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linux_dsm_epyc7002/sound/soc/codecs/ad1980.c
Lars-Peter Clausen cf7b71f46b ASoC: ad1980: Replace goto loop with do-while loop 
Using a proper do-while loop here instead of a open-coded goto loop is both
cleaner and shorter.

Also fixes the following warnings from smatch:
	sound/soc/codecs/ad1980.c:213 ad1980_reset() info: loop could be replaced with if statement.
	sound/soc/codecs/ad1980.c:212 ad1980_reset() info: ignoring unreachable code.
	sound/soc/codecs/ad1980.c:215 ad1980_reset() info: ignoring unreachable code.

While we are at it also change retry_cnt to unsigned int, using u16 for a
on-stack loop counter doesn't make that much sense.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-20 22:53:36 +01:00

327 lines
8.7 KiB
C
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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 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 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)
{
unsigned int retry_cnt = 0;
do {
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)
return 0;
} while (retry_cnt++ < 10);
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,
.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",
.owner = THIS_MODULE,
},
.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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