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linux/sound/soc/codecs/ssm2602.c
Lars-Peter Clausen 0289053526 ASoC: ssm2602: Support setting the oscillator and the clock output state 
Currently the oscillator is always enabled and the clock output is always
disabled. This patch adds support for controlling the oscillator and clock
output state through snd_soc_dai_set_sysclk. Which makes it possible to
disable or enable them dynamically according to the requirements of the board
on which the CODEC is used.

This patch also slightly modifies the behavior as to when the oscillator is
going to be disabled in low-power states. Previously it would only be disabled
in BIAS_OFF, now it is also going to be disabled in BIAS_STANDBY, since no
components which depend on it should be active in this state.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 13:30:48 +01:00

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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/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/spi/spi.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include "ssm2602.h"
#define SSM2602_VERSION "0.1"
enum ssm2602_type {
SSM2602,
SSM2604,
};
/* codec private data */
struct ssm2602_priv {
unsigned int sysclk;
enum snd_soc_control_type control_type;
struct snd_pcm_substream *master_substream;
struct snd_pcm_substream *slave_substream;
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
};
#define ssm2602_reset(c) snd_soc_write(c, SSM2602_RESET, 0)
/*Appending several "None"s just for OSS mixer use*/
static const char *ssm2602_input_select[] = {
"Line", "Mic", "None", "None", "None",
"None", "None", "None",
};
static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
static const struct soc_enum ssm2602_enum[] = {
SOC_ENUM_SINGLE(SSM2602_APANA, 2, 2, ssm2602_input_select),
SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph),
};
static const unsigned int ssm260x_outmix_tlv[] = {
TLV_DB_RANGE_HEAD(2),
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"},
};
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)},
/* 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)},
/* 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_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
u16 iface = snd_soc_read(codec, SSM2602_IFACE) & 0xfff3;
int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params));
if (substream == ssm2602->slave_substream) {
dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n");
return 0;
}
if (srate < 0)
return srate;
snd_soc_write(codec, SSM2602_SRATE, srate);
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
iface |= 0x0004;
break;
case SNDRV_PCM_FORMAT_S24_LE:
iface |= 0x0008;
break;
case SNDRV_PCM_FORMAT_S32_LE:
iface |= 0x000c;
break;
}
snd_soc_write(codec, SSM2602_IFACE, iface);
return 0;
}
static int ssm2602_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
struct snd_pcm_runtime *master_runtime;
/* The DAI has shared clocks so if we already have a playback or
* capture going then constrain this substream to match it.
* TODO: the ssm2602 allows pairs of non-matching PB/REC rates
*/
if (ssm2602->master_substream) {
master_runtime = ssm2602->master_substream->runtime;
dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n",
master_runtime->sample_bits,
master_runtime->rate);
if (master_runtime->rate != 0)
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
master_runtime->rate,
master_runtime->rate);
if (master_runtime->sample_bits != 0)
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
master_runtime->sample_bits,
master_runtime->sample_bits);
ssm2602->slave_substream = substream;
} else
ssm2602->master_substream = substream;
return 0;
}
static void ssm2602_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
if (ssm2602->master_substream == substream)
ssm2602->master_substream = ssm2602->slave_substream;
ssm2602->slave_substream = NULL;
}
static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 mute_reg = snd_soc_read(codec, SSM2602_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE;
if (mute)
snd_soc_write(codec, SSM2602_APDIGI,
mute_reg | APDIGI_ENABLE_DAC_MUTE);
else
snd_soc_write(codec, SSM2602_APDIGI, mute_reg);
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 11289600:
case 12000000:
case 12288000:
case 16934400:
case 18432000:
ssm2602->sysclk = freq;
break;
default:
return -EINVAL;
}
} 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;
snd_soc_update_bits(codec, 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 snd_soc_codec *codec = codec_dai->codec;
u16 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 */
snd_soc_write(codec, 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 */
snd_soc_update_bits(codec, 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, */
snd_soc_update_bits(codec, 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 */
snd_soc_update_bits(codec, SSM2602_PWR,
PWR_POWER_OFF, PWR_POWER_OFF);
break;
}
codec->dapm.bias_level = level;
return 0;
}
#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | 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 struct snd_soc_dai_ops ssm2602_dai_ops = {
.startup = ssm2602_startup,
.hw_params = ssm2602_hw_params,
.shutdown = ssm2602_shutdown,
.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,
};
static int ssm2602_suspend(struct snd_soc_codec *codec, pm_message_t state)
{
ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int ssm2602_resume(struct snd_soc_codec *codec)
{
snd_soc_cache_sync(codec);
ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
static int ssm2602_probe(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret, reg;
reg = snd_soc_read(codec, SSM2602_LOUT1V);
snd_soc_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH);
reg = snd_soc_read(codec, SSM2602_ROUT1V);
snd_soc_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH);
ret = snd_soc_add_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_probe(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
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_probe(struct snd_soc_codec *codec)
{
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
int ret, reg;
pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION);
ret = snd_soc_codec_set_cache_io(codec, 7, 9, ssm2602->control_type);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
ret = ssm2602_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
return ret;
}
/* set the update bits */
reg = snd_soc_read(codec, SSM2602_LINVOL);
snd_soc_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH);
reg = snd_soc_read(codec, SSM2602_RINVOL);
snd_soc_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH);
/*select Line in as default input*/
snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC |
APANA_ENABLE_MIC_BOOST);
switch (ssm2602->type) {
case SSM2602:
ret = ssm2602_probe(codec);
break;
case SSM2604:
ret = ssm2604_probe(codec);
break;
}
if (ret)
return ret;
ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
/* remove everything here */
static int ssm2602_remove(struct snd_soc_codec *codec)
{
ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
.probe = ssm260x_probe,
.remove = ssm2602_remove,
.suspend = ssm2602_suspend,
.resume = ssm2602_resume,
.set_bias_level = ssm2602_set_bias_level,
.reg_cache_size = ARRAY_SIZE(ssm2602_reg),
.reg_word_size = sizeof(u16),
.reg_cache_default = ssm2602_reg,
.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),
};
#if defined(CONFIG_SPI_MASTER)
static int __devinit ssm2602_spi_probe(struct spi_device *spi)
{
struct ssm2602_priv *ssm2602;
int ret;
ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL);
if (ssm2602 == NULL)
return -ENOMEM;
spi_set_drvdata(spi, ssm2602);
ssm2602->control_type = SND_SOC_SPI;
ssm2602->type = SSM2602;
ret = snd_soc_register_codec(&spi->dev,
&soc_codec_dev_ssm2602, &ssm2602_dai, 1);
if (ret < 0)
kfree(ssm2602);
return ret;
}
static int __devexit ssm2602_spi_remove(struct spi_device *spi)
{
snd_soc_unregister_codec(&spi->dev);
kfree(spi_get_drvdata(spi));
return 0;
}
static struct spi_driver ssm2602_spi_driver = {
.driver = {
.name = "ssm2602",
.owner = THIS_MODULE,
},
.probe = ssm2602_spi_probe,
.remove = __devexit_p(ssm2602_spi_remove),
};
#endif
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
/*
* ssm2602 2 wire address is determined by GPIO5
* state during powerup.
* low = 0x1a
* high = 0x1b
*/
static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct ssm2602_priv *ssm2602;
int ret;
ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL);
if (ssm2602 == NULL)
return -ENOMEM;
i2c_set_clientdata(i2c, ssm2602);
ssm2602->control_type = SND_SOC_I2C;
ssm2602->type = id->driver_data;
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_ssm2602, &ssm2602_dai, 1);
if (ret < 0)
kfree(ssm2602);
return ret;
}
static int __devexit ssm2602_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
kfree(i2c_get_clientdata(client));
return 0;
}
static const struct i2c_device_id ssm2602_i2c_id[] = {
{ "ssm2602", SSM2602 },
{ "ssm2603", SSM2602 },
{ "ssm2604", SSM2604 },
{ }
};
MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id);
/* corgi i2c codec control layer */
static struct i2c_driver ssm2602_i2c_driver = {
.driver = {
.name = "ssm2602",
.owner = THIS_MODULE,
},
.probe = ssm2602_i2c_probe,
.remove = __devexit_p(ssm2602_i2c_remove),
.id_table = ssm2602_i2c_id,
};
#endif
static int __init ssm2602_modinit(void)
{
int ret = 0;
#if defined(CONFIG_SPI_MASTER)
ret = spi_register_driver(&ssm2602_spi_driver);
if (ret)
return ret;
#endif
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&ssm2602_i2c_driver);
if (ret)
return ret;
#endif
return ret;
}
module_init(ssm2602_modinit);
static void __exit ssm2602_exit(void)
{
#if defined(CONFIG_SPI_MASTER)
spi_unregister_driver(&ssm2602_spi_driver);
#endif
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&ssm2602_i2c_driver);
#endif
}
module_exit(ssm2602_exit);
MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 driver");
MODULE_AUTHOR("Cliff Cai");
MODULE_LICENSE("GPL");
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