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linux/sound/soc/codecs/da7210.c
Ashish Chavan 5eda19497b ASoC: da7210: Add support for mute and zero cross controls 
This patch adds support for below set of controls,
(1) Mute controls for MIC, AUX and ADC
(2) Zero cross controls for head phone, AUX, INPGA and line out
(3) Head phone mode selection - class H or G

It also adds digital_mute() call back.

Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-19 17:58:58 +01:00

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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_STATUS 0x02
#define DA7210_STARTUP1 0x03
#define DA7210_MIC_L 0x07
#define DA7210_MIC_R 0x08
#define DA7210_AUX2 0x0B
#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_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_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)
#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);
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),
};
/* 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;
}
}
static int da7210_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
struct snd_soc_codec *codec = dai->codec;
if (is_play) {
/* Enable Out */
snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10);
snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10);
} else {
/* Volume 7 */
snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7);
snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7);
/* Enable Mic */
snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1);
snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1);
}
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 = {
.startup = da7210_startup,
.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),
};
#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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