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kernel-fxtec-pro1x/sound/soc/codecs/ad1836.c
Liam Girdwood ce6120cca2 ASoC: Decouple DAPM from CODECs 
Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.

This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.

This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.

Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.

Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-06 11:28:29 -04:00

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/*
* File: sound/soc/codecs/ad1836.c
* Author: Barry Song <Barry.Song@analog.com>
*
* Created: Aug 04 2009
* Description: Driver for AD1836 sound chip
*
* Modified:
* Copyright 2009 Analog Devices Inc.
*
* Bugs: Enter bugs at http://blackfin.uclinux.org/
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*/
#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 <sound/soc-dapm.h>
#include <linux/spi/spi.h>
#include "ad1836.h"
/* codec private data */
struct ad1836_priv {
enum snd_soc_control_type control_type;
void *control_data;
};
/*
* 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);
static const struct snd_kcontrol_new ad1836_snd_controls[] = {
/* DAC volume control */
SOC_DOUBLE_R("DAC1 Volume", AD1836_DAC_L1_VOL,
AD1836_DAC_R1_VOL, 0, 0x3FF, 0),
SOC_DOUBLE_R("DAC2 Volume", AD1836_DAC_L2_VOL,
AD1836_DAC_R2_VOL, 0, 0x3FF, 0),
SOC_DOUBLE_R("DAC3 Volume", AD1836_DAC_L3_VOL,
AD1836_DAC_R3_VOL, 0, 0x3FF, 0),
/* ADC switch control */
SOC_DOUBLE("ADC1 Switch", AD1836_ADC_CTRL2, AD1836_ADCL1_MUTE,
AD1836_ADCR1_MUTE, 1, 1),
SOC_DOUBLE("ADC2 Switch", AD1836_ADC_CTRL2, AD1836_ADCL2_MUTE,
AD1836_ADCR2_MUTE, 1, 1),
/* DAC switch control */
SOC_DOUBLE("DAC1 Switch", AD1836_DAC_CTRL2, AD1836_DACL1_MUTE,
AD1836_DACR1_MUTE, 1, 1),
SOC_DOUBLE("DAC2 Switch", AD1836_DAC_CTRL2, AD1836_DACL2_MUTE,
AD1836_DACR2_MUTE, 1, 1),
SOC_DOUBLE("DAC3 Switch", AD1836_DAC_CTRL2, AD1836_DACL3_MUTE,
AD1836_DACR3_MUTE, 1, 1),
/* 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 ad1836_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),
SND_SOC_DAPM_OUTPUT("DAC1OUT"),
SND_SOC_DAPM_OUTPUT("DAC2OUT"),
SND_SOC_DAPM_OUTPUT("DAC3OUT"),
SND_SOC_DAPM_INPUT("ADC1IN"),
SND_SOC_DAPM_INPUT("ADC2IN"),
};
static const struct snd_soc_dapm_route audio_paths[] = {
{ "DAC", NULL, "ADC_PWR" },
{ "ADC", NULL, "ADC_PWR" },
{ "DAC1OUT", "DAC1 Switch", "DAC" },
{ "DAC2OUT", "DAC2 Switch", "DAC" },
{ "DAC3OUT", "DAC3 Switch", "DAC" },
{ "ADC", "ADC1 Switch", "ADC1IN" },
{ "ADC", "ADC2 Switch", "ADC2IN" },
};
/*
* 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_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
word_len = 3;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
word_len = 1;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case SNDRV_PCM_FORMAT_S32_LE:
word_len = 0;
break;
}
snd_soc_update_bits(codec, AD1836_DAC_CTRL1,
AD1836_DAC_WORD_LEN_MASK, word_len);
snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
AD1836_ADC_WORD_LEN_MASK, word_len);
return 0;
}
#ifdef CONFIG_PM
static int ad1836_soc_suspend(struct snd_soc_codec *codec,
pm_message_t state)
{
/* reset clock control mode */
u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK;
return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
}
static int ad1836_soc_resume(struct snd_soc_codec *codec)
{
/* restore clock control mode */
u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
adc_ctrl2 |= AD1836_ADC_AUX;
return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
}
#else
#define ad1836_soc_suspend NULL
#define ad1836_soc_resume NULL
#endif
static struct snd_soc_dai_ops ad1836_dai_ops = {
.hw_params = ad1836_hw_params,
.set_fmt = ad1836_set_dai_fmt,
};
/* codec DAI instance */
static struct snd_soc_dai_driver ad1836_dai = {
.name = "ad1836-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 6,
.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 = 4,
.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 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 ret = 0;
codec->control_data = ad1836->control_data;
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);
kfree(ad1836);
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);
/* left/right diff:PGA/MUX */
snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A);
/* volume */
snd_soc_write(codec, AD1836_DAC_L1_VOL, 0x3FF);
snd_soc_write(codec, AD1836_DAC_R1_VOL, 0x3FF);
snd_soc_write(codec, AD1836_DAC_L2_VOL, 0x3FF);
snd_soc_write(codec, AD1836_DAC_R2_VOL, 0x3FF);
snd_soc_write(codec, AD1836_DAC_L3_VOL, 0x3FF);
snd_soc_write(codec, AD1836_DAC_R3_VOL, 0x3FF);
snd_soc_add_controls(codec, ad1836_snd_controls,
ARRAY_SIZE(ad1836_snd_controls));
snd_soc_dapm_new_controls(dapm, ad1836_dapm_widgets,
ARRAY_SIZE(ad1836_dapm_widgets));
snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
return ret;
}
/* power down chip */
static int ad1836_remove(struct snd_soc_codec *codec)
{
/* reset clock control mode */
u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK;
return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
}
static struct snd_soc_codec_driver soc_codec_dev_ad1836 = {
.probe = ad1836_probe,
.remove = ad1836_remove,
.suspend = ad1836_soc_suspend,
.resume = ad1836_soc_resume,
.reg_cache_size = AD1836_NUM_REGS,
.reg_word_size = sizeof(u16),
};
static int __devinit ad1836_spi_probe(struct spi_device *spi)
{
struct ad1836_priv *ad1836;
int ret;
ad1836 = kzalloc(sizeof(struct ad1836_priv), GFP_KERNEL);
if (ad1836 == NULL)
return -ENOMEM;
spi_set_drvdata(spi, ad1836);
ad1836->control_data = spi;
ad1836->control_type = SND_SOC_SPI;
ret = snd_soc_register_codec(&spi->dev,
&soc_codec_dev_ad1836, &ad1836_dai, 1);
if (ret < 0)
kfree(ad1836);
return ret;
}
static int __devexit ad1836_spi_remove(struct spi_device *spi)
{
snd_soc_unregister_codec(&spi->dev);
kfree(spi_get_drvdata(spi));
return 0;
}
static struct spi_driver ad1836_spi_driver = {
.driver = {
.name = "ad1836-codec",
.owner = THIS_MODULE,
},
.probe = ad1836_spi_probe,
.remove = __devexit_p(ad1836_spi_remove),
};
static int __init ad1836_init(void)
{
int ret;
ret = spi_register_driver(&ad1836_spi_driver);
if (ret != 0) {
printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n",
ret);
}
return ret;
}
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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