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/*
* max9850.c -- codec driver for max9850
*
* Copyright (C) 2011 taskit GmbH
*
* Author: Christian Glindkamp <christian.glindkamp@taskit.de>
*
* Initial development of this code was funded by
* MICRONIC Computer Systeme GmbH, http://www.mcsberlin.de/
*
* 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.
*
*/
#include <linux/module.h>
#include <linux/init.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#include "max9850.h"
struct max9850_priv {
struct regmap *regmap;
unsigned int sysclk;
};
/* max9850 register cache */
static const struct reg_default max9850_reg[] = {
{ 2, 0x0c },
{ 3, 0x00 },
{ 4, 0x00 },
{ 5, 0x00 },
{ 6, 0x00 },
{ 7, 0x00 },
{ 8, 0x00 },
{ 9, 0x00 },
{ 10, 0x00 },
};
/* these registers are not used at the moment but provided for the sake of
* completeness */
static bool max9850_volatile_register(struct device *dev, unsigned int reg)
{
switch (reg) {
case MAX9850_STATUSA:
case MAX9850_STATUSB:
return 1;
default:
return 0;
}
}
static const struct regmap_config max9850_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = MAX9850_DIGITAL_AUDIO,
.volatile_reg = max9850_volatile_register,
.cache_type = REGCACHE_RBTREE,
};
static const unsigned int max9850_tlv[] = {
TLV_DB_RANGE_HEAD(4),
0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0),
0x20, 0x33, TLV_DB_SCALE_ITEM(-4150, 200, 0),
0x34, 0x37, TLV_DB_SCALE_ITEM(-150, 100, 0),
0x38, 0x3f, TLV_DB_SCALE_ITEM(250, 50, 0),
};
static const struct snd_kcontrol_new max9850_controls[] = {
SOC_SINGLE_TLV("Headphone Volume", MAX9850_VOLUME, 0, 0x3f, 1, max9850_tlv),
SOC_SINGLE("Headphone Switch", MAX9850_VOLUME, 7, 1, 1),
SOC_SINGLE("Mono Switch", MAX9850_GENERAL_PURPOSE, 2, 1, 0),
};
static const struct snd_kcontrol_new max9850_mixer_controls[] = {
SOC_DAPM_SINGLE("Line In Switch", MAX9850_ENABLE, 1, 1, 0),
};
static const struct snd_soc_dapm_widget max9850_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("Charge Pump 1", MAX9850_ENABLE, 4, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Charge Pump 2", MAX9850_ENABLE, 5, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("MCLK", MAX9850_ENABLE, 6, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("SHDN", MAX9850_ENABLE, 7, 0, NULL, 0),
SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", MAX9850_ENABLE, 2, 0,
&max9850_mixer_controls[0],
ARRAY_SIZE(max9850_mixer_controls)),
SND_SOC_DAPM_PGA("Headphone Output", MAX9850_ENABLE, 3, 0, NULL, 0),
SND_SOC_DAPM_DAC("DAC", "HiFi Playback", MAX9850_ENABLE, 0, 0),
SND_SOC_DAPM_OUTPUT("OUTL"),
SND_SOC_DAPM_OUTPUT("HPL"),
SND_SOC_DAPM_OUTPUT("OUTR"),
SND_SOC_DAPM_OUTPUT("HPR"),
SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_INPUT("INL"),
SND_SOC_DAPM_INPUT("INR"),
};
static const struct snd_soc_dapm_route max9850_dapm_routes[] = {
/* output mixer */
{"Output Mixer", NULL, "DAC"},
{"Output Mixer", "Line In Switch", "Line Input"},
/* outputs */
{"Headphone Output", NULL, "Output Mixer"},
{"HPL", NULL, "Headphone Output"},
{"HPR", NULL, "Headphone Output"},
{"OUTL", NULL, "Output Mixer"},
{"OUTR", NULL, "Output Mixer"},
/* inputs */
{"Line Input", NULL, "INL"},
{"Line Input", NULL, "INR"},
/* supplies */
{"Output Mixer", NULL, "Charge Pump 1"},
{"Output Mixer", NULL, "Charge Pump 2"},
{"Output Mixer", NULL, "SHDN"},
{"DAC", NULL, "MCLK"},
};
static int max9850_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;
struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec);
u64 lrclk_div;
u8 sf, da;
if (!max9850->sysclk)
return -EINVAL;
/* lrclk_div = 2^22 * rate / iclk with iclk = mclk / sf */
sf = (snd_soc_read(codec, MAX9850_CLOCK) >> 2) + 1;
lrclk_div = (1 << 22);
lrclk_div *= params_rate(params);
lrclk_div *= sf;
do_div(lrclk_div, max9850->sysclk);
snd_soc_write(codec, MAX9850_LRCLK_MSB, (lrclk_div >> 8) & 0x7f);
snd_soc_write(codec, MAX9850_LRCLK_LSB, lrclk_div & 0xff);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
da = 0;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
da = 0x2;
break;
case SNDRV_PCM_FORMAT_S24_LE:
da = 0x3;
break;
default:
return -EINVAL;
}
snd_soc_update_bits(codec, MAX9850_DIGITAL_AUDIO, 0x3, da);
return 0;
}
static int max9850_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec);
/* calculate mclk -> iclk divider */
if (freq <= 13000000)
snd_soc_write(codec, MAX9850_CLOCK, 0x0);
else if (freq <= 26000000)
snd_soc_write(codec, MAX9850_CLOCK, 0x4);
else if (freq <= 40000000)
snd_soc_write(codec, MAX9850_CLOCK, 0x8);
else
return -EINVAL;
max9850->sysclk = freq;
return 0;
}
static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
u8 da = 0;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
da |= MAX9850_MASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
return -EINVAL;
}
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
da |= MAX9850_DLY;
break;
case SND_SOC_DAIFMT_RIGHT_J:
da |= MAX9850_RTJ;
break;
case SND_SOC_DAIFMT_LEFT_J:
break;
default:
return -EINVAL;
}
/* clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_IF:
da |= MAX9850_BCINV | MAX9850_INV;
break;
case SND_SOC_DAIFMT_IB_NF:
da |= MAX9850_BCINV;
break;
case SND_SOC_DAIFMT_NB_IF:
da |= MAX9850_INV;
break;
default:
return -EINVAL;
}
/* set da */
snd_soc_write(codec, MAX9850_DIGITAL_AUDIO, da);
return 0;
}
static int max9850_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec);
int ret;
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
ret
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