ak4642.c

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/*
 * ak4642.c  --  AK4642/AK4643 ALSA Soc Audio driver
 *
 * Copyright (C) 2009 Renesas Solutions Corp.
 * Kuninori Morimoto <[email protected]>
 *
 * 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 tested.
 * AK4643 is tested.
 * AK4648 is tested.
 */

#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/slab.h>
#include <linux/module.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>

#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

/* 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),
};

static const struct snd_kcontrol_new ak4642_headphone_control =
    SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);

static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
    SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
};

static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {

    /* Outputs */
    SND_SOC_DAPM_OUTPUT("HPOUTL"),
    SND_SOC_DAPM_OUTPUT("HPOUTR"),
    SND_SOC_DAPM_OUTPUT("LINEOUT"),

    SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
    SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
    SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
                &ak4642_headphone_control),

    SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),

    SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
               &ak4642_lout_mixer_controls[0],
               ARRAY_SIZE(ak4642_lout_mixer_controls)),

    /* DAC */
    SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0),
};

static const struct snd_soc_dapm_route ak4642_intercon[] = {

    /* Outputs */
    {"HPOUTL", NULL, "HPL Out"},
    {"HPOUTR", NULL, "HPR Out"},
    {"LINEOUT", NULL, "LINEOUT Mixer"},

    {"HPL Out", NULL, "Headphone Enable"},
    {"HPR Out", NULL, "Headphone Enable"},

    {"Headphone Enable", "Switch", "DACH"},

    {"DACH", NULL, "DAC"},

    {"LINEOUT Mixer", "DACL", "DAC"},
};

/* codec private data */
struct ak4642_priv {
    unsigned int sysclk;
    enum snd_soc_control_type control_type;
};

/*
 * ak4642 register cache
 */
static const u8 ak4642_reg[] = {
    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,
};

static const u8 ak4648_reg[] = {
    0x00, 0x00, 0x01, 0x00,
    0x02, 0x00, 0x00, 0x00,
    0xe1, 0xe1, 0x18, 0x00,
    0xe1, 0x18, 0x11, 0xb8,
    0x00, 0x00, 0x00, 0x00,
    0x00, 0x00, 0x00, 0x00,
    0x00, 0x00, 0x00, 0x00,
    0x00, 0x00, 0x00, 0x00,
    0x00, 0x00, 0x00, 0x00,
    0x00, 0x88, 0x88, 0x08,
};

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_write(codec, L_IVC, 0x91); /* volume */
        snd_soc_write(codec, R_IVC, 0x91); /* volume */
    } 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.
         */
        snd_soc_write(codec, SG_SL1, PMMP | MGAIN0);
        snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3));
        snd_soc_write(codec, ALC_CTL1, ALC | LMTH0);
        snd_soc_update_bits(codec, PW_MGMT1, PMADL, 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) {
    } 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 int ak4642_set_bias_level(struct snd_soc_codec *codec,
                 enum snd_soc_bias_level level)
{
    switch (level) {
    case SND_SOC_BIAS_OFF:
        snd_soc_write(codec, PW_MGMT1, 0x00);
        break;
    default:
        snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM);
        break;
    }
    codec->dapm.bias_level = level;

    return 0;
}

static const 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)
{
    snd_soc_cache_sync(codec);
    return 0;
}


static int ak4642_probe(struct snd_soc_codec *codec)
{
    struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec);
    int ret;

    ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type);
    if (ret < 0) {
        dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
        return ret;
    }

    snd_soc_add_codec_controls(codec, ak4642_snd_controls,
                 ARRAY_SIZE(ak4642_snd_controls));

    ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

    return 0;
}

static int ak4642_remove(struct snd_soc_codec *codec)
{
    ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF);
    return 0;
}

static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
    .probe          = ak4642_probe,
    .remove         = ak4642_remove,
    .resume         = ak4642_resume,
    .set_bias_level     = ak4642_set_bias_level,
    .reg_cache_default  = ak4642_reg,           /* ak4642 reg */
    .reg_cache_size     = ARRAY_SIZE(ak4642_reg),   /* ak4642 reg */
    .reg_word_size      = sizeof(u8),
    .dapm_widgets       = ak4642_dapm_widgets,
    .num_dapm_widgets   = ARRAY_SIZE(ak4642_dapm_widgets),
    .dapm_routes        = ak4642_intercon,
    .num_dapm_routes    = ARRAY_SIZE(ak4642_intercon),
};

static struct snd_soc_codec_driver soc_codec_dev_ak4648 = {
    .probe          = ak4642_probe,
    .remove         = ak4642_remove,
    .resume         = ak4642_resume,
    .set_bias_level     = ak4642_set_bias_level,
    .reg_cache_default  = ak4648_reg,           /* ak4648 reg */
    .reg_cache_size     = ARRAY_SIZE(ak4648_reg),   /* ak4648 reg */
    .reg_word_size      = sizeof(u8),
    .dapm_widgets       = ak4642_dapm_widgets,
    .num_dapm_widgets   = ARRAY_SIZE(ak4642_dapm_widgets),
    .dapm_routes        = ak4642_intercon,
    .num_dapm_routes    = ARRAY_SIZE(ak4642_intercon),
};

#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 = devm_kzalloc(&i2c->dev, sizeof(struct ak4642_priv),
                  GFP_KERNEL);
    if (!ak4642)
        return -ENOMEM;

    i2c_set_clientdata(i2c, ak4642);
    ak4642->control_type = SND_SOC_I2C;

    ret =  snd_soc_register_codec(&i2c->dev,
                (struct snd_soc_codec_driver *)id->driver_data,
                &ak4642_dai, 1);
    return ret;
}

static __devexit int ak4642_i2c_remove(struct i2c_client *client)
{
    snd_soc_unregister_codec(&client->dev);
    return 0;
}

static const struct i2c_device_id ak4642_i2c_id[] = {
    { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 },
    { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 },
    { "ak4648", (kernel_ulong_t)&soc_codec_dev_ak4648 },
    { }
};
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 <[email protected]>");
MODULE_LICENSE("GPL");