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linux_dsm_epyc7002/sound/soc/codecs/ad1980.c
Lars-Peter Clausen 7361fbeaea ASoC: ac97: Add support for resetting device before registration 
AC97 devices need to be initially reset before they can be used. Currently
each driver does this on its own.

Add support for resetting the device to core in snd_soc_new_ac97_codec().
If the caller supplies a device ID and device ID mask the function will
reset the device and verify that it has the correct ID, if it does not a
error is returned.

This will allow to remove custom code with similar functionality from
individual drivers.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-07-23 17:33:28 +01:00

348 lines
9.0 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 <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/initval.h>
#include <sound/soc.h>
static const struct reg_default ad1980_reg_defaults[] = {
{ 0x02, 0x8000 },
{ 0x04, 0x8000 },
{ 0x06, 0x8000 },
{ 0x0c, 0x8008 },
{ 0x0e, 0x8008 },
{ 0x10, 0x8808 },
{ 0x12, 0x8808 },
{ 0x16, 0x8808 },
{ 0x18, 0x8808 },
{ 0x1a, 0x0000 },
{ 0x1c, 0x8000 },
{ 0x20, 0x0000 },
{ 0x28, 0x03c7 },
{ 0x2c, 0xbb80 },
{ 0x2e, 0xbb80 },
{ 0x30, 0xbb80 },
{ 0x32, 0xbb80 },
{ 0x36, 0x8080 },
{ 0x38, 0x8080 },
{ 0x3a, 0x2000 },
{ 0x60, 0x0000 },
{ 0x62, 0x0000 },
{ 0x72, 0x0000 },
{ 0x74, 0x1001 },
{ 0x76, 0x0000 },
};
static bool ad1980_readable_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
case AC97_RESET ... AC97_MASTER_MONO:
case AC97_PHONE ... AC97_CD:
case AC97_AUX ... AC97_GENERAL_PURPOSE:
case AC97_POWERDOWN ... AC97_PCM_LR_ADC_RATE:
case AC97_SPDIF:
case AC97_CODEC_CLASS_REV:
case AC97_PCI_SVID:
case AC97_AD_CODEC_CFG:
case AC97_AD_JACK_SPDIF:
case AC97_AD_SERIAL_CFG:
case AC97_VENDOR_ID1:
case AC97_VENDOR_ID2:
return true;
default:
return false;
}
}
static bool ad1980_writeable_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
case AC97_VENDOR_ID1:
case AC97_VENDOR_ID2:
return false;
default:
return ad1980_readable_reg(dev, reg);
}
}
static const struct regmap_config ad1980_regmap_config = {
.reg_bits = 16,
.reg_stride = 2,
.val_bits = 16,
.max_register = 0x7e,
.cache_type = REGCACHE_RBTREE,
.volatile_reg = regmap_ac97_default_volatile,
.readable_reg = ad1980_readable_reg,
.writeable_reg = ad1980_writeable_reg,
.reg_defaults = ad1980_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(ad1980_reg_defaults),
};
static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line",
"Stereo Mix", "Mono Mix", "Phone"};
static SOC_ENUM_DOUBLE_DECL(ad1980_cap_src,
AC97_REC_SEL, 8, 0, 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 const struct snd_soc_dapm_widget ad1980_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
SND_SOC_DAPM_INPUT("CD_L"),
SND_SOC_DAPM_INPUT("CD_R"),
SND_SOC_DAPM_INPUT("AUX_L"),
SND_SOC_DAPM_INPUT("AUX_R"),
SND_SOC_DAPM_INPUT("LINE_IN_L"),
SND_SOC_DAPM_INPUT("LINE_IN_R"),
SND_SOC_DAPM_OUTPUT("LFE_OUT"),
SND_SOC_DAPM_OUTPUT("CENTER_OUT"),
SND_SOC_DAPM_OUTPUT("LINE_OUT_L"),
SND_SOC_DAPM_OUTPUT("LINE_OUT_R"),
SND_SOC_DAPM_OUTPUT("MONO_OUT"),
SND_SOC_DAPM_OUTPUT("HP_OUT_L"),
SND_SOC_DAPM_OUTPUT("HP_OUT_R"),
};
static const struct snd_soc_dapm_route ad1980_dapm_routes[] = {
{ "Capture", NULL, "MIC1" },
{ "Capture", NULL, "MIC2" },
{ "Capture", NULL, "CD_L" },
{ "Capture", NULL, "CD_R" },
{ "Capture", NULL, "AUX_L" },
{ "Capture", NULL, "AUX_R" },
{ "Capture", NULL, "LINE_IN_L" },
{ "Capture", NULL, "LINE_IN_R" },
{ "LFE_OUT", NULL, "Playback" },
{ "CENTER_OUT", NULL, "Playback" },
{ "LINE_OUT_L", NULL, "Playback" },
{ "LINE_OUT_R", NULL, "Playback" },
{ "MONO_OUT", NULL, "Playback" },
{ "HP_OUT_L", NULL, "Playback" },
{ "HP_OUT_R", NULL, "Playback" },
};
static struct snd_soc_dai_driver ad1980_dai = {
.name = "ad1980-hifi",
.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)
{
struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
unsigned int retry_cnt = 0;
do {
if (try_warm && soc_ac97_ops->warm_reset) {
soc_ac97_ops->warm_reset(ac97);
if (snd_soc_read(codec, AC97_RESET) == 0x0090)
return 1;
}
soc_ac97_ops->reset(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)
*/
snd_soc_write(codec, AC97_AD_SERIAL_CFG, 0x9900);
if (snd_soc_read(codec, AC97_RESET) == 0x0090)
return 0;
} while (retry_cnt++ < 10);
dev_err(codec->dev, "Failed to reset: AC97 link error\n");
return -EIO;
}
static int ad1980_soc_probe(struct snd_soc_codec *codec)
{
struct snd_ac97 *ac97;
struct regmap *regmap;
int ret;
u16 vendor_id2;
u16 ext_status;
ac97 = snd_soc_new_ac97_codec(codec, 0, 0);
if (IS_ERR(ac97)) {
ret = PTR_ERR(ac97);
dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret);
return ret;
}
regmap = regmap_init_ac97(ac97, &ad1980_regmap_config);
if (IS_ERR(regmap)) {
ret = PTR_ERR(regmap);
goto err_free_ac97;
}
snd_soc_codec_init_regmap(codec, regmap);
snd_soc_codec_set_drvdata(codec, ac97);
ret = ad1980_reset(codec, 0);
if (ret < 0)
goto reset_err;
/* Read out vendor ID to make sure it is ad1980 */
if (snd_soc_read(codec, AC97_VENDOR_ID1) != 0x4144) {
ret = -ENODEV;
goto reset_err;
}
vendor_id2 = snd_soc_read(codec, AC97_VENDOR_ID2);
if (vendor_id2 != 0x5370) {
if (vendor_id2 != 0x5374) {
ret = -ENODEV;
goto reset_err;
} else {
dev_warn(codec->dev,
"Found AD1981 - only 2/2 IN/OUT Channels supported\n");
}
}
/* unmute captures and playbacks volume */
snd_soc_write(codec, AC97_MASTER, 0x0000);
snd_soc_write(codec, AC97_PCM, 0x0000);
snd_soc_write(codec, AC97_REC_GAIN, 0x0000);
snd_soc_write(codec, AC97_CENTER_LFE_MASTER, 0x0000);
snd_soc_write(codec, AC97_SURROUND_MASTER, 0x0000);
/*power on LFE/CENTER/Surround DACs*/
ext_status = snd_soc_read(codec, AC97_EXTENDED_STATUS);
snd_soc_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
return 0;
reset_err:
snd_soc_codec_exit_regmap(codec);
err_free_ac97:
snd_soc_free_ac97_codec(ac97);
return ret;
}
static int ad1980_soc_remove(struct snd_soc_codec *codec)
{
struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
snd_soc_codec_exit_regmap(codec);
snd_soc_free_ac97_codec(ac97);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_ad1980 = {
.probe = ad1980_soc_probe,
.remove = ad1980_soc_remove,
.controls = ad1980_snd_ac97_controls,
.num_controls = ARRAY_SIZE(ad1980_snd_ac97_controls),
.dapm_widgets = ad1980_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ad1980_dapm_widgets),
.dapm_routes = ad1980_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(ad1980_dapm_routes),
};
static int ad1980_probe(struct platform_device *pdev)
{
return snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_ad1980, &ad1980_dai, 1);
}
static int ad1980_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static struct platform_driver ad1980_codec_driver = {
.driver = {
.name = "ad1980",
},
.probe = ad1980_probe,
.remove = 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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