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WSL2-Linux-Kernel/sound/soc/codecs/tlv320aic23.c

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16 KiB
C
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/*
* ALSA SoC TLV320AIC23 codec driver
*
* Author: Arun KS, <arunks@mistralsolutions.com>
* Copyright: (C) 2008 Mistral Solutions Pvt Ltd.,
*
* Based on sound/soc/codecs/wm8731.c by Richard Purdie
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
* Notes:
* The AIC23 is a driver for a low power stereo audio
* codec tlv320aic23
*
* The machine layer should disable unsupported inputs/outputs by
* snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#include <sound/initval.h>
#include "tlv320aic23.h"
/*
* AIC23 register cache
*/
static const struct reg_default tlv320aic23_reg[] = {
{ 0, 0x0097 },
{ 1, 0x0097 },
{ 2, 0x00F9 },
{ 3, 0x00F9 },
{ 4, 0x001A },
{ 5, 0x0004 },
{ 6, 0x0007 },
{ 7, 0x0001 },
{ 8, 0x0020 },
{ 9, 0x0000 },
};
const struct regmap_config tlv320aic23_regmap = {
.reg_bits = 7,
.val_bits = 9,
.max_register = TLV320AIC23_RESET,
.reg_defaults = tlv320aic23_reg,
.num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg),
.cache_type = REGCACHE_RBTREE,
};
EXPORT_SYMBOL(tlv320aic23_regmap);
static const char *rec_src_text[] = { "Line", "Mic" };
static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
static SOC_ENUM_SINGLE_DECL(rec_src_enum,
TLV320AIC23_ANLG, 2, rec_src_text);
static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
SOC_DAPM_ENUM("Input Select", rec_src_enum);
static SOC_ENUM_SINGLE_DECL(tlv320aic23_rec_src,
TLV320AIC23_ANLG, 2, rec_src_text);
static SOC_ENUM_SINGLE_DECL(tlv320aic23_deemph,
TLV320AIC23_DIGT, 1, deemph_text);
static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0);
static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
u16 val, reg;
val = (ucontrol->value.integer.value[0] & 0x07);
/* linear conversion to userspace
* 000 = -6db
* 001 = -9db
* 010 = -12db
* 011 = -18db (Min)
* 100 = 0db (Max)
*/
val = (val >= 4) ? 4 : (3 - val);
reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0);
snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
return 0;
}
static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
u16 val;
val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0);
val = val >> 6;
val = (val >= 4) ? 4 : (3 - val);
ucontrol->value.integer.value[0] = val;
return 0;
}
static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1),
SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL,
TLV320AIC23_RINVOL, 7, 1, 0),
SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL,
TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
SOC_SINGLE_EXT_TLV("Sidetone Volume", TLV320AIC23_ANLG, 6, 4, 0,
snd_soc_tlv320aic23_get_volsw,
snd_soc_tlv320aic23_put_volsw, sidetone_vol_tlv),
SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
};
/* PGA Mixer controls for Line and Mic switch */
static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0),
SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0),
};
static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1),
SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1),
SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
&tlv320aic23_rec_src_mux_controls),
SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1,
&tlv320aic23_output_mixer_controls[0],
ARRAY_SIZE(tlv320aic23_output_mixer_controls)),
SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0),
SND_SOC_DAPM_OUTPUT("LHPOUT"),
SND_SOC_DAPM_OUTPUT("RHPOUT"),
SND_SOC_DAPM_OUTPUT("LOUT"),
SND_SOC_DAPM_OUTPUT("ROUT"),
SND_SOC_DAPM_INPUT("LLINEIN"),
SND_SOC_DAPM_INPUT("RLINEIN"),
SND_SOC_DAPM_INPUT("MICIN"),
};
static const struct snd_soc_dapm_route tlv320aic23_intercon[] = {
/* Output Mixer */
{"Output Mixer", "Line Bypass Switch", "Line Input"},
{"Output Mixer", "Playback Switch", "DAC"},
{"Output Mixer", "Mic Sidetone Switch", "Mic Input"},
/* Outputs */
{"RHPOUT", NULL, "Output Mixer"},
{"LHPOUT", NULL, "Output Mixer"},
{"LOUT", NULL, "Output Mixer"},
{"ROUT", NULL, "Output Mixer"},
/* Inputs */
{"Line Input", "NULL", "LLINEIN"},
{"Line Input", "NULL", "RLINEIN"},
{"Mic Input", "NULL", "MICIN"},
/* input mux */
{"Capture Source", "Line", "Line Input"},
{"Capture Source", "Mic", "Mic Input"},
{"ADC", NULL, "Capture Source"},
};
/* AIC23 driver data */
struct aic23 {
struct regmap *regmap;
int mclk;
int requested_adc;
int requested_dac;
};
/*
* Common Crystals used
* 11.2896 Mhz /128 = *88.2k /192 = 58.8k
* 12.0000 Mhz /125 = *96k /136 = 88.235K
* 12.2880 Mhz /128 = *96k /192 = 64k
* 16.9344 Mhz /128 = 132.3k /192 = *88.2k
* 18.4320 Mhz /128 = 144k /192 = *96k
*/
/*
* Normal BOSR 0-256/2 = 128, 1-384/2 = 192
* USB BOSR 0-250/2 = 125, 1-272/2 = 136
*/
static const int bosr_usb_divisor_table[] = {
128, 125, 192, 136
};
#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7))
#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15))
static const unsigned short sr_valid_mask[] = {
LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/
LOWER_GROUP, /* Usb, bosr - 0*/
LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
UPPER_GROUP, /* Usb, bosr - 1*/
};
/*
* Every divisor is a factor of 11*12
*/
#define SR_MULT (11*12)
#define A(x) (SR_MULT/x)
static const unsigned char sr_adc_mult_table[] = {
A(2), A(2), A(12), A(12), 0, 0, A(3), A(1),
A(2), A(2), A(11), A(11), 0, 0, 0, A(1)
};
static const unsigned char sr_dac_mult_table[] = {
A(2), A(12), A(2), A(12), 0, 0, A(3), A(1),
A(2), A(11), A(2), A(11), 0, 0, 0, A(1)
};
static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
int dac, int dac_l, int dac_h, int need_dac)
{
if ((adc >= adc_l) && (adc <= adc_h) &&
(dac >= dac_l) && (dac <= dac_h)) {
int diff_adc = need_adc - adc;
int diff_dac = need_dac - dac;
return abs(diff_adc) + abs(diff_dac);
}
return UINT_MAX;
}
static int find_rate(int mclk, u32 need_adc, u32 need_dac)
{
int i, j;
int best_i = -1;
int best_j = -1;
int best_div = 0;
unsigned best_score = UINT_MAX;
int adc_l, adc_h, dac_l, dac_h;
need_adc *= SR_MULT;
need_dac *= SR_MULT;
/*
* rates given are +/- 1/32
*/
adc_l = need_adc - (need_adc >> 5);
adc_h = need_adc + (need_adc >> 5);
dac_l = need_dac - (need_dac >> 5);
dac_h = need_dac + (need_dac >> 5);
for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) {
int base = mclk / bosr_usb_divisor_table[i];
int mask = sr_valid_mask[i];
for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table);
j++, mask >>= 1) {
int adc;
int dac;
int score;
if ((mask & 1) == 0)
continue;
adc = base * sr_adc_mult_table[j];
dac = base * sr_dac_mult_table[j];
score = get_score(adc, adc_l, adc_h, need_adc,
dac, dac_l, dac_h, need_dac);
if (best_score > score) {
best_score = score;
best_i = i;
best_j = j;
best_div = 0;
}
score = get_score((adc >> 1), adc_l, adc_h, need_adc,
(dac >> 1), dac_l, dac_h, need_dac);
/* prefer to have a /2 */
if ((score != UINT_MAX) && (best_score >= score)) {
best_score = score;
best_i = i;
best_j = j;
best_div = 1;
}
}
}
return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT);
}
#ifdef DEBUG
static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
u32 *sample_rate_adc, u32 *sample_rate_dac)
{
int src = snd_soc_read(codec, TLV320AIC23_SRATE);
int sr = (src >> 2) & 0x0f;
int val = (mclk / bosr_usb_divisor_table[src & 3]);
int adc = (val * sr_adc_mult_table[sr]) / SR_MULT;
int dac = (val * sr_dac_mult_table[sr]) / SR_MULT;
if (src & TLV320AIC23_CLKIN_HALF) {
adc >>= 1;
dac >>= 1;
}
*sample_rate_adc = adc;
*sample_rate_dac = dac;
}
#endif
static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
u32 sample_rate_adc, u32 sample_rate_dac)
{
/* Search for the right sample rate */
int data = find_rate(mclk, sample_rate_adc, sample_rate_dac);
if (data < 0) {
printk(KERN_ERR "%s:Invalid rate %u,%u requested\n",
__func__, sample_rate_adc, sample_rate_dac);
return -EINVAL;
}
snd_soc_write(codec, TLV320AIC23_SRATE, data);
#ifdef DEBUG
{
u32 adc, dac;
get_current_sample_rates(codec, mclk, &adc, &dac);
printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n",
adc, dac, data);
}
#endif
return 0;
}
static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
u16 iface_reg;
int ret;
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
u32 sample_rate_adc = aic23->requested_adc;
u32 sample_rate_dac = aic23->requested_dac;
u32 sample_rate = params_rate(params);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
aic23->requested_dac = sample_rate_dac = sample_rate;
if (!sample_rate_adc)
sample_rate_adc = sample_rate;
} else {
aic23->requested_adc = sample_rate_adc = sample_rate;
if (!sample_rate_dac)
sample_rate_dac = sample_rate;
}
ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc,
sample_rate_dac);
if (ret < 0)
return ret;
iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
switch (params_width(params)) {
case 16:
break;
case 20:
iface_reg |= (0x01 << 2);
break;
case 24:
iface_reg |= (0x02 << 2);
break;
case 32:
iface_reg |= (0x03 << 2);
break;
}
snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
return 0;
}
static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
/* set active */
snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001);
return 0;
}
static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
/* deactivate */
if (!snd_soc_codec_is_active(codec)) {
udelay(50);
snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
aic23->requested_dac = 0;
else
aic23->requested_adc = 0;
}
static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 reg;
reg = snd_soc_read(codec, TLV320AIC23_DIGT);
if (mute)
reg |= TLV320AIC23_DACM_MUTE;
else
reg &= ~TLV320AIC23_DACM_MUTE;
snd_soc_write(codec, TLV320AIC23_DIGT, reg);
return 0;
}
static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 iface_reg;
iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
iface_reg |= TLV320AIC23_MS_MASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
iface_reg &= ~TLV320AIC23_MS_MASTER;
break;
default:
return -EINVAL;
}
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
iface_reg |= TLV320AIC23_FOR_I2S;
break;
case SND_SOC_DAIFMT_DSP_A:
iface_reg |= TLV320AIC23_LRP_ON;
case SND_SOC_DAIFMT_DSP_B:
iface_reg |= TLV320AIC23_FOR_DSP;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
iface_reg |= TLV320AIC23_FOR_LJUST;
break;
default:
return -EINVAL;
}
snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
return 0;
}
static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct aic23 *aic23 = snd_soc_dai_get_drvdata(codec_dai);
aic23->mclk = freq;
return 0;
}
static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f;
switch (level) {
case SND_SOC_BIAS_ON:
/* vref/mid, osc on, dac unmute */
reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \
TLV320AIC23_DAC_OFF);
snd_soc_write(codec, TLV320AIC23_PWR, reg);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
snd_soc_write(codec, TLV320AIC23_PWR,
reg | TLV320AIC23_CLK_OFF);
break;
case SND_SOC_BIAS_OFF:
/* everything off, dac mute, inactive */
snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff);
break;
}
return 0;
}
#define AIC23_RATES SNDRV_PCM_RATE_8000_96000
#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
static const struct snd_soc_dai_ops tlv320aic23_dai_ops = {
.prepare = tlv320aic23_pcm_prepare,
.hw_params = tlv320aic23_hw_params,
.shutdown = tlv320aic23_shutdown,
.digital_mute = tlv320aic23_mute,
.set_fmt = tlv320aic23_set_dai_fmt,
.set_sysclk = tlv320aic23_set_dai_sysclk,
};
static struct snd_soc_dai_driver tlv320aic23_dai = {
.name = "tlv320aic23-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = AIC23_RATES,
.formats = AIC23_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = AIC23_RATES,
.formats = AIC23_FORMATS,},
.ops = &tlv320aic23_dai_ops,
};
static int tlv320aic23_resume(struct snd_soc_codec *codec)
{
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
regcache_mark_dirty(aic23->regmap);
regcache_sync(aic23->regmap);
return 0;
}
static int tlv320aic23_codec_probe(struct snd_soc_codec *codec)
{
/* Reset codec */
snd_soc_write(codec, TLV320AIC23_RESET, 0);
snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
/* Unmute input */
snd_soc_update_bits(codec, TLV320AIC23_LINVOL,
TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED);
snd_soc_update_bits(codec, TLV320AIC23_RINVOL,
TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED);
snd_soc_update_bits(codec, TLV320AIC23_ANLG,
TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED,
0);
/* Default output volume */
snd_soc_write(codec, TLV320AIC23_LCHNVOL,
TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK);
snd_soc_write(codec, TLV320AIC23_RCHNVOL,
TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK);
snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
.probe = tlv320aic23_codec_probe,
.resume = tlv320aic23_resume,
.set_bias_level = tlv320aic23_set_bias_level,
.suspend_bias_off = true,
.controls = tlv320aic23_snd_controls,
.num_controls = ARRAY_SIZE(tlv320aic23_snd_controls),
.dapm_widgets = tlv320aic23_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
.dapm_routes = tlv320aic23_intercon,
.num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon),
};
int tlv320aic23_probe(struct device *dev, struct regmap *regmap)
{
struct aic23 *aic23;
if (IS_ERR(regmap))
return PTR_ERR(regmap);
aic23 = devm_kzalloc(dev, sizeof(struct aic23), GFP_KERNEL);
if (aic23 == NULL)
return -ENOMEM;
aic23->regmap = regmap;
dev_set_drvdata(dev, aic23);
return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23,
&tlv320aic23_dai, 1);
}
EXPORT_SYMBOL(tlv320aic23_probe);
MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
MODULE_LICENSE("GPL");
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