read(codec, reg2) >> shift;
ucontrol->value.integer.value[0] = (max - val) & mask;
ucontrol->value.integer.value[1] = (max - val2) & mask;
return 0;
}
static int snd_soc_put_volsw_2r_out(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
unsigned int reg2 = mc->rreg;
unsigned int shift = mc->shift;
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
int err;
unsigned int val, val2, val_mask;
val_mask = mask << shift;
val = ((max - ucontrol->value.integer.value[0]) & mask);
val2 = ((max - ucontrol->value.integer.value[1]) & mask);
val = val << shift;
val2 = val2 << shift;
err = snd_soc_update_bits(codec, reg, val_mask, val);
if (err < 0)
return err;
err = snd_soc_update_bits(codec, reg2, val_mask, val2);
return err;
}
/* DAPM Widget Events */
/*
* A lot registers are belong to RSYNC domain. It requires enabling RSYNC bit
* after updating these registers. Otherwise, these updated registers won't
* be effective.
*/
static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
/*
* In order to avoid current on the load, mute power-on and power-off
* should be transients.
* Unmute by DAC_MUTE. It should be unmuted when DAPM sequence is
* finished.
*/
snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0);
snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
RSYNC_CHANGE, RSYNC_CHANGE);
return 0;
}
static int pm860x_dac_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
unsigned int dac = 0;
int data;
if (!strcmp(w->name, "Left DAC"))
dac = DAC_LEFT;
if (!strcmp(w->name, "Right DAC"))
dac = DAC_RIGHT;
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (dac) {
/* Auto mute in power-on sequence. */
dac |= MODULATOR;
snd_soc_update_bits(codec, PM860X_DAC_OFFSET,
DAC_MUTE, DAC_MUTE);
snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
RSYNC_CHANGE, RSYNC_CHANGE);
/* update dac */
snd_soc_update_bits(codec, PM860X_DAC_EN_2,
dac, dac);
}
break;
case SND_SOC_DAPM_PRE_PMD:
if (dac) {
/* Auto mute in power-off sequence. */
snd_soc_update_bits(codec, PM860X_DAC_OFFSET,
DAC_MUTE, DAC_MUTE);
snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
RSYNC_CHANGE, RSYNC_CHANGE);
/* update dac */
data = snd_soc_read(codec, PM860X_DAC_EN_2);
data &= ~dac;
if (!(data & (DAC_LEFT | DAC_RIGHT)))
data &= ~MODULATOR;
snd_soc_write(codec, PM860X_DAC_EN_2, data);
}
break;
}
return 0;
}
static const char *pm860x_opamp_texts[] = {"-50%", "-25%", "0%", "75%"};
static const char *pm860x_pa_texts[] = {"-33%", "0%", "33%", "66%"};
static const struct soc_enum pm860x_hs1_opamp_enum =
SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts);
static const struct soc_enum pm860x_hs2_opamp_enum =
SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts);
static const struct soc_enum pm860x_hs1_pa_enum =
SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts);
static const struct soc_enum pm860x_hs2_pa_enum =
SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts);
static const struct soc_enum pm860x_lo1_opamp_enum =
SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts);
static const struct soc_enum pm860x_lo2_opamp_enum =
SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts);
static const struct soc_enum pm860x_lo1_pa_enum =
SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts);
static const struct soc_enum pm860x_lo2_pa_enum =
SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts);
static const struct soc_enum pm860x_spk_pa_enum =
SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts);
static const struct soc_enum pm860x_ear_pa_enum =
SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_texts);
static const struct soc_enum pm860x_spk_ear_opamp_enum =
SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts);
static const struct snd_kcontrol_new pm860x_snd_controls[] = {
SOC_DOUBLE_R_TLV("ADC Capture Volume", PM860X_ADC_ANA_2,
PM860X_ADC_ANA_3, 6, 3, 0, adc_tlv),
SOC_DOUBLE_TLV("AUX Capture Volume", PM860X_ADC_ANA_3, 0, 3, 7, 0,
aux_tlv),
SOC_SINGLE_TLV("MIC1 Capture Volume", PM860X_ADC_ANA_2, 0, 7, 0,
mic_tlv),
SOC_SINGLE_TLV("MIC3 Capture Volume", PM860X_ADC_ANA_2, 3, 7, 0,
mic_tlv),
SOC_DOUBLE_R_EXT_TLV("Sidetone Volume", PM860X_SIDETONE_L_GAIN,
PM860X_SIDETONE_R_GAIN, 0, ARRAY_SIZE(st_table)-1,
0, snd_soc_get_volsw_2r_st,
snd_soc_put_volsw_2r_st, st_tlv),
SOC_SINGLE_TLV("Speaker Playback Volume", PM860X_EAR_CTRL_1,
0, 7, 0, out_tlv),
SOC_DOUBLE_R_TLV("Line Playback Volume", PM860X_LO1_CTRL,
PM860X_LO2_CTRL, 0, 7, 0, out_tlv),
SOC_DOUBLE_R_TLV("Headset Playback Volume", PM860X_HS1_CTRL,
PM860X_HS2_CTRL, 0, 7, 0, out_tlv),
SOC_DOUBLE_R_EXT_TLV("Hifi Left Playback Volume",
PM860X_HIFIL_GAIN_LEFT,
PM860X_HIFIL_GAIN_RIGHT, 0, 63, 0,
snd_soc_get_volsw_2r_out,
snd_soc_put_volsw_2r_out, dpga_tlv),
SOC_DOUBLE_R_EXT_TLV("Hifi Right Playback Volume",
PM860X_HIFIR_GAIN_LEFT,
PM860X_HIFIR_GAIN_RIGHT, 0, 63, 0,
snd_soc_get_volsw_2r_out,
snd_soc_put_volsw_2r_out, dpga_tlv),
SOC_DOUBLE_R_EXT_TLV("Lofi Playback Volume", PM860X_LOFI_GAIN_LEFT,
PM860X_LOFI_GAIN_RIGHT, 0, 63, 0,
snd_soc_get_volsw_2r_out,
snd_soc_put_volsw_2r_out, dpga_tlv),
SOC_ENUM("Headset1 Operational Amplifier Current",
pm860x_hs1_opamp_enum),
SOC_ENUM("Headset2 Operational Amplifier Current",
pm860x_hs2_opamp_enum),
SOC_ENUM("Headset1 Amplifier Current", pm860x_hs1_pa_enum),
SOC_ENUM("Headset2 Amplifier Current", pm860x_hs2_pa_enum),
SOC_ENUM("Lineout1 Operational Amplifier Current",
pm860x_lo1_opamp_enum),
SOC_ENUM("Lineout2 Operational Amplifier Current",
pm860x_lo2_opamp_enum),
SOC_ENUM("Lineout1 Amplifier Current", pm860x_lo1_pa_enum),
SOC_ENUM("Lineout2 Amplifier Current", pm860x_lo2_pa_enum),
SOC_ENUM("Speaker Operational Amplifier Current",
pm860x_spk_ear_opamp_enum),
SOC_ENUM("Speaker Amplifier Current", pm860x_spk_pa_enum),
SOC_ENUM("Earpiece Amplifier Current", pm860x_ear_pa_enum),
};
/*
* DAPM Controls
*/
/* PCM Switch / PCM Interface */
static const struct snd_kcontrol_new pcm_switch_controls =
SOC_DAPM_SINGLE("Switch", PM860X_ADC_EN_2, 0, 1, 0);
/* AUX1 Switch */
static const struct snd_kcontrol_new aux1_switch_controls =
SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 4, 1, 0);
/* AUX2 Switch */
static const struct snd_kcontrol_new aux2_switch_controls =
SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 5, 1, 0);
/* Left Ex. PA Switch */
static const struct snd_kcontrol_new lepa_switch_controls =
SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 2, 1, 0);
/* Right Ex. PA Switch */
static const struct snd_kcontrol_new repa_switch_controls =
SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 1, 1, 0);
/* PCM Mux / Mux7 */
static const char *aif1_text[] = {
"PCM L", "PCM R",
};
static const struct soc_enum aif1_enum =
SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 6, 2, aif1_text);
static const struct snd_kcontrol_new aif1_mux =
SOC_DAPM_ENUM("PCM Mux", aif1_enum);
/* I2S Mux / Mux9 */
static const char *i2s_din_text[] = {
"DIN", "DIN1",
};
static const struct soc_enum i2s_din_enum =
SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 1, 2, i2s_din_text);
static const struct snd_kcontrol_new i2s_din_mux =
SOC_DAPM_ENUM("I2S DIN Mux", i2s_din_enum);
/* I2S Mic Mux / Mux8 */
static const char *i2s_mic_text[] = {
"Ex PA", "ADC",
};
static const struct soc_enum i2s_mic_enum =
SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 4, 2, i2s_mic_text);
static const struct snd_kcontrol_new i2s_mic_mux =
SOC_DAPM_ENUM("I2S Mic Mux", i2s_mic_enum);
/* ADCL Mux / Mux2 */
static const char *adcl_text[] = {
"ADCR", "ADCL",
};
static const struct soc_enum adcl_enum =
SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 4, 2, adcl_text);
static const struct snd_kcontrol_new adcl_mux =
SOC_DAPM_ENUM("ADC Left Mux", adcl_enum);
/* ADCR Mux / Mux3 */
static const char *adcr_text[] = {
"ADCL", "ADCR",
};
static const struct soc_enum adcr_enum =
SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 2, 2, adcr_text);
static const struct snd_kcontrol_new adcr_mux =
SOC_DAPM_ENUM("ADC Right Mux", adcr_enum);
/* ADCR EC Mux / Mux6 */
static const char *adcr_ec_text[] = {
"ADCR", "EC",
};
static const struct soc_enum adcr_ec_enum =
SOC_ENUM_SINGLE(PM860X_ADC_EN_2, 3, 2, adcr_ec_text);
static const struct snd_kcontrol_new adcr_ec_mux =
SOC_DAPM_ENUM("ADCR EC Mux", adcr_ec_enum);
/* EC Mux / Mux4 */
static const char *ec_text[] = {
"Left", "Right", "Left + Right",
};
static const struct soc_enum ec_enum =
SOC_ENUM_SINGLE(PM860X_EC_PATH, 1, 3, ec_text);
static const struct snd_kcontrol_new ec_mux =
SOC_DAPM_ENUM("EC Mux", ec_enum);
static const char *dac_text[] = {
"No input", "Right", "Left", "No input",
};
/* DAC Headset 1 Mux / Mux10 */
static const struct soc_enum dac_hs1_enum =
SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 0, 4, dac_text);
static const struct snd_kcontrol_new dac_hs1_mux =
SOC_DAPM_ENUM("DAC HS1 Mux", dac_hs1_enum);
/* DAC Headset 2 Mux / Mux11 */
static const struct soc_enum dac_hs2_enum =
SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 2, 4, dac_text);
static const struct snd_kcontrol_new dac_hs2_mux =
SOC_DAPM_ENUM("DAC HS2 Mux", dac_hs2_enum);
/* DAC Lineout 1 Mux / Mux12 */
static const struct soc_enum dac_lo1_enum =
SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 4, 4, dac_text);
static const struct snd_kcontrol_new dac_lo1_mux =
SOC_DAPM_ENUM("DAC LO1 Mux", dac_lo1_enum);
/* DAC Lineout 2 Mux / Mux13 */
static const struct soc_enum dac_lo2_enum =
SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 6, 4, dac_text);
static const struct snd_kcontrol_new dac_lo2_mux =
SOC_DAPM_ENUM("DAC LO2 Mux", dac_lo2_enum);
/* DAC Spearker Earphone Mux / Mux14 */
static const struct soc_enum dac_spk_ear_enum =
SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_2, 0, 4, dac_text);
static const struct snd_kcontrol_new dac_spk_ear_mux =
SOC_DAPM_ENUM("DAC SP Mux", dac_spk_ear_enum);
/* Headset 1 Mux / Mux15 */
static const char *in_text[] = {
"Digital", "Analog",
};
static const struct soc_enum hs1_enum =
SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 0, 2, in_text);
static const struct snd_kcontrol_new hs1_mux =
SOC_DAPM_ENUM("Headset1 Mux", hs1_enum);
/* Headset 2 Mux / Mux16 */
static const struct soc_enum hs2_enum =
SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 1, 2, in_text);
static const struct snd_kcontrol_new hs2_mux =
SOC_DAPM_ENUM("Headset2 Mux", hs2_enum);
/* Lineout 1 Mux / Mux17 */
static const struct soc_enum lo1_enum =
SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 2, 2, in_text);
static const struct snd_kcontrol_new lo1_mux =
SOC_DAPM_ENUM("Lineout1 Mux", lo1_enum);
/* Lineout 2 Mux / Mux18 */
static const struct soc_enum lo2_enum =
SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 3, 2, in_text);
static const struct snd_kcontrol_new lo2_mux =
SOC_DAPM_ENUM("Lineout2 Mux", lo2_enum);
/* Speaker Earpiece Demux */
static const char *spk_text[] = {
"Earpiece", "Speaker",
};
static const struct soc_enum spk_enum =
SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 6, 2, spk_text);
static const struct snd_kcontrol_new spk_demux =
SOC_DAPM_ENUM("Speaker Earpiece Demux", spk_enum);
/* MIC Mux / Mux1 */
static const char *mic_text[] = {
"Mic 1", "Mic 2",
};
static const struct soc_enum mic_enum =
SOC_ENUM_SINGLE(PM860X_ADC_ANA_4, 4, 2, mic_text);
static const struct snd_kcontrol_new mic_mux =
SOC_DAPM_ENUM("MIC Mux", mic_enum);
static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = {
SND_SOC_DAPM_AIF_IN("PCM SDI", "PCM Playback", 0,
PM860X_ADC_EN_2, 0, 0),
SND_SOC_DAPM_AIF_OUT("PCM SDO", "PCM Capture", 0,
PM860X_PCM_IFACE_3, 1, 1),
SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0,
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0,
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0,
PM860X_I2S_IFACE_3, 5, 1),
SND_SOC_DAPM_SUPPLY("I2S CLK", PM860X_DAC_EN_2, 0, 0, NULL, 0),
SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux),
SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux),
SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux),
SND_SOC_DAPM_MUX("EC Mux", SND_SOC_NOPM, 0, 0, &ec_mux),
SND_SOC_DAPM_MUX("ADCR EC Mux", SND_SOC_NOPM, 0, 0, &adcr_ec_mux),
SND_SOC_DAPM_SWITCH("Left EPA", SND_SOC_NOPM, 0, 0,
&lepa_switch_controls),
SND_SOC_DAPM_SWITCH("Right EPA", SND_SOC_NOPM, 0, 0,
&repa_switch_controls),
SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Left ADC MOD", PM860X_ADC_EN_1,
0, 1, 1, 0),
SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Right ADC MOD", PM860X_ADC_EN_1,
1, 1, 1, 0),
SND_SOC_DAPM_ADC("Left ADC", NULL, PM860X_ADC_EN_2, 5, 0),
SND_SOC_DAPM_ADC("Right ADC", NULL, PM860X_ADC_EN_2, 4, 0),
SND_SOC_DAPM_SWITCH("AUX1 Switch", SND_SOC_NOPM, 0, 0,
&aux1_switch_controls),
SND_SOC_DAPM_SWITCH("AUX2 Switch", SND_SOC_NOPM, 0, 0,
&aux2_switch_controls),
SND_SOC_DAPM_MUX("MIC Mux", SND_SOC_NOPM, 0, 0, &mic_mux),
SND_SOC_DAPM_MICBIAS("Mic1 Bias", PM860X_ADC_ANA_1, 2, 0),
SND_SOC_DAPM_MICBIAS("Mic3 Bias", PM860X_ADC_ANA_1, 7, 0),
SND_SOC_DAPM_PGA("MIC1 Volume", PM860X_ADC_EN_1, 2, 0, NULL, 0),
SND_SOC_DAPM_PGA("MIC3 Volume", PM860X_ADC_EN_1, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("AUX1 Volume", PM860X_ADC_EN_1, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("AUX2 Volume", PM860X_ADC_EN_1, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("Sidetone PGA", PM860X_ADC_EN_2, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("Lofi PGA", PM860X_ADC_EN_2, 2, 0, NULL, 0),
SND_SOC_DAPM_INPUT("AUX1"),
SND_SOC_DAPM_INPUT("AUX2"),
SND_SOC_DAPM_INPUT("MIC1P"),
SND_SOC_DAPM_INPUT("MIC1N"),
SND_SOC_DAPM_INPUT("MIC2P"),
SND_SOC_DAPM_INPUT("MIC2N"),
SND_SOC_DAPM_INPUT("MIC3P"),
SND_SOC_DAPM_INPUT("MIC3N"),
SND_SOC_DAPM_DAC_E("Left DAC", NULL, SND_SOC_NOPM, 0, 0,
pm860x_dac_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_DAC_E("Right DAC", NULL, SND_SOC_NOPM, 0, 0,
pm860x_dac_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_MUX("I2S DIN Mux", SND_SOC_NOPM, 0, 0, &i2s_din_mux),
SND_SOC_DAPM_MUX("DAC HS1 Mux", SND_SOC_NOPM, 0, 0, &dac_hs1_mux),
SND_SOC_DAPM_MUX("DAC HS2 Mux", SND_SOC_NOPM, 0, 0, &dac_hs2_mux),
SND_SOC_DAPM_MUX("DAC LO1 Mux", SND_SOC_NOPM, 0, 0, &dac_lo1_mux),
SND_SOC_DAPM_MUX("DAC LO2 Mux", SND_SOC_NOPM, 0, 0, &dac_lo2_mux),
SND_SOC_DAPM_MUX("DAC SP Mux", SND_SOC_NOPM, 0, 0, &dac_spk_ear_mux),
SND_SOC_DAPM_MUX("Headset1 Mux", SND_SOC_NOPM, 0, 0, &hs1_mux),
SND_SOC_DAPM_MUX("Headset2 Mux", SND_SOC_NOPM, 0, 0, &hs2_mux),
SND_SOC_DAPM_MUX("Lineout1 Mux", SND_SOC_NOPM, 0, 0, &lo1_mux),
SND_SOC_DAPM_MUX("Lineout2 Mux", SND_SOC_NOPM, 0, 0, &lo2_mux),
SND_SOC_DAPM_MUX("Speaker Earpiece Demux", SND_SOC_NOPM, 0, 0,
&spk_demux),
SND_SOC_DAPM_PGA("Headset1 PGA", PM860X_DAC_EN_1, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Headset2 PGA", PM860X_DAC_EN_1, 1, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("HS1"),
SND_SOC_DAPM_OUTPUT("HS2"),
SND_SOC_DAPM_PGA("Lineout1 PGA", PM860X_DAC_EN_1, 2, 0, NULL, 0),
SND_SOC_DAPM_PGA("Lineout2 PGA", PM860X_DAC_EN_1, 3, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LINEOUT1"),
SND_SOC_DAPM_OUTPUT("LINEOUT2"),
SND_SOC_DAPM_PGA("Earpiece PGA", PM860X_DAC_EN_1, 4, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("EARP"),
SND_SOC_DAPM_OUTPUT("EARN"),
SND_SOC_DAPM_PGA("Speaker PGA", PM860X_DAC_EN_1, 5, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LSP"),
SND_SOC_DAPM_OUTPUT("LSN"),
SND_SOC_DAPM_REG(snd_soc_dapm_supply, "VCODEC", PM860X_AUDIO_SUPPLIES_2,
0, SUPPLY_MASK, SUPPLY_MASK, 0),
PM860X_DAPM_OUTPUT("RSYNC", pm860x_rsync_event),
};
static const struct snd_soc_dapm_route audio_map[] = {
/* supply */
{"Left DAC", NULL, "VCODEC"},
{"Right DAC", NULL, "VCODEC"},
{"Left ADC", NULL, "VCODEC"},
{"Right ADC", NULL, "VCODEC"},
{"Left ADC", NULL, "Left ADC MOD"},
{"Right ADC", NULL, "Right ADC MOD"},
/* I2S Clock */
{"I2S DIN", NULL, "I2S CLK"},
{"I2S DIN1", NULL, "I2S CLK"},
{"I2S DOUT", NULL, "I2S CLK"},
/* PCM/AIF1 Inputs */
{"PCM SDO", NULL, "ADC Left Mux"},
{"PCM SDO", NULL, "ADCR EC Mux"},
/* PCM/AFI2 Outputs */
{"Lofi PGA", NULL, "PCM SDI"},
{"Lofi PGA", NULL, "Sidetone PGA"},
{"Left DAC", NULL, "Lofi PGA"},
{"Right DAC", NULL, "Lofi PGA"},
/* I2S/AIF2 Inputs */
{"MIC Mux", "Mic 1", "MIC1P"},
{"MIC Mux", "Mic 1", "MIC1N"},
{"MIC Mux", "Mic 2", "MIC2P"},
{"MIC Mux", "Mic 2", "MIC2N"},
{"MIC1 Volume", NULL, "MIC Mux"},
{"MIC3 Volume", NULL, "MIC3P"},
{"MIC3 Volume", NULL, "MIC3N"},
{"Left ADC", NULL, "MIC1 Volume"},
{"Right ADC", NULL, "MIC3 Volume"},
{"ADC Left Mux", "ADCR", "Right ADC"},
{"ADC Left Mux", "ADCL", "Left ADC"},
{"ADC Right Mux", "ADCL", "Left ADC"},
{"ADC Right Mux", "ADCR", "Right ADC"},
{"Left EPA", "Switch", "Left DAC"},
{"Right EPA", "Switch", "Right DAC"},
{"EC Mux", "Left", "Left DAC"},
{"EC Mux", "Right", "Right DAC"},
{"EC Mux", "Left + Right", "Left DAC"},
{"EC Mux", "Left + Right", "Right DAC"},
{"ADCR EC Mux", "ADCR", "ADC Right Mux"},
{"ADCR EC Mux", "EC", "EC Mux"},
{"I2S Mic Mux", "Ex PA", "Left EPA"},
{"I2S Mic Mux", "Ex PA", "Right EPA"},
{"I2S Mic Mux", "ADC", "ADC Left Mux"},
{"I2S Mic Mux", "ADC", "ADCR EC Mux"},
{"I2S DOUT", NULL, "I2S Mic Mux"},
/* I2S/AIF2 Outputs */
{"I2S DIN Mux", "DIN", "I2S DIN"},
{"I2S DIN Mux", "DIN1", "I2S DIN1"},
{"Left DAC", NULL, "I2S DIN Mux"},
{"Right DAC", NULL, "I2S DIN Mux"},
{"DAC HS1 Mux", "Left", "Left DAC"},
{"DAC HS1 Mux", "Right", "Right DAC"},
{"DAC HS2 Mux", "Left", "Left DAC"},
{"DAC HS2 Mux", "Right", "Right DAC"},
{"DAC LO1 Mux", "Left", "Left DAC"},
{"DAC LO1 Mux", "Right", "Right DAC"},
{"DAC LO2 Mux", "Left", "Left DAC"},
{"DAC LO2 Mux", "Right", "Right DAC"},
{"Headset1 Mux", "Digital", "DAC HS1 Mux"},
{"Headset2 Mux", "Digital", "DAC HS2 Mux"},
{"Lineout1 Mux", "Digital", "DAC LO1 Mux"},
{"Lineout2 Mux", "Digital", "DAC LO2 Mux"},
{"Headset1 PGA", NULL, "Headset1 Mux"},
{"Headset2 PGA", NULL, "Headset2 Mux"},
{"Lineout1 PGA", NULL, "Lineout1 Mux"},
{"Lineout2 PGA", NULL, "Lineout2 Mux"},
{"DAC SP Mux", "Left", "Left DAC"},
{"DAC SP Mux", "Right", "Right DAC"},
{"Speaker Earpiece Demux", "Speaker", "DAC SP Mux"},
{"Speaker PGA", NULL, "Speaker Earpiece Demux"},
{"Earpiece PGA", NULL, "Speaker Earpiece Demux"},
{"RSYNC", NULL, "Headset1 PGA"},
{"RSYNC", NULL, "Headset2 PGA"},
{"RSYNC", NULL, "Lineout1 PGA"},
{"RSYNC", NULL, "Lineout2 PGA"},
{"RSYNC", NULL, "Speaker PGA"},
{"RSYNC", NULL, "Speaker PGA"},
{"RSYNC", NULL, "Earpiece PGA"},
{"RSYNC", NULL, "Earpiece PGA"},
{"HS1", NULL, "RSYNC"},
{"HS2", NULL, "RSYNC"},
{"LINEOUT1", NULL, "RSYNC"},
{"LINEOUT2", NULL, "RSYNC"},
{"LSP", NULL, "RSYNC"},
{"LSN", NULL, "RSYNC"},
{"EARP", NULL, "RSYNC"},
{"EARN", NULL, "RSYNC"},
};
/*
* Use MUTE_LEFT & MUTE_RIGHT to implement digital mute.
* These bits can also be used to mute.
*/
static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
int data = 0, mask = MUTE_LEFT | MUTE_RIGHT;
if (mute)
data = mask;
snd_soc_update_bits(codec, PM860X_DAC_OFFSET, mask, data);
snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
RSYNC_CHANGE, RSYNC_CHANGE);
return 0;
}
static int pm860x_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
unsigned char inf = 0, mask = 0;
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
inf &= ~PCM_INF2_18WL;
break;
case SNDRV_PCM_FORMAT_S18_3LE:
inf |= PCM_INF2_18WL;
break;
default:
return -EINVAL;
}
mask |= PCM_INF2_18WL;
snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf);
/* sample rate */
switch (params_rate(params)) {
case 8000:
inf = 0;
break;
case 16000:
inf = 3;
break;
case 32000:
inf = 6;
break;
case 48000:
inf = 8;
break;
default:
return -EINVAL;
}
snd_soc_update_bits(codec, PM860X_PCM_RATE, 0x0f, inf);
return 0;
}
static int pm860x_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
unsigned char inf = 0, mask = 0;
int ret = -EINVAL;
mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
case SND_SOC_DAIFMT_CBM_CFS:
if (pm860x->dir == PM860X_CLK_DIR_OUT) {
inf |= PCM_INF2_MASTER;
ret = 0;
}
break;
case SND_SOC_DAIFMT_CBS_CFS:
if (pm860x->dir == PM860X_CLK_DIR_IN) {
inf &= ~PCM_INF2_MASTER;
ret = 0;
}
break;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
inf |= PCM_EXACT_I2S;
ret = 0;
break;
}
mask |= PCM_MODE_MASK;
if (ret)
return ret;
snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf);
return 0;
}
static int pm860x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
if (dir == PM860X_CLK_DIR_OUT)
pm860x->dir = PM860X_CLK_DIR_OUT;
else {
pm860x->dir = PM860X_CLK_DIR_IN;
return -EINVAL;
}
return 0;
}
static int pm860x_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
unsigned char inf;
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
inf = 0;
break;
case SNDRV_PCM_FORMAT_S18_3LE:
inf = PCM_INF2_18WL;
break;
default:
return -EINVAL;
}
snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, PCM_INF2_18WL, inf);
/* sample rate */
switch (params_rate(params)) {
case 8000:
inf = 0;
break;
case 11025:
inf = 1;
break;
case 16000:
inf = 3;
break;
case 22050:
inf = 4;
break;
case 32000:
inf = 6;
break;
case 44100:
inf = 7;
break;
case 48000:
inf = 8;
break;
default:
return -EINVAL;
}
snd_soc_update_bits(codec, PM860X_I2S_IFACE_4, 0xf, inf);
return 0;
}
static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
unsigned char inf = 0, mask = 0;
mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
if (pm860x->dir == PM860X_CLK_DIR_OUT)
inf |= PCM_INF2_MASTER;
else
return -EINVAL;
break;
case SND_SOC_DAIFMT_CBS_CFS:
if (pm860x->dir == PM860X_CLK_DIR_IN)
inf &= ~PCM_INF2_MASTER;
else
return -EINVAL;
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
inf |= PCM_EXACT_I2S;
break;
default:
return -EINVAL;
}
mask |= PCM_MODE_MASK;
snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, mask, inf);
return 0;
}
static int pm860x_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
int data;
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Enable Audio PLL & Audio section */
data = AUDIO_PLL | AUDIO_SECTION_ON;
pm860x_reg_write(codec->control_data, REG_MISC2, data);
udelay(300);
data = AUDIO_PLL | AUDIO_SECTION_RESET
| AUDIO_SECTION_ON;
pm860x_reg_write(codec->control_data, REG_MISC2, data);
}
break;
case SND_SOC_BIAS_OFF:
data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON;
pm860x_set_bits(codec->control_data, REG_MISC2, data, 0);
break;
}
codec->dapm.bias_level = level;
return 0;
}
static const struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
.digital_mute = pm860x_digital_mute,
.hw_params = pm860x_pcm_hw_params,
.set_fmt = pm860x_pcm_set_dai_fmt,
.set_sysclk = pm860x_set_dai_sysclk,
};
static const struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
.digital_mute = pm860x_digital_mute,
.hw_params = pm860x_i2s_hw_params,
.set_fmt = pm860x_i2s_set_dai_fmt,
.set_sysclk = pm860x_set_dai_sysclk,
};
#define PM860X_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \
SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000)
static struct snd_soc_dai_driver pm860x_dai[] = {
{
/* DAI PCM */
.name = "88pm860x-pcm",
.id = 1,
.playback = {
.stream_name = "PCM Playback",
.channels_min = 2,
.channels_max = 2,
.rates = PM860X_RATES,
.formats = SNDRV_PCM_FORMAT_S16_LE | \
SNDRV_PCM_FORMAT_S18_3LE,
},
.capture = {
.stream_name = "PCM Capture",
.channels_min = 2,
.channels_max = 2,
.rates = PM860X_RATES,
.formats = SNDRV_PCM_FORMAT_S16_LE | \
SNDRV_PCM_FORMAT_S18_3LE,
},
.ops = &pm860x_pcm_dai_ops,
}, {
/* DAI I2S */
.name = "88pm860x-i2s",
.id = 2,
.playback = {
.stream_name = "I2S Playback",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FORMAT_S16_LE | \
SNDRV_PCM_FORMAT_S18_3LE,
},
.capture = {
.stream_name = "I2S Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FORMAT_S16_LE | \
SNDRV_PCM_FORMAT_S18_3LE,
},
.ops = &pm860x_i2s_dai_ops,
},
};
static irqreturn_t pm860x_codec_handler(int irq, void *data)
{
struct pm860x_priv *pm860x = data;
int status, shrt, report = 0, mic_report = 0;
int mask;
status = pm860x_reg_read(pm860x->i2c, REG_STATUS_1);
shrt = pm860x_reg_read(pm860x->i2c, REG_SHORTS);
mask = pm860x->det.hs_shrt | pm860x->det.hook_det | pm860x->det.lo_shrt
| pm860x->det.hp_det;
#ifndef CONFIG_SND_SOC_88PM860X_MODULE
if (status & (HEADSET_STATUS | MIC_STATUS | SHORT_HS1 | SHORT_HS2 |
SHORT_LO1 | SHORT_LO2))
trace_snd_soc_jack_irq(dev_name(pm860x->codec->dev));
#endif
if ((pm860x->det.hp_det & SND_JACK_HEADPHONE)
&& (status & HEADSET_STATUS))
report |= SND_JACK_HEADPHONE;
if ((pm860x->det.mic_det & SND_JACK_MICROPHONE)
&& (status & MIC_STATUS))
mic_report |= SND_JACK_MICROPHONE;
if (pm860x->det.hs_shrt && (shrt & (SHORT_HS1 | SHORT_HS2)))
report |= pm860x->det.hs_shrt;
if (pm860x->det.hook_det && (status & HOOK_STATUS))
report |= pm860x->det.hook_det;
if (pm860x->det.lo_shrt && (shrt & (SHORT_LO1 | SHORT_LO2)))
report |= pm860x->det.lo_shrt;
if (report)
snd_soc_jack_report(pm860x->det.hp_jack, report, mask);
if (mic_report)
snd_soc_jack_report(pm860x->det.mic_jack, SND_JACK_MICROPHONE,
SND_JACK_MICROPHONE);
dev_dbg(pm860x->codec->dev, "headphone report:0x%x, mask:%x\n",
report, mask);
dev_dbg(pm860x->codec->dev, "microphone report:0x%x\n", mic_report);
return IRQ_HANDLED;
}
int pm860x_hs_jack_detect(struct snd_soc_codec *codec,
struct snd_soc_jack *jack,
int det, int hook, int hs_shrt, int lo_shrt)
{
struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
int data;
pm860x->det.hp_jack = jack;
pm860x->det.hp_det = det;
pm860x->det.hook_det = hook;
pm860x->det.hs_shrt = hs_shrt;
pm860x->det.lo_shrt = lo_shrt;
if (det & SND_JACK_HEADPHONE)
pm860x_set_bits(codec->control_data, REG_HS_DET,
EN_HS_DET, EN_HS_DET);
/* headset short detect */
if (hs_shrt) {
data = CLR_SHORT_HS2 | CLR_SHORT_HS1;
pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
}
/* Lineout short detect */
if (lo_shrt) {
data = CLR_SHORT_LO2 | CLR_SHORT_LO1;
pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
}
/* sync status */
pm860x_codec_handler(0, pm860x);
return 0;
}
EXPORT_SYMBOL_GPL(pm860x_hs_jack_detect);
int pm860x_mic_jack_detect(struct snd_soc_codec *codec,
struct snd_soc_jack *jack, int det)
{
struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
pm860x->det.mic_jack = jack;
pm860x->det.mic_det = det;
if (det & SND_JACK_MICROPHONE)
pm860x_set_bits(codec->control_data, REG_MIC_DET,
MICDET_MASK, MICDET_MASK);
/* sync status */
pm860x_codec_handler(0, pm860x);
return 0;
}
EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect);
static int pm860x_probe(struct snd_soc_codec *codec)
{
struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
int i, ret;
pm860x->codec = codec;
codec->control_data = pm860x->i2c;
for (i = 0; i < 4; i++) {
ret = request_threaded_irq(pm860x->irq[i], NULL,
pm860x_codec_handler, IRQF_ONESHOT,
pm860x->name[i], pm860x);
if (ret < 0) {
dev_err(codec->dev, "Failed to request IRQ!\n");
goto out;
}
}
pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE,
REG_CACHE_SIZE, codec->reg_cache);
if (ret < 0) {
dev_err(codec->dev, "Failed to fill register cache: %d\n",
ret);
goto out;
}
snd_soc_add_controls(codec, pm860x_snd_controls,
ARRAY_SIZE(pm860x_snd_controls));
snd_soc_dapm_new_controls(dapm, pm860x_dapm_widgets,
ARRAY_SIZE(pm860x_dapm_widgets));
snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
out:
while (--i >= 0)
free_irq(pm860x->irq[i], pm860x);
return ret;
}
static int pm860x_remove(struct snd_soc_codec *codec)
{
struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
int i;
for (i = 3; i >= 0; i--)
free_irq(pm860x->irq[i], pm860x);
pm860x_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_pm860x = {
.probe = pm860x_probe,
.remove = pm860x_remove,
.read = pm860x_read_reg_cache,
.write = pm860x_write_reg_cache,
.reg_cache_size = REG_CACHE_SIZE,
.reg_word_size = sizeof(u8),
.set_bias_level = pm860x_set_bias_level,
};
static int __devinit pm860x_codec_probe(struct platform_device *pdev)
{
struct pm860x_chip *chip = dev_get_drvdata(pdev->dev.parent);
struct pm860x_priv *pm860x;
struct resource *res;
int i, ret;
pm860x = kzalloc(sizeof(struct pm860x_priv), GFP_KERNEL);
if (pm860x == NULL)
return -ENOMEM;
pm860x->chip = chip;
pm860x->i2c = (chip->id == CHIP_PM8607) ? chip->client
: chip->companion;
platform_set_drvdata(pdev, pm860x);
for (i = 0; i < 4; i++) {
res = platform_get_resource(pdev, IORESOURCE_IRQ, i);
if (!res) {
dev_err(&pdev->dev, "Failed to get IRQ resources\n");
goto out;
}
pm860x->irq[i] = res->start + chip->irq_base;
strncpy(pm860x->name[i], res->name, MAX_NAME_LEN);
}
ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pm860x,
pm860x_dai, ARRAY_SIZE(pm860x_dai));
if (ret) {
dev_err(&pdev->dev, "Failed to register codec\n");
goto out;
}
return ret;
out:
platform_set_drvdata(pdev, NULL);
kfree(pm860x);
return -EINVAL;
}
static int __devexit pm860x_codec_remove(struct platform_device *pdev)
{
struct pm860x_priv *pm860x = platform_get_drvdata(pdev);
snd_soc_unregister_codec(&pdev->dev);
platform_set_drvdata(pdev, NULL);
kfree(pm860x);
return 0;
}
static struct platform_driver pm860x_codec_driver = {
.driver = {
.name = "88pm860x-codec",
.owner = THIS_MODULE,
},
.probe = pm860x_codec_probe,
.remove = __devexit_p(pm860x_codec_remove),
};
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");
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