seeed-voicecard/ac101.c

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/*
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* ac101.c
*
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* (C) Copyright 2017-2018
* Seeed Technology Co., Ltd. <www.seeedstudio.com>
*
* PeterYang <linsheng.yang@seeed.cc>
*
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* (C) Copyright 2014-2017
* Reuuimlla Technology Co., Ltd. <www.reuuimllatech.com>
*
* huangxin <huangxin@Reuuimllatech.com>
* liushaohua <liushaohua@allwinnertech.com>
*
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* X-Powers AC101 codec driver
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*
* 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
*/
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#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>
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#include <linux/gpio/consumer.h>
#include <linux/regmap.h>
#include <linux/input.h>
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#include <linux/delay.h>
#include "ac101_regs.h"
#include "ac10x.h"
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/*
* *** 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()
*/
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/*Default initialize configuration*/
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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;
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#define AC101_RATES (SNDRV_PCM_RATE_8000_96000 & \
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~(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)
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static struct ac10x_priv* static_ac10x;
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int ac101_read(struct snd_soc_codec *codec, unsigned reg) {
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
int r, v = 0;
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if ((r = regmap_read(ac10x->regmap101, reg, &v)) < 0) {
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dev_err(codec->dev, "read reg %02X fail\n",
reg);
return r;
}
return v;
}
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int ac101_write(struct snd_soc_codec *codec, unsigned reg, unsigned val) {
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
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int v;
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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);
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int v;
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v = regmap_update_bits(ac10x->regmap101, reg, mask, value);
return v;
}
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#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;
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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)
{
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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++;
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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
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int r, d;
d = GET_HMIC_DATA(ac101_read(codec, HMIC_STS));
r = 0x1 << HMIC_DATA_PEND;
ac101_write(codec, HMIC_STS, r);
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/* prevent i2c accessing too frequently */
usleep_range(1500, 3000);
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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);
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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);
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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);
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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);
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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_warn("[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_warn("[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) {
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devm_free_irq(&i2c->dev, ac10x->irq, ac10x);
ac10x->irq = 0;
}
_err_irq:
return ret;
}
/******************************************************************************/
/********************************switch****************************************/
/******************************************************************************/
#endif
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void drc_config(struct snd_soc_codec *codec)
{
int reg_val;
reg_val = ac101_read(codec, 0xa3);
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reg_val &= ~(0x7ff<<0);
reg_val |= 1<<0;
ac101_write(codec, 0xa3, reg_val);
ac101_write(codec, 0xa4, 0x2baf);
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reg_val = ac101_read(codec, 0xa5);
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reg_val &= ~(0x7ff<<0);
reg_val |= 1<<0;
ac101_write(codec, 0xa5, reg_val);
ac101_write(codec, 0xa6, 0x2baf);
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reg_val = ac101_read(codec, 0xa7);
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reg_val &= ~(0x7ff<<0);
ac101_write(codec, 0xa7, reg_val);
ac101_write(codec, 0xa8, 0x44a);
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reg_val = ac101_read(codec, 0xa9);
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reg_val &= ~(0x7ff<<0);
ac101_write(codec, 0xa9, reg_val);
ac101_write(codec, 0xaa, 0x1e06);
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reg_val = ac101_read(codec, 0xab);
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reg_val &= ~(0x7ff<<0);
reg_val |= (0x352<<0);
ac101_write(codec, 0xab, reg_val);
ac101_write(codec, 0xac, 0x6910);
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reg_val = ac101_read(codec, 0xad);
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reg_val &= ~(0x7ff<<0);
reg_val |= (0x77a<<0);
ac101_write(codec, 0xad, reg_val);
ac101_write(codec, 0xae, 0xaaaa);
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reg_val = ac101_read(codec, 0xaf);
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reg_val &= ~(0x7ff<<0);
reg_val |= (0x2de<<0);
ac101_write(codec, 0xaf, reg_val);
ac101_write(codec, 0xb0, 0xc982);
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ac101_write(codec, 0x16, 0x9f9f);
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}
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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);
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reg_val |= (0x1<<6);
ac101_write(codec, MOD_CLK_ENA, reg_val);
reg_val = ac101_read(codec, MOD_RST_CTRL);
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reg_val |= (0x1<<6);
ac101_write(codec, MOD_RST_CTRL, reg_val);
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reg_val = ac101_read(codec, 0xa0);
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reg_val |= (0x7<<0);
ac101_write(codec, 0xa0, reg_val);
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} else {
ac101_write(codec, 0xb5, 0x0);
reg_val = ac101_read(codec, MOD_CLK_ENA);
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reg_val &= ~(0x1<<6);
ac101_write(codec, MOD_CLK_ENA, reg_val);
reg_val = ac101_read(codec, MOD_RST_CTRL);
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reg_val &= ~(0x1<<6);
ac101_write(codec, MOD_RST_CTRL, reg_val);
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reg_val = ac101_read(codec, 0xa0);
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reg_val &= ~(0x7<<0);
ac101_write(codec, 0xa0, reg_val);
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}
}
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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));
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} else {
ac101_update_bits(codec, SPKOUT_CTRL, (0x1f<<SPK_VOL), (single_speaker_val<<SPK_VOL));
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}
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));
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if (dmic_used) {
ac101_write(codec, ADC_VOL_CTRL, adc_digital_val);
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}
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;
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}
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static int late_enable_dac(struct snd_soc_codec* codec, int event) {
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struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
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mutex_lock(&ac10x->dac_mutex);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
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AC101_DBG();
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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));
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}
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));
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}
break;
}
mutex_unlock(&ac10x->dac_mutex);
return 0;
}
static int ac101_headphone_event(struct snd_soc_codec* codec, int event) {
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switch (event) {
case SND_SOC_DAPM_POST_PMU:
/*open*/
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AC101_DBG("post:open\n");
ac101_update_bits(codec, OMIXER_DACA_CTRL, (0xf<<HPOUTPUTENABLE), (0xf<<HPOUTPUTENABLE));
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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));
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break;
case SND_SOC_DAPM_PRE_PMD:
/*close*/
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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));
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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) {
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struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
int ret = 0;
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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) {
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AC101_DBG("start sysclk failed\n");
} else {
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AC101_DBG("hw sysclk enable\n");
ac10x->aif1_clken++;
}
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}
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) {
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AC101_DBG("stop sysclk failed\n");
} else {
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AC101_DBG("hw sysclk disable\n");
ac10x->aif1_clken = 0;
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}
break;
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}
}
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AC101_DBG("event=%d pre_up/%d post_down/%d\n", event, SND_SOC_DAPM_PRE_PMU, SND_SOC_DAPM_POST_PMD);
return ret;
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}
/**
* 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;
}
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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);
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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);
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static struct snd_kcontrol_new ac101_controls[] = {
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/*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 Playback Volume", SPKOUT_CTRL, SPK_VOL, 0x1f, 0, speaker_vol_tlv),
SOC_SINGLE_TLV("Headphone Playback Volume", HPOUT_CTRL, HP_VOL, 0x3f, 0, headphone_vol_tlv),
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};
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/* 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;
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};
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struct kv_map {
int val;
int bit;
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};
/*
* 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]
*/
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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 */
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};
static const struct aif1_fs codec_aif1_fs[] = {
{8000, 12, 0},
{11025, 8, 1, _SERIES_22_579K},
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{12000, 8, 2},
{16000, 6, 3},
{22050, 4, 4, _SERIES_22_579K},
{24000, 4, 5},
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/* {32000, 3, 6}, dividing by 3 is not support */
{44100, 2, 7, _SERIES_22_579K},
{48000, 2, 8},
{96000, 1, 9},
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};
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static const struct kv_map codec_aif1_lrck[] = {
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{16, 0},
{32, 1},
{64, 2},
{128, 3},
{256, 4},
};
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static const struct kv_map codec_aif1_wsize[] = {
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{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,
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};
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)
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{
struct snd_soc_codec *codec = codec_dai->codec;
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struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
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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
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if (ac10x->gpiod_spk_amp_gate) {
gpiod_set_value(ac10x->gpiod_spk_amp_gate, 0);
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}
/* 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) {
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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
}
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return 0;
}
void ac101_aif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai)
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{
struct snd_soc_codec *codec = codec_dai->codec;
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
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AC101_DBG("stream = %s, play: %d, capt: %d, active: %d\n",
snd_pcm_stream_str(substream),
codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE],
snd_soc_dai_active(codec_dai));
if (!snd_soc_dai_active(codec_dai)) {
ac10x->aif1_clken = 1;
ac101_aif1clk(codec, SND_SOC_DAPM_POST_PMD, 0);
} else {
ac101_aif1clk(codec, SND_SOC_DAPM_PRE_PMU, 0);
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}
}
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;
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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));
break;
case AC101_BCLK1:
/*pll source from BCLK1*/
ac101_update_bits(codec, SYSCLK_CTRL, (0x3<<PLLCLK_SRC), (0x2<<PLLCLK_SRC));
break;
default:
return -EINVAL;
}
/* freq_out = freq_in * n/(m*(2k+1)) , k=1,N=N_i+N_f */
for (i = m = n_i = n_f = 0; i < ARRAY_SIZE(codec_pll_div); i++) {
if ((codec_pll_div[i].pll_in == freq_in) && (codec_pll_div[i].pll_out == freq_out)) {
m = codec_pll_div[i].m;
n_i = codec_pll_div[i].n_i;
n_f = codec_pll_div[i].n_f;
break;
}
}
/* config pll m */
if (m == 64) m = 0;
ac101_update_bits(codec, PLL_CTRL1, (0x3f<<PLL_POSTDIV_M), (m<<PLL_POSTDIV_M));
/* config pll n */
ac101_update_bits(codec, PLL_CTRL2, (0x3ff<<PLL_PREDIV_NI), (n_i<<PLL_PREDIV_NI));
ac101_update_bits(codec, PLL_CTRL2, (0x7<<PLL_POSTDIV_NF), (n_f<<PLL_POSTDIV_NF));
/* enable pll */
ac101_update_bits(codec, PLL_CTRL2, (0x1<<PLL_EN), (1<<PLL_EN));
ac101_update_bits(codec, SYSCLK_CTRL, (0x1<<PLLCLK_ENA), (0x1<<PLLCLK_ENA));
ac101_update_bits(codec, SYSCLK_CTRL, (0x3<<AIF1CLK_SRC), (0x3<<AIF1CLK_SRC));
return 0;
}
int ac101_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params,
struct snd_soc_dai *codec_dai)
{
int i = 0;
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int AIF_CLK_CTRL = AIF1_CLK_CTRL;
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int aif1_word_size = 24;
int aif1_slot_size = 32;
int aif1_lrck_div;
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struct snd_soc_codec *codec = codec_dai->codec;
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
int reg_val, freq_out;
unsigned channels;
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AC101_DBG("+++\n");
if (_MASTER_MULTI_CODEC == _MASTER_AC101 && ac101_sysclk_started()) {
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/* not configure hw_param twice if stream is playback, tell the caller it's started */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
return 1;
}
}
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/* get channels count & slot size */
channels = params_channels(params);
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switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S24_LE:
case SNDRV_PCM_FORMAT_S32_LE:
aif1_slot_size = 32;
break;
case SNDRV_PCM_FORMAT_S16_LE:
default:
aif1_slot_size = 16;
break;
}
/* set LRCK/BCLK ratio */
aif1_lrck_div = aif1_slot_size * channels;
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for (i = 0; i < ARRAY_SIZE(codec_aif1_lrck); i++) {
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if (codec_aif1_lrck[i].val == aif1_lrck_div) {
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break;
}
}
ac101_update_bits(codec, AIF_CLK_CTRL, (0x7<<AIF1_LRCK_DIV), codec_aif1_lrck[i].bit<<AIF1_LRCK_DIV);
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/* set PLL output freq */
freq_out = _FREQ_24_576K;
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for (i = 0; i < ARRAY_SIZE(codec_aif1_fs); i++) {
if (codec_aif1_fs[i].samp_rate == params_rate(params)) {
if (codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] && dmic_used && codec_aif1_fs[i].samp_rate == 44100) {
ac101_update_bits(codec, AIF_SR_CTRL, (0xf<<AIF1_FS), (0x4<<AIF1_FS));
} else {
ac101_update_bits(codec, AIF_SR_CTRL, (0xf<<AIF1_FS), ((codec_aif1_fs[i].srbit)<<AIF1_FS));
}
if (codec_aif1_fs[i].series == _SERIES_22_579K)
freq_out = _FREQ_22_579K;
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break;
}
}
/* set I2S word size */
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for (i = 0; i < ARRAY_SIZE(codec_aif1_wsize); i++) {
if (codec_aif1_wsize[i].val == aif1_word_size) {
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break;
}
}
ac101_update_bits(codec, AIF_CLK_CTRL, (0x3<<AIF1_WORK_SIZ), ((codec_aif1_wsize[i].bit)<<AIF1_WORK_SIZ));
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/* set TDM slot size */
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if ((reg_val = codec_aif1_wsize[i].bit) > 2) reg_val = 2;
ac101_update_bits(codec, AIF1_ADCDAT_CTRL, 0x3 << AIF1_SLOT_SIZ, reg_val << AIF1_SLOT_SIZ);
/* setting pll if it's master mode */
reg_val = ac101_read(codec, AIF_CLK_CTRL);
if ((reg_val & (0x1 << AIF1_MSTR_MOD)) == 0) {
unsigned bclkdiv;
ac101_set_pll(codec_dai, AC101_MCLK1, 0, ac10x->sysclk, freq_out);
bclkdiv = freq_out / (aif1_lrck_div * params_rate(params));
for (i = 0; i < ARRAY_SIZE(ac101_bclkdivs) - 1; i++) {
if (ac101_bclkdivs[i] >= bclkdiv) {
break;
}
}
ac101_update_bits(codec, AIF_CLK_CTRL, (0xf<<AIF1_BCLK_DIV), i<<AIF1_BCLK_DIV);
} else {
/* set pll clock source to BCLK if slave mode */
ac101_set_pll(codec_dai, AC101_BCLK1, 0, aif1_lrck_div * params_rate(params), freq_out);
}
#if _MASTER_MULTI_CODEC == _MASTER_AC101
/* Master mode, to clear cpu_dai fifos, disable output bclk & lrck */
ac101_aif1clk(codec, SND_SOC_DAPM_POST_PMD, 0);
#endif
AC101_DBG("rate: %d , channels: %d , samp_res: %d",
params_rate(params), channels, aif1_slot_size);
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AC101_DBG("---\n");
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return 0;
}
int ac101_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
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{
int reg_val;
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int AIF_CLK_CTRL = AIF1_CLK_CTRL;
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struct snd_soc_codec *codec = codec_dai->codec;
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AC101_DBG();
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/*
* master or slave selection
* 0 = Master mode
* 1 = Slave mode
*/
reg_val = ac101_read(codec, AIF_CLK_CTRL);
reg_val &= ~(0x1<<AIF1_MSTR_MOD);
switch(fmt & SND_SOC_DAIFMT_MASTER_MASK) {
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case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master, ap is slave*/
#if _MASTER_MULTI_CODEC == _MASTER_AC101
pr_info("AC101 as Master\n");
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reg_val |= (0x0<<AIF1_MSTR_MOD);
break;
#else
pr_info("AC108 as Master\n");
#endif
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case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & frm slave, ap is master*/
pr_info("AC101 as Slave\n");
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reg_val |= (0x1<<AIF1_MSTR_MOD);
break;
default:
pr_err("unknwon master/slave format\n");
return -EINVAL;
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}
/*
* Enable TDM mode
*/
reg_val |= (0x1 << AIF1_TDMM_ENA);
ac101_write(codec, AIF_CLK_CTRL, reg_val);
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/* i2s mode selection */
reg_val = ac101_read(codec, AIF_CLK_CTRL);
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reg_val&=~(3<<AIF1_DATA_FMT);
switch(fmt & SND_SOC_DAIFMT_FORMAT_MASK){
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case SND_SOC_DAIFMT_I2S: /* I2S1 mode */
reg_val |= (0x0<<AIF1_DATA_FMT);
break;
case SND_SOC_DAIFMT_RIGHT_J: /* Right Justified mode */
reg_val |= (0x2<<AIF1_DATA_FMT);
break;
case SND_SOC_DAIFMT_LEFT_J: /* Left Justified mode */
reg_val |= (0x1<<AIF1_DATA_FMT);
break;
case SND_SOC_DAIFMT_DSP_A: /* L reg_val msb after FRM LRC */
reg_val |= (0x3<<AIF1_DATA_FMT);
break;
case SND_SOC_DAIFMT_DSP_B:
/* TODO: data offset set to 0 */
reg_val |= (0x3<<AIF1_DATA_FMT);
break;
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default:
pr_err("%s, line:%d\n", __func__, __LINE__);
return -EINVAL;
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}
ac101_write(codec, AIF_CLK_CTRL, reg_val);
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/* DAI signal inversions */
reg_val = ac101_read(codec, AIF_CLK_CTRL);
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switch(fmt & SND_SOC_DAIFMT_INV_MASK){
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case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + nor frame */
reg_val &= ~(0x1<<AIF1_LRCK_INV);
reg_val &= ~(0x1<<AIF1_BCLK_INV);
break;
case SND_SOC_DAIFMT_NB_IF: /* normal bclk + inv frm */
reg_val |= (0x1<<AIF1_LRCK_INV);
reg_val &= ~(0x1<<AIF1_BCLK_INV);
break;
case SND_SOC_DAIFMT_IB_NF: /* invert bclk + nor frm */
reg_val &= ~(0x1<<AIF1_LRCK_INV);
reg_val |= (0x1<<AIF1_BCLK_INV);
break;
case SND_SOC_DAIFMT_IB_IF: /* invert bclk + inv frm */
reg_val |= (0x1<<AIF1_LRCK_INV);
reg_val |= (0x1<<AIF1_BCLK_INV);
break;
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}
ac101_write(codec, AIF_CLK_CTRL, reg_val);
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return 0;
}
int ac101_audio_startup(struct snd_pcm_substream *substream,
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struct snd_soc_dai *codec_dai)
{
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// struct snd_soc_codec *codec = codec_dai->codec;
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AC101_DBG("\n\n\n");
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if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
}
return 0;
}
#if _MASTER_MULTI_CODEC == _MASTER_AC101
Second/Alternative change to correct for 5.5+ change in trigger order The idea is the same as the previous attempt: calls ac101_trigger() just before set_clock(1). https://github.com/respeaker/seeed-voicecard/issues/290 6mics / linear 4 mics: fails to record against v5.10 kernel https://github.com/raspberrypi/linux/issues/4279 [regression] alsa system call blocks on record between 5.4.83 and 5.5.19 In v5.5, commit 4378f1fbe924054a09ff0d4e39e1a581b9245252 Author: Peter Ujfalusi <peter.ujfalusi@ti.com> Date: Fri Sep 27 10:16:46 2019 +0300 ASoC: soc-pcm: Use different sequence for start/stop trigger On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20190927071646.22319-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie@kernel.org>
2021-04-27 02:35:13 +02:00
static int ac101_set_clock(int y_start_n_stop, struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) {
int r;
if (y_start_n_stop) {
/* enable global clock */
r = ac101_aif1clk(static_ac10x->codec, SND_SOC_DAPM_PRE_PMU, 1);
} else {
/* disable global clock */
static_ac10x->aif1_clken = 1;
r = ac101_aif1clk(static_ac10x->codec, SND_SOC_DAPM_POST_PMD, 0);
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}
return r;
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}
#endif
int ac101_trigger(struct snd_pcm_substream *substream, int cmd,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
int ret = 0;
unsigned long flags;
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AC101_DBG("stream=%s cmd=%d\n",
snd_pcm_stream_str(substream),
cmd);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
#if _MASTER_MULTI_CODEC == _MASTER_AC101
spin_lock_irqsave(&ac10x->lock, flags);
if (ac10x->aif1_clken == 0){
/*
* enable aif1clk, it' here due to reduce time between 'AC108 Sysclk Enable' and 'AC101 Sysclk Enable'
* Or else the two AC108 chips lost the sync.
*/
ret = 0;
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));
}
spin_unlock_irqrestore(&ac10x->lock, flags);
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ac101_set_clock(1, substream, cmd, dai);
#endif
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
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ac101_set_clock(0, NULL, 0, NULL);
break;
default:
ret = -EINVAL;
}
AC101_DBG("stream=%s cmd=%d;finished %d\n",
snd_pcm_stream_str(substream),
cmd, ret);
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return ret;
}
#if 0
static int ac101_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 ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
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AC101_DBG("id=%d freq=%d, dir=%d\n",
clk_id, freq, dir);
ac10x->sysclk = freq;
return 0;
}
static const struct snd_soc_dai_ops ac101_aif1_dai_ops = {
//.startup = ac101_audio_startup,
//.shutdown = ac101_aif_shutdown,
//.set_sysclk = ac101_set_dai_sysclk,
//.set_pll = ac101_set_pll,
//.set_fmt = ac101_set_dai_fmt,
//.hw_params = ac101_hw_params,
//.trigger = ac101_trigger,
//.digital_mute = ac101_aif_mute,
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};
static struct snd_soc_dai_driver ac101_dai[] = {
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{
.name = "ac10x-aif1",
.id = AIF1_CLK,
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.playback = {
.stream_name = "Playback",
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.channels_min = 1,
.channels_max = 8,
.rates = AC101_RATES,
.formats = AC101_FORMATS,
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},
#if 0
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.capture = {
.stream_name = "Capture",
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.channels_min = 1,
.channels_max = 8,
.rates = AC101_RATES,
.formats = AC101_FORMATS,
},
#endif
.ops = &ac101_aif1_dai_ops,
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}
};
#endif
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static void codec_resume_work(struct work_struct *work)
{
struct ac10x_priv *ac10x = container_of(work, struct ac10x_priv, codec_resume);
struct snd_soc_codec *codec = ac10x->codec;
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AC101_DBG("+++\n");
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set_configuration(codec);
if (drc_used) {
drc_config(codec);
}
/*enable this bit to prevent leakage from ldoin*/
ac101_update_bits(codec, ADDA_TUNE3, (0x1<<OSCEN), (0x1<<OSCEN));
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AC101_DBG("---\n");
return;
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}
int ac101_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level)
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{
switch (level) {
case SND_SOC_BIAS_ON:
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AC101_DBG("SND_SOC_BIAS_ON\n");
break;
case SND_SOC_BIAS_PREPARE:
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AC101_DBG("SND_SOC_BIAS_PREPARE\n");
break;
case SND_SOC_BIAS_STANDBY:
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AC101_DBG("SND_SOC_BIAS_STANDBY\n");
#ifdef CONFIG_AC101_SWITCH_DETECT
switch_hw_config(codec);
#endif
break;
case SND_SOC_BIAS_OFF:
#ifdef CONFIG_AC101_SWITCH_DETECT
ac101_update_bits(codec, ADC_APC_CTRL, (0x1<<HBIASEN), (0<<HBIASEN));
ac101_update_bits(codec, ADC_APC_CTRL, (0x1<<HBIASADCEN), (0<<HBIASADCEN));
#endif
ac101_update_bits(codec, OMIXER_DACA_CTRL, (0xf<<HPOUTPUTENABLE), (0<<HPOUTPUTENABLE));
ac101_update_bits(codec, ADDA_TUNE3, (0x1<<OSCEN), (0<<OSCEN));
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AC101_DBG("SND_SOC_BIAS_OFF\n");
break;
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}
snd_soc_codec_get_dapm(codec)->bias_level = level;
return 0;
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}
int ac101_codec_probe(struct snd_soc_codec *codec)
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{
int ret = 0;
struct ac10x_priv *ac10x;
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ac10x = dev_get_drvdata(codec->dev);
if (ac10x == NULL) {
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AC101_DBG("not set client data!\n");
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return -ENOMEM;
}
ac10x->codec = codec;
INIT_DELAYED_WORK(&ac10x->dlywork, ac10x_work_aif_play);
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INIT_WORK(&ac10x->codec_resume, codec_resume_work);
ac10x->dac_enable = 0;
ac10x->aif1_clken = 0;
mutex_init(&ac10x->dac_mutex);
#if _MASTER_MULTI_CODEC == _MASTER_AC101
seeed_voice_card_register_set_clock(SNDRV_PCM_STREAM_PLAYBACK, ac101_set_clock);
#endif
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set_configuration(ac10x->codec);
/*enable this bit to prevent leakage from ldoin*/
ac101_update_bits(codec, ADDA_TUNE3, (0x1<<OSCEN), (0x1<<OSCEN));
ac101_write(codec, DAC_VOL_CTRL, 0);
/* customized get/put inteface */
for (ret = 0; ret < ARRAY_SIZE(ac101_controls); ret++) {
struct snd_kcontrol_new* skn = &ac101_controls[ret];
skn->get = snd_ac101_get_volsw;
skn->put = snd_ac101_put_volsw;
}
ret = snd_soc_add_codec_controls(codec, ac101_controls, ARRAY_SIZE(ac101_controls));
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if (ret) {
pr_err("[ac10x] Failed to register audio mode control, "
"will continue without it.\n");
}
#ifdef CONFIG_AC101_SWITCH_DETECT
ret = ac101_switch_probe(ac10x);
if (ret) {
// not care the switch return value
}
#endif
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return 0;
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}
/* power down chip */
int ac101_codec_remove(struct snd_soc_codec *codec)
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{
#ifdef CONFIG_AC101_SWITCH_DETECT
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
if (ac10x->irq) {
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devm_free_irq(codec->dev, ac10x->irq, ac10x);
ac10x->irq = 0;
}
if (cancel_work_sync(&ac10x->work_switch) != 0) {
}
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if (cancel_work_sync(&ac10x->work_clear_irq) != 0) {
}
if (ac10x->inpdev) {
input_unregister_device(ac10x->inpdev);
ac10x->inpdev = NULL;
}
#endif
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return 0;
}
int ac101_codec_suspend(struct snd_soc_codec *codec)
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{
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
AC101_DBG("[codec]:suspend\n");
regcache_cache_only(ac10x->regmap101, true);
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return 0;
}
int ac101_codec_resume(struct snd_soc_codec *codec)
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{
struct ac10x_priv *ac10x = snd_soc_codec_get_drvdata(codec);
int ret;
AC101_DBG("[codec]:resume");
/* Sync reg_cache with the hardware */
regcache_cache_only(ac10x->regmap101, false);
ret = regcache_sync(ac10x->regmap101);
if (ret != 0) {
dev_err(codec->dev, "Failed to sync register cache: %d\n", ret);
regcache_cache_only(ac10x->regmap101, true);
return ret;
}
#ifdef CONFIG_AC101_SWITCH_DETECT
ac10x->mode = HEADPHONE_IDLE;
ac10x->state = -1;
#endif
ac101_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
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schedule_work(&ac10x->codec_resume);
return 0;
}
/***************************************************************************/
static ssize_t ac101_debug_store(struct device *dev,
struct device_attribute *attr, const char *buf, size_t count)
{
struct ac10x_priv *ac10x = dev_get_drvdata(dev);
int val = 0, flag = 0;
u16 value_w, value_r;
u8 reg, num, i=0;
val = simple_strtol(buf, NULL, 16);
flag = (val >> 24) & 0xF;
if (flag) {
reg = (val >> 16) & 0xFF;
value_w = val & 0xFFFF;
ac101_write(ac10x->codec, reg, value_w);
printk("write 0x%x to reg:0x%x\n", value_w, reg);
} else {
reg = (val >> 8) & 0xFF;
num = val & 0xff;
printk("\n");
printk("read:start add:0x%x,count:0x%x\n", reg, num);
regcache_cache_bypass(ac10x->regmap101, true);
do {
value_r = ac101_read(ac10x->codec, reg);
printk("0x%x: 0x%04x ", reg++, value_r);
if (++i % 4 == 0 || i == num)
printk("\n");
} while (i < num);
regcache_cache_bypass(ac10x->regmap101, false);
}
return count;
}
static ssize_t ac101_debug_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
printk("echo flag|reg|val > ac10x\n");
printk("eg read star addres=0x06,count 0x10:echo 0610 >ac10x\n");
printk("eg write value:0x13fe to address:0x06 :echo 10613fe > ac10x\n");
return 0;
}
static DEVICE_ATTR(ac10x, 0644, ac101_debug_show, ac101_debug_store);
static struct attribute *audio_debug_attrs[] = {
&dev_attr_ac10x.attr,
NULL,
};
static struct attribute_group audio_debug_attr_group = {
.name = "ac101_debug",
.attrs = audio_debug_attrs,
};
/***************************************************************************/
/************************************************************/
static bool ac101_volatile_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
case PLL_CTRL2:
case HMIC_STS:
return true;
}
return false;
}
static const struct regmap_config ac101_regmap = {
.reg_bits = 8,
.val_bits = 16,
.reg_stride = 1,
.max_register = 0xB5,
.cache_type = REGCACHE_FLAT,
.volatile_reg = ac101_volatile_reg,
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};
/* Sync reg_cache from the hardware */
int ac10x_fill_regcache(struct device* dev, struct regmap* map) {
int r, i, n;
int v;
n = regmap_get_max_register(map);
for (i = 0; i < n; i++) {
regcache_cache_bypass(map, true);
r = regmap_read(map, i, &v);
if (r) {
dev_err(dev, "failed to read register %d\n", i);
continue;
}
regcache_cache_bypass(map, false);
regcache_cache_only(map, true);
r = regmap_write(map, i, v);
regcache_cache_only(map, false);
}
regcache_cache_bypass(map, false);
regcache_cache_only(map, false);
return 0;
}
int ac101_probe(struct i2c_client *i2c, const struct i2c_device_id *id)
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{
struct ac10x_priv *ac10x = i2c_get_clientdata(i2c);
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int ret = 0;
unsigned v = 0;
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AC101_DBG();
static_ac10x = ac10x;
ac10x->regmap101 = devm_regmap_init_i2c(i2c, &ac101_regmap);
if (IS_ERR(ac10x->regmap101)) {
ret = PTR_ERR(ac10x->regmap101);
dev_err(&i2c->dev, "Fail to initialize I/O: %d\n", ret);
return ret;
}
/* Chip reset */
regcache_cache_only(ac10x->regmap101, false);
ret = regmap_write(ac10x->regmap101, CHIP_AUDIO_RST, 0);
msleep(50);
/* sync regcache for FLAT type */
ac10x_fill_regcache(&i2c->dev, ac10x->regmap101);
ret = regmap_read(ac10x->regmap101, CHIP_AUDIO_RST, &v);
if (ret < 0) {
dev_err(&i2c->dev, "failed to read vendor ID: %d\n", ret);
return ret;
}
if (v != AC101_CHIP_ID) {
dev_err(&i2c->dev, "chip is not AC101 (%X)\n", v);
dev_err(&i2c->dev, "Expected %X\n", AC101_CHIP_ID);
return -ENODEV;
}
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ret = sysfs_create_group(&i2c->dev.kobj, &audio_debug_attr_group);
if (ret) {
pr_err("failed to create attr group\n");
}
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ac10x->gpiod_spk_amp_gate = devm_gpiod_get_optional(&i2c->dev, "spk-amp-switch", GPIOD_OUT_LOW);
if (IS_ERR(ac10x->gpiod_spk_amp_gate)) {
ac10x->gpiod_spk_amp_gate = NULL;
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dev_err(&i2c->dev, "failed get spk-amp-switch in device tree\n");
}
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return 0;
}
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void ac101_shutdown(struct i2c_client *i2c)
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{
struct ac10x_priv *ac10x = i2c_get_clientdata(i2c);
struct snd_soc_codec *codec = ac10x->codec;
int reg_val;
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if (codec == NULL) {
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pr_err(": no sound card.\n");
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return;
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}
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/*set headphone volume to 0*/
reg_val = ac101_read(codec, HPOUT_CTRL);
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reg_val &= ~(0x3f<<HP_VOL);
ac101_write(codec, HPOUT_CTRL, reg_val);
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/*disable pa*/
reg_val = ac101_read(codec, HPOUT_CTRL);
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reg_val &= ~(0x1<<HPPA_EN);
ac101_write(codec, HPOUT_CTRL, reg_val);
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/*hardware xzh support*/
reg_val = ac101_read(codec, OMIXER_DACA_CTRL);
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reg_val &= ~(0xf<<HPOUTPUTENABLE);
ac101_write(codec, OMIXER_DACA_CTRL, reg_val);
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/*unmute l/r headphone pa*/
reg_val = ac101_read(codec, HPOUT_CTRL);
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reg_val &= ~((0x1<<RHPPA_MUTE)|(0x1<<LHPPA_MUTE));
ac101_write(codec, HPOUT_CTRL, reg_val);
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return;
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}
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int ac101_remove(struct i2c_client *i2c)
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{
sysfs_remove_group(&i2c->dev.kobj, &audio_debug_attr_group);
return 0;
}
MODULE_DESCRIPTION("ASoC ac10x driver");
MODULE_AUTHOR("huangxin,liushaohua");
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MODULE_AUTHOR("PeterYang<linsheng.yang@seeed.cc>");