/*
* ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver
*
* Copyright (C) 2009 Renesas Solutions Corp.
* Kuninori Morimoto <morimoto.kuninori@renesas.com>
*
* Based on wm8731.c by Richard Purdie
* Based on ak4535.c by Richard Purdie
* Based on wm8753.c by Liam Girdwood
*
* 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.
*/
/* ** CAUTION **
*
* This is very simple driver.
* It can use headphone output / stereo input only
*
* AK4642 is not tested.
* AK4643 is tested.
*/
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#define AK4642_VERSION "0.0.1"
#define PW_MGMT1 0x00
#define PW_MGMT2 0x01
#define SG_SL1 0x02
#define SG_SL2 0x03
#define MD_CTL1 0x04
#define MD_CTL2 0x05
#define TIMER 0x06
#define ALC_CTL1 0x07
#define ALC_CTL2 0x08
#define L_IVC 0x09
#define L_DVC 0x0a
#define ALC_CTL3 0x0b
#define R_IVC 0x0c
#define R_DVC 0x0d
#define MD_CTL3 0x0e
#define MD_CTL4 0x0f
#define PW_MGMT3 0x10
#define DF_S 0x11
#define FIL3_0 0x12
#define FIL3_1 0x13
#define FIL3_2 0x14
#define FIL3_3 0x15
#define EQ_0 0x16
#define EQ_1 0x17
#define EQ_2 0x18
#define EQ_3 0x19
#define EQ_4 0x1a
#define EQ_5 0x1b
#define FIL1_0 0x1c
#define FIL1_1 0x1d
#define FIL1_2 0x1e
#define FIL1_3 0x1f
#define PW_MGMT4 0x20
#define MD_CTL5 0x21
#define LO_MS 0x22
#define HP_MS 0x23
#define SPK_MS 0x24
#define AK4642_CACHEREGNUM 0x25
/* PW_MGMT1*/
#define PMVCM (1 << 6) /* VCOM Power Management */
#define PMMIN (1 << 5) /* MIN Input Power Management */
#define PMDAC (1 << 2) /* DAC Power Management */
#define PMADL (1 << 0) /* MIC Amp Lch and ADC Lch Power Management */
/* PW_MGMT2 */
#define HPMTN (1 << 6)
#define PMHPL (1 << 5)
#define PMHPR (1 << 4)
#define MS (1 << 3) /* master/slave select */
#define MCKO (1 << 1)
#define PMPLL (1 << 0)
#define PMHP_MASK (PMHPL | PMHPR)
#define PMHP PMHP_MASK
/* PW_MGMT3 */
#define PMADR (1 << 0) /* MIC L / ADC R Power Management */
/* SG_SL1 */
#define MINS (1 << 6) /* Switch from MIN to Speaker */
#define DACL (1 << 4) /* Switch from DAC to Stereo or Receiver */
#define PMMP (1 << 2) /* MPWR pin Power Management */
#define MGAIN0 (1 << 0) /* MIC amp gain*/
/* TIMER */
#define ZTM(param) ((param & 0x3) << 4) /* ALC Zoro Crossing TimeOut */
#define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2))
/* ALC_CTL1 */
#define ALC (1 << 5) /* ALC Enable */
#define LMTH0 (1 << 0) /* ALC Limiter / Recovery Level */
/* MD_CTL1 */
#define PLL3 (1 << 7)
#define PLL2 (1 << 6)
#define PLL1 (1 << 5)
#define PLL0 (1 << 4)
#define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0)
#define BCKO_MASK (1 << 3)
#define BCKO_64 BCKO_MASK
#define DIF_MASK (3 << 0)
#define DSP (0 << 0)
#define RIGHT_J (1 << 0)
#define LEFT_J (2 << 0)
#define I2S (3 << 0)
/* MD_CTL2 */
#define FS0 (1 << 0)
#define FS1 (1 << 1)
#define FS2 (1 << 2)
#define FS3 (1 << 5)
#define FS_MASK (FS0 | FS1 | FS2 | FS3)
/* MD_CTL3 */
#define BST1 (1 << 3)
/* MD_CTL4 */
#define DACH (1 << 0)
/*
* Playback Volume (table 39)
*
* max : 0x00 : +12.0 dB
* ( 0.5 dB step )
* min : 0xFE : -115.0 dB
* mute: 0xFF
*/
static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
static const struct snd_kcontrol_new ak4642_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
0, 0xFF, 1, out_tlv),
};
/* codec private data */
struct ak4642_priv {
unsigned int sysclk;
enum snd_soc_control_type control_type;
void *control_data;
};
/*
* ak4642 register cache
*/
static const u8 ak4642_reg[AK4642_CACHEREGNUM] = {
0x00, 0x00, 0x01, 0x00,
0x02, 0x00, 0x00, 0x00,
0xe1, 0xe1, 0x18, 0x00,
0xe1, 0x18, 0x11, 0x08,
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00,
0x00,
};
/*
* read ak4642 register cache
*/
static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec,
unsigned int reg)
{
u16 *cache = codec->reg_cache;
if (reg >= AK4642_CACHEREGNUM)
return -1;
return cache[reg];
}
/*
* write ak4642 register cache
*/
static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec,
u16 reg, unsigned int value)
{
u16 *cache = codec->reg_cache;
if (reg >= AK4642_CACHEREGNUM)
return;
cache[reg] = value;
}
/*
* write to the AK4642 register space
*/
static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
u8 data[2];
/* data is
* D15..D8 AK4642 register offset
* D7...D0 register data
*/
data[0] = reg & 0xff;
data[1] = value & 0xff;
if (codec->hw_write(codec->control_data, data, 2) == 2) {
ak4642_write_reg_cache(codec, reg, value);
return 0;
} else
return -EIO;
}
static int ak4642_sync(struct snd_soc_codec *codec)
{
u16 *cache = codec->reg_cache;
int i, r = 0;
for (i = 0; i < AK4642_CACHEREGNUM; i++)
r |= ak4642_write(codec, i, cache[i]);
return r;
};
static int ak4642_dai_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) {
/*
* start headphone output
*
* PLL, Master Mode
* Audio I/F Format :MSB justified (ADC & DAC)
* Bass Boost Level : Middle
*
* This operation came from example code of
* "ASAHI KASEI AK4642" (japanese) manual p97.
*/
snd_soc_update_bits(codec, MD_CTL4, DACH, DACH);
snd_soc_update_bits(codec, MD_CTL3, BST1, BST1);
ak4642_write(codec, L_IVC, 0x91); /* volume */
ak4642_write(codec, R_IVC, 0x91); /* volume */
snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMMIN | PMDAC,
PMVCM | PMMIN | PMDAC);
snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP);
snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN);
} else {
/*
* start stereo input
*
* PLL Master Mode
* Audio I/F Format:MSB justified (ADC & DAC)
* Pre MIC AMP:+20dB
* MIC Power On
* ALC setting:Refer to Table 35
* ALC bit=“1”
*
* This operation came from example code of
* "ASAHI KASEI AK4642" (japanese) manual p94.
*/
ak4642_write(codec, SG_SL1, PMMP | MGAIN0);
ak4642_write(codec, TIMER, ZTM(0x3) | WTM(0x3));
ak4642_write(codec, ALC_CTL1, ALC | LMTH0);
snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMADL,
PMVCM | PMADL);
snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR);
}
return 0;
}
static void ak4642_dai_shutdown(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) {
/* stop headphone output */
snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0);
snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, 0);
snd_soc_update_bits(codec, PW_MGMT1, PMMIN | PMDAC, 0);
snd_soc_update_bits(codec, MD_CTL3, BST1, 0);
snd_soc_update_bits(codec, MD_CTL4, DACH, 0);
} else {
/* stop stereo input */
snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0);
snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0);
snd_soc_update_bits(codec, ALC_CTL1, ALC, 0);
}
}
static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
u8 pll;
switch (freq) {
case 11289600:
pll = PLL2;
break;
case 12288000:
pll = PLL2 | PLL0;
break;
case 12000000:
pll = PLL2 | PLL1;
break;
case 24000000:
pll = PLL2 | PLL1 | PLL0;
break;
case 13500000:
pll = PLL3 | PLL2;
break;
case 27000000:
pll = PLL3 | PLL2 | PLL0;
break;
default:
return -EINVAL;
}
snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll);
return 0;
}
static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_codec *codec = dai->codec;
u8 data;
u8 bcko;
data = MCKO | PMPLL; /* use MCKO */
bcko = 0;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
data |= MS;
bcko = BCKO_64;
break;
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
return -EINVAL;
}
snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
/* format type */
data = 0;
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_LEFT_J:
data = LEFT_J;
break;
case SND_SOC_DAIFMT_I2S:
data = I2S;
break;
/* FIXME
* Please add RIGHT_J / DSP support here
*/
default:
return -EINVAL;
break;
}
snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data);
return 0;
}
static int ak4642_dai_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;
u8 rate;
switch (params_rate(params)) {
case 7350:
rate = FS2;
break;
case 8000:
rate = 0;
break;
case 11025:
rate = FS2 | FS0;
break;
case 12000:
rate = FS0;
break;
case 14700:
rate = FS2 | FS1;
break;
case 16000:
rate = FS1;
break;
case 22050:
rate = FS2 | FS1 | FS0;
break;
case 24000:
rate = FS1 | FS0;
break;
case 29400:
rate = FS3 | FS2 | FS1;
break;
case 32000:
rate = FS3 | FS1;
break;
case 44100:
rate = FS3 | FS2 | FS1 | FS0;
break;
case 48000:
rate = FS3 | FS1 | FS0;
break;
default:
return -EINVAL;
break;
}
snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
return 0;
}
static struct snd_soc_dai_ops ak4642_dai_ops = {
.startup = ak4642_dai_startup,
.shutdown = ak4642_dai_shutdown,
.set_sysclk = ak4642_dai_set_sysclk,
.set_fmt = ak4642_dai_set_fmt,
.hw_params = ak4642_dai_hw_params,
};
static struct snd_soc_dai_driver ak4642_dai = {
.name = "ak4642-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE },
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE },
.ops = &ak4642_dai_ops,
.symmetric_rates = 1,
};
static int ak4642_resume(struct snd_soc_codec *codec)
{
ak4642_sync(codec);
return 0;
}
static int ak4642_probe(struct snd_soc_codec *codec)
{
struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec);
dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION);
codec->hw_write = (hw_write_t)i2c_master_send;
codec->control_data = ak4642->control_data;
snd_soc_add_controls(codec, ak4642_snd_controls,
ARRAY_SIZE(ak4642_snd_controls));
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
.probe = ak4642_probe,
.resume = ak4642_resume,
.read = ak4642_read_reg_cache,
.write = ak4642_write,
.reg_cache_size = ARRAY_SIZE(ak4642_reg),
.reg_word_size = sizeof(u8),
.reg_cache_default = ak4642_reg,
};
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static __devinit int ak4642_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct ak4642_priv *ak4642;
int ret;
ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL);
if (!ak4642)
return -ENOMEM;
i2c_set_clientdata(i2c, ak4642);
ak4642->control_data = i2c;
ak4642->control_type = SND_SOC_I2C;
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_ak4642, &ak4642_dai, 1);
if (ret < 0)
kfree(ak4642);
return ret;
}
static __devexit int ak4642_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 ak4642_i2c_id[] = {
{ "ak4642", 0 },
{ "ak4643", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
static struct i2c_driver ak4642_i2c_driver = {
.driver = {
.name = "ak4642-codec",
.owner = THIS_MODULE,
},
.probe = ak4642_i2c_probe,
.remove = __devexit_p(ak4642_i2c_remove),
.id_table = ak4642_i2c_id,
};
#endif
static int __init ak4642_modinit(void)
{
int ret = 0;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&ak4642_i2c_driver);
#endif
return ret;
}
module_init(ak4642_modinit);
static void __exit ak4642_exit(void)
{
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&ak4642_i2c_driver);
#endif
}
module_exit(ak4642_exit);
MODULE_DESCRIPTION("Soc AK4642 driver");
MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
MODULE_LICENSE("GPL");