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OpenCloudOS-Kernel/sound/soc/codecs/da7210.c

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/*
* DA7210 ALSA Soc codec driver
*
* Copyright (c) 2009 Dialog Semiconductor
* Written by David Chen <Dajun.chen@diasemi.com>
*
* Copyright (C) 2009 Renesas Solutions Corp.
* Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com>
*
* Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S
*
* 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.
*/
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
/* DA7210 register space */
#define DA7210_CONTROL 0x01
#define DA7210_STATUS 0x02
#define DA7210_STARTUP1 0x03
#define DA7210_STARTUP2 0x04
#define DA7210_STARTUP3 0x05
#define DA7210_MIC_L 0x07
#define DA7210_MIC_R 0x08
#define DA7210_AUX1_L 0x09
#define DA7210_AUX1_R 0x0A
#define DA7210_AUX2 0x0B
#define DA7210_IN_GAIN 0x0C
#define DA7210_INMIX_L 0x0D
#define DA7210_INMIX_R 0x0E
#define DA7210_ADC_HPF 0x0F
#define DA7210_ADC 0x10
#define DA7210_ADC_EQ1_2 0X11
#define DA7210_ADC_EQ3_4 0x12
#define DA7210_ADC_EQ5 0x13
#define DA7210_DAC_HPF 0x14
#define DA7210_DAC_L 0x15
#define DA7210_DAC_R 0x16
#define DA7210_DAC_SEL 0x17
#define DA7210_SOFTMUTE 0x18
#define DA7210_DAC_EQ1_2 0x19
#define DA7210_DAC_EQ3_4 0x1A
#define DA7210_DAC_EQ5 0x1B
#define DA7210_OUTMIX_L 0x1C
#define DA7210_OUTMIX_R 0x1D
#define DA7210_HP_L_VOL 0x21
#define DA7210_HP_R_VOL 0x22
#define DA7210_HP_CFG 0x23
#define DA7210_ZERO_CROSS 0x24
#define DA7210_DAI_SRC_SEL 0x25
#define DA7210_DAI_CFG1 0x26
#define DA7210_DAI_CFG3 0x28
#define DA7210_PLL_DIV1 0x29
#define DA7210_PLL_DIV2 0x2A
#define DA7210_PLL_DIV3 0x2B
#define DA7210_PLL 0x2C
#define DA7210_ALC_MAX 0x83
#define DA7210_ALC_MIN 0x84
#define DA7210_ALC_NOIS 0x85
#define DA7210_ALC_ATT 0x86
#define DA7210_ALC_REL 0x87
#define DA7210_ALC_DEL 0x88
#define DA7210_A_HID_UNLOCK 0x8A
#define DA7210_A_TEST_UNLOCK 0x8B
#define DA7210_A_PLL1 0x90
#define DA7210_A_CP_MODE 0xA7
/* STARTUP1 bit fields */
#define DA7210_SC_MST_EN (1 << 0)
/* MIC_L bit fields */
#define DA7210_MICBIAS_EN (1 << 6)
#define DA7210_MIC_L_EN (1 << 7)
/* MIC_R bit fields */
#define DA7210_MIC_R_EN (1 << 7)
/* INMIX_L bit fields */
#define DA7210_IN_L_EN (1 << 7)
/* INMIX_R bit fields */
#define DA7210_IN_R_EN (1 << 7)
/* ADC bit fields */
#define DA7210_ADC_ALC_EN (1 << 0)
#define DA7210_ADC_L_EN (1 << 3)
#define DA7210_ADC_R_EN (1 << 7)
/* DAC/ADC HPF fields */
#define DA7210_VOICE_F0_MASK (0x7 << 4)
#define DA7210_VOICE_F0_25 (1 << 4)
#define DA7210_VOICE_EN (1 << 7)
/* DAC_SEL bit fields */
#define DA7210_DAC_L_SRC_DAI_L (4 << 0)
#define DA7210_DAC_L_EN (1 << 3)
#define DA7210_DAC_R_SRC_DAI_R (5 << 4)
#define DA7210_DAC_R_EN (1 << 7)
/* OUTMIX_L bit fields */
#define DA7210_OUT_L_EN (1 << 7)
/* OUTMIX_R bit fields */
#define DA7210_OUT_R_EN (1 << 7)
/* HP_CFG bit fields */
#define DA7210_HP_2CAP_MODE (1 << 1)
#define DA7210_HP_SENSE_EN (1 << 2)
#define DA7210_HP_L_EN (1 << 3)
#define DA7210_HP_MODE (1 << 6)
#define DA7210_HP_R_EN (1 << 7)
/* DAI_SRC_SEL bit fields */
#define DA7210_DAI_OUT_L_SRC (6 << 0)
#define DA7210_DAI_OUT_R_SRC (7 << 4)
/* DAI_CFG1 bit fields */
#define DA7210_DAI_WORD_S16_LE (0 << 0)
#define DA7210_DAI_WORD_S20_3LE (1 << 0)
#define DA7210_DAI_WORD_S24_LE (2 << 0)
#define DA7210_DAI_WORD_S32_LE (3 << 0)
#define DA7210_DAI_FLEN_64BIT (1 << 2)
#define DA7210_DAI_MODE_SLAVE (0 << 7)
#define DA7210_DAI_MODE_MASTER (1 << 7)
/* DAI_CFG3 bit fields */
#define DA7210_DAI_FORMAT_I2SMODE (0 << 0)
#define DA7210_DAI_FORMAT_LEFT_J (1 << 0)
#define DA7210_DAI_FORMAT_RIGHT_J (2 << 0)
#define DA7210_DAI_OE (1 << 3)
#define DA7210_DAI_EN (1 << 7)
/*PLL_DIV3 bit fields */
#define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4)
#define DA7210_PLL_BYP (1 << 6)
/* PLL bit fields */
#define DA7210_PLL_FS_MASK (0xF << 0)
#define DA7210_PLL_FS_8000 (0x1 << 0)
#define DA7210_PLL_FS_11025 (0x2 << 0)
#define DA7210_PLL_FS_12000 (0x3 << 0)
#define DA7210_PLL_FS_16000 (0x5 << 0)
#define DA7210_PLL_FS_22050 (0x6 << 0)
#define DA7210_PLL_FS_24000 (0x7 << 0)
#define DA7210_PLL_FS_32000 (0x9 << 0)
#define DA7210_PLL_FS_44100 (0xA << 0)
#define DA7210_PLL_FS_48000 (0xB << 0)
#define DA7210_PLL_FS_88200 (0xE << 0)
#define DA7210_PLL_FS_96000 (0xF << 0)
#define DA7210_PLL_EN (0x1 << 7)
/* SOFTMUTE bit fields */
#define DA7210_RAMP_EN (1 << 6)
/* CONTROL bit fields */
#define DA7210_NOISE_SUP_EN (1 << 3)
/* IN_GAIN bit fields */
#define DA7210_INPGA_L_VOL (0x0F << 0)
#define DA7210_INPGA_R_VOL (0xF0 << 0)
/* ZERO_CROSS bit fields */
#define DA7210_AUX1_L_ZC (1 << 0)
#define DA7210_AUX1_R_ZC (1 << 1)
#define DA7210_HP_L_ZC (1 << 6)
#define DA7210_HP_R_ZC (1 << 7)
/* AUX1_L bit fields */
#define DA7210_AUX1_L_VOL (0x3F << 0)
/* AUX1_R bit fields */
#define DA7210_AUX1_R_VOL (0x3F << 0)
/* Minimum INPGA and AUX1 volume to enable noise suppression */
#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */
#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */
#define DA7210_VERSION "0.0.1"
/*
* Playback Volume
*
* max : 0x3F (+15.0 dB)
* (1.5 dB step)
* min : 0x11 (-54.0 dB)
* mute : 0x10
* reserved : 0x00 - 0x0F
*
* Reserved area are considered as "mute".
*/
static const unsigned int hp_out_tlv[] = {
TLV_DB_RANGE_HEAD(2),
0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1),
/* -54 dB to +15 dB */
0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0),
};
static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0);
static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1);
/* ADC and DAC high pass filter f0 value */
static const char const *da7210_hpf_cutoff_txt[] = {
"Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi"
};
static const struct soc_enum da7210_dac_hpf_cutoff =
SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt);
static const struct soc_enum da7210_adc_hpf_cutoff =
SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt);
/* ADC and DAC voice (8kHz) high pass cutoff value */
static const char const *da7210_vf_cutoff_txt[] = {
"2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
};
static const struct soc_enum da7210_dac_vf_cutoff =
SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt);
static const struct soc_enum da7210_adc_vf_cutoff =
SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt);
static const char *da7210_hp_mode_txt[] = {
"Class H", "Class G"
};
static const struct soc_enum da7210_hp_mode_sel =
SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt);
/* ALC can be enabled only if noise suppression is disabled */
static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
if (ucontrol->value.integer.value[0]) {
/* Check if noise suppression is enabled */
if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) {
dev_dbg(codec->dev,
"Disable noise suppression to enable ALC\n");
return -EINVAL;
}
}
/* If all conditions are met or we are actually disabling ALC */
return snd_soc_put_volsw(kcontrol, ucontrol);
}
/* Noise suppression can be enabled only if following conditions are met
* ALC disabled
* ZC enabled for HP and AUX1 PGA
* INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB
* AUX1_L_VOL and AUX1_R_VOL >= 6 dB
*/
static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
u8 val;
if (ucontrol->value.integer.value[0]) {
/* Check if ALC is enabled */
if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN)
goto err;
/* Check ZC for HP and AUX1 PGA */
if ((snd_soc_read(codec, DA7210_ZERO_CROSS) &
(DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC |
DA7210_HP_R_ZC)) != (DA7210_AUX1_L_ZC |
DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | DA7210_HP_R_ZC))
goto err;
/* Check INPGA_L_VOL and INPGA_R_VOL */
val = snd_soc_read(codec, DA7210_IN_GAIN);
if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) ||
(((val & DA7210_INPGA_R_VOL) >> 4) <
DA7210_INPGA_MIN_VOL_NS))
goto err;
/* Check AUX1_L_VOL and AUX1_R_VOL */
if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) <
DA7210_AUX1_MIN_VOL_NS) ||
((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) <
DA7210_AUX1_MIN_VOL_NS))
goto err;
}
/* If all conditions are met or we are actually disabling Noise sup */
return snd_soc_put_volsw(kcontrol, ucontrol);
err:
return -EINVAL;
}
static const struct snd_kcontrol_new da7210_snd_controls[] = {
SOC_DOUBLE_R_TLV("HeadPhone Playback Volume",
DA7210_HP_L_VOL, DA7210_HP_R_VOL,
0, 0x3F, 0, hp_out_tlv),
/* DAC Equalizer controls */
SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0),
SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1,
eq_gain_tlv),
SOC_SINGLE_TLV("DAC EQ2 Volume", DA7210_DAC_EQ1_2, 4, 0xf, 1,
eq_gain_tlv),
SOC_SINGLE_TLV("DAC EQ3 Volume", DA7210_DAC_EQ3_4, 0, 0xf, 1,
eq_gain_tlv),
SOC_SINGLE_TLV("DAC EQ4 Volume", DA7210_DAC_EQ3_4, 4, 0xf, 1,
eq_gain_tlv),
SOC_SINGLE_TLV("DAC EQ5 Volume", DA7210_DAC_EQ5, 0, 0xf, 1,
eq_gain_tlv),
/* ADC Equalizer controls */
SOC_SINGLE("ADC EQ Switch", DA7210_ADC_EQ5, 7, 1, 0),
SOC_SINGLE_TLV("ADC EQ Master Volume", DA7210_ADC_EQ5, 4, 0x3,
1, adc_eq_master_gain_tlv),
SOC_SINGLE_TLV("ADC EQ1 Volume", DA7210_ADC_EQ1_2, 0, 0xf, 1,
eq_gain_tlv),
SOC_SINGLE_TLV("ADC EQ2 Volume", DA7210_ADC_EQ1_2, 4, 0xf, 1,
eq_gain_tlv),
SOC_SINGLE_TLV("ADC EQ3 Volume", DA7210_ADC_EQ3_4, 0, 0xf, 1,
eq_gain_tlv),
SOC_SINGLE_TLV("ADC EQ4 Volume", DA7210_ADC_EQ3_4, 4, 0xf, 1,
eq_gain_tlv),
SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1,
eq_gain_tlv),
SOC_SINGLE("DAC HPF Switch", DA7210_DAC_HPF, 3, 1, 0),
SOC_ENUM("DAC HPF Cutoff", da7210_dac_hpf_cutoff),
SOC_SINGLE("DAC Voice Mode Switch", DA7210_DAC_HPF, 7, 1, 0),
SOC_ENUM("DAC Voice Cutoff", da7210_dac_vf_cutoff),
SOC_SINGLE("ADC HPF Switch", DA7210_ADC_HPF, 3, 1, 0),
SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff),
SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0),
SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff),
/* Mute controls */
SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0),
SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0),
SOC_DOUBLE("ADC Capture Switch", DA7210_ADC, 2, 6, 1, 0),
SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0),
SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0),
/* Zero cross controls */
SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0),
SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0),
SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0),
SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0),
SOC_ENUM("Headphone Class", da7210_hp_mode_sel),
/* ALC controls */
SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0,
snd_soc_get_volsw, da7210_put_alc_sw),
SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0),
SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0),
SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0),
SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0),
SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0),
SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0),
SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1,
0, snd_soc_get_volsw, da7210_put_noise_sup_sw),
};
/*
* DAPM Controls
*
* Current DAPM implementation covers almost all codec components e.g. IOs,
* mixers, PGAs,ADC and DAC.
*/
/* In Mixer Left */
static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = {
SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0),
SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0),
};
/* In Mixer Right */
static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = {
SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0),
SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0),
};
/* Out Mixer Left */
static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = {
SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0),
};
/* Out Mixer Right */
static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = {
SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0),
};
/* DAPM widgets */
static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = {
/* Input Side */
/* Input Lines */
SND_SOC_DAPM_INPUT("MICL"),
SND_SOC_DAPM_INPUT("MICR"),
/* Input PGAs */
SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0),
SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0),
SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0),
/* Input Mixers */
SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0,
&da7210_dapm_inmixl_controls[0],
ARRAY_SIZE(da7210_dapm_inmixl_controls)),
SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0,
&da7210_dapm_inmixr_controls[0],
ARRAY_SIZE(da7210_dapm_inmixr_controls)),
/* ADCs */
SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1),
SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1),
/* Output Side */
/* DACs */
SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1),
SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1),
/* Output Mixers */
SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0,
&da7210_dapm_outmixl_controls[0],
ARRAY_SIZE(da7210_dapm_outmixl_controls)),
SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0,
&da7210_dapm_outmixr_controls[0],
ARRAY_SIZE(da7210_dapm_outmixr_controls)),
/* Output PGAs */
SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0),
SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0),
SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0),
SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0),
/* Output Lines */
SND_SOC_DAPM_OUTPUT("HPL"),
SND_SOC_DAPM_OUTPUT("HPR"),
};
/* DAPM audio route definition */
static const struct snd_soc_dapm_route da7210_audio_map[] = {
/* Dest Connecting Widget source */
/* Input path */
{"Mic Left", NULL, "MICL"},
{"Mic Right", NULL, "MICR"},
{"In Mixer Left", "Mic Left Switch", "Mic Left"},
{"In Mixer Left", "Mic Right Switch", "Mic Right"},
{"In Mixer Right", "Mic Right Switch", "Mic Right"},
{"In Mixer Right", "Mic Left Switch", "Mic Left"},
{"INPGA Left", NULL, "In Mixer Left"},
{"ADC Left", NULL, "INPGA Left"},
{"INPGA Right", NULL, "In Mixer Right"},
{"ADC Right", NULL, "INPGA Right"},
/* Output path */
{"Out Mixer Left", "DAC Left Switch", "DAC Left"},
{"Out Mixer Right", "DAC Right Switch", "DAC Right"},
{"OUTPGA Left Enable", NULL, "Out Mixer Left"},
{"OUTPGA Right Enable", NULL, "Out Mixer Right"},
{"Headphone Left", NULL, "OUTPGA Left Enable"},
{"HPL", NULL, "Headphone Left"},
{"Headphone Right", NULL, "OUTPGA Right Enable"},
{"HPR", NULL, "Headphone Right"},
};
/* Codec private data */
struct da7210_priv {
enum snd_soc_control_type control_type;
};
/*
* Register cache
*/
static const u8 da7210_reg[] = {
0x00, 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R0 - R7 */
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x08, /* R8 - RF */
0x00, 0x00, 0x00, 0x00, 0x08, 0x10, 0x10, 0x54, /* R10 - R17 */
0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R18 - R1F */
0x00, 0x00, 0x00, 0x02, 0x00, 0x76, 0x00, 0x00, /* R20 - R27 */
0x04, 0x00, 0x00, 0x30, 0x2A, 0x00, 0x40, 0x00, /* R28 - R2F */
0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, /* R30 - R37 */
0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x00, 0x00, /* R38 - R3F */
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R40 - R4F */
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R48 - R4F */
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R50 - R57 */
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R58 - R5F */
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R60 - R67 */
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R68 - R6F */
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R70 - R77 */
0x00, 0x00, 0x00, 0x00, 0x00, 0x54, 0x54, 0x00, /* R78 - R7F */
0x00, 0x00, 0x2C, 0x00, 0x00, 0x00, 0x00, 0x00, /* R80 - R87 */
0x00, /* R88 */
};
static int da7210_volatile_register(struct snd_soc_codec *codec,
unsigned int reg)
{
switch (reg) {
case DA7210_STATUS:
return 1;
default:
return 0;
}
}
/*
* Set PCM DAI word length.
*/
static int da7210_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
u32 dai_cfg1;
u32 fs, bypass;
/* set DAI source to Left and Right ADC */
snd_soc_write(codec, DA7210_DAI_SRC_SEL,
DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC);
/* Enable DAI */
snd_soc_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN);
dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
dai_cfg1 |= DA7210_DAI_WORD_S16_LE;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
dai_cfg1 |= DA7210_DAI_WORD_S20_3LE;
break;
case SNDRV_PCM_FORMAT_S24_LE:
dai_cfg1 |= DA7210_DAI_WORD_S24_LE;
break;
case SNDRV_PCM_FORMAT_S32_LE:
dai_cfg1 |= DA7210_DAI_WORD_S32_LE;
break;
default:
return -EINVAL;
}
snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1);
switch (params_rate(params)) {
case 8000:
fs = DA7210_PLL_FS_8000;
bypass = DA7210_PLL_BYP;
break;
case 11025:
fs = DA7210_PLL_FS_11025;
bypass = 0;
break;
case 12000:
fs = DA7210_PLL_FS_12000;
bypass = DA7210_PLL_BYP;
break;
case 16000:
fs = DA7210_PLL_FS_16000;
bypass = DA7210_PLL_BYP;
break;
case 22050:
fs = DA7210_PLL_FS_22050;
bypass = 0;
break;
case 32000:
fs = DA7210_PLL_FS_32000;
bypass = DA7210_PLL_BYP;
break;
case 44100:
fs = DA7210_PLL_FS_44100;
bypass = 0;
break;
case 48000:
fs = DA7210_PLL_FS_48000;
bypass = DA7210_PLL_BYP;
break;
case 88200:
fs = DA7210_PLL_FS_88200;
bypass = 0;
break;
case 96000:
fs = DA7210_PLL_FS_96000;
bypass = DA7210_PLL_BYP;
break;
default:
return -EINVAL;
}
/* Disable active mode */
snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0);
snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs);
snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass);
/* Enable active mode */
snd_soc_update_bits(codec, DA7210_STARTUP1,
DA7210_SC_MST_EN, DA7210_SC_MST_EN);
return 0;
}
/*
* Set DAI mode and Format
*/
static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
u32 dai_cfg1;
u32 dai_cfg3;
dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1);
dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3);
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
dai_cfg1 |= DA7210_DAI_MODE_MASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
dai_cfg1 |= DA7210_DAI_MODE_SLAVE;
break;
default:
return -EINVAL;
}
/* FIXME
*
* It support I2S only now
*/
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE;
break;
case SND_SOC_DAIFMT_LEFT_J:
dai_cfg3 |= DA7210_DAI_FORMAT_LEFT_J;
break;
case SND_SOC_DAIFMT_RIGHT_J:
dai_cfg3 |= DA7210_DAI_FORMAT_RIGHT_J;
break;
default:
return -EINVAL;
}
/* FIXME
*
* It support 64bit data transmission only now
*/
dai_cfg1 |= DA7210_DAI_FLEN_64BIT;
snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1);
snd_soc_write(codec, DA7210_DAI_CFG3, dai_cfg3);
return 0;
}
static int da7210_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB;
if (mute)
snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4);
else
snd_soc_write(codec, DA7210_DAC_HPF, mute_reg);
return 0;
}
#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
/* DAI operations */
static struct snd_soc_dai_ops da7210_dai_ops = {
.hw_params = da7210_hw_params,
.set_fmt = da7210_set_dai_fmt,
.digital_mute = da7210_mute,
};
static struct snd_soc_dai_driver da7210_dai = {
.name = "da7210-hifi",
/* playback capabilities */
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = DA7210_FORMATS,
},
/* capture capabilities */
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = DA7210_FORMATS,
},
.ops = &da7210_dai_ops,
.symmetric_rates = 1,
};
static int da7210_probe(struct snd_soc_codec *codec)
{
struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
int ret;
dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION);
ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
/* FIXME
*
* This driver use fixed value here
* And below settings expects MCLK = 12.288MHz
*
* When you select different MCLK, please check...
* DA7210_PLL_DIV1 val
* DA7210_PLL_DIV2 val
* DA7210_PLL_DIV3 val
* DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx
*/
/*
* make sure that DA7210 use bypass mode before start up
*/
snd_soc_write(codec, DA7210_STARTUP1, 0);
snd_soc_write(codec, DA7210_PLL_DIV3,
DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
/*
* ADC settings
*/
/* Enable Left & Right MIC PGA and Mic Bias */
snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN);
snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN);
/* Enable Left and Right input PGA */
snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN);
snd_soc_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN);
/* Enable Left and Right ADC */
snd_soc_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN);
/*
* DAC settings
*/
/* Enable Left and Right DAC */
snd_soc_write(codec, DA7210_DAC_SEL,
DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN |
DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN);
/* Enable Left and Right out PGA */
snd_soc_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN);
snd_soc_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN);
/* Enable Left and Right HeadPhone PGA */
snd_soc_write(codec, DA7210_HP_CFG,
DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN |
DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN);
/* Enable ramp mode for DAC gain update */
snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN);
/* Diable PLL and bypass it */
snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000);
/*
* If 48kHz sound came, it use bypass mode,
* and when it is 44.1kHz, it use PLL.
*
* This time, this driver sets PLL always ON
* and controls bypass/PLL mode by switching
* DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit.
* see da7210_hw_params
*/
snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */
snd_soc_write(codec, DA7210_PLL_DIV2, 0x99);
snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A |
DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN);
/* As suggested by Dialog */
snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */
snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4);
snd_soc_write(codec, DA7210_A_PLL1, 0x01);
snd_soc_write(codec, DA7210_A_CP_MODE, 0x7C);
snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */
snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0x00);
/* Activate all enabled subsystem */
snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN);
dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_da7210 = {
.probe = da7210_probe,
.reg_cache_size = ARRAY_SIZE(da7210_reg),
.reg_word_size = sizeof(u8),
.reg_cache_default = da7210_reg,
.volatile_register = da7210_volatile_register,
.controls = da7210_snd_controls,
.num_controls = ARRAY_SIZE(da7210_snd_controls),
.dapm_widgets = da7210_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(da7210_dapm_widgets),
.dapm_routes = da7210_audio_map,
.num_dapm_routes = ARRAY_SIZE(da7210_audio_map),
};
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct da7210_priv *da7210;
int ret;
da7210 = kzalloc(sizeof(struct da7210_priv), GFP_KERNEL);
if (!da7210)
return -ENOMEM;
i2c_set_clientdata(i2c, da7210);
da7210->control_type = SND_SOC_I2C;
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_da7210, &da7210_dai, 1);
if (ret < 0)
kfree(da7210);
return ret;
}
static int __devexit da7210_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
kfree(i2c_get_clientdata(client));
return 0;
}
static const struct i2c_device_id da7210_i2c_id[] = {
{ "da7210", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, da7210_i2c_id);
/* I2C codec control layer */
static struct i2c_driver da7210_i2c_driver = {
.driver = {
.name = "da7210-codec",
.owner = THIS_MODULE,
},
.probe = da7210_i2c_probe,
.remove = __devexit_p(da7210_i2c_remove),
.id_table = da7210_i2c_id,
};
#endif
static int __init da7210_modinit(void)
{
int ret = 0;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&da7210_i2c_driver);
#endif
return ret;
}
module_init(da7210_modinit);
static void __exit da7210_exit(void)
{
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&da7210_i2c_driver);
#endif
}
module_exit(da7210_exit);
MODULE_DESCRIPTION("ASoC DA7210 driver");
MODULE_AUTHOR("David Chen, Kuninori Morimoto");
MODULE_LICENSE("GPL");
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