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linux/sound/soc/codecs/stac9766.c

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/*
* stac9766.c -- ALSA SoC STAC9766 codec support
*
* Copyright 2009 Jon Smirl, Digispeaker
* Author: Jon Smirl <jonsmirl@gmail.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.
*
* Features:-
*
* o Support for AC97 Codec, S/PDIF
*/
#include <linux/init.h>
#include <linux/slab.h>
#include <linux/module.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/pcm_params.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#define STAC9766_VENDOR_ID 0x83847666
#define STAC9766_VENDOR_ID_MASK 0xffffffff
#define AC97_STAC_DA_CONTROL 0x6A
#define AC97_STAC_ANALOG_SPECIAL 0x6E
#define AC97_STAC_STEREO_MIC 0x78
static const struct reg_default stac9766_reg_defaults[] = {
{ 0x02, 0x8000 },
{ 0x04, 0x8000 },
{ 0x06, 0x8000 },
{ 0x0a, 0x0000 },
{ 0x0c, 0x8008 },
{ 0x0e, 0x8008 },
{ 0x10, 0x8808 },
{ 0x12, 0x8808 },
{ 0x14, 0x8808 },
{ 0x16, 0x8808 },
{ 0x18, 0x8808 },
{ 0x1a, 0x0000 },
{ 0x1c, 0x8000 },
{ 0x20, 0x0000 },
{ 0x22, 0x0000 },
{ 0x28, 0x0a05 },
{ 0x2c, 0xbb80 },
{ 0x32, 0xbb80 },
{ 0x3a, 0x2000 },
{ 0x3e, 0x0100 },
{ 0x4c, 0x0300 },
{ 0x4e, 0xffff },
{ 0x50, 0x0000 },
{ 0x52, 0x0000 },
{ 0x54, 0x0000 },
{ 0x6a, 0x0000 },
{ 0x6e, 0x1000 },
{ 0x72, 0x0000 },
{ 0x78, 0x0000 },
};
static const struct regmap_config stac9766_regmap_config = {
.reg_bits = 16,
.reg_stride = 2,
.val_bits = 16,
.max_register = 0x78,
.cache_type = REGCACHE_RBTREE,
.volatile_reg = regmap_ac97_default_volatile,
.reg_defaults = stac9766_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(stac9766_reg_defaults),
};
static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
"Line", "Stereo Mix", "Mono Mix", "Phone"};
static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
static const char *stac9766_record_all_mux[] = {"All analog",
"Analog plus DAC"};
static const char *stac9766_boost1[] = {"0dB", "10dB"};
static const char *stac9766_boost2[] = {"0dB", "20dB"};
static const char *stac9766_stereo_mic[] = {"Off", "On"};
static SOC_ENUM_DOUBLE_DECL(stac9766_record_enum,
AC97_REC_SEL, 8, 0, stac9766_record_mux);
static SOC_ENUM_SINGLE_DECL(stac9766_mono_enum,
AC97_GENERAL_PURPOSE, 9, stac9766_mono_mux);
static SOC_ENUM_SINGLE_DECL(stac9766_mic_enum,
AC97_GENERAL_PURPOSE, 8, stac9766_mic_mux);
static SOC_ENUM_SINGLE_DECL(stac9766_SPDIF_enum,
AC97_STAC_DA_CONTROL, 1, stac9766_SPDIF_mux);
static SOC_ENUM_SINGLE_DECL(stac9766_popbypass_enum,
AC97_GENERAL_PURPOSE, 15, stac9766_popbypass_mux);
static SOC_ENUM_SINGLE_DECL(stac9766_record_all_enum,
AC97_STAC_ANALOG_SPECIAL, 12,
stac9766_record_all_mux);
static SOC_ENUM_SINGLE_DECL(stac9766_boost1_enum,
AC97_MIC, 6, stac9766_boost1); /* 0/10dB */
static SOC_ENUM_SINGLE_DECL(stac9766_boost2_enum,
AC97_STAC_ANALOG_SPECIAL, 2, stac9766_boost2); /* 0/20dB */
static SOC_ENUM_SINGLE_DECL(stac9766_stereo_mic_enum,
AC97_STAC_STEREO_MIC, 2, stac9766_stereo_mic);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(master_tlv, -4650, 150, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(record_tlv, 0, 150, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(beep_tlv, -4500, 300, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(mix_tlv, -3450, 150, 0);
static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
master_tlv),
SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
master_tlv),
SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
SOC_ENUM("Record All Mux", stac9766_record_all_enum),
SOC_ENUM("Record Mux", stac9766_record_enum),
SOC_ENUM("Mono Mux", stac9766_mono_enum),
SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
};
static int ac97_analog_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned short reg;
/* enable variable rate audio, disable SPDIF output */
snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x5, 0x1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
reg = AC97_PCM_FRONT_DAC_RATE;
else
reg = AC97_PCM_LR_ADC_RATE;
return snd_soc_write(codec, reg, runtime->rate);
}
static int ac97_digital_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned short reg;
snd_soc_write(codec, AC97_SPDIF, 0x2002);
/* Enable VRA and SPDIF out */
snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x5, 0x5);
reg = AC97_PCM_FRONT_DAC_RATE;
return snd_soc_write(codec, reg, runtime->rate);
}
static int stac9766_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
switch (level) {
case SND_SOC_BIAS_ON: /* full On */
case SND_SOC_BIAS_PREPARE: /* partial On */
case SND_SOC_BIAS_STANDBY: /* Off, with power */
snd_soc_write(codec, AC97_POWERDOWN, 0x0000);
break;
case SND_SOC_BIAS_OFF: /* Off, without power */
/* disable everything including AC link */
snd_soc_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
return 0;
}
static int stac9766_codec_resume(struct snd_soc_codec *codec)
{
struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
return snd_ac97_reset(ac97, true, STAC9766_VENDOR_ID,
STAC9766_VENDOR_ID_MASK);
}
static const struct snd_soc_dai_ops stac9766_dai_ops_analog = {
.prepare = ac97_analog_prepare,
};
static const struct snd_soc_dai_ops stac9766_dai_ops_digital = {
.prepare = ac97_digital_prepare,
};
static struct snd_soc_dai_driver stac9766_dai[] = {
{
.name = "stac9766-hifi-analog",
/* stream cababilities */
.playback = {
.stream_name = "stac9766 analog",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SND_SOC_STD_AC97_FMTS,
},
.capture = {
.stream_name = "stac9766 analog",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SND_SOC_STD_AC97_FMTS,
},
/* alsa ops */
.ops = &stac9766_dai_ops_analog,
},
{
.name = "stac9766-hifi-IEC958",
/* stream cababilities */
.playback = {
.stream_name = "stac9766 IEC958",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_32000 | \
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
},
/* alsa ops */
.ops = &stac9766_dai_ops_digital,
}
};
static int stac9766_codec_probe(struct snd_soc_codec *codec)
{
struct snd_ac97 *ac97;
struct regmap *regmap;
int ret;
ac97 = snd_soc_new_ac97_codec(codec, STAC9766_VENDOR_ID,
STAC9766_VENDOR_ID_MASK);
if (IS_ERR(ac97))
return PTR_ERR(ac97);
regmap = regmap_init_ac97(ac97, &stac9766_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);
return 0;
err_free_ac97:
snd_soc_free_ac97_codec(ac97);
return ret;
}
static int stac9766_codec_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_stac9766 = {
.component_driver = {
.controls = stac9766_snd_ac97_controls,
.num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls),
},
.set_bias_level = stac9766_set_bias_level,
.suspend_bias_off = true,
.probe = stac9766_codec_probe,
.remove = stac9766_codec_remove,
.resume = stac9766_codec_resume,
};
static int stac9766_probe(struct platform_device *pdev)
{
return snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_stac9766, stac9766_dai, ARRAY_SIZE(stac9766_dai));
}
static int stac9766_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static struct platform_driver stac9766_codec_driver = {
.driver = {
.name = "stac9766-codec",
},
.probe = stac9766_probe,
.remove = stac9766_remove,
};
module_platform_driver(stac9766_codec_driver);
MODULE_DESCRIPTION("ASoC stac9766 driver");
MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
MODULE_LICENSE("GPL");
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