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linux/sound/soc/codecs/tlv320aic23.c
Kuninori Morimoto e88e495064 ASoC: codec duplicated callback function goes to component on tlv320aic23 
codec driver and component driver has duplicated callback functions,
and codec side functions are just copied to component side when
register timing. This was quick-hack, but no longer needed.
This patch moves these functions from codec driver to component driver.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2016-08-08 11:57:57 +01:00

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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,
.component_driver = {
.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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