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linux_dsm_epyc7002/sound/soc/codecs/ssm2602.c
Bhumika Goyal a180ba45b1 ASoC: codecs: add const to snd_soc_codec_driver structures 
Declare snd_soc_codec_driver structures as const as they are only passed
as an argument to the function snd_soc_register_codec. This argument is
of type const, so declare the structures with this property as const.
In file codecs/sn95031.c, snd_soc_codec_driver structure is also used in
a copy operation along with getting passed to snd_soc_register_codec.
So, it can be made const too.
Done using Coccinelle:

@match disable optional_qualifier@
identifier s;
position p;
@@
static struct snd_soc_codec_driver s@p={...};

@good1@
identifier match.s;
position p;
@@
snd_soc_register_codec(...,&s@p,...)

@bad@
identifier match.s;
position p!={match.p,good1.p};
@@
s@p

@depends on !bad disable optional_qualifier@
identifier match.s;
@@
static
+const
struct snd_soc_codec_driver s={...};

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-10 16:10:50 +01:00

652 lines
17 KiB
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/*
* File: sound/soc/codecs/ssm2602.c
* Author: Cliff Cai <Cliff.Cai@analog.com>
*
* Created: Tue June 06 2008
* Description: Driver for ssm2602 sound chip
*
* Modified:
* Copyright 2008 Analog Devices Inc.
*
* Bugs: Enter bugs at http://blackfin.uclinux.org/
*
* 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.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, see the file COPYING, or write
* to the Free Software Foundation, Inc.,
* 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <linux/module.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 "ssm2602.h"
/* codec private data */
struct ssm2602_priv {
unsigned int sysclk;
const struct snd_pcm_hw_constraint_list *sysclk_constraints;
struct regmap *regmap;
enum ssm2602_type type;
unsigned int clk_out_pwr;
};
/*
* ssm2602 register cache
* We can't read the ssm2602 register space when we are
* using 2 wire for device control, so we cache them instead.
* There is no point in caching the reset register
*/
static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = {
0x0097, 0x0097, 0x0079, 0x0079,
0x000a, 0x0008, 0x009f, 0x000a,
0x0000, 0x0000
};
/*Appending several "None"s just for OSS mixer use*/
static const char *ssm2602_input_select[] = {
"Line", "Mic",
};
static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
static const struct soc_enum ssm2602_enum[] = {
SOC_ENUM_SINGLE(SSM2602_APANA, 2, ARRAY_SIZE(ssm2602_input_select),
ssm2602_input_select),
SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, ARRAY_SIZE(ssm2602_deemph),
ssm2602_deemph),
};
static const DECLARE_TLV_DB_RANGE(ssm260x_outmix_tlv,
0, 47, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0),
48, 127, TLV_DB_SCALE_ITEM(-7400, 100, 0)
);
static const DECLARE_TLV_DB_SCALE(ssm260x_inpga_tlv, -3450, 150, 0);
static const DECLARE_TLV_DB_SCALE(ssm260x_sidetone_tlv, -1500, 300, 0);
static const struct snd_kcontrol_new ssm260x_snd_controls[] = {
SOC_DOUBLE_R_TLV("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 45, 0,
ssm260x_inpga_tlv),
SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1),
SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1),
SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0),
SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]),
};
static const struct snd_kcontrol_new ssm2602_snd_controls[] = {
SOC_DOUBLE_R_TLV("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V,
0, 127, 0, ssm260x_outmix_tlv),
SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V,
7, 1, 0),
SOC_SINGLE_TLV("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1,
ssm260x_sidetone_tlv),
SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0),
SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0),
SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1),
};
/* Output Mixer */
static const struct snd_kcontrol_new ssm260x_output_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0),
SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0),
SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0),
};
/* Input mux */
static const struct snd_kcontrol_new ssm2602_input_mux_controls =
SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]);
static const struct snd_soc_dapm_widget ssm260x_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1),
SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1),
SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LOUT"),
SND_SOC_DAPM_OUTPUT("ROUT"),
SND_SOC_DAPM_INPUT("RLINEIN"),
SND_SOC_DAPM_INPUT("LLINEIN"),
};
static const struct snd_soc_dapm_widget ssm2602_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1,
ssm260x_output_mixer_controls,
ARRAY_SIZE(ssm260x_output_mixer_controls)),
SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls),
SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1),
SND_SOC_DAPM_OUTPUT("LHPOUT"),
SND_SOC_DAPM_OUTPUT("RHPOUT"),
SND_SOC_DAPM_INPUT("MICIN"),
};
static const struct snd_soc_dapm_widget ssm2604_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0,
ssm260x_output_mixer_controls,
ARRAY_SIZE(ssm260x_output_mixer_controls) - 1), /* Last element is the mic */
};
static const struct snd_soc_dapm_route ssm260x_routes[] = {
{"DAC", NULL, "Digital Core Power"},
{"ADC", NULL, "Digital Core Power"},
{"Output Mixer", "Line Bypass Switch", "Line Input"},
{"Output Mixer", "HiFi Playback Switch", "DAC"},
{"ROUT", NULL, "Output Mixer"},
{"LOUT", NULL, "Output Mixer"},
{"Line Input", NULL, "LLINEIN"},
{"Line Input", NULL, "RLINEIN"},
};
static const struct snd_soc_dapm_route ssm2602_routes[] = {
{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
{"RHPOUT", NULL, "Output Mixer"},
{"LHPOUT", NULL, "Output Mixer"},
{"Input Mux", "Line", "Line Input"},
{"Input Mux", "Mic", "Mic Bias"},
{"ADC", NULL, "Input Mux"},
{"Mic Bias", NULL, "MICIN"},
};
static const struct snd_soc_dapm_route ssm2604_routes[] = {
{"ADC", NULL, "Line Input"},
};
static const unsigned int ssm2602_rates_12288000[] = {
8000, 16000, 32000, 48000, 96000,
};
static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
.list = ssm2602_rates_12288000,
.count = ARRAY_SIZE(ssm2602_rates_12288000),
};
static const unsigned int ssm2602_rates_11289600[] = {
8000, 11025, 22050, 44100, 88200,
};
static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = {
.list = ssm2602_rates_11289600,
.count = ARRAY_SIZE(ssm2602_rates_11289600),
};
struct ssm2602_coeff {
u32 mclk;
u32 rate;
u8 srate;
};
#define SSM2602_COEFF_SRATE(sr, bosr, usb) (((sr) << 2) | ((bosr) << 1) | (usb))
/* codec mclk clock coefficients */
static const struct ssm2602_coeff ssm2602_coeff_table[] = {
/* 48k */
{12288000, 48000, SSM2602_COEFF_SRATE(0x0, 0x0, 0x0)},
{18432000, 48000, SSM2602_COEFF_SRATE(0x0, 0x1, 0x0)},
{12000000, 48000, SSM2602_COEFF_SRATE(0x0, 0x0, 0x1)},
/* 32k */
{12288000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x0)},
{18432000, 32000, SSM2602_COEFF_SRATE(0x6, 0x1, 0x0)},
{12000000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x1)},
/* 16k */
{12288000, 16000, SSM2602_COEFF_SRATE(0x5, 0x0, 0x0)},
{18432000, 16000, SSM2602_COEFF_SRATE(0x5, 0x1, 0x0)},
{12000000, 16000, SSM2602_COEFF_SRATE(0xa, 0x0, 0x1)},
/* 8k */
{12288000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x0)},
{18432000, 8000, SSM2602_COEFF_SRATE(0x3, 0x1, 0x0)},
{11289600, 8000, SSM2602_COEFF_SRATE(0xb, 0x0, 0x0)},
{16934400, 8000, SSM2602_COEFF_SRATE(0xb, 0x1, 0x0)},
{12000000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x1)},
/* 96k */
{12288000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x0)},
{18432000, 96000, SSM2602_COEFF_SRATE(0x7, 0x1, 0x0)},
{12000000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x1)},
/* 11.025k */
{11289600, 11025, SSM2602_COEFF_SRATE(0xc, 0x0, 0x0)},
{16934400, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x0)},
{12000000, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x1)},
/* 22.05k */
{11289600, 22050, SSM2602_COEFF_SRATE(0xd, 0x0, 0x0)},
{16934400, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x0)},
{12000000, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x1)},
/* 44.1k */
{11289600, 44100, SSM2602_COEFF_SRATE(0x8, 0x0, 0x0)},
{16934400, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x0)},
{12000000, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x1)},
/* 88.2k */
{11289600, 88200, SSM2602_COEFF_SRATE(0xf, 0x0, 0x0)},
{16934400, 88200, SSM2602_COEFF_SRATE(0xf, 0x1, 0x0)},
{12000000, 88200, SSM2602_COEFF_SRATE(0xf, 0x1, 0x1)},
};
static inline int ssm2602_get_coeff(int mclk, int rate)
{
int i;
for (i = 0; i < ARRAY_SIZE(ssm2602_coeff_table); i++) {
if (ssm2602_coeff_table[i].rate == rate &&
ssm2602_coeff_table[i].mclk == mclk)
return ssm2602_coeff_table[i].srate;
}
return -EINVAL;
}
static int ssm2602_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 ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params));
unsigned int iface;
if (srate < 0)
return srate;
regmap_write(ssm2602->regmap, SSM2602_SRATE, srate);
/* bit size */
switch (params_width(params)) {
case 16:
iface = 0x0;
break;
case 20:
iface = 0x4;
break;
case 24:
iface = 0x8;
break;
case 32:
iface = 0xc;
break;
default:
return -EINVAL;
}
regmap_update_bits(ssm2602->regmap, SSM2602_IFACE,
IFACE_AUDIO_DATA_LEN, iface);
return 0;
}
static int ssm2602_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
if (ssm2602->sysclk_constraints) {
snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
ssm2602->sysclk_constraints);
}
return 0;
}
static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
{
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec);
if (mute)
regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
APDIGI_ENABLE_DAC_MUTE,
APDIGI_ENABLE_DAC_MUTE);
else
regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
APDIGI_ENABLE_DAC_MUTE, 0);
return 0;
}
static int ssm2602_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 ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
if (dir == SND_SOC_CLOCK_IN) {
if (clk_id != SSM2602_SYSCLK)
return -EINVAL;
switch (freq) {
case 12288000:
case 18432000:
ssm2602->sysclk_constraints = &ssm2602_constraints_12288000;
break;
case 11289600:
case 16934400:
ssm2602->sysclk_constraints = &ssm2602_constraints_11289600;
break;
case 12000000:
ssm2602->sysclk_constraints = NULL;
break;
default:
return -EINVAL;
}
ssm2602->sysclk = freq;
} else {
unsigned int mask;
switch (clk_id) {
case SSM2602_CLK_CLKOUT:
mask = PWR_CLK_OUT_PDN;
break;
case SSM2602_CLK_XTO:
mask = PWR_OSC_PDN;
break;
default:
return -EINVAL;
}
if (freq == 0)
ssm2602->clk_out_pwr |= mask;
else
ssm2602->clk_out_pwr &= ~mask;
regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr);
}
return 0;
}
static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec_dai->codec);
unsigned int iface = 0;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
iface |= 0x0040;
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:
iface |= 0x0002;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
iface |= 0x0001;
break;
case SND_SOC_DAIFMT_DSP_A:
iface |= 0x0013;
break;
case SND_SOC_DAIFMT_DSP_B:
iface |= 0x0003;
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:
iface |= 0x0090;
break;
case SND_SOC_DAIFMT_IB_NF:
iface |= 0x0080;
break;
case SND_SOC_DAIFMT_NB_IF:
iface |= 0x0010;
break;
default:
return -EINVAL;
}
/* set iface */
regmap_write(ssm2602->regmap, SSM2602_IFACE, iface);
return 0;
}
static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
switch (level) {
case SND_SOC_BIAS_ON:
/* vref/mid on, osc and clkout on if enabled */
regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
ssm2602->clk_out_pwr);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
PWR_CLK_OUT_PDN | PWR_OSC_PDN);
break;
case SND_SOC_BIAS_OFF:
/* everything off */
regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
PWR_POWER_OFF, PWR_POWER_OFF);
break;
}
return 0;
}
#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
SNDRV_PCM_RATE_96000)
#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
static const struct snd_soc_dai_ops ssm2602_dai_ops = {
.startup = ssm2602_startup,
.hw_params = ssm2602_hw_params,
.digital_mute = ssm2602_mute,
.set_sysclk = ssm2602_set_dai_sysclk,
.set_fmt = ssm2602_set_dai_fmt,
};
static struct snd_soc_dai_driver ssm2602_dai = {
.name = "ssm2602-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = SSM2602_RATES,
.formats = SSM2602_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SSM2602_RATES,
.formats = SSM2602_FORMATS,},
.ops = &ssm2602_dai_ops,
.symmetric_rates = 1,
.symmetric_samplebits = 1,
};
static int ssm2602_resume(struct snd_soc_codec *codec)
{
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
regcache_sync(ssm2602->regmap);
return 0;
}
static int ssm2602_codec_probe(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
int ret;
regmap_update_bits(ssm2602->regmap, SSM2602_LOUT1V,
LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH);
regmap_update_bits(ssm2602->regmap, SSM2602_ROUT1V,
ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH);
ret = snd_soc_add_codec_controls(codec, ssm2602_snd_controls,
ARRAY_SIZE(ssm2602_snd_controls));
if (ret)
return ret;
ret = snd_soc_dapm_new_controls(dapm, ssm2602_dapm_widgets,
ARRAY_SIZE(ssm2602_dapm_widgets));
if (ret)
return ret;
return snd_soc_dapm_add_routes(dapm, ssm2602_routes,
ARRAY_SIZE(ssm2602_routes));
}
static int ssm2604_codec_probe(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
int ret;
ret = snd_soc_dapm_new_controls(dapm, ssm2604_dapm_widgets,
ARRAY_SIZE(ssm2604_dapm_widgets));
if (ret)
return ret;
return snd_soc_dapm_add_routes(dapm, ssm2604_routes,
ARRAY_SIZE(ssm2604_routes));
}
static int ssm260x_codec_probe(struct snd_soc_codec *codec)
{
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
int ret;
ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
return ret;
}
/* set the update bits */
regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL,
LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);
regmap_update_bits(ssm2602->regmap, SSM2602_RINVOL,
RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH);
/*select Line in as default input*/
regmap_write(ssm2602->regmap, SSM2602_APANA, APANA_SELECT_DAC |
APANA_ENABLE_MIC_BOOST);
switch (ssm2602->type) {
case SSM2602:
ret = ssm2602_codec_probe(codec);
break;
case SSM2604:
ret = ssm2604_codec_probe(codec);
break;
}
return ret;
}
static const struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
.probe = ssm260x_codec_probe,
.resume = ssm2602_resume,
.set_bias_level = ssm2602_set_bias_level,
.suspend_bias_off = true,
.component_driver = {
.controls = ssm260x_snd_controls,
.num_controls = ARRAY_SIZE(ssm260x_snd_controls),
.dapm_widgets = ssm260x_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ssm260x_dapm_widgets),
.dapm_routes = ssm260x_routes,
.num_dapm_routes = ARRAY_SIZE(ssm260x_routes),
},
};
static bool ssm2602_register_volatile(struct device *dev, unsigned int reg)
{
return reg == SSM2602_RESET;
}
const struct regmap_config ssm2602_regmap_config = {
.val_bits = 9,
.reg_bits = 7,
.max_register = SSM2602_RESET,
.volatile_reg = ssm2602_register_volatile,
.cache_type = REGCACHE_RBTREE,
.reg_defaults_raw = ssm2602_reg,
.num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg),
};
EXPORT_SYMBOL_GPL(ssm2602_regmap_config);
int ssm2602_probe(struct device *dev, enum ssm2602_type type,
struct regmap *regmap)
{
struct ssm2602_priv *ssm2602;
if (IS_ERR(regmap))
return PTR_ERR(regmap);
ssm2602 = devm_kzalloc(dev, sizeof(*ssm2602), GFP_KERNEL);
if (ssm2602 == NULL)
return -ENOMEM;
dev_set_drvdata(dev, ssm2602);
ssm2602->type = type;
ssm2602->regmap = regmap;
return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602,
&ssm2602_dai, 1);
}
EXPORT_SYMBOL_GPL(ssm2602_probe);
MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 driver");
MODULE_AUTHOR("Cliff Cai");
MODULE_LICENSE("GPL");
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