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linux-stable/sound/soc/codecs/ak4642.c
Lars-Peter Clausen 85e7652d89 ASoC: Constify snd_soc_dai_ops structs 
Commit 1ee46ebd("ASoC: Make the DAI ops constant in the DAI structure")
introduced the possibility to have constant DAI ops structures, yet this is
barley used in both existing drivers and also new drivers being submitted,
although none of them modifies its DAI ops structure. The later is not
surprising since existing drivers are often used as templates for new drivers.
So this patch just constifies all existing snd_soc_dai_ops structs to eliminate
the issue altogether.

The patch was generated with the following coccinelle semantic patch:
// <smpl>
@@
identifier ops;
@@
-struct snd_soc_dai_ops ops =
+const struct snd_soc_dai_ops ops =
{ ... };
// </smpl>

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-23 10:40:46 +00:00

596 lines
14 KiB
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/*
* 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/platform_device.h>
#include <linux/slab.h>
#include <linux/module.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
/* 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, -11500, 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),
SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0),
};
static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = {
SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 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_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0,
&ak4642_hpout_mixer_controls[0],
ARRAY_SIZE(ak4642_hpout_mixer_controls)),
SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0,
&ak4642_hpout_mixer_controls[0],
ARRAY_SIZE(ak4642_hpout_mixer_controls)),
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, "HPOUTL Mixer"},
{"HPOUTR", NULL, "HPOUTR Mixer"},
{"LINEOUT", NULL, "LINEOUT Mixer"},
{"HPOUTL Mixer", "DACH", "DAC"},
{"HPOUTR Mixer", "DACH", "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;
dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION);
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_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 = kzalloc(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);
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", (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 <morimoto.kuninori@renesas.com>");
MODULE_LICENSE("GPL");
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