- 根目录:
- sound
- soc
- codecs
- ak4642.c
/*
* 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 tested.
* AK4643 is tested.
* AK4648 is tested.
*/
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/slab.h>
#include <linux/of_device.h>
#include <linux/module.h>
#include <linux/regmap.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"},
};
/*
* ak4642 register cache
*/
static const struct reg_default ak4642_reg[] = {
{ 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 },
{ 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 },
{ 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
{ 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0x08 },
{ 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
{ 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
{ 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
{ 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
{ 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
{ 36, 0x00 },
};
static const struct reg_default ak4648_reg[] = {
{ 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 },
{ 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 },
{ 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
{ 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0xb8 },
{ 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
{ 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
{ 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
{ 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
{ 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
{ 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 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_update_bits(codec, SG_SL1, PMMP | MGAIN0, 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;
}
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;
}
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)
{
struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
regcache_mark_dirty(regmap);
regcache_sync(regmap);
return 0;
}
static int ak4642_probe(struct snd_soc_codec *codec)
{
int ret;
ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
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,
.controls = ak4642_snd_controls,
.num_controls = ARRAY_SIZE(ak4642_snd_controls),
.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 const struct regmap_config ak4642_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = ARRAY_SIZE(ak4642_reg) + 1,
.reg_defaults = ak4642_reg,
.num_reg_defaults = ARRAY_SIZE(ak4642_reg),
};
static const struct regmap_config ak4648_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = ARRAY_SIZE(ak4648_reg) + 1,
.reg_defaults = ak4648_reg,
.num_reg_defaults = ARRAY_SIZE(ak4648_reg),
};
static struct of_device_id ak4642_of_match[];
static int ak4642_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct device_node *np = i2c->dev.of_node;
const struct regmap_config *regmap_config = NULL;
struct regmap *regmap;
if (np) {
const struct of_device_id *of_id;
of_id = of_match_device(ak4642_of_match, &i2c->dev);
if (of_id)
regmap_config = of_id->data;
} else {
regmap_config = (const struct regmap_config *)id->driver_data;
}
if (!regmap_config) {
dev_err(&i2c->dev, "Unknown device type\n");
return -EINVAL;
}
regmap = devm_regmap_init_i2c(i2c, regmap_config);
if (IS_ERR(regmap))
return PTR_ERR(regmap);
return snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_ak4642, &ak4642_dai, 1);
}
static int ak4642_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
return 0;
}
static struct of_device_id ak4642_of_match[] = {
{ .compatible = "asahi-kasei,ak4642", .data = &ak4642_regmap},
{ .compatible = "asahi-kasei,ak4643", .data = &ak4642_regmap},
{ .compatible = "asahi-kasei,ak4648", .data = &ak4648_regmap},
{},
};
MODULE_DEVICE_TABLE(of, ak4642_of_match);
static const struct i2c_device_id ak4642_i2c_id[] = {
{ "ak4642", (kernel_ulong_t)&ak4642_regmap },
{ "ak4643", (kernel_ulong_t)&ak4642_regmap },
{ "ak4648", (kernel_ulong_t)&ak4648_regmap },
{ }
};
MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
static struct i2c_driver ak4642_i2c_driver = {
.driver = {
.name = "ak4642-codec",
.owner = THIS_MODULE,
.of_match_table = ak4642_of_match,
},
.probe = ak4642_i2c_probe,
.remove = ak4642_i2c_remove,
.id_table = ak4642_i2c_id,
};
module_i2c_driver(ak4642_i2c_driver);
MODULE_DESCRIPTION("Soc AK4642 driver");
MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
MODULE_LICENSE("GPL");