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SmartAudio/lichee/linux-4.9/sound/soc/codecs/acx00.c

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/*
* acx00.c -- ACX00 ALSA Soc Audio Codec driver
*
* (C) Copyright 2010-2016 Allwinnertech Technology., Ltd.
*
* Author: Wolfgang Huang <huangjinhui@allwinner.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/module.h>
#include <linux/moduleparam.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/firmware.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/clk.h>
#include <linux/of_gpio.h>
#include <linux/sunxi-gpio.h>
#include <linux/of_device.h>
#include <linux/of_platform.h>
#include <linux/debugfs.h>
#include <linux/slab.h>
#include <linux/mfd/acx00-mfd.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include <linux/workqueue.h>
#include "acx00.h"
#define ACX00_DEF_VOL 0x9F9F
#undef ACX00_DAPM_LINEOUT
struct acx00_priv {
struct acx00 *acx00; /* parent mfd device struct */
struct snd_soc_codec *codec;
struct clk *clk;
unsigned int sample_rate;
unsigned int fmt;
unsigned int enable;
unsigned int spk_gpio;
bool spk_gpio_used;
struct mutex mutex;
struct delayed_work spk_work;
struct delayed_work resume_work;
};
struct sample_rate {
unsigned int samplerate;
unsigned int rate_bit;
};
static const struct sample_rate sample_rate_conv[] = {
{44100, 7},
{48000, 8},
{8000, 0},
{32000, 6},
{22050, 4},
{24000, 5},
{16000, 3},
{11025, 1},
{12000, 2},
{192000, 10},
{96000, 9},
};
void __iomem *io_stat_addr;
static const DECLARE_TLV_DB_SCALE(i2s_mixer_adc_tlv, -600, 600, 1);
static const DECLARE_TLV_DB_SCALE(i2s_mixer_dac_tlv, -600, 600, 1);
static const DECLARE_TLV_DB_SCALE(dac_mixer_adc_tlv, -600, 600, 1);
static const DECLARE_TLV_DB_SCALE(dac_mixer_dac_tlv, -600, 600, 1);
static const DECLARE_TLV_DB_SCALE(line_out_tlv, -450, 150, 0);
static const DECLARE_TLV_DB_SCALE(mic_out_tlv, -450, 150, 0);
static const DECLARE_TLV_DB_SCALE(phoneout_tlv, -450, 150, 0);
static const DECLARE_TLV_DB_SCALE(adc_input_tlv, -450, 150, 0);
static const DECLARE_TLV_DB_SCALE(lineout_tlv, -4800, 150, 1);
static const unsigned int mic_boost_tlv[] = {
TLV_DB_RANGE_HEAD(2),
0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0),
};
static const struct snd_kcontrol_new acx00_codec_controls[] = {
SOC_DOUBLE_TLV("I2S Mixer ADC Volume", AC_I2S_MIXER_GAIN,
I2S_MIXERL_GAIN_ADC, I2S_MIXERR_GAIN_ADC,
0x1, 0, i2s_mixer_adc_tlv),
SOC_DOUBLE_TLV("I2S Mixer DAC Volume", AC_I2S_MIXER_GAIN,
I2S_MIXERR_GAIN_DAC, I2S_MIXERR_GAIN_DAC,
0x1, 0, i2s_mixer_dac_tlv),
SOC_DOUBLE_TLV("DAC Mixer ADC Volume", AC_DAC_MIXER_GAIN,
DAC_MIXERL_GAIN_ADC, DAC_MIXERR_GAIN_ADC,
0x1, 0, dac_mixer_adc_tlv),
SOC_DOUBLE_TLV("DAC Mxier DAC Volume", AC_DAC_MIXER_GAIN,
DAC_MIXERL_GAIN_DAC, DAC_MIXERR_GAIN_DAC,
0x1, 0, dac_mixer_dac_tlv),
SOC_SINGLE_TLV("Line Out Mixer Volume", AC_OUT_MIXER_CTL,
OUT_MIXER_LINE_VOL, 0x7, 0, line_out_tlv),
SOC_DOUBLE_TLV("MIC Out Mixer Volume", AC_OUT_MIXER_CTL,
OUT_MIXER_MIC1_VOL, OUT_MIXER_MIC2_VOL,
0x7, 0, mic_out_tlv),
SOC_SINGLE_TLV("ADC Input Volume", AC_ADC_MIC_CTL,
ADC_GAIN, 0x07, 0, adc_input_tlv),
SOC_SINGLE_TLV("LINEOUT Volume", AC_LINEOUT_CTL,
LINEOUT_VOL, 0x1f, 0, lineout_tlv),
SOC_SINGLE_TLV("MIC1 Boost Volume", AC_ADC_MIC_CTL,
MIC1_BOOST, 0x07, 0, mic_boost_tlv),
SOC_SINGLE_TLV("MIC2 Boost Volume", AC_ADC_MIC_CTL,
MIC2_BOOST, 0x07, 0, mic_boost_tlv),
};
/* Enable I2S & DAC clk, then enable the DAC digital part */
static int acx00_playback_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
snd_soc_update_bits(codec, AC_SYS_CLK_CTL,
(0x1<<SYS_CLK_DAC), (0x1<<SYS_CLK_DAC));
snd_soc_update_bits(codec, AC_SYS_MOD_RST,
(0x1<<MOD_RST_DAC), (0x1<<MOD_RST_DAC));
snd_soc_update_bits(codec, AC_DAC_CTL,
(0x1<<DAC_CTL_DAC_EN), (0x1<<DAC_CTL_DAC_EN));
break;
case SND_SOC_DAPM_POST_PMD:
snd_soc_update_bits(codec, AC_SYS_CLK_CTL,
(0x1<<SYS_CLK_DAC), (0x0<<SYS_CLK_DAC));
snd_soc_update_bits(codec, AC_SYS_MOD_RST,
(0x1<<MOD_RST_DAC), (0x0<<MOD_RST_DAC));
snd_soc_update_bits(codec, AC_DAC_CTL,
(0x1<<DAC_CTL_DAC_EN), (0x0<<DAC_CTL_DAC_EN));
break;
default:
break;
}
return 0;
}
/* Enable I2S & ADC clk, then enable the ADC digital part */
static int acx00_capture_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
snd_soc_update_bits(codec, AC_SYS_CLK_CTL,
(0x1<<SYS_CLK_ADC), (0x1<<SYS_CLK_ADC));
snd_soc_update_bits(codec, AC_SYS_MOD_RST,
(0x1<<MOD_RST_ADC), (0x1<<MOD_RST_ADC));
snd_soc_update_bits(codec, AC_ADC_CTL,
(0x1<<ADC_EN), (0x1<<ADC_EN));
break;
case SND_SOC_DAPM_POST_PMD:
snd_soc_update_bits(codec, AC_SYS_CLK_CTL,
(0x1<<SYS_CLK_ADC), (0x0<<SYS_CLK_ADC));
snd_soc_update_bits(codec, AC_SYS_MOD_RST,
(0x1<<MOD_RST_ADC), (0x0<<MOD_RST_ADC));
snd_soc_update_bits(codec, AC_ADC_CTL,
(0x1<<ADC_EN), (0x0<<ADC_EN));
break;
default:
break;
}
return 0;
}
/*
* we used for three scene:
* 1. No external Spker & DAPM LINEOUT used, we just enable the LINEOUT in the
* ALSA codec probe(acx00_codec_probe) and resume, and we shutdown the LINEOUT
* in device shutdown or suspend.
* 2. No external Spker, but DAPM LINEOUT used, we just using the LINEOUT
* enable or disable throught the DAPM control.
* 3. External Spker & DAPM LINEOUT used, we just using the LINEOUT and
* External Spker control GPIO enable or disable through DAPM control.
*/
static unsigned int spk_delay = 100;
module_param(spk_delay, int, 0644);
MODULE_PARM_DESC(spk_delay, "ACX00-Codec spk mute delay time");
static void acx00_spk_enable(struct work_struct *work)
{
struct acx00_priv *priv = container_of(work,
struct acx00_priv, spk_work.work);
gpio_set_value(priv->spk_gpio, 1);
}
static int acx00_lineout_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
if (!priv->enable) {
snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
(1<<LINEL_SRC_EN), (1<<LINEL_SRC_EN));
snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
(1<<LINER_SRC_EN), (1<<LINER_SRC_EN));
msleep(100);
priv->enable = 1;
}
#ifdef ACX00_DAPM_LINEOUT
snd_soc_update_bits(codec, AC_LINEOUT_CTL,
(1<<LINEOUT_EN), (1<<LINEOUT_EN));
mdelay(50);
#endif
if (priv->spk_gpio_used) {
if (spk_delay == 0) {
gpio_set_value(priv->spk_gpio, 1);
/*
* time delay to wait spk pa work fine,
* general setting 50ms
*/
mdelay(50);
} else
schedule_delayed_work(&priv->spk_work,
msecs_to_jiffies(spk_delay));
}
break;
case SND_SOC_DAPM_PRE_PMD:
mdelay(50);
if (priv->spk_gpio_used) {
gpio_set_value(priv->spk_gpio, 0);
msleep(50);
}
#ifdef ACX00_DAPM_LINEOUT
snd_soc_update_bits(codec, AC_LINEOUT_CTL,
(1<<LINEOUT_EN), (0<<LINEOUT_EN));
#endif
break;
default:
break;
}
return 0;
}
/* AC_I2S_MIXER_SRC : 0x2114 */
static const struct snd_kcontrol_new i2sl_mixer_src[] = {
SOC_DAPM_SINGLE("I2SDACL Switch", AC_I2S_MIXER_SRC,
I2S_MIXERL_SRC_DAC, 1, 0),
SOC_DAPM_SINGLE("ADCL Switch", AC_I2S_MIXER_SRC,
I2S_MIXERL_SRC_ADC, 1, 0),
};
static const struct snd_kcontrol_new i2sr_mixer_src[] = {
SOC_DAPM_SINGLE("I2SDACR Switch", AC_I2S_MIXER_SRC,
I2S_MIXERR_SRC_DAC, 1, 0),
SOC_DAPM_SINGLE("ADCR Switch", AC_I2S_MIXER_SRC,
I2S_MIXERR_SRC_ADC, 1, 0),
};
/* AC_DAC_MIXER_SRC : 0x2202 */
static const struct snd_kcontrol_new dacl_mixer_src[] = {
SOC_DAPM_SINGLE("I2SDACL Switch", AC_DAC_MIXER_SRC,
DAC_MIXERL_SRC_DAC, 1, 0),
SOC_DAPM_SINGLE("ADCL Switch", AC_DAC_MIXER_SRC,
DAC_MIXERL_SRC_ADC, 1, 0),
};
static const struct snd_kcontrol_new dacr_mixer_src[] = {
SOC_DAPM_SINGLE("I2SDACR Switch", AC_DAC_MIXER_SRC,
DAC_MIXERR_SRC_DAC, 1, 0),
SOC_DAPM_SINGLE("ADCR Switch", AC_DAC_MIXER_SRC,
DAC_MIXERR_SRC_ADC, 1, 0),
};
/* AC_OUT_MIXER_SRC : 0x2222 */
static const struct snd_kcontrol_new left_output_mixer[] = {
SOC_DAPM_SINGLE("MIC1 Switch", AC_OUT_MIXER_SRC,
OUT_MIXERL_SRC_MIC1, 1, 0),
SOC_DAPM_SINGLE("MIC2 Switch", AC_OUT_MIXER_SRC,
OUT_MIXERL_SRC_MIC2, 1, 0),
SOC_DAPM_SINGLE("PhonePN Switch", AC_OUT_MIXER_SRC,
OUT_MIXERL_SRC_PHPN, 1, 0),
SOC_DAPM_SINGLE("PhoneN Switch", AC_OUT_MIXER_SRC,
OUT_MIXERL_SRC_PHN, 1, 0),
SOC_DAPM_SINGLE("LINEINL Switch", AC_OUT_MIXER_SRC,
OUT_MIXERL_SRC_LINEL, 1, 0),
SOC_DAPM_SINGLE("DACL Switch", AC_OUT_MIXER_SRC,
OUT_MIXERL_SRC_DACL, 1, 0),
SOC_DAPM_SINGLE("DACR Switch", AC_OUT_MIXER_SRC,
OUT_MIXERL_SRC_DACR, 1, 0),
};
static const struct snd_kcontrol_new right_output_mixer[] = {
SOC_DAPM_SINGLE("MIC1 Switch", AC_OUT_MIXER_SRC,
OUT_MIXERR_SRC_MIC1, 1, 0),
SOC_DAPM_SINGLE("MIC2 Switch", AC_OUT_MIXER_SRC,
OUT_MIXERR_SRC_MIC2, 1, 0),
SOC_DAPM_SINGLE("PhonePN Switch", AC_OUT_MIXER_SRC,
OUT_MIXERR_SRC_PHPN, 1, 0),
SOC_DAPM_SINGLE("PhoneP Switch", AC_OUT_MIXER_SRC,
OUT_MIXERR_SRC_PHP, 1, 0),
SOC_DAPM_SINGLE("LINEINR Switch", AC_OUT_MIXER_SRC,
OUT_MIXERR_SRC_LINER, 1, 0),
SOC_DAPM_SINGLE("DACR Switch", AC_OUT_MIXER_SRC,
OUT_MIXERR_SRC_DACR, 1, 0),
SOC_DAPM_SINGLE("DACL Switch", AC_OUT_MIXER_SRC,
OUT_MIXERR_SRC_DACL, 1, 0),
};
/* AC_LINEOUT_CTL : 0x2224 */
const char * const left_lineout_text[] = {
"Left OMixer", "LR OMixer",
};
static const struct soc_enum left_lineout_enum =
SOC_ENUM_SINGLE(AC_LINEOUT_CTL, LINEL_SRC,
ARRAY_SIZE(left_lineout_text), left_lineout_text);
static const struct snd_kcontrol_new left_lineout_mux =
SOC_DAPM_ENUM("Left LINEOUT Mux", left_lineout_enum);
const char * const right_lineout_text[] = {
"Right OMixer", "LR OMixer",
};
static const struct soc_enum right_lineout_enum =
SOC_ENUM_SINGLE(AC_LINEOUT_CTL, LINER_SRC,
ARRAY_SIZE(right_lineout_text), right_lineout_text);
static const struct snd_kcontrol_new right_lineout_mux =
SOC_DAPM_ENUM("Right LINEOUT Mux", right_lineout_enum);
/* AC_ADC_MIXER_SRC : 0x2322 */
static const struct snd_kcontrol_new left_input_mixer[] = {
SOC_DAPM_SINGLE("MIC1 Switch", AC_ADC_MIXER_SRC,
ADC_MIXERL_MIC1, 1, 0),
SOC_DAPM_SINGLE("MIC2 Switch", AC_ADC_MIXER_SRC,
ADC_MIXERL_MIC2, 1, 0),
SOC_DAPM_SINGLE("PhonePN Switch", AC_ADC_MIXER_SRC,
ADC_MIXERL_PHPN, 1, 0),
SOC_DAPM_SINGLE("PhoneN Switch", AC_ADC_MIXER_SRC,
ADC_MIXERL_PHN, 1, 0),
SOC_DAPM_SINGLE("LINEINL Switch", AC_ADC_MIXER_SRC,
ADC_MIXERL_LINEL, 1, 0),
SOC_DAPM_SINGLE("OMixerL Switch", AC_ADC_MIXER_SRC,
ADC_MIXERL_MIXL, 1, 0),
SOC_DAPM_SINGLE("OMixerR Switch", AC_ADC_MIXER_SRC,
ADC_MIXERL_MIXR, 1, 0),
};
static const struct snd_kcontrol_new right_input_mixer[] = {
SOC_DAPM_SINGLE("MIC1 Switch", AC_ADC_MIXER_SRC,
ADC_MIXERR_MIC1, 1, 0),
SOC_DAPM_SINGLE("MIC2 Switch", AC_ADC_MIXER_SRC,
ADC_MIXERR_MIC2, 1, 0),
SOC_DAPM_SINGLE("PhonePN Switch", AC_ADC_MIXER_SRC,
ADC_MIXERR_PHPN, 1, 0),
SOC_DAPM_SINGLE("PhoneP Switch", AC_ADC_MIXER_SRC,
ADC_MIXERR_PHP, 1, 0),
SOC_DAPM_SINGLE("LINEINR Switch", AC_ADC_MIXER_SRC,
ADC_MIXERR_LINER, 1, 0),
SOC_DAPM_SINGLE("OMixerR Switch", AC_ADC_MIXER_SRC,
ADC_MIXERR_MIXR, 1, 0),
SOC_DAPM_SINGLE("OMixerL Switch", AC_ADC_MIXER_SRC,
ADC_MIXERR_MIXL, 1, 0),
};
static const struct snd_soc_dapm_widget acx00_codec_dapm_widgets[] = {
SND_SOC_DAPM_AIF_IN_E("DACL", "Playback", 0, AC_DAC_CTL,
OUT_MIXER_DACL_EN, 0,
acx00_playback_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_IN_E("DACR", "Playback", 0,
AC_DAC_CTL, OUT_MIXER_DACR_EN, 0,
acx00_playback_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_OUT_E("ADCL", "Capture", 0,
AC_ADC_MIC_CTL, ADCL_EN, 0,
acx00_capture_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_OUT_E("ADCR", "Capture", 0,
AC_ADC_MIC_CTL, ADCR_EN, 0,
acx00_capture_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MIXER("Left Output Mixer", AC_OUT_MIXER_CTL,
OUT_MIXER_LMIX_EN, 0,
left_output_mixer, ARRAY_SIZE(left_output_mixer)),
SND_SOC_DAPM_MIXER("Right Output Mixer", AC_OUT_MIXER_CTL,
OUT_MIXER_RMIX_EN, 0, right_output_mixer,
ARRAY_SIZE(right_output_mixer)),
SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0,
left_input_mixer, ARRAY_SIZE(left_input_mixer)),
SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0,
right_input_mixer, ARRAY_SIZE(right_input_mixer)),
SND_SOC_DAPM_MIXER("Left DAC Mixer", AC_OUT_MIXER_CTL,
OUT_MIXER_DACL_EN, 0, dacl_mixer_src,
ARRAY_SIZE(dacl_mixer_src)),
SND_SOC_DAPM_MIXER("Right DAC Mixer", AC_OUT_MIXER_CTL,
OUT_MIXER_DACR_EN, 0, dacr_mixer_src,
ARRAY_SIZE(dacr_mixer_src)),
SND_SOC_DAPM_MIXER("Left I2S Mixer", SND_SOC_NOPM,
0, 0, i2sl_mixer_src, ARRAY_SIZE(i2sl_mixer_src)),
SND_SOC_DAPM_MIXER("Right I2S Mixer", SND_SOC_NOPM,
0, 0, i2sr_mixer_src, ARRAY_SIZE(i2sr_mixer_src)),
SND_SOC_DAPM_MUX("Left LINEOUT Mux", SND_SOC_NOPM,
0, 0, &left_lineout_mux),
SND_SOC_DAPM_MUX("Right LINEOUT Mux", SND_SOC_NOPM,
0, 0, &right_lineout_mux),
SND_SOC_DAPM_PGA("MIC1 PGA", AC_ADC_MIC_CTL,
MIC1_GAIN_EN, 0, NULL, 0),
SND_SOC_DAPM_PGA("MIC2 PGA", AC_ADC_MIC_CTL,
MIC2_GAIN_EN, 0, NULL, 0),
SND_SOC_DAPM_MICBIAS("MIC Bias", AC_MICBIAS_CTL,
MMBIAS_EN, 0),
/* PHONEIN & PHONEOUT not enable in pin assign */
SND_SOC_DAPM_INPUT("PHONEINP"),
SND_SOC_DAPM_INPUT("PHONEINN"),
SND_SOC_DAPM_INPUT("PHONEINPN"),
/* endpoint define */
SND_SOC_DAPM_LINE("LINEIN", NULL),
SND_SOC_DAPM_LINE("LINEOUT", acx00_lineout_event),
SND_SOC_DAPM_MIC("MIC1", NULL),
SND_SOC_DAPM_MIC("MIC2", NULL),
};
static const struct snd_soc_dapm_route acx00_codec_dapm_routes[] = {
{"Left Output Mixer", "MIC1 Switch", "MIC1 PGA"},
{"Left Output Mixer", "MIC2 Switch", "MIC2 PGA"},
{"Left Output Mixer", "PhonePN Switch", "PHONEINPN"},
{"Left Output Mixer", "PhoneN Switch", "PHONEINN"},
{"Left Output Mixer", "LINEINL Switch", "LINEIN"},
{"Left Output Mixer", "DACR Switch", "Right DAC Mixer"},
{"Left Output Mixer", "DACL Switch", "Left DAC Mixer"},
{"Right Output Mixer", "MIC1 Switch", "MIC1 PGA"},
{"Right Output Mixer", "MIC2 Switch", "MIC2 PGA"},
{"Right Output Mixer", "PhonePN Switch", "PHONEINPN"},
{"Right Output Mixer", "PhoneP Switch", "PHONEINP"},
{"Right Output Mixer", "LINEINR Switch", "LINEIN"},
{"Right Output Mixer", "DACR Switch", "Right DAC Mixer"},
{"Right Output Mixer", "DACL Switch", "Left DAC Mixer"},
{"Left LINEOUT Mux", NULL, "Left Output Mixer"},
{"Left LINEOUT Mux", "LR OMixer", "Right Output Mixer"},
{"Right LINEOUT Mux", NULL, "Right Output Mixer"},
{"Right LINEOUT Mux", "LR OMixer", "Left Output Mixer"},
{"Left Input Mixer", "MIC1 Switch", "MIC1 PGA"},
{"Left Input Mixer", "MIC2 Switch", "MIC2 PGA"},
{"Left Input Mixer", "PhonePN Switch", "PHONEINPN"},
{"Left Input Mixer", "PhoneN Switch", "PHONEINN"},
{"Left Input Mixer", "LINEINL Switch", "LINEIN"},
{"Left Input Mixer", "OMixerL Switch", "Left Output Mixer"},
{"Left Input Mixer", "OMixerR Switch", "Right Output Mixer"},
{"Right Input Mixer", "MIC1 Switch", "MIC1 PGA"},
{"Right Input Mixer", "MIC2 Switch", "MIC2 PGA"},
{"Right Input Mixer", "PhonePN Switch", "PHONEINPN"},
{"Right Input Mixer", "PhoneP Switch", "PHONEINP"},
{"Right Input Mixer", "LINEINR Switch", "LINEIN"},
{"Right Input Mixer", "OMixerR Switch", "Right Output Mixer"},
{"Right Input Mixer", "OMixerL Switch", "Left Output Mixer"},
{"Left I2S Mixer", "I2SDACL Switch", "DACL"},
{"Left I2S Mixer", "ADCL Switch", "Left Input Mixer"},
{"Right I2S Mixer", "I2SDACR Switch", "DACR"},
{"Right I2S Mixer", "ADCR Switch", "Right Input Mixer"},
{"Left DAC Mixer", "I2SDACL Switch", "DACL"},
{"Left DAC Mixer", "ADCL Switch", "Left Input Mixer"},
{"Right DAC Mixer", "I2SDACR Switch", "DACR"},
{"Right DAC Mixer", "ADCR Switch", "Right Input Mixer"},
{"ADCL", NULL, "Left I2S Mixer"},
{"ADCR", NULL, "Right I2S Mixer"},
{"LINEOUT", NULL, "Left LINEOUT Mux"},
{"LINEOUT", NULL, "Right LINEOUT Mux"},
{"MIC Bias", NULL, "MIC1"},
{"MIC Bias", NULL, "MIC2"},
{"MIC1 PGA", NULL, "MIC Bias"},
{"MIC2 PGA", NULL, "MIC Bias"},
};
static void acx00_codec_txctrl_enable(struct snd_soc_codec *codec,
int enable)
{
pr_debug("Enter %s, enable %d\n", __func__, enable);
if (enable) {
snd_soc_update_bits(codec, AC_I2S_CTL,
(1<<I2S_RX_EN), (1<<I2S_RX_EN));
} else {
snd_soc_update_bits(codec, AC_I2S_CTL,
(1<<I2S_RX_EN), (0<<I2S_RX_EN));
}
pr_debug("End %s, enable %d\n", __func__, enable);
}
static void acx00_codec_rxctrl_enable(struct snd_soc_codec *codec,
int enable)
{
pr_debug("Enter %s, enable %d\n", __func__, enable);
if (enable) {
snd_soc_update_bits(codec, AC_I2S_CTL,
(1<<I2S_TX_EN), (1<<I2S_TX_EN));
} else {
snd_soc_update_bits(codec, AC_I2S_CTL,
(1<<I2S_TX_EN), (0<<I2S_TX_EN));
}
pr_debug("End %s, enable %d\n", __func__, enable);
}
static void acx00_codec_init(struct snd_soc_codec *codec)
{
struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
/* acx00_codec sysctl init */
acx00_reg_write(priv->acx00, 0x0010, 0x03);
acx00_reg_write(priv->acx00, 0x0012, 0x01);
/* The bit3 need to setup to 1 for bias current. */
snd_soc_update_bits(codec, AC_MICBIAS_CTL,
(0x1 << ADDA_BIAS_CUR), (0x1 << ADDA_BIAS_CUR));
/* enable the output & global enable bit */
snd_soc_update_bits(codec, AC_I2S_CTL,
(1<<I2S_SDO0_EN), (1<<I2S_SDO0_EN));
snd_soc_update_bits(codec, AC_I2S_CTL, (1<<I2S_GEN), (1<<I2S_GEN));
/* Default setting slot width as 32 bit for I2S */
snd_soc_update_bits(codec, AC_I2S_FMT0,
(7<<I2S_FMT_SLOT_WIDTH), (7<<I2S_FMT_SLOT_WIDTH));
/* default setting 0xA0A0 for ADC & DAC Volume */
snd_soc_write(codec, AC_I2S_DAC_VOL, ACX00_DEF_VOL);
snd_soc_write(codec, AC_I2S_ADC_VOL, ACX00_DEF_VOL);
/* Enable HPF for high pass filter */
snd_soc_update_bits(codec, AC_DAC_CTL,
(1<<DAC_CTL_HPF_EN), (1<<DAC_CTL_HPF_EN));
/* LINEOUT ANTI POP & Click noise */
snd_soc_update_bits(codec, AC_LINEOUT_CTL,
(0x7<<LINE_ANTI_TIME), (0x3<<LINE_ANTI_TIME));
snd_soc_update_bits(codec, AC_LINEOUT_CTL,
(0x3<<LINE_SLOPE_SEL), (0x3<<LINE_SLOPE_SEL));
/* enable & setting adc convert delay time */
snd_soc_update_bits(codec, AC_ADC_CTL, (0x3<<ADC_DELAY_TIME),
(0x3<<ADC_DELAY_TIME));
snd_soc_update_bits(codec, AC_ADC_CTL, (1<<ADC_DELAY_EN),
(1<<ADC_DELAY_EN));
if (priv->spk_gpio_used) {
snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
(1<<LINEL_SRC_EN), (1<<LINEL_SRC_EN));
snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
(1<<LINER_SRC_EN), (1<<LINER_SRC_EN));
priv->enable = 1;
}
#ifndef ACX00_DAPM_LINEOUT
snd_soc_update_bits(codec, AC_LINEOUT_CTL, (1<<LINEOUT_EN),
(1<<LINEOUT_EN));
#endif
}
static int acx00_codec_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;
int i;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
snd_soc_update_bits(codec, AC_I2S_FMT0,
(7<<I2S_FMT_SAMPLE), (3<<I2S_FMT_SAMPLE));
break;
case SNDRV_PCM_FORMAT_S24_LE:
snd_soc_update_bits(codec, AC_I2S_FMT0,
(7<<I2S_FMT_SAMPLE), (5<<I2S_FMT_SAMPLE));
break;
case SNDRV_PCM_FORMAT_S32_LE:
snd_soc_update_bits(codec, AC_I2S_FMT0,
(7<<I2S_FMT_SAMPLE), (7<<I2S_FMT_SAMPLE));
break;
default:
dev_err(codec->dev, "unrecognized format support\n");
break;
}
for (i = 0; i < ARRAY_SIZE(sample_rate_conv); i++) {
if (sample_rate_conv[i].samplerate == params_rate(params)) {
snd_soc_update_bits(codec, AC_SYS_SR_CTL,
(SYS_SR_MASK<<SYS_SR_BIT),
(sample_rate_conv[i].rate_bit<<SYS_SR_BIT));
}
}
return 0;
}
static int acx00_codec_dai_set_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
return 0;
}
static int acx00_codec_dai_set_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct acx00_priv *priv = snd_soc_dai_get_drvdata(codec_dai);
struct snd_soc_codec *codec = priv->codec;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
/* codec clk & FRM master */
case SND_SOC_DAIFMT_CBM_CFM:
snd_soc_update_bits(codec, AC_I2S_CLK,
(0x1<<I2S_BCLK_OUT), (0x1<<I2S_BCLK_OUT));
snd_soc_update_bits(codec, AC_I2S_CLK,
(0x1<<I2S_LRCK_OUT), (0x1<<I2S_LRCK_OUT));
break;
/* codec clk & FRM slave */
case SND_SOC_DAIFMT_CBS_CFS:
snd_soc_update_bits(codec, AC_I2S_CLK,
(0x1<<I2S_BCLK_OUT), 0x0<<I2S_BCLK_OUT);
snd_soc_update_bits(codec, AC_I2S_CLK,
(0x1<<I2S_LRCK_OUT), 0x0<<I2S_LRCK_OUT);
break;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
snd_soc_update_bits(codec, AC_I2S_CLK,
(0x3FF<<I2S_LRCK_PERIOD),
(0x1F<<I2S_LRCK_PERIOD));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x3<<I2S_FMT_MODE), (0x1<<I2S_FMT_MODE));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x1<<I2S_FMT_TX_OFFSET),
(0x1<<I2S_FMT_TX_OFFSET));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x1<<I2S_FMT_RX_OFFSET),
(0x1<<I2S_FMT_RX_OFFSET));
break;
case SND_SOC_DAIFMT_RIGHT_J:
snd_soc_update_bits(codec, AC_I2S_CLK,
(0x3FF<<I2S_LRCK_PERIOD),
(0x1F<<I2S_LRCK_PERIOD));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x3<<I2S_FMT_MODE), (0x2<<I2S_FMT_MODE));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x1<<I2S_FMT_TX_OFFSET),
(0x0<<I2S_FMT_TX_OFFSET));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x1<<I2S_FMT_TX_OFFSET),
(0x0<<I2S_FMT_RX_OFFSET));
break;
case SND_SOC_DAIFMT_LEFT_J:
snd_soc_update_bits(codec, AC_I2S_CLK,
(0x3FF<<I2S_LRCK_PERIOD),
(0x1F<<I2S_LRCK_PERIOD));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x3<<I2S_FMT_MODE), (0x1<<I2S_FMT_MODE));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x1<<I2S_FMT_TX_OFFSET),
(0x0<<I2S_FMT_TX_OFFSET));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x1<<I2S_FMT_RX_OFFSET),
(0x0<<I2S_FMT_RX_OFFSET));
break;
case SND_SOC_DAIFMT_DSP_A:
snd_soc_update_bits(codec, AC_I2S_CLK,
(0x3FF<<I2S_LRCK_PERIOD),
(0x3F<<I2S_LRCK_PERIOD));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x3<<I2S_FMT_MODE), (0x0<<I2S_FMT_MODE));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x1<<I2S_FMT_TX_OFFSET),
(0x1<<I2S_FMT_TX_OFFSET));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x1<<I2S_FMT_RX_OFFSET),
(0x1<<I2S_FMT_RX_OFFSET));
break;
case SND_SOC_DAIFMT_DSP_B:
snd_soc_update_bits(codec, AC_I2S_CLK,
(0x3FF<<I2S_LRCK_PERIOD),
(0x3F<<I2S_LRCK_PERIOD));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x3<<I2S_FMT_MODE), (0x0<<I2S_FMT_MODE));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x1<<I2S_FMT_TX_OFFSET),
(0x0<<I2S_FMT_TX_OFFSET));
snd_soc_update_bits(codec, AC_I2S_FMT0,
(0x1<<I2S_FMT_RX_OFFSET),
(0x0<<I2S_FMT_RX_OFFSET));
break;
default:
dev_err(codec->dev, "format setting failed\n");
break;
}
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
snd_soc_update_bits(codec, AC_I2S_FMT1,
(0x1<<I2S_FMT_BCLK_POLAR),
(0x0<<I2S_FMT_BCLK_POLAR));
snd_soc_update_bits(codec, AC_I2S_FMT1,
(0x1<<I2S_FMT_LRCK_POLAR),
(0x0<<I2S_FMT_LRCK_POLAR));
break;
case SND_SOC_DAIFMT_NB_IF:
snd_soc_update_bits(codec, AC_I2S_FMT1,
(0x1<<I2S_FMT_BCLK_POLAR),
(0x0<<I2S_FMT_BCLK_POLAR));
snd_soc_update_bits(codec, AC_I2S_FMT1,
(0x1<<I2S_FMT_LRCK_POLAR),
(0x1<<I2S_FMT_LRCK_POLAR));
break;
case SND_SOC_DAIFMT_IB_NF:
snd_soc_update_bits(codec, AC_I2S_FMT1,
(0x1<<I2S_FMT_BCLK_POLAR),
(0x1<<I2S_FMT_BCLK_POLAR));
snd_soc_update_bits(codec, AC_I2S_FMT1,
(0x1<<I2S_FMT_LRCK_POLAR),
(0x0<<I2S_FMT_LRCK_POLAR));
break;
case SND_SOC_DAIFMT_IB_IF:
snd_soc_update_bits(codec, AC_I2S_FMT1,
(0x1<<I2S_FMT_BCLK_POLAR),
(0x1<<I2S_FMT_BCLK_POLAR));
snd_soc_update_bits(codec, AC_I2S_FMT1,
(0x1<<I2S_FMT_LRCK_POLAR),
(0x1<<I2S_FMT_LRCK_POLAR));
break;
default:
dev_err(codec->dev, "invert clk setting failed\n");
return -EINVAL;
}
return 0;
}
static int acx00_codec_dai_set_clkdiv(struct snd_soc_dai *codec_dai,
int clk_id, int clk_div)
{
struct acx00_priv *priv = snd_soc_dai_get_drvdata(codec_dai);
struct snd_soc_codec *codec = priv->codec;
unsigned int bclk_div;
/*
* when PCM mode, setting as 64fs, when I2S mode as 32fs,
* then two channel, then just as 64fs
*/
unsigned int div_ratio = clk_div / 64;
switch (div_ratio) {
case 1:
bclk_div = I2S_BCLK_DIV_1;
break;
case 2:
bclk_div = I2S_BCLK_DIV_2;
break;
case 4:
bclk_div = I2S_BCLK_DIV_3;
break;
case 6:
bclk_div = I2S_BCLK_DIV_4;
break;
case 8:
bclk_div = I2S_BCLK_DIV_5;
break;
case 12:
bclk_div = I2S_BCLK_DIV_6;
break;
case 16:
bclk_div = I2S_BCLK_DIV_7;
break;
case 24:
bclk_div = I2S_BCLK_DIV_8;
break;
case 32:
bclk_div = I2S_BCLK_DIV_9;
break;
case 48:
bclk_div = I2S_BCLK_DIV_10;
break;
case 64:
bclk_div = I2S_BCLK_DIV_11;
break;
case 96:
bclk_div = I2S_BCLK_DIV_12;
break;
case 128:
bclk_div = I2S_BCLK_DIV_13;
break;
case 176:
bclk_div = I2S_BCLK_DIV_14;
break;
case 192:
bclk_div = I2S_BCLK_DIV_15;
break;
default:
dev_err(codec->dev, "setting blck div failed\n");
break;
}
snd_soc_update_bits(codec, AC_I2S_CLK,
(I2S_BCLK_DIV_MASK<<I2S_BLCK_DIV),
(bclk_div<<I2S_BLCK_DIV));
return 0;
}
static int acx00_codec_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai)
{
return 0;
}
static bool acx00_loop_en;
module_param(acx00_loop_en, bool, 0644);
MODULE_PARM_DESC(acx00_loop_en, "ACX00-Codec audio loopback debug(Y=enable, N=disable)");
static int acx00_codec_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *codec_dai)
{
return 0;
}
static int acx00_codec_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *codec_dai)
{
snd_soc_update_bits(codec_dai->codec, AC_SYS_CLK_CTL,
(0x1<<SYS_CLK_I2S), (0x1<<SYS_CLK_I2S));
snd_soc_update_bits(codec_dai->codec, AC_SYS_MOD_RST,
(0x1<<MOD_RST_I2S), (0x1<<MOD_RST_I2S));
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (acx00_loop_en)
snd_soc_update_bits(codec_dai->codec, AC_I2S_FMT0,
(0x1<<I2S_FMT_LOOP),
(0x1<<I2S_FMT_LOOP));
else
snd_soc_update_bits(codec_dai->codec, AC_I2S_FMT0,
(0x1<<I2S_FMT_LOOP),
(0x0<<I2S_FMT_LOOP));
acx00_codec_txctrl_enable(codec_dai->codec, 1);
} else
acx00_codec_rxctrl_enable(codec_dai->codec, 1);
return 0;
}
static int acx00_codec_digital_mute(struct snd_soc_dai *codec_dai,
int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
if (mute)
snd_soc_write(codec, AC_I2S_DAC_VOL, 0);
else
snd_soc_write(codec, AC_I2S_DAC_VOL, ACX00_DEF_VOL);
return 0;
}
static void acx00_codec_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
acx00_codec_txctrl_enable(codec, 0);
else
acx00_codec_rxctrl_enable(codec, 0);
}
static const struct snd_soc_dai_ops acx00_codec_dai_ops = {
.hw_params = acx00_codec_hw_params,
.shutdown = acx00_codec_shutdown,
.digital_mute = acx00_codec_digital_mute,
.set_sysclk = acx00_codec_dai_set_sysclk,
.set_fmt = acx00_codec_dai_set_fmt,
.set_clkdiv = acx00_codec_dai_set_clkdiv,
.startup = acx00_codec_startup,
.trigger = acx00_codec_trigger,
.prepare = acx00_codec_prepare,
};
static struct snd_soc_dai_driver acx00_codec_dai[] = {
{
.name = "acx00-dai",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000
| SNDRV_PCM_RATE_KNOT,
.formats = SNDRV_PCM_FMTBIT_S16_LE
| SNDRV_PCM_FMTBIT_S24_LE
| SNDRV_PCM_FMTBIT_S32_LE,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000
| SNDRV_PCM_RATE_KNOT,
.formats = SNDRV_PCM_FMTBIT_S16_LE
| SNDRV_PCM_FMTBIT_S24_LE
| SNDRV_PCM_FMTBIT_S32_LE,
},
.ops = &acx00_codec_dai_ops,
},
};
static void acx00_codec_resume_work(struct work_struct *work)
{
struct acx00_priv *priv = container_of(work,
struct acx00_priv, resume_work.work);
acx00_codec_init(priv->codec);
}
static int acx00_codec_probe(struct snd_soc_codec *codec)
{
struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
int ret = 0;
mutex_init(&priv->mutex);
priv->codec = codec;
/* Add virtual switch */
ret = snd_soc_add_codec_controls(codec, acx00_codec_controls,
ARRAY_SIZE(acx00_codec_controls));
if (ret) {
pr_err("[audio-codec] Failed to register audio mode control, will continue without it.\n");
}
snd_soc_dapm_new_controls(dapm, acx00_codec_dapm_widgets, ARRAY_SIZE(acx00_codec_dapm_widgets));
snd_soc_dapm_add_routes(dapm, acx00_codec_dapm_routes, ARRAY_SIZE(acx00_codec_dapm_routes));
/* using late_initcall to wait 120ms acx00-core to make chip reset */
acx00_codec_init(codec);
INIT_DELAYED_WORK(&priv->spk_work, acx00_spk_enable);
INIT_DELAYED_WORK(&priv->resume_work, acx00_codec_resume_work);
return 0;
}
static int acx00_codec_remove(struct snd_soc_codec *codec)
{
struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
cancel_delayed_work_sync(&priv->spk_work);
cancel_delayed_work_sync(&priv->resume_work);
return 0;
}
static unsigned int acx00_codec_read(struct snd_soc_codec *codec,
unsigned int reg)
{
unsigned int data;
struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
/* Device I/O API */
data = acx00_reg_read(priv->acx00, reg);
return data;
}
static int acx00_codec_write(struct snd_soc_codec *codec,
unsigned int reg, unsigned int value)
{
struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
return acx00_reg_write(priv->acx00, reg, value);
}
static int sunxi_gpio_iodisable(u32 gpio)
{
char pin_name[8];
u32 config, ret;
sunxi_gpio_to_name(gpio, pin_name);
config = 7 << 16;
ret = pin_config_set(SUNXI_PINCTRL, pin_name, config);
return ret;
}
static int acx00_codec_suspend(struct snd_soc_codec *codec)
{
struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
pr_debug("Enter %s\n", __func__);
clk_disable_unprepare(priv->clk);
/* PA_CTRL first setting low state, then make it iodisabled */
if (priv->spk_gpio_used) {
sunxi_gpio_iodisable(priv->spk_gpio);
msleep(30);
}
/*
* when codec suspend, then the register reset, if auto reset produce
* Pop & Click noise, then we should cut down the LINEOUT in this town.
*/
if (priv->enable) {
snd_soc_update_bits(codec, AC_LINEOUT_CTL,
(1<<LINEOUT_EN), (0<<LINEOUT_EN));
snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
(1<<LINEL_SRC_EN), (0<<LINEL_SRC_EN));
snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
(1<<LINER_SRC_EN), (0<<LINER_SRC_EN));
priv->enable = 0;
}
pr_debug("Exit %s\n", __func__);
return 0;
}
static int acx00_codec_resume(struct snd_soc_codec *codec)
{
struct acx00_priv *priv = snd_soc_codec_get_drvdata(codec);
pr_debug("Enter %s\n", __func__);
if (clk_prepare_enable(priv->clk)) {
dev_err(codec->dev, "codec resume clk failed\n");
return -EBUSY;
}
schedule_delayed_work(&priv->resume_work, msecs_to_jiffies(300));
if (priv->spk_gpio_used) {
gpio_direction_output(priv->spk_gpio, 1);
gpio_set_value(priv->spk_gpio, 0);
}
pr_debug("Exit %s\n", __func__);
return 0;
}
static int acx00_codec_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
codec->component.dapm.bias_level = level;
return 0;
}
struct label {
const char *name;
int value;
};
#define LABEL(constant) { #constant, constant }
#define LABEL_END { NULL, -1 }
static struct label reg_labels[] = {
LABEL(AC_SYS_CLK_CTL),
LABEL(AC_SYS_MOD_RST),
LABEL(AC_SYS_SR_CTL),
LABEL(AC_I2S_CTL),
LABEL(AC_I2S_CLK),
LABEL(AC_I2S_FMT0),
LABEL(AC_I2S_FMT1),
LABEL(AC_I2S_MIXER_SRC),
LABEL(AC_I2S_MIXER_GAIN),
LABEL(AC_I2S_DAC_VOL),
LABEL(AC_I2S_ADC_VOL),
LABEL(AC_DAC_CTL),
LABEL(AC_DAC_MIXER_SRC),
LABEL(AC_DAC_MIXER_GAIN),
LABEL(AC_OUT_MIXER_CTL),
LABEL(AC_OUT_MIXER_SRC),
LABEL(AC_LINEOUT_CTL),
LABEL(AC_ADC_CTL),
LABEL(AC_MICBIAS_CTL),
LABEL(AC_ADC_MIC_CTL),
LABEL(AC_ADC_MIXER_SRC),
LABEL(AC_BIAS_CTL),
LABEL(AC_ANALOG_PROF_CTL),
LABEL_END,
};
static ssize_t show_audio_reg(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct acx00_priv *priv = dev_get_drvdata(dev);
int count = 0, i = 0;
unsigned int reg_val;
count += sprintf(buf, "dump audio reg:\n");
while (reg_labels[i].name != NULL) {
reg_val = acx00_reg_read(priv->acx00, reg_labels[i].value);
count += sprintf(buf + count, "%s 0x%x: 0x%04x\n",
reg_labels[i].name, (reg_labels[i].value), reg_val);
i++;
}
return count;
}
/*
* param 1: 0 read;1 write
* param 2: 1 digital reg; 2 analog reg
* param 3: reg value;
* param 4: write value;
* read:
* echo 0,1,0x00> audio_reg
* echo 0,2,0x00> audio_reg
* write:
* echo 1,1,0x00,0xa > audio_reg
* echo 1,2,0x00,0xff > audio_reg
*/
static ssize_t store_audio_reg(struct device *dev,
struct device_attribute *attr, const char *buf, size_t count)
{
int ret;
int rw_flag;
unsigned int input_reg_val = 0;
int input_reg_group = 0;
unsigned int input_reg_offset = 0;
struct acx00_priv *priv = dev_get_drvdata(dev);
ret = sscanf(buf, "%d,%d,0x%x,0x%x", &rw_flag, &input_reg_group,
&input_reg_offset, &input_reg_val);
dev_info(dev, "ret:%d, reg_group:%d, reg_offset:%d, reg_val:0x%x\n",
ret, input_reg_group, input_reg_offset, input_reg_val);
if (input_reg_group != 1) {
pr_err("not exist reg group\n");
ret = count;
goto out;
}
if (!(rw_flag == 1 || rw_flag == 0)) {
pr_err("not rw_flag\n");
ret = count;
goto out;
}
if (rw_flag) {
acx00_reg_write(priv->acx00, input_reg_offset, input_reg_val);
} else {
input_reg_val = acx00_reg_read(priv->acx00, input_reg_offset);
dev_info(dev, "\n\n Reg[0x%x] : 0x%04x\n\n",
input_reg_offset, input_reg_val);
}
ret = count;
out:
return ret;
}
static DEVICE_ATTR(audio_reg, 0644, show_audio_reg, store_audio_reg);
static struct attribute *audio_debug_attrs[] = {
&dev_attr_audio_reg.attr,
NULL,
};
static struct attribute_group audio_debug_attr_group = {
.name = "audio_reg_debug",
.attrs = audio_debug_attrs,
};
static struct snd_soc_codec_driver soc_codec_driver_acx00 = {
.probe = acx00_codec_probe,
.remove = acx00_codec_remove,
.suspend = acx00_codec_suspend,
.resume = acx00_codec_resume,
.read = acx00_codec_read,
.write = acx00_codec_write,
.ignore_pmdown_time = 1,
.set_bias_level = acx00_codec_set_bias_level,
};
/* through acx00 is part of mfd devices, after the mfd */
static int acx00_codec_dev_probe(struct platform_device *pdev)
{
struct acx00_priv *priv;
int ret;
struct device_node *np = of_find_compatible_node(NULL, NULL,
"allwinner,ac200_codec");
priv = devm_kzalloc(&pdev->dev, sizeof(struct acx00_priv), GFP_KERNEL);
if (!priv) {
dev_err(&pdev->dev, "acx00 codec priv mem alloc failed\n");
return -ENOMEM;
}
platform_set_drvdata(pdev, priv);
priv->acx00 = dev_get_drvdata(pdev->dev.parent);
if (np) {
ret = of_get_named_gpio(np, "gpio-spk", 0);
if (ret >= 0) {
priv->spk_gpio_used = 1;
priv->spk_gpio = ret;
if (!gpio_is_valid(priv->spk_gpio)) {
dev_err(&pdev->dev, "gpio-spk is valid\n");
ret = -EINVAL;
goto err_devm_kfree;
} else {
ret = devm_gpio_request(&pdev->dev,
priv->spk_gpio, "SPK");
if (ret) {
dev_err(&pdev->dev,
"failed request gpio-spk\n");
ret = -EBUSY;
goto err_devm_kfree;
} else {
gpio_direction_output(priv->spk_gpio,
1);
gpio_set_value(priv->spk_gpio, 0);
}
}
} else {
priv->spk_gpio_used = 0;
}
}
ret = snd_soc_register_codec(&pdev->dev, &soc_codec_driver_acx00,
acx00_codec_dai, ARRAY_SIZE(acx00_codec_dai));
if (ret < 0)
dev_err(&pdev->dev, "Failed register acx00: %d\n", ret);
ret = sysfs_create_group(&pdev->dev.kobj, &audio_debug_attr_group);
if (ret)
dev_warn(&pdev->dev, "failed to create attr group\n");
return 0;
err_devm_kfree:
devm_kfree(&pdev->dev, priv);
return ret;
}
/* Mark this space to clear the LINEOUT & gpio */
static void acx00_codec_dev_shutdown(struct platform_device *pdev)
{
struct acx00_priv *priv = platform_get_drvdata(pdev);
if (priv->spk_gpio_used)
gpio_set_value(priv->spk_gpio, 0);
}
static int acx00_codec_dev_remove(struct platform_device *pdev)
{
struct acx00_priv *priv = platform_get_drvdata(pdev);
#ifndef ACX00_DAPM_LINEOUT
snd_soc_update_bits(priv->codec, AC_LINEOUT_CTL,
(1<<LINEOUT_EN), (0<<LINEOUT_EN));
#endif
snd_soc_unregister_codec(&pdev->dev);
clk_disable_unprepare(priv->clk);
devm_kfree(&pdev->dev, priv);
return 0;
}
static struct platform_driver acx00_codec_driver = {
.driver = {
.name = "acx00-codec",
.owner = THIS_MODULE,
},
.probe = acx00_codec_dev_probe,
.remove = acx00_codec_dev_remove,
.shutdown = acx00_codec_dev_shutdown,
};
static int __init acx00_codec_driver_init(void)
{
return platform_driver_register(&acx00_codec_driver);
}
static void __exit acx00_codec_driver_exit(void)
{
platform_driver_unregister(&acx00_codec_driver);
}
late_initcall(acx00_codec_driver_init);
module_exit(acx00_codec_driver_exit);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("SUNXI ASoC ACX00 Codec Driver");
MODULE_AUTHOR("wolfgang huang");
MODULE_ALIAS("platform:acx00-codec");
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