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linux/sound/soc/codecs/ad1980.c
Liam Girdwood 022658beab ASoC: core: Add support for DAI and machine kcontrols. 
Currently ASoC can only add kcontrols using codec and platform component device
handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for
SoC card machine drivers too. This allows the kcontrol to have a direct handle to
the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily
get it's private data.

This change makes snd_soc_add_controls() static and wraps it in the folowing
calls (card and dai are new) :-

snd_soc_add_card_controls()
snd_soc_add_codec_controls()
snd_soc_add_dai_controls()
snd_soc_add_platform_controls()

This patch also does a lot of small mechanical changes in individual codec drivers
to replace snd_soc_add_controls() with snd_soc_add_codec_controls().

It also updates the McBSP DAI driver to use snd_soc_add_dai_controls().

Finally, it updates the existing machine drivers that register controls to either :-

1) Use snd_soc_add_card_controls() where no direct codec control is required.
2) Use snd_soc_add_codec_controls() where there is direct codec control.

In the case of 1) above we also update the machine drivers to get the correct
component data pointers from the kcontrol (rather than getting the machine pointer
via the codec pointer).

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-02-04 12:40:11 +00:00

285 lines
7.5 KiB
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/*
* ad1980.c -- ALSA Soc AD1980 codec support
*
* Copyright: Analog Device Inc.
* Author: Roy Huang <roy.huang@analog.com>
* Cliff Cai <cliff.cai@analog.com>
*
* 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.
*/
/*
* WARNING:
*
* Because Analog Devices Inc. discontinued the ad1980 sound chip since
* Sep. 2009, this ad1980 driver is not maintained, tested and supported
* by ADI now.
*/
#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/ac97_codec.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include "ad1980.h"
/*
* AD1980 register cache
*/
static const u16 ad1980_reg[] = {
0x0090, 0x8000, 0x8000, 0x8000, /* 0 - 6 */
0x0000, 0x0000, 0x8008, 0x8008, /* 8 - e */
0x8808, 0x8808, 0x0000, 0x8808, /* 10 - 16 */
0x8808, 0x0000, 0x8000, 0x0000, /* 18 - 1e */
0x0000, 0x0000, 0x0000, 0x0000, /* 20 - 26 */
0x03c7, 0x0000, 0xbb80, 0xbb80, /* 28 - 2e */
0xbb80, 0xbb80, 0x0000, 0x8080, /* 30 - 36 */
0x8080, 0x2000, 0x0000, 0x0000, /* 38 - 3e */
0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
0x8080, 0x0000, 0x0000, 0x0000, /* 60 - 66 */
0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
0x0000, 0x0000, 0x1001, 0x0000, /* 70 - 76 */
0x0000, 0x0000, 0x4144, 0x5370 /* 78 - 7e */
};
static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line",
"Stereo Mix", "Mono Mix", "Phone"};
static const struct soc_enum ad1980_cap_src =
SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel);
static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = {
SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1),
SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1),
SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 31, 0),
SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1),
SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
SOC_SINGLE("Phone Capture Volume", AC97_PHONE, 0, 31, 1),
SOC_SINGLE("Phone Capture Switch", AC97_PHONE, 15, 1, 1),
SOC_SINGLE("Mic Volume", AC97_MIC, 0, 31, 1),
SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
SOC_SINGLE("Stereo Mic Switch", AC97_AD_MISC, 6, 1, 0),
SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0),
SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1),
SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1),
SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1),
SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1),
SOC_ENUM("Capture Source", ad1980_cap_src),
SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
};
static unsigned int ac97_read(struct snd_soc_codec *codec,
unsigned int reg)
{
u16 *cache = codec->reg_cache;
switch (reg) {
case AC97_RESET:
case AC97_INT_PAGING:
case AC97_POWERDOWN:
case AC97_EXTENDED_STATUS:
case AC97_VENDOR_ID1:
case AC97_VENDOR_ID2:
return soc_ac97_ops.read(codec->ac97, reg);
default:
reg = reg >> 1;
if (reg >= ARRAY_SIZE(ad1980_reg))
return -EINVAL;
return cache[reg];
}
}
static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int val)
{
u16 *cache = codec->reg_cache;
soc_ac97_ops.write(codec->ac97, reg, val);
reg = reg >> 1;
if (reg < ARRAY_SIZE(ad1980_reg))
cache[reg] = val;
return 0;
}
static struct snd_soc_dai_driver ad1980_dai = {
.name = "ad1980-hifi",
.ac97_control = 1,
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 6,
.rates = SNDRV_PCM_RATE_48000,
.formats = SND_SOC_STD_AC97_FMTS, },
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_48000,
.formats = SND_SOC_STD_AC97_FMTS, },
};
static int ad1980_reset(struct snd_soc_codec *codec, int try_warm)
{
u16 retry_cnt = 0;
retry:
if (try_warm && soc_ac97_ops.warm_reset) {
soc_ac97_ops.warm_reset(codec->ac97);
if (ac97_read(codec, AC97_RESET) == 0x0090)
return 1;
}
soc_ac97_ops.reset(codec->ac97);
/* Set bit 16slot in register 74h, then every slot will has only 16
* bits. This command is sent out in 20bit mode, in which case the
* first nibble of data is eaten by the addr. (Tag is always 16 bit)*/
ac97_write(codec, AC97_AD_SERIAL_CFG, 0x9900);
if (ac97_read(codec, AC97_RESET) != 0x0090)
goto err;
return 0;
err:
while (retry_cnt++ < 10)
goto retry;
printk(KERN_ERR "AD1980 AC97 reset failed\n");
return -EIO;
}
static int ad1980_soc_probe(struct snd_soc_codec *codec)
{
int ret;
u16 vendor_id2;
u16 ext_status;
printk(KERN_INFO "AD1980 SoC Audio Codec\n");
ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
if (ret < 0) {
printk(KERN_ERR "ad1980: failed to register AC97 codec\n");
return ret;
}
ret = ad1980_reset(codec, 0);
if (ret < 0) {
printk(KERN_ERR "Failed to reset AD1980: AC97 link error\n");
goto reset_err;
}
/* Read out vendor ID to make sure it is ad1980 */
if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144) {
ret = -ENODEV;
goto reset_err;
}
vendor_id2 = ac97_read(codec, AC97_VENDOR_ID2);
if (vendor_id2 != 0x5370) {
if (vendor_id2 != 0x5374) {
ret = -ENODEV;
goto reset_err;
} else {
printk(KERN_WARNING "ad1980: "
"Found AD1981 - only 2/2 IN/OUT Channels "
"supported\n");
}
}
/* unmute captures and playbacks volume */
ac97_write(codec, AC97_MASTER, 0x0000);
ac97_write(codec, AC97_PCM, 0x0000);
ac97_write(codec, AC97_REC_GAIN, 0x0000);
ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000);
ac97_write(codec, AC97_SURROUND_MASTER, 0x0000);
/*power on LFE/CENTER/Surround DACs*/
ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
snd_soc_add_codec_controls(codec, ad1980_snd_ac97_controls,
ARRAY_SIZE(ad1980_snd_ac97_controls));
return 0;
reset_err:
snd_soc_free_ac97_codec(codec);
return ret;
}
static int ad1980_soc_remove(struct snd_soc_codec *codec)
{
snd_soc_free_ac97_codec(codec);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_ad1980 = {
.probe = ad1980_soc_probe,
.remove = ad1980_soc_remove,
.reg_cache_size = ARRAY_SIZE(ad1980_reg),
.reg_word_size = sizeof(u16),
.reg_cache_default = ad1980_reg,
.reg_cache_step = 2,
.write = ac97_write,
.read = ac97_read,
};
static __devinit int ad1980_probe(struct platform_device *pdev)
{
return snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_ad1980, &ad1980_dai, 1);
}
static int __devexit ad1980_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static struct platform_driver ad1980_codec_driver = {
.driver = {
.name = "ad1980",
.owner = THIS_MODULE,
},
.probe = ad1980_probe,
.remove = __devexit_p(ad1980_remove),
};
module_platform_driver(ad1980_codec_driver);
MODULE_DESCRIPTION("ASoC ad1980 driver (Obsolete)");
MODULE_AUTHOR("Roy Huang, Cliff Cai");
MODULE_LICENSE("GPL");
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