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ac101.c
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ac101.c
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/*
* ac101.c
*
* (C) Copyright 2017-2018
* Seeed Technology Co., Ltd. <www.seeedstudio.com>
*
* PeterYang <[email protected]>
*
* (C) Copyright 2014-2017
* Reuuimlla Technology Co., Ltd. <www.reuuimllatech.com>
*
* huangxin <[email protected]>
* liushaohua <[email protected]>
*
* X-Powers AC101 codec driver
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License as
* published by the Free Software Foundation; either version 2 of
* the License, or (at your option) any later version.
*
*/
/* #undef AC101_DEBG
* use 'make DEBUG=1' to enable debugging
*/
#include <linux/module.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
#include <linux/i2c.h>
#include <linux/irq.h>
#include <linux/workqueue.h>
#include <linux/clk.h>
#include <linux/gpio/consumer.h>
#include <linux/regmap.h>
#include <linux/input.h>
#include <linux/delay.h>
#include "ac101_regs.h"
#include "ac10x.h"
/*
* *** To sync channels ***
*
* 1. disable clock in codec hw_params()
* 2. clear fifo in bcm2835 hw_params()
* 3. clear fifo in bcm2385 prepare()
* 4. enable RX in bcm2835 trigger()
* 5. enable clock in machine trigger()
*/
/*Default initialize configuration*/
static bool speaker_double_used = 1;
static int double_speaker_val = 0x1B;
static int single_speaker_val = 0x19;
static int headset_val = 0x3B;
static int mainmic_val = 0x4;
static int headsetmic_val = 0x4;
static bool dmic_used = 0;
static int adc_digital_val = 0xb0b0;
static bool drc_used = false;
#define AC101_RATES (SNDRV_PCM_RATE_8000_96000 & \
~(SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_64000 | \
SNDRV_PCM_RATE_88200))
#define AC101_FORMATS (/*SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE |*/ \
SNDRV_PCM_FMTBIT_S32_LE | \
0)
static struct ac10x_priv* static_ac10x;
int ac101_read(struct snd_soc_codec *codec, unsigned reg) {
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
int r, v = 0;
if ((r = regmap_read(ac10x->regmap101, reg, &v)) < 0) {
dev_err(codec->dev, "read reg %02X fail\n",
reg);
return r;
}
return v;
}
int ac101_write(struct snd_soc_codec *codec, unsigned reg, unsigned val) {
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
int v;
v = regmap_write(ac10x->regmap101, reg, val);
return v;
}
int ac101_update_bits(struct snd_soc_codec *codec, unsigned reg,
unsigned mask, unsigned value
) {
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
int v;
v = regmap_update_bits(ac10x->regmap101, reg, mask, value);
return v;
}
#ifdef CONFIG_AC101_SWITCH_DETECT
/******************************************************************************/
/********************************switch****************************************/
/******************************************************************************/
#define KEY_HEADSETHOOK 226 /* key define */
#define HEADSET_FILTER_CNT (10)
/*
* switch_hw_config:config the 53 codec register
*/
static void switch_hw_config(struct snd_soc_codec *codec)
{
int r;
AC101_DBG();
/*HMIC/MMIC BIAS voltage level select:2.5v*/
ac101_update_bits(codec, OMIXER_BST1_CTRL, (0xf<<BIASVOLTAGE), (0xf<<BIASVOLTAGE));
/*debounce when Key down or keyup*/
ac101_update_bits(codec, HMIC_CTRL1, (0xf<<HMIC_M), (0x0<<HMIC_M));
/*debounce when earphone plugin or pullout*/
ac101_update_bits(codec, HMIC_CTRL1, (0xf<<HMIC_N), (0x0<<HMIC_N));
/*Down Sample Setting Select: Downby 4,32Hz*/
ac101_update_bits(codec, HMIC_CTRL2, (0x3<<HMIC_SAMPLE_SELECT), (0x02<<HMIC_SAMPLE_SELECT));
/*Hmic_th2 for detecting Keydown or Keyup.*/
ac101_update_bits(codec, HMIC_CTRL2, (0x1f<<HMIC_TH2), (0x8<<HMIC_TH2));
/*Hmic_th1[4:0],detecting eraphone plugin or pullout*/
ac101_update_bits(codec, HMIC_CTRL2, (0x1f<<HMIC_TH1), (0x1<<HMIC_TH1));
/*Headset microphone BIAS working mode: when HBIASEN = 1 */
ac101_update_bits(codec, ADC_APC_CTRL, (0x1<<HBIASMOD), (0x1<<HBIASMOD));
/*Headset microphone BIAS Enable*/
ac101_update_bits(codec, ADC_APC_CTRL, (0x1<<HBIASEN), (0x1<<HBIASEN));
/*Headset microphone BIAS Current sensor & ADC Enable*/
ac101_update_bits(codec, ADC_APC_CTRL, (0x1<<HBIASADCEN), (0x1<<HBIASADCEN));
/*Earphone Plugin/out Irq Enable*/
ac101_update_bits(codec, HMIC_CTRL1, (0x1<<HMIC_PULLOUT_IRQ), (0x1<<HMIC_PULLOUT_IRQ));
ac101_update_bits(codec, HMIC_CTRL1, (0x1<<HMIC_PLUGIN_IRQ), (0x1<<HMIC_PLUGIN_IRQ));
/*Hmic KeyUp/key down Irq Enable*/
ac101_update_bits(codec, HMIC_CTRL1, (0x1<<HMIC_KEYDOWN_IRQ), (0x1<<HMIC_KEYDOWN_IRQ));
ac101_update_bits(codec, HMIC_CTRL1, (0x1<<HMIC_KEYUP_IRQ), (0x1<<HMIC_KEYUP_IRQ));
/*headphone calibration clock frequency select*/
ac101_update_bits(codec, SPKOUT_CTRL, (0x7<<HPCALICKS), (0x7<<HPCALICKS));
/*clear hmic interrupt */
r = HMIC_PEND_ALL;
ac101_write(codec, HMIC_STS, r);
return;
}
/*
* switch_status_update: update the switch state.
*/
static void switch_status_update(struct ac10x_priv *ac10x)
{
AC101_DBG("ac10x->state:%d\n", ac10x->state);
input_report_switch(ac10x->inpdev, SW_HEADPHONE_INSERT, ac10x->state);
input_sync(ac10x->inpdev);
return;
}
/*
* work_cb_clear_irq: clear audiocodec pending and Record the interrupt.
*/
static void work_cb_clear_irq(struct work_struct *work)
{
int reg_val = 0;
struct ac10x_priv *ac10x = container_of(work, struct ac10x_priv, work_clear_irq);
struct snd_soc_codec *codec = ac10x->codec;
ac10x->irq_cntr++;
reg_val = ac101_read(codec, HMIC_STS);
if (BIT(HMIC_PULLOUT_PEND) & reg_val) {
ac10x->pullout_cntr++;
AC101_DBG("ac10x->pullout_cntr: %d\n", ac10x->pullout_cntr);
}
reg_val |= HMIC_PEND_ALL;
ac101_write(codec, HMIC_STS, reg_val);
reg_val = ac101_read(codec, HMIC_STS);
if ((reg_val & HMIC_PEND_ALL) != 0){
reg_val |= HMIC_PEND_ALL;
ac101_write(codec, HMIC_STS, reg_val);
}
if (cancel_work_sync(&ac10x->work_switch) != 0) {
ac10x->irq_cntr--;
}
if (0 == schedule_work(&ac10x->work_switch)) {
ac10x->irq_cntr--;
AC101_DBG("[work_cb_clear_irq] add work struct failed!\n");
}
}
enum {
HBIAS_LEVEL_1 = 0x02,
HBIAS_LEVEL_2 = 0x0B,
HBIAS_LEVEL_3 = 0x13,
HBIAS_LEVEL_4 = 0x17,
HBIAS_LEVEL_5 = 0x19,
};
static int __ac101_get_hmic_data(struct snd_soc_codec *codec) {
#ifdef AC101_DEBG
static long counter;
#endif
int r, d;
d = GET_HMIC_DATA(ac101_read(codec, HMIC_STS));
r = 0x1 << HMIC_DATA_PEND;
ac101_write(codec, HMIC_STS, r);
/* prevent i2c accessing too frequently */
usleep_range(1500, 3000);
AC101_DBG("HMIC_DATA(%3ld): %02X\n", counter++, d);
return d;
}
/*
* work_cb_earphone_switch: judge the status of the headphone
*/
static void work_cb_earphone_switch(struct work_struct *work)
{
struct ac10x_priv *ac10x = container_of(work, struct ac10x_priv, work_switch);
struct snd_soc_codec *codec = ac10x->codec;
static int hook_flag1 = 0, hook_flag2 = 0;
static int KEY_VOLUME_FLAG = 0;
unsigned filter_buf = 0;
int filt_index = 0;
int t = 0;
ac10x->irq_cntr--;
/* read HMIC_DATA */
t = __ac101_get_hmic_data(codec);
if ((t >= HBIAS_LEVEL_2) && (ac10x->mode == FOUR_HEADPHONE_PLUGIN)) {
t = __ac101_get_hmic_data(codec);
if (t >= HBIAS_LEVEL_5){
msleep(150);
t = __ac101_get_hmic_data(codec);
if (((t < HBIAS_LEVEL_2 && t >= HBIAS_LEVEL_1 - 1) || t >= HBIAS_LEVEL_5)
&& (ac10x->pullout_cntr == 0)) {
input_report_key(ac10x->inpdev, KEY_HEADSETHOOK, 1);
input_sync(ac10x->inpdev);
AC101_DBG("KEY_HEADSETHOOK1\n");
if (hook_flag1 != hook_flag2)
hook_flag1 = hook_flag2 = 0;
hook_flag1++;
}
if (ac10x->pullout_cntr)
ac10x->pullout_cntr--;
} else if (t >= HBIAS_LEVEL_4) {
msleep(80);
t = __ac101_get_hmic_data(codec);
if (t < HBIAS_LEVEL_5 && t >= HBIAS_LEVEL_4 && (ac10x->pullout_cntr == 0)) {
KEY_VOLUME_FLAG = 1;
input_report_key(ac10x->inpdev, KEY_VOLUMEUP, 1);
input_sync(ac10x->inpdev);
input_report_key(ac10x->inpdev, KEY_VOLUMEUP, 0);
input_sync(ac10x->inpdev);
AC101_DBG("HMIC_DATA: %d KEY_VOLUMEUP\n", t);
}
if (ac10x->pullout_cntr)
ac10x->pullout_cntr--;
} else if (t >= HBIAS_LEVEL_3){
msleep(80);
t = __ac101_get_hmic_data(codec);
if (t < HBIAS_LEVEL_4 && t >= HBIAS_LEVEL_3 && (ac10x->pullout_cntr == 0)){
KEY_VOLUME_FLAG = 1;
input_report_key(ac10x->inpdev, KEY_VOLUMEDOWN, 1);
input_sync(ac10x->inpdev);
input_report_key(ac10x->inpdev, KEY_VOLUMEDOWN, 0);
input_sync(ac10x->inpdev);
AC101_DBG("KEY_VOLUMEDOWN\n");
}
if (ac10x->pullout_cntr)
ac10x->pullout_cntr--;
}
} else if ((t < HBIAS_LEVEL_2 && t >= HBIAS_LEVEL_1) && (ac10x->mode == FOUR_HEADPHONE_PLUGIN)) {
t = __ac101_get_hmic_data(codec);
if (t < HBIAS_LEVEL_2 && t >= HBIAS_LEVEL_1) {
if (KEY_VOLUME_FLAG) {
KEY_VOLUME_FLAG = 0;
}
if (hook_flag1 == (++hook_flag2)) {
hook_flag1 = hook_flag2 = 0;
input_report_key(ac10x->inpdev, KEY_HEADSETHOOK, 0);
input_sync(ac10x->inpdev);
AC101_DBG("KEY_HEADSETHOOK0\n");
}
}
} else {
while (ac10x->irq_cntr == 0 && ac10x->irq != 0) {
msleep(20);
t = __ac101_get_hmic_data(codec);
if (filt_index <= HEADSET_FILTER_CNT) {
if (filt_index++ == 0) {
filter_buf = t;
} else if (filter_buf != t) {
filt_index = 0;
}
continue;
}
filt_index = 0;
if (filter_buf >= HBIAS_LEVEL_2) {
ac10x->mode = THREE_HEADPHONE_PLUGIN;
ac10x->state = 2;
} else if (filter_buf >= HBIAS_LEVEL_1 - 1) {
ac10x->mode = FOUR_HEADPHONE_PLUGIN;
ac10x->state = 1;
} else {
ac10x->mode = HEADPHONE_IDLE;
ac10x->state = 0;
}
switch_status_update(ac10x);
ac10x->pullout_cntr = 0;
break;
}
}
}
/*
* audio_hmic_irq: the interrupt handlers
*/
static irqreturn_t audio_hmic_irq(int irq, void *para)
{
struct ac10x_priv *ac10x = (struct ac10x_priv *)para;
if (ac10x == NULL) {
return -EINVAL;
}
if (0 == schedule_work(&ac10x->work_clear_irq)){
AC101_DBG("[audio_hmic_irq] work already in queue_codec_irq, adding failed!\n");
}
return IRQ_HANDLED;
}
static int ac101_switch_probe(struct ac10x_priv *ac10x) {
struct i2c_client *i2c = ac10x->i2c101;
long ret;
ac10x->gpiod_irq = devm_gpiod_get_optional(&i2c->dev, "switch-irq", GPIOD_IN);
if (IS_ERR(ac10x->gpiod_irq)) {
ac10x->gpiod_irq = NULL;
dev_err(&i2c->dev, "failed get switch-irq in device tree\n");
goto _err_irq;
}
gpiod_direction_input(ac10x->gpiod_irq);
ac10x->irq = gpiod_to_irq(ac10x->gpiod_irq);
if (IS_ERR_VALUE(ac10x->irq)) {
pr_info("[ac101] map gpio to irq failed, errno = %ld\n", ac10x->irq);
ac10x->irq = 0;
goto _err_irq;
}
/* request irq, set irq type to falling edge trigger */
ret = devm_request_irq(ac10x->codec->dev, ac10x->irq, audio_hmic_irq, IRQF_TRIGGER_FALLING, "SWTICH_EINT", ac10x);
if (IS_ERR_VALUE(ret)) {
pr_info("[ac101] request virq %ld failed, errno = %ld\n", ac10x->irq, ret);
goto _err_irq;
}
ac10x->mode = HEADPHONE_IDLE;
ac10x->state = -1;
/*use for judge the state of switch*/
INIT_WORK(&ac10x->work_switch, work_cb_earphone_switch);
INIT_WORK(&ac10x->work_clear_irq, work_cb_clear_irq);
/********************create input device************************/
ac10x->inpdev = devm_input_allocate_device(ac10x->codec->dev);
if (!ac10x->inpdev) {
AC101_DBG("input_allocate_device: not enough memory for input device\n");
ret = -ENOMEM;
goto _err_input_allocate_device;
}
ac10x->inpdev->name = "seed-voicecard-headset";
ac10x->inpdev->phys = dev_name(ac10x->codec->dev);
ac10x->inpdev->id.bustype = BUS_I2C;
ac10x->inpdev->dev.parent = ac10x->codec->dev;
input_set_drvdata(ac10x->inpdev, ac10x->codec);
ac10x->inpdev->evbit[0] = BIT_MASK(EV_KEY) | BIT(EV_SW);
set_bit(KEY_HEADSETHOOK, ac10x->inpdev->keybit);
set_bit(KEY_VOLUMEUP, ac10x->inpdev->keybit);
set_bit(KEY_VOLUMEDOWN, ac10x->inpdev->keybit);
input_set_capability(ac10x->inpdev, EV_SW, SW_HEADPHONE_INSERT);
ret = input_register_device(ac10x->inpdev);
if (ret) {
AC101_DBG("input_register_device: input_register_device failed\n");
goto _err_input_register_device;
}
/* the first headset state checking */
switch_hw_config(ac10x->codec);
ac10x->irq_cntr = 1;
schedule_work(&ac10x->work_switch);
return 0;
_err_input_register_device:
_err_input_allocate_device:
if (ac10x->irq) {
devm_free_irq(&i2c->dev, ac10x->irq, ac10x);
ac10x->irq = 0;
}
_err_irq:
return ret;
}
/******************************************************************************/
/********************************switch****************************************/
/******************************************************************************/
#endif
void drc_config(struct snd_soc_codec *codec)
{
int reg_val;
reg_val = ac101_read(codec, 0xa3);
reg_val &= ~(0x7ff<<0);
reg_val |= 1<<0;
ac101_write(codec, 0xa3, reg_val);
ac101_write(codec, 0xa4, 0x2baf);
reg_val = ac101_read(codec, 0xa5);
reg_val &= ~(0x7ff<<0);
reg_val |= 1<<0;
ac101_write(codec, 0xa5, reg_val);
ac101_write(codec, 0xa6, 0x2baf);
reg_val = ac101_read(codec, 0xa7);
reg_val &= ~(0x7ff<<0);
ac101_write(codec, 0xa7, reg_val);
ac101_write(codec, 0xa8, 0x44a);
reg_val = ac101_read(codec, 0xa9);
reg_val &= ~(0x7ff<<0);
ac101_write(codec, 0xa9, reg_val);
ac101_write(codec, 0xaa, 0x1e06);
reg_val = ac101_read(codec, 0xab);
reg_val &= ~(0x7ff<<0);
reg_val |= (0x352<<0);
ac101_write(codec, 0xab, reg_val);
ac101_write(codec, 0xac, 0x6910);
reg_val = ac101_read(codec, 0xad);
reg_val &= ~(0x7ff<<0);
reg_val |= (0x77a<<0);
ac101_write(codec, 0xad, reg_val);
ac101_write(codec, 0xae, 0xaaaa);
reg_val = ac101_read(codec, 0xaf);
reg_val &= ~(0x7ff<<0);
reg_val |= (0x2de<<0);
ac101_write(codec, 0xaf, reg_val);
ac101_write(codec, 0xb0, 0xc982);
ac101_write(codec, 0x16, 0x9f9f);
}
void drc_enable(struct snd_soc_codec *codec,bool on)
{
int reg_val;
if (on) {
ac101_write(codec, 0xb5, 0xA080);
reg_val = ac101_read(codec, MOD_CLK_ENA);
reg_val |= (0x1<<6);
ac101_write(codec, MOD_CLK_ENA, reg_val);
reg_val = ac101_read(codec, MOD_RST_CTRL);
reg_val |= (0x1<<6);
ac101_write(codec, MOD_RST_CTRL, reg_val);
reg_val = ac101_read(codec, 0xa0);
reg_val |= (0x7<<0);
ac101_write(codec, 0xa0, reg_val);
} else {
ac101_write(codec, 0xb5, 0x0);
reg_val = ac101_read(codec, MOD_CLK_ENA);
reg_val &= ~(0x1<<6);
ac101_write(codec, MOD_CLK_ENA, reg_val);
reg_val = ac101_read(codec, MOD_RST_CTRL);
reg_val &= ~(0x1<<6);
ac101_write(codec, MOD_RST_CTRL, reg_val);
reg_val = ac101_read(codec, 0xa0);
reg_val &= ~(0x7<<0);
ac101_write(codec, 0xa0, reg_val);
}
}
void set_configuration(struct snd_soc_codec *codec)
{
if (speaker_double_used) {
ac101_update_bits(codec, SPKOUT_CTRL, (0x1f<<SPK_VOL), (double_speaker_val<<SPK_VOL));
} else {
ac101_update_bits(codec, SPKOUT_CTRL, (0x1f<<SPK_VOL), (single_speaker_val<<SPK_VOL));
}
ac101_update_bits(codec, HPOUT_CTRL, (0x3f<<HP_VOL), (headset_val<<HP_VOL));
ac101_update_bits(codec, ADC_SRCBST_CTRL, (0x7<<ADC_MIC1G), (mainmic_val<<ADC_MIC1G));
ac101_update_bits(codec, ADC_SRCBST_CTRL, (0x7<<ADC_MIC2G), (headsetmic_val<<ADC_MIC2G));
if (dmic_used) {
ac101_write(codec, ADC_VOL_CTRL, adc_digital_val);
}
if (drc_used) {
drc_config(codec);
}
/*headphone calibration clock frequency select*/
ac101_update_bits(codec, SPKOUT_CTRL, (0x7<<HPCALICKS), (0x7<<HPCALICKS));
/* I2S1 DAC Timeslot 0 data <- I2S1 DAC channel 0 */
// "AIF1IN0L Mux" <= "AIF1DACL"
// "AIF1IN0R Mux" <= "AIF1DACR"
ac101_update_bits(codec, AIF1_DACDAT_CTRL, 0x3 << AIF1_DA0L_SRC, 0x0 << AIF1_DA0L_SRC);
ac101_update_bits(codec, AIF1_DACDAT_CTRL, 0x3 << AIF1_DA0R_SRC, 0x0 << AIF1_DA0R_SRC);
/* Timeslot 0 Left & Right Channel enable */
ac101_update_bits(codec, AIF1_DACDAT_CTRL, 0x3 << AIF1_DA0R_ENA, 0x3 << AIF1_DA0R_ENA);
/* DAC Digital Mixer Source Select <- I2S1 DA0 */
// "DACL Mixer" += "AIF1IN0L Mux"
// "DACR Mixer" += "AIF1IN0R Mux"
ac101_update_bits(codec, DAC_MXR_SRC, 0xF << DACL_MXR_ADCL, 0x8 << DACL_MXR_ADCL);
ac101_update_bits(codec, DAC_MXR_SRC, 0xF << DACR_MXR_ADCR, 0x8 << DACR_MXR_ADCR);
/* Internal DAC Analog Left & Right Channel enable */
ac101_update_bits(codec, OMIXER_DACA_CTRL, 0x3 << DACALEN, 0x3 << DACALEN);
/* Output Mixer Source Select */
// "Left Output Mixer" += "DACL Mixer"
// "Right Output Mixer" += "DACR Mixer"
ac101_update_bits(codec, OMIXER_SR, 0x1 << LMIXMUTEDACL, 0x1 << LMIXMUTEDACL);
ac101_update_bits(codec, OMIXER_SR, 0x1 << RMIXMUTEDACR, 0x1 << RMIXMUTEDACR);
/* Left & Right Analog Output Mixer enable */
ac101_update_bits(codec, OMIXER_DACA_CTRL, 0x3 << LMIXEN, 0x3 << LMIXEN);
/* Headphone Ouput Control */
// "HP_R Mux" <= "DACR Mixer"
// "HP_L Mux" <= "DACL Mixer"
ac101_update_bits(codec, HPOUT_CTRL, 0x1 << LHPS, 0x0 << LHPS);
ac101_update_bits(codec, HPOUT_CTRL, 0x1 << RHPS, 0x0 << RHPS);
/* Speaker Output Control */
// "SPK_L Mux" <= "SPK_LR Adder"
// "SPK_R Mux" <= "SPK_LR Adder"
ac101_update_bits(codec, SPKOUT_CTRL, (0x1 << LSPKS) | (0x1 << RSPKS), (0x1 << LSPKS) | (0x1 << RSPKS));
/* Enable Left & Right Speaker */
ac101_update_bits(codec, SPKOUT_CTRL, (0x1 << LSPK_EN) | (0x1 << RSPK_EN), (0x1 << LSPK_EN) | (0x1 << RSPK_EN));
return;
}
static int late_enable_dac(struct snd_soc_codec* codec, int event) {
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
mutex_lock(&ac10x->dac_mutex);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
AC101_DBG();
if (ac10x->dac_enable == 0){
/*enable dac module clk*/
ac101_update_bits(codec, MOD_CLK_ENA, (0x1<<MOD_CLK_DAC_DIG), (0x1<<MOD_CLK_DAC_DIG));
ac101_update_bits(codec, MOD_RST_CTRL, (0x1<<MOD_RESET_DAC_DIG), (0x1<<MOD_RESET_DAC_DIG));
ac101_update_bits(codec, DAC_DIG_CTRL, (0x1<<ENDA), (0x1<<ENDA));
ac101_update_bits(codec, DAC_DIG_CTRL, (0x1<<ENHPF),(0x1<<ENHPF));
}
ac10x->dac_enable++;
break;
case SND_SOC_DAPM_POST_PMD:
if (ac10x->dac_enable != 0){
ac10x->dac_enable = 0;
ac101_update_bits(codec, DAC_DIG_CTRL, (0x1<<ENHPF),(0x0<<ENHPF));
ac101_update_bits(codec, DAC_DIG_CTRL, (0x1<<ENDA), (0x0<<ENDA));
/*disable dac module clk*/
ac101_update_bits(codec, MOD_CLK_ENA, (0x1<<MOD_CLK_DAC_DIG), (0x0<<MOD_CLK_DAC_DIG));
ac101_update_bits(codec, MOD_RST_CTRL, (0x1<<MOD_RESET_DAC_DIG), (0x0<<MOD_RESET_DAC_DIG));
}
break;
}
mutex_unlock(&ac10x->dac_mutex);
return 0;
}
static int ac101_headphone_event(struct snd_soc_codec* codec, int event) {
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/*open*/
AC101_DBG("post:open\n");
ac101_update_bits(codec, OMIXER_DACA_CTRL, (0xf<<HPOUTPUTENABLE), (0xf<<HPOUTPUTENABLE));
msleep(10);
ac101_update_bits(codec, HPOUT_CTRL, (0x1<<HPPA_EN), (0x1<<HPPA_EN));
ac101_update_bits(codec, HPOUT_CTRL, (0x3<<LHPPA_MUTE), (0x3<<LHPPA_MUTE));
break;
case SND_SOC_DAPM_PRE_PMD:
/*close*/
AC101_DBG("pre:close\n");
ac101_update_bits(codec, HPOUT_CTRL, (0x3<<LHPPA_MUTE), (0x0<<LHPPA_MUTE));
msleep(10);
ac101_update_bits(codec, OMIXER_DACA_CTRL, (0xf<<HPOUTPUTENABLE), (0x0<<HPOUTPUTENABLE));
ac101_update_bits(codec, HPOUT_CTRL, (0x1<<HPPA_EN), (0x0<<HPPA_EN));
break;
}
return 0;
}
static int ac101_sysclk_started(void) {
int reg_val;
reg_val = ac101_read(static_ac10x->codec, SYSCLK_CTRL);
return (reg_val & (0x1<<SYSCLK_ENA));
}
static int ac101_aif1clk(struct snd_soc_codec* codec, int event, int quick) {
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
int ret = 0;
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (ac10x->aif1_clken == 0){
ret = ac101_update_bits(codec, SYSCLK_CTRL, (0x1<<AIF1CLK_ENA), (0x1<<AIF1CLK_ENA));
if(!quick || _MASTER_MULTI_CODEC != _MASTER_AC101) {
/* enable aif1clk & sysclk */
ret = ret || ac101_update_bits(codec, MOD_CLK_ENA, (0x1<<MOD_CLK_AIF1), (0x1<<MOD_CLK_AIF1));
ret = ret || ac101_update_bits(codec, MOD_RST_CTRL, (0x1<<MOD_RESET_AIF1), (0x1<<MOD_RESET_AIF1));
}
ret = ret || ac101_update_bits(codec, SYSCLK_CTRL, (0x1<<SYSCLK_ENA), (0x1<<SYSCLK_ENA));
if (ret) {
AC101_DBG("start sysclk failed\n");
} else {
AC101_DBG("hw sysclk enable\n");
ac10x->aif1_clken++;
}
}
break;
case SND_SOC_DAPM_POST_PMD:
if (ac10x->aif1_clken != 0) {
/* disable aif1clk & sysclk */
ret = ac101_update_bits(codec, SYSCLK_CTRL, (0x1<<AIF1CLK_ENA),(0x0<<AIF1CLK_ENA));
ret = ret || ac101_update_bits(codec, MOD_CLK_ENA, (0x1<<MOD_CLK_AIF1), (0x0<<MOD_CLK_AIF1));
ret = ret || ac101_update_bits(codec, MOD_RST_CTRL, (0x1<<MOD_RESET_AIF1), (0x0<<MOD_RESET_AIF1));
ret = ret || ac101_update_bits(codec, SYSCLK_CTRL, (0x1<<SYSCLK_ENA), (0x0<<SYSCLK_ENA));
if (ret) {
AC101_DBG("stop sysclk failed\n");
} else {
AC101_DBG("hw sysclk disable\n");
ac10x->aif1_clken = 0;
}
break;
}
}
AC101_DBG("event=%d pre_up/%d post_down/%d\n", event, SND_SOC_DAPM_PRE_PMU, SND_SOC_DAPM_POST_PMD);
return ret;
}
/**
* snd_ac101_get_volsw - single mixer get callback
* @kcontrol: mixer control
* @ucontrol: control element information
*
* Callback to get the value of a single mixer control, or a double mixer
* control that spans 2 registers.
*
* Returns 0 for success.
*/
static int snd_ac101_get_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol
){
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int val, mask = (1 << fls(mc->max)) - 1;
unsigned int invert = mc->invert;
int ret;
if ((ret = ac101_read(static_ac10x->codec, mc->reg)) < 0)
return ret;
val = (ret >> mc->shift) & mask;
ucontrol->value.integer.value[0] = val - mc->min;
if (invert) {
ucontrol->value.integer.value[0] =
mc->max - ucontrol->value.integer.value[0];
}
if (snd_soc_volsw_is_stereo(mc)) {
val = (ret >> mc->rshift) & mask;
ucontrol->value.integer.value[1] = val - mc->min;
if (invert) {
ucontrol->value.integer.value[1] =
mc->max - ucontrol->value.integer.value[1];
}
}
return 0;
}
/**
* snd_ac101_put_volsw - single mixer put callback
* @kcontrol: mixer control
* @ucontrol: control element information
*
* Callback to set the value of a single mixer control, or a double mixer
* control that spans 2 registers.
*
* Returns 0 for success.
*/
static int snd_ac101_put_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol
){
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int sign_bit = mc->sign_bit;
unsigned int val, mask = (1 << fls(mc->max)) - 1;
unsigned int invert = mc->invert;
int ret;
if (sign_bit)
mask = BIT(sign_bit + 1) - 1;
val = ((ucontrol->value.integer.value[0] + mc->min) & mask);
if (invert) {
val = mc->max - val;
}
ret = ac101_update_bits(static_ac10x->codec, mc->reg, mask << mc->shift, val << mc->shift);
if (! snd_soc_volsw_is_stereo(mc)) {
return ret;
}
val = ((ucontrol->value.integer.value[1] + mc->min) & mask);
if (invert) {
val = mc->max - val;
}
ret = ac101_update_bits(static_ac10x->codec, mc->reg, mask << mc->rshift, val << mc->rshift);
return ret;
}
static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -11925, 75, 0);
static const DECLARE_TLV_DB_SCALE(dac_mix_vol_tlv, -600, 600, 0);
static const DECLARE_TLV_DB_SCALE(dig_vol_tlv, -7308, 116, 0);
static const DECLARE_TLV_DB_SCALE(speaker_vol_tlv, -4800, 150, 0);
static const DECLARE_TLV_DB_SCALE(headphone_vol_tlv, -6300, 100, 0);
static struct snd_kcontrol_new ac101_controls[] = {
/*DAC*/
SOC_DOUBLE_TLV("DAC volume", DAC_VOL_CTRL, DAC_VOL_L, DAC_VOL_R, 0xff, 0, dac_vol_tlv),
SOC_DOUBLE_TLV("DAC mixer gain", DAC_MXR_GAIN, DACL_MXR_GAIN, DACR_MXR_GAIN, 0xf, 0, dac_mix_vol_tlv),
SOC_SINGLE_TLV("digital volume", DAC_DBG_CTRL, DVC, 0x3f, 1, dig_vol_tlv),
SOC_SINGLE_TLV("speaker volume", SPKOUT_CTRL, SPK_VOL, 0x1f, 0, speaker_vol_tlv),
SOC_SINGLE_TLV("headphone volume", HPOUT_CTRL, HP_VOL, 0x3f, 0, headphone_vol_tlv),
};
/* PLL divisors */
struct pll_div {
unsigned int pll_in;
unsigned int pll_out;
int m;
int n_i;
int n_f;
};
struct aif1_fs {
unsigned samp_rate;
int bclk_div;
int srbit;
#define _SERIES_24_576K 0
#define _SERIES_22_579K 1
int series;
};
struct kv_map {
int val;
int bit;
};
/*
* Note : pll code from original tdm/i2s driver.
* freq_out = freq_in * N/(M*(2k+1)) , k=1,N=N_i+N_f,N_f=factor*0.2;
* N_i[0,1023], N_f_factor[0,7], m[1,64]=REG_VAL[1-63,0]
*/
static const struct pll_div codec_pll_div[] = {
{128000, _FREQ_22_579K, 1, 529, 1},
{192000, _FREQ_22_579K, 1, 352, 4},
{256000, _FREQ_22_579K, 1, 264, 3},
{384000, _FREQ_22_579K, 1, 176, 2}, /*((176+2*0.2)*6000000)/(38*(2*1+1))*/
{1411200, _FREQ_22_579K, 1, 48, 0},
{2822400, _FREQ_22_579K, 1, 24, 0}, /* accurate, 11025 * 256 */
{5644800, _FREQ_22_579K, 1, 12, 0}, /* accurate, 22050 * 256 */
{6000000, _FREQ_22_579K, 38, 429, 0}, /*((429+0*0.2)*6000000)/(38*(2*1+1))*/
{11289600, _FREQ_22_579K, 1, 6, 0}, /* accurate, 44100 * 256 */
{13000000, _FREQ_22_579K, 19, 99, 0},
{19200000, _FREQ_22_579K, 25, 88, 1},
{24000000, _FREQ_22_579K, 63, 177, 4}, /* 22577778 Hz */
{128000, _FREQ_24_576K, 1, 576, 0},
{192000, _FREQ_24_576K, 1, 384, 0},
{256000, _FREQ_24_576K, 1, 288, 0},
{384000, _FREQ_24_576K, 1, 192, 0},
{2048000, _FREQ_24_576K, 1, 36, 0}, /* accurate, 8000 * 256 */
{3072000, _FREQ_24_576K, 1, 24, 0}, /* accurate, 12000 * 256 */
{4096000, _FREQ_24_576K, 1, 18, 0}, /* accurate, 16000 * 256 */
{6000000, _FREQ_24_576K, 25, 307, 1},
{6144000, _FREQ_24_576K, 4, 48, 0}, /* accurate, 24000 * 256 */
{12288000, _FREQ_24_576K, 8, 48, 0}, /* accurate, 48000 * 256 */
{13000000, _FREQ_24_576K, 42, 238, 1},
{19200000, _FREQ_24_576K, 25, 96, 0},
{24000000, _FREQ_24_576K, 25, 76, 4}, /* accurate */
{_FREQ_22_579K, _FREQ_22_579K, 8, 24, 0}, /* accurate, 88200 * 256 */
{_FREQ_24_576K, _FREQ_24_576K, 8, 24, 0}, /* accurate, 96000 * 256 */
};
static const struct aif1_fs codec_aif1_fs[] = {
{8000, 12, 0},
{11025, 8, 1, _SERIES_22_579K},
{12000, 8, 2},
{16000, 6, 3},
{22050, 4, 4, _SERIES_22_579K},
{24000, 4, 5},
/* {32000, 3, 6}, dividing by 3 is not support */
{44100, 2, 7, _SERIES_22_579K},
{48000, 2, 8},
{96000, 1, 9},
};
static const struct kv_map codec_aif1_lrck[] = {
{16, 0},
{32, 1},
{64, 2},
{128, 3},
{256, 4},
};
static const struct kv_map codec_aif1_wsize[] = {
{8, 0},
{16, 1},
{20, 2},
{24, 3},
{32, 3},
};
static const unsigned ac101_bclkdivs[] = {
1, 2, 4, 6,
8, 12, 16, 24,
32, 48, 64, 96,
128, 192, 0, 0,
};
static int ac101_aif_play(struct ac10x_priv* ac10x) {
struct snd_soc_codec * codec = ac10x->codec;
late_enable_dac(codec, SND_SOC_DAPM_PRE_PMU);
ac101_headphone_event(codec, SND_SOC_DAPM_POST_PMU);
if (drc_used) {
drc_enable(codec, 1);
}
/* Enable Left & Right Speaker */
ac101_update_bits(codec, SPKOUT_CTRL, (0x1 << LSPK_EN) | (0x1 << RSPK_EN), (0x1 << LSPK_EN) | (0x1 << RSPK_EN));
if (ac10x->gpiod_spk_amp_gate) {
gpiod_set_value(ac10x->gpiod_spk_amp_gate, 1);
}
return 0;
}
static void ac10x_work_aif_play(struct work_struct *work) {
struct ac10x_priv *ac10x = container_of(work, struct ac10x_priv, dlywork.work);
ac101_aif_play(ac10x);
return;
}
int ac101_aif_mute(struct snd_soc_dai *codec_dai, int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
AC101_DBG("mute=%d\n", mute);
ac101_write(codec, DAC_VOL_CTRL, mute? 0: 0xA0A0);
if (!mute) {
#if _MASTER_MULTI_CODEC != _MASTER_AC101
/* enable global clock */
ac10x->aif1_clken = 0;
ac101_aif1clk(codec, SND_SOC_DAPM_PRE_PMU, 0);
ac101_aif_play(ac10x);
#else
schedule_delayed_work(&ac10x->dlywork, msecs_to_jiffies(50));
#endif
} else {
#if _MASTER_MULTI_CODEC == _MASTER_AC101
cancel_delayed_work_sync(&ac10x->dlywork);
#endif
if (ac10x->gpiod_spk_amp_gate) {
gpiod_set_value(ac10x->gpiod_spk_amp_gate, 0);
}
/* Disable Left & Right Speaker */
ac101_update_bits(codec, SPKOUT_CTRL, (0x1 << LSPK_EN) | (0x1 << RSPK_EN), (0x0 << LSPK_EN) | (0x0 << RSPK_EN));
if (drc_used) {
drc_enable(codec, 0);
}
ac101_headphone_event(codec, SND_SOC_DAPM_PRE_PMD);
late_enable_dac(codec, SND_SOC_DAPM_POST_PMD);
#if _MASTER_MULTI_CODEC != _MASTER_AC101
ac10x->aif1_clken = 1;
ac101_aif1clk(codec, SND_SOC_DAPM_POST_PMD, 0);
#endif
}
return 0;
}
void ac101_aif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
AC101_DBG("stream = %s, play: %d, capt: %d, active: %d\n",
snd_pcm_stream_str(substream),
codec_dai->playback_active, codec_dai->capture_active,
codec_dai->active);
if (!codec_dai->active) {
ac10x->aif1_clken = 1;
ac101_aif1clk(codec, SND_SOC_DAPM_POST_PMD, 0);
} else {
ac101_aif1clk(codec, SND_SOC_DAPM_PRE_PMU, 0);
}
}
static int ac101_set_pll(struct snd_soc_dai *codec_dai, int pll_id, int source,
unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_codec *codec = codec_dai->codec;
int i, m, n_i, n_f;
AC101_DBG("pll_id:%d\n", pll_id);
/* clear volatile reserved bits*/
ac101_update_bits(codec, SYSCLK_CTRL, 0xFF & ~(0x1 << SYSCLK_ENA), 0x0);
/* select aif1 clk srouce from mclk1 */
ac101_update_bits(codec, SYSCLK_CTRL, (0x3<<AIF1CLK_SRC), (0x0<<AIF1CLK_SRC));
/* disable pll */
ac101_update_bits(codec, PLL_CTRL2, (0x1<<PLL_EN), (0<<PLL_EN));
if (!freq_out)
return 0;
if ((freq_in < 128000) || (freq_in > _FREQ_24_576K)) {
return -EINVAL;
} else if ((freq_in == _FREQ_24_576K) || (freq_in == _FREQ_22_579K)) {
if (pll_id == AC101_MCLK1) {
/*select aif1 clk source from mclk1*/
ac101_update_bits(codec, SYSCLK_CTRL, (0x3<<AIF1CLK_SRC), (0x0<<AIF1CLK_SRC));
return 0;
}
}
switch (pll_id) {
case AC101_MCLK1:
/*pll source from MCLK1*/
ac101_update_bits(codec, SYSCLK_CTRL, (0x3<<PLLCLK_SRC), (0x0<<PLLCLK_SRC));