FEX-Emu/linux
1
0
Fork 0
mirror of https://github.com/FEX-Emu/linux.git synced 2024-12-20 08:22:39 +00:00
Code Issues Actions Packages Projects Releases Wiki Activity
9d5a643439
linux/sound/soc/codecs/ad1836.c
Liam Girdwood f0fba2ad1b ASoC: multi-component - ASoC Multi-Component Support 
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.

 struct snd_soc_codec    --->  struct snd_soc_codec (device data)
                          +->  struct snd_soc_codec_driver (driver data)

 struct snd_soc_platform --->  struct snd_soc_platform (device data)
                          +->  struct snd_soc_platform_driver (driver data)

 struct snd_soc_dai      --->  struct snd_soc_dai (device data)
                          +->  struct snd_soc_dai_driver (driver data)

 struct snd_soc_device   --->  deleted

This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.

The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.

This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.

Other notable multi-component changes:-

 * Stream operations now de-reference less structures.
 * close_delayed work() now runs on a DAI basis rather than looping all DAIs
   in a card.
 * PM suspend()/resume() operations can now handle N CODECs and Platforms
   per sound card.
 * Added soc_bind_dai_link() to bind the component devices to the sound card.
 * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
   DAI link components.
 * sysfs entries can now be registered per component per card.
 * snd_soc_new_pcms() functionailty rolled into dai_link_probe().
 * snd_soc_register_codec() now does all the codec list and mutex init.

This patch changes the probe() and remove() of the CODEC drivers as follows:-

 o Make CODEC driver a platform driver
 o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
 o Removed all static codec pointers (drivers now support > 1 codec dev)
 o snd_soc_register_pcms() now done by core.
 o snd_soc_register_dai() folded into snd_soc_register_codec().

CS4270 portions:
Acked-by: Timur Tabi <timur@freescale.com>

Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>

TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>

Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>

MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>

i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>

J4740 platform fixes:-
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>

CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>

Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-08-12 14:00:00 +01:00

340 lines
8.7 KiB
C
Raw Blame History

/*
* 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);
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(codec, ad1836_dapm_widgets,
ARRAY_SIZE(ad1836_dapm_widgets));
snd_soc_dapm_add_routes(codec, 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");
Reference in New Issue View Git Blame Copy Permalink