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kernel-ark/sound/soc/codecs/ad1836.c
Mark Brown e6968a1719 ASoC: codecs: Remove rtd->codec usage from CODEC drivers 
In order to support CODEC<->CODEC links remove the assumption that there
is only a single CODEC on a DAI link by removing the use of the CODEC
pointer in the rtd from the CODEC drivers. They are already being passed
their DAI whenever they are passed an rtd and can get the CODEC from
there.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-04 15:59:12 +01:00

397 lines
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/*
* Audio Codec driver supporting:
* AD1835A, AD1836, AD1837A, AD1838A, AD1839A
*
* Copyright 2009-2011 Analog Devices Inc.
*
* Licensed under the GPL-2 or later.
*/
#include <linux/init.h>
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#include <linux/spi/spi.h>
#include "ad1836.h"
enum ad1836_type {
AD1835,
AD1836,
AD1838,
};
/* codec private data */
struct ad1836_priv {
enum ad1836_type type;
};
/*
* AD1836 volume/mute/de-emphasis etc. controls
*/
static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"};
static const struct soc_enum ad1836_deemp_enum =
SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp);
#define AD1836_DAC_VOLUME(x) \
SOC_DOUBLE_R("DAC" #x " Playback Volume", AD1836_DAC_L_VOL(x), \
AD1836_DAC_R_VOL(x), 0, 0x3FF, 0)
#define AD1836_DAC_SWITCH(x) \
SOC_DOUBLE("DAC" #x " Playback Switch", AD1836_DAC_CTRL2, \
AD1836_MUTE_LEFT(x), AD1836_MUTE_RIGHT(x), 1, 1)
#define AD1836_ADC_SWITCH(x) \
SOC_DOUBLE("ADC" #x " Capture Switch", AD1836_ADC_CTRL2, \
AD1836_MUTE_LEFT(x), AD1836_MUTE_RIGHT(x), 1, 1)
static const struct snd_kcontrol_new ad183x_dac_controls[] = {
AD1836_DAC_VOLUME(1),
AD1836_DAC_SWITCH(1),
AD1836_DAC_VOLUME(2),
AD1836_DAC_SWITCH(2),
AD1836_DAC_VOLUME(3),
AD1836_DAC_SWITCH(3),
AD1836_DAC_VOLUME(4),
AD1836_DAC_SWITCH(4),
};
static const struct snd_soc_dapm_widget ad183x_dac_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("DAC1OUT"),
SND_SOC_DAPM_OUTPUT("DAC2OUT"),
SND_SOC_DAPM_OUTPUT("DAC3OUT"),
SND_SOC_DAPM_OUTPUT("DAC4OUT"),
};
static const struct snd_soc_dapm_route ad183x_dac_routes[] = {
{ "DAC1OUT", NULL, "DAC" },
{ "DAC2OUT", NULL, "DAC" },
{ "DAC3OUT", NULL, "DAC" },
{ "DAC4OUT", NULL, "DAC" },
};
static const struct snd_kcontrol_new ad183x_adc_controls[] = {
AD1836_ADC_SWITCH(1),
AD1836_ADC_SWITCH(2),
AD1836_ADC_SWITCH(3),
};
static const struct snd_soc_dapm_widget ad183x_adc_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("ADC1IN"),
SND_SOC_DAPM_INPUT("ADC2IN"),
};
static const struct snd_soc_dapm_route ad183x_adc_routes[] = {
{ "ADC", NULL, "ADC1IN" },
{ "ADC", NULL, "ADC2IN" },
};
static const struct snd_kcontrol_new ad183x_controls[] = {
/* ADC high-pass filter */
SOC_SINGLE("ADC High Pass Filter Switch", AD1836_ADC_CTRL1,
AD1836_ADC_HIGHPASS_FILTER, 1, 0),
/* DAC de-emphasis */
SOC_ENUM("Playback Deemphasis", ad1836_deemp_enum),
};
static const struct snd_soc_dapm_widget ad183x_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "Playback", AD1836_DAC_CTRL1,
AD1836_DAC_POWERDOWN, 1),
SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1836_ADC_CTRL1,
AD1836_ADC_POWERDOWN, 1, NULL, 0),
};
static const struct snd_soc_dapm_route ad183x_dapm_routes[] = {
{ "DAC", NULL, "ADC_PWR" },
{ "ADC", NULL, "ADC_PWR" },
};
static const DECLARE_TLV_DB_SCALE(ad1836_in_tlv, 0, 300, 0);
static const struct snd_kcontrol_new ad1836_controls[] = {
SOC_DOUBLE_TLV("ADC2 Capture Volume", AD1836_ADC_CTRL1, 3, 0, 4, 0,
ad1836_in_tlv),
};
/*
* DAI ops entries
*/
static int ad1836_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
/* at present, we support adc aux mode to interface with
* blackfin sport tdm mode
*/
case SND_SOC_DAIFMT_DSP_A:
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_IB_IF:
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
/* ALCLK,ABCLK are both output, AD1836 can only be master */
case SND_SOC_DAIFMT_CBM_CFM:
break;
default:
return -EINVAL;
}
return 0;
}
static int ad1836_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
int word_len = 0;
struct snd_soc_codec *codec = dai->codec;
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
word_len = AD1836_WORD_LEN_16;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
word_len = AD1836_WORD_LEN_20;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case SNDRV_PCM_FORMAT_S32_LE:
word_len = AD1836_WORD_LEN_24;
break;
}
snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK,
word_len << AD1836_DAC_WORD_LEN_OFFSET);
snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK,
word_len << AD1836_ADC_WORD_OFFSET);
return 0;
}
static const struct snd_soc_dai_ops ad1836_dai_ops = {
.hw_params = ad1836_hw_params,
.set_fmt = ad1836_set_dai_fmt,
};
#define AD183X_DAI(_name, num_dacs, num_adcs) \
{ \
.name = _name "-hifi", \
.playback = { \
.stream_name = "Playback", \
.channels_min = 2, \
.channels_max = (num_dacs) * 2, \
.rates = SNDRV_PCM_RATE_48000, \
.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, \
}, \
.capture = { \
.stream_name = "Capture", \
.channels_min = 2, \
.channels_max = (num_adcs) * 2, \
.rates = SNDRV_PCM_RATE_48000, \
.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, \
}, \
.ops = &ad1836_dai_ops, \
}
static struct snd_soc_dai_driver ad183x_dais[] = {
[AD1835] = AD183X_DAI("ad1835", 4, 1),
[AD1836] = AD183X_DAI("ad1836", 3, 2),
[AD1838] = AD183X_DAI("ad1838", 3, 1),
};
#ifdef CONFIG_PM
static int ad1836_suspend(struct snd_soc_codec *codec)
{
/* reset clock control mode */
return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
AD1836_ADC_SERFMT_MASK, 0);
}
static int ad1836_resume(struct snd_soc_codec *codec)
{
/* restore clock control mode */
return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
AD1836_ADC_SERFMT_MASK, AD1836_ADC_AUX);
}
#else
#define ad1836_suspend NULL
#define ad1836_resume NULL
#endif
static int ad1836_probe(struct snd_soc_codec *codec)
{
struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
int num_dacs, num_adcs;
int ret = 0;
int i;
num_dacs = ad183x_dais[ad1836->type].playback.channels_max / 2;
num_adcs = ad183x_dais[ad1836->type].capture.channels_max / 2;
ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI);
if (ret < 0) {
dev_err(codec->dev, "failed to set cache I/O: %d\n",
ret);
return ret;
}
/* default setting for ad1836 */
/* de-emphasis: 48kHz, power-on dac */
snd_soc_write(codec, AD1836_DAC_CTRL1, 0x300);
/* unmute dac channels */
snd_soc_write(codec, AD1836_DAC_CTRL2, 0x0);
/* high-pass filter enable, power-on adc */
snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100);
/* unmute adc channles, adc aux mode */
snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180);
/* volume */
for (i = 1; i <= num_dacs; ++i) {
snd_soc_write(codec, AD1836_DAC_L_VOL(i), 0x3FF);
snd_soc_write(codec, AD1836_DAC_R_VOL(i), 0x3FF);
}
if (ad1836->type == AD1836) {
/* left/right diff:PGA/MUX */
snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A);
ret = snd_soc_add_codec_controls(codec, ad1836_controls,
ARRAY_SIZE(ad1836_controls));
if (ret)
return ret;
} else {
snd_soc_write(codec, AD1836_ADC_CTRL3, 0x00);
}
ret = snd_soc_add_codec_controls(codec, ad183x_dac_controls, num_dacs * 2);
if (ret)
return ret;
ret = snd_soc_add_codec_controls(codec, ad183x_adc_controls, num_adcs);
if (ret)
return ret;
ret = snd_soc_dapm_new_controls(dapm, ad183x_dac_dapm_widgets, num_dacs);
if (ret)
return ret;
ret = snd_soc_dapm_new_controls(dapm, ad183x_adc_dapm_widgets, num_adcs);
if (ret)
return ret;
ret = snd_soc_dapm_add_routes(dapm, ad183x_dac_routes, num_dacs);
if (ret)
return ret;
ret = snd_soc_dapm_add_routes(dapm, ad183x_adc_routes, num_adcs);
if (ret)
return ret;
return ret;
}
/* power down chip */
static int ad1836_remove(struct snd_soc_codec *codec)
{
/* reset clock control mode */
return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
AD1836_ADC_SERFMT_MASK, 0);
}
static struct snd_soc_codec_driver soc_codec_dev_ad1836 = {
.probe = ad1836_probe,
.remove = ad1836_remove,
.suspend = ad1836_suspend,
.resume = ad1836_resume,
.reg_cache_size = AD1836_NUM_REGS,
.reg_word_size = sizeof(u16),
.controls = ad183x_controls,
.num_controls = ARRAY_SIZE(ad183x_controls),
.dapm_widgets = ad183x_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ad183x_dapm_widgets),
.dapm_routes = ad183x_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(ad183x_dapm_routes),
};
static int __devinit ad1836_spi_probe(struct spi_device *spi)
{
struct ad1836_priv *ad1836;
int ret;
ad1836 = devm_kzalloc(&spi->dev, sizeof(struct ad1836_priv),
GFP_KERNEL);
if (ad1836 == NULL)
return -ENOMEM;
ad1836->type = spi_get_device_id(spi)->driver_data;
spi_set_drvdata(spi, ad1836);
ret = snd_soc_register_codec(&spi->dev,
&soc_codec_dev_ad1836, &ad183x_dais[ad1836->type], 1);
return ret;
}
static int __devexit ad1836_spi_remove(struct spi_device *spi)
{
snd_soc_unregister_codec(&spi->dev);
return 0;
}
static const struct spi_device_id ad1836_ids[] = {
{ "ad1835", AD1835 },
{ "ad1836", AD1836 },
{ "ad1837", AD1835 },
{ "ad1838", AD1838 },
{ "ad1839", AD1838 },
{ },
};
MODULE_DEVICE_TABLE(spi, ad1836_ids);
static struct spi_driver ad1836_spi_driver = {
.driver = {
.name = "ad1836",
.owner = THIS_MODULE,
},
.probe = ad1836_spi_probe,
.remove = __devexit_p(ad1836_spi_remove),
.id_table = ad1836_ids,
};
static int __init ad1836_init(void)
{
return spi_register_driver(&ad1836_spi_driver);
}
module_init(ad1836_init);
static void __exit ad1836_exit(void)
{
spi_unregister_driver(&ad1836_spi_driver);
}
module_exit(ad1836_exit);
MODULE_DESCRIPTION("ASoC ad1836 driver");
MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>");
MODULE_LICENSE("GPL");
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