提交 c1f27190 编写于 作者: A Arun KS 提交者: Takashi Iwai

ALSA: ASoC: Add TLV320AIC23 codec driver

ASoC codec driver for TLV320AIC23 device
Signed-off-by: NArun KS <arunks@mistralsolutions.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
上级 869fbb36
......@@ -7,6 +7,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_AK4535
select SND_SOC_CS4270
select SND_SOC_SSM2602
select SND_SOC_TLV320AIC23
select SND_SOC_TLV320AIC26
select SND_SOC_TLV320AIC3X
select SND_SOC_UDA1380
......@@ -62,6 +63,10 @@ config SND_SOC_CS4270_VD33_ERRATA
config SND_SOC_SSM2602
tristate
config SND_SOC_TLV320AIC23
tristate
depends on I2C
config SND_SOC_TLV320AIC26
tristate "TI TLV320AIC26 Codec support"
depends on SND_SOC && SPI
......
......@@ -4,6 +4,7 @@ snd-soc-ad73311-objs := ad73311.o
snd-soc-ak4535-objs := ak4535.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-ssm2602-objs := ssm2602.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-uda1380-objs := uda1380.o
......@@ -25,6 +26,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
......
/*
* 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/i2c.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
#include <sound/initval.h>
#include "tlv320aic23.h"
#define AUDIO_NAME "tlv320aic23"
#define AIC23_VERSION "0.1"
struct tlv320aic23_srate_reg_info {
u32 sample_rate;
u8 control; /* SR3, SR2, SR1, SR0 and BOSR */
u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
};
/*
* AIC23 register cache
*/
static const u16 tlv320aic23_reg[] = {
0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */
0x001A, 0x0004, 0x0007, 0x0001, /* 4 */
0x0020, 0x0000, 0x0000, 0x0000, /* 8 */
0x0000, 0x0000, 0x0000, 0x0000, /* 12 */
};
/*
* read tlv320aic23 register cache
*/
static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec
*codec, unsigned int reg)
{
u16 *cache = codec->reg_cache;
if (reg >= ARRAY_SIZE(tlv320aic23_reg))
return -1;
return cache[reg];
}
/*
* write tlv320aic23 register cache
*/
static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec,
u8 reg, u16 value)
{
u16 *cache = codec->reg_cache;
if (reg >= ARRAY_SIZE(tlv320aic23_reg))
return;
cache[reg] = value;
}
/*
* write to the tlv320aic23 register space
*/
static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
u8 data;
/* TLV320AIC23 has 7 bit address and 9 bits of data
* so we need to switch one data bit into reg and rest
* of data into val
*/
if ((reg < 0 || reg > 9) && (reg != 15)) {
printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
return -1;
}
data = (reg << 1) | (value >> 8 & 0x01);
tlv320aic23_write_reg_cache(codec, reg, value);
if (codec->hw_write(codec->control_data, data,
(value & 0xff)) == 0)
return 0;
printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
value, reg);
return -EIO;
}
static const char *rec_src_text[] = { "Line", "Mic" };
static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
static const char *sidetone_text[] = {"-6db", "-9db", "-12db", "-18db", "0db"};
static const struct soc_enum rec_src_enum =
SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
SOC_DAPM_ENUM("Input Select", rec_src_enum);
static const struct soc_enum tlv320aic23_rec_src =
SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
static const struct soc_enum tlv320aic23_deemph =
SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text);
static const struct soc_enum tlv320aic23_sidetone =
SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 6, 5, sidetone_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 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_ENUM("Sidetone Gain", tlv320aic23_sidetone),
SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
};
/* add non dapm controls */
static int tlv320aic23_add_controls(struct snd_soc_codec *codec)
{
int err, i;
for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) {
err = snd_ctl_add(codec->card,
snd_soc_cnew(&tlv320aic23_snd_controls[i],
codec, NULL));
if (err < 0)
return err;
}
return 0;
}
/* 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 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"},
};
/* tlv320aic23 related */
static const struct tlv320aic23_srate_reg_info srate_reg_info[] = {
{4000, 0x06, 1}, /* 4000 */
{8000, 0x06, 0}, /* 8000 */
{16000, 0x0C, 1}, /* 16000 */
{22050, 0x11, 1}, /* 22050 */
{24000, 0x00, 1}, /* 24000 */
{32000, 0x0C, 0}, /* 32000 */
{44100, 0x11, 0}, /* 44100 */
{48000, 0x00, 0}, /* 48000 */
{88200, 0x1F, 0}, /* 88200 */
{96000, 0x0E, 0}, /* 96000 */
};
static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* set up audio path interconnects */
snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
snd_soc_dapm_new_widgets(codec);
return 0;
}
static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
u16 iface_reg, data;
u8 count = 0;
iface_reg =
tlv320aic23_read_reg_cache(codec,
TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
/* Search for the right sample rate */
/* Verify what happens if the rate is not supported
* now it goes to 96Khz */
while ((srate_reg_info[count].sample_rate != params_rate(params)) &&
(count < ARRAY_SIZE(srate_reg_info))) {
count++;
}
data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) |
(srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) |
TLV320AIC23_USB_CLK_ON;
tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
iface_reg |= (0x01 << 2);
break;
case SNDRV_PCM_FORMAT_S24_LE:
iface_reg |= (0x02 << 2);
break;
case SNDRV_PCM_FORMAT_S32_LE:
iface_reg |= (0x03 << 2);
break;
}
tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
return 0;
}
static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
/* set active */
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
return 0;
}
static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
/* deactivate */
if (!codec->active) {
udelay(50);
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
}
}
static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 reg;
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT);
if (mute)
reg |= TLV320AIC23_DACM_MUTE;
else
reg &= ~TLV320AIC23_DACM_MUTE;
tlv320aic23_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 =
tlv320aic23_read_reg_cache(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:
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_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;
}
tlv320aic23_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 snd_soc_codec *codec = codec_dai->codec;
switch (freq) {
case 12000000:
return 0;
}
return -EINVAL;
}
static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f;
switch (level) {
case SND_SOC_BIAS_ON:
/* vref/mid, osc on, dac unmute */
tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040);
break;
case SND_SOC_BIAS_OFF:
/* everything off, dac mute, inactive */
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
break;
}
codec->bias_level = level;
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)
struct snd_soc_dai tlv320aic23_dai = {
.name = "tlv320aic23",
.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 = {
.prepare = tlv320aic23_pcm_prepare,
.hw_params = tlv320aic23_hw_params,
.shutdown = tlv320aic23_shutdown,
},
.dai_ops = {
.digital_mute = tlv320aic23_mute,
.set_fmt = tlv320aic23_set_dai_fmt,
.set_sysclk = tlv320aic23_set_dai_sysclk,
}
};
EXPORT_SYMBOL_GPL(tlv320aic23_dai);
static int tlv320aic23_suspend(struct platform_device *pdev,
pm_message_t state)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int tlv320aic23_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
int i;
u16 reg;
/* Sync reg_cache with the hardware */
for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
u16 val = tlv320aic23_read_reg_cache(codec, reg);
tlv320aic23_write(codec, reg, val);
}
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
tlv320aic23_set_bias_level(codec, codec->suspend_bias_level);
return 0;
}
/*
* initialise the AIC23 driver
* register the mixer and dsp interfaces with the kernel
*/
static int tlv320aic23_init(struct snd_soc_device *socdev)
{
struct snd_soc_codec *codec = socdev->codec;
int ret = 0;
u16 reg;
codec->name = "tlv320aic23";
codec->owner = THIS_MODULE;
codec->read = tlv320aic23_read_reg_cache;
codec->write = tlv320aic23_write;
codec->set_bias_level = tlv320aic23_set_bias_level;
codec->dai = &tlv320aic23_dai;
codec->num_dai = 1;
codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg);
codec->reg_cache =
kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL);
if (codec->reg_cache == NULL)
return -ENOMEM;
/* Reset codec */
tlv320aic23_write(codec, TLV320AIC23_RESET, 0);
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
printk(KERN_ERR "tlv320aic23: failed to create pcms\n");
goto pcm_err;
}
/* power on device */
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
/* Unmute input */
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL);
tlv320aic23_write(codec, TLV320AIC23_LINVOL,
(reg & (~TLV320AIC23_LIM_MUTED)) |
(TLV320AIC23_LRS_ENABLED));
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL);
tlv320aic23_write(codec, TLV320AIC23_RINVOL,
(reg & (~TLV320AIC23_LIM_MUTED)) |
TLV320AIC23_LRS_ENABLED);
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG);
tlv320aic23_write(codec, TLV320AIC23_ANLG,
(reg) & (~TLV320AIC23_BYPASS_ON) &
(~TLV320AIC23_MICM_MUTED));
/* Default output volume */
tlv320aic23_write(codec, TLV320AIC23_LCHNVOL,
TLV320AIC23_DEFAULT_OUT_VOL &
TLV320AIC23_OUT_VOL_MASK);
tlv320aic23_write(codec, TLV320AIC23_RCHNVOL,
TLV320AIC23_DEFAULT_OUT_VOL &
TLV320AIC23_OUT_VOL_MASK);
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
tlv320aic23_add_controls(codec);
tlv320aic23_add_widgets(codec);
ret = snd_soc_register_card(socdev);
if (ret < 0) {
printk(KERN_ERR "tlv320aic23: failed to register card\n");
goto card_err;
}
return ret;
card_err:
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
pcm_err:
kfree(codec->reg_cache);
return ret;
}
static struct snd_soc_device *tlv320aic23_socdev;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
/*
* If the i2c layer weren't so broken, we could pass this kind of data
* around
*/
static int tlv320aic23_codec_probe(struct i2c_client *i2c,
const struct i2c_device_id *i2c_id)
{
struct snd_soc_device *socdev = tlv320aic23_socdev;
struct snd_soc_codec *codec = socdev->codec;
int ret;
if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
return -EINVAL;
i2c_set_clientdata(i2c, codec);
codec->control_data = i2c;
ret = tlv320aic23_init(socdev);
if (ret < 0) {
printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n");
goto err;
}
return ret;
err:
kfree(codec);
kfree(i2c);
return ret;
}
static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
{
put_device(&i2c->dev);
return 0;
}
static const struct i2c_device_id tlv320aic23_id[] = {
{"tlv320aic23", 0},
{}
};
MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
static struct i2c_driver tlv320aic23_i2c_driver = {
.driver = {
.name = "tlv320aic23",
},
.probe = tlv320aic23_codec_probe,
.remove = __exit_p(tlv320aic23_i2c_remove),
.id_table = tlv320aic23_id,
};
#endif
static int tlv320aic23_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec;
int ret = 0;
printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (codec == NULL)
return -ENOMEM;
socdev->codec = codec;
mutex_init(&codec->mutex);
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
tlv320aic23_socdev = socdev;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data;
codec->hw_read = NULL;
ret = i2c_add_driver(&tlv320aic23_i2c_driver);
if (ret != 0)
printk(KERN_ERR "can't add i2c driver");
#endif
return ret;
}
static int tlv320aic23_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
if (codec->control_data)
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&tlv320aic23_i2c_driver);
#endif
kfree(codec->reg_cache);
kfree(codec);
return 0;
}
struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
.probe = tlv320aic23_probe,
.remove = tlv320aic23_remove,
.suspend = tlv320aic23_suspend,
.resume = tlv320aic23_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
MODULE_LICENSE("GPL");
/*
* ALSA SoC TLV320AIC23 codec driver
*
* Author: Arun KS, <arunks@mistralsolutions.com>
* Copyright: (C) 2008 Mistral Solutions Pvt Ltd
*
* 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.
*/
#ifndef _TLV320AIC23_H
#define _TLV320AIC23_H
/* Codec TLV320AIC23 */
#define TLV320AIC23_LINVOL 0x00
#define TLV320AIC23_RINVOL 0x01
#define TLV320AIC23_LCHNVOL 0x02
#define TLV320AIC23_RCHNVOL 0x03
#define TLV320AIC23_ANLG 0x04
#define TLV320AIC23_DIGT 0x05
#define TLV320AIC23_PWR 0x06
#define TLV320AIC23_DIGT_FMT 0x07
#define TLV320AIC23_SRATE 0x08
#define TLV320AIC23_ACTIVE 0x09
#define TLV320AIC23_RESET 0x0F
/* Left (right) line input volume control register */
#define TLV320AIC23_LRS_ENABLED 0x0100
#define TLV320AIC23_LIM_MUTED 0x0080
#define TLV320AIC23_LIV_DEFAULT 0x0017
#define TLV320AIC23_LIV_MAX 0x001f
#define TLV320AIC23_LIV_MIN 0x0000
/* Left (right) channel headphone volume control register */
#define TLV320AIC23_LZC_ON 0x0080
#define TLV320AIC23_LHV_DEFAULT 0x0079
#define TLV320AIC23_LHV_MAX 0x007f
#define TLV320AIC23_LHV_MIN 0x0000
/* Analog audio path control register */
#define TLV320AIC23_STA_REG(x) ((x)<<6)
#define TLV320AIC23_STE_ENABLED 0x0020
#define TLV320AIC23_DAC_SELECTED 0x0010
#define TLV320AIC23_BYPASS_ON 0x0008
#define TLV320AIC23_INSEL_MIC 0x0004
#define TLV320AIC23_MICM_MUTED 0x0002
#define TLV320AIC23_MICB_20DB 0x0001
/* Digital audio path control register */
#define TLV320AIC23_DACM_MUTE 0x0008
#define TLV320AIC23_DEEMP_32K 0x0002
#define TLV320AIC23_DEEMP_44K 0x0004
#define TLV320AIC23_DEEMP_48K 0x0006
#define TLV320AIC23_ADCHP_ON 0x0001
/* Power control down register */
#define TLV320AIC23_DEVICE_PWR_OFF 0x0080
#define TLV320AIC23_CLK_OFF 0x0040
#define TLV320AIC23_OSC_OFF 0x0020
#define TLV320AIC23_OUT_OFF 0x0010
#define TLV320AIC23_DAC_OFF 0x0008
#define TLV320AIC23_ADC_OFF 0x0004
#define TLV320AIC23_MIC_OFF 0x0002
#define TLV320AIC23_LINE_OFF 0x0001
/* Digital audio interface register */
#define TLV320AIC23_MS_MASTER 0x0040
#define TLV320AIC23_LRSWAP_ON 0x0020
#define TLV320AIC23_LRP_ON 0x0010
#define TLV320AIC23_IWL_16 0x0000
#define TLV320AIC23_IWL_20 0x0004
#define TLV320AIC23_IWL_24 0x0008
#define TLV320AIC23_IWL_32 0x000C
#define TLV320AIC23_FOR_I2S 0x0002
#define TLV320AIC23_FOR_DSP 0x0003
#define TLV320AIC23_FOR_LJUST 0x0001
/* Sample rate control register */
#define TLV320AIC23_CLKOUT_HALF 0x0080
#define TLV320AIC23_CLKIN_HALF 0x0040
#define TLV320AIC23_BOSR_384fs 0x0002 /* BOSR_272fs in USB mode */
#define TLV320AIC23_USB_CLK_ON 0x0001
#define TLV320AIC23_SR_MASK 0xf
#define TLV320AIC23_CLKOUT_SHIFT 7
#define TLV320AIC23_CLKIN_SHIFT 6
#define TLV320AIC23_SR_SHIFT 2
#define TLV320AIC23_BOSR_SHIFT 1
/* Digital interface register */
#define TLV320AIC23_ACT_ON 0x0001
/*
* AUDIO related MACROS
*/
#define TLV320AIC23_DEFAULT_OUT_VOL 0x70
#define TLV320AIC23_DEFAULT_IN_VOLUME 0x10
#define TLV320AIC23_OUT_VOL_MIN TLV320AIC23_LHV_MIN
#define TLV320AIC23_OUT_VOL_MAX TLV320AIC23_LHV_MAX
#define TLV320AIC23_OUT_VO_RANGE (TLV320AIC23_OUT_VOL_MAX - \
TLV320AIC23_OUT_VOL_MIN)
#define TLV320AIC23_OUT_VOL_MASK TLV320AIC23_OUT_VOL_MAX
#define TLV320AIC23_IN_VOL_MIN TLV320AIC23_LIV_MIN
#define TLV320AIC23_IN_VOL_MAX TLV320AIC23_LIV_MAX
#define TLV320AIC23_IN_VOL_RANGE (TLV320AIC23_IN_VOL_MAX - \
TLV320AIC23_IN_VOL_MIN)
#define TLV320AIC23_IN_VOL_MASK TLV320AIC23_IN_VOL_MAX
#define TLV320AIC23_SIDETONE_MASK 0x1c0
#define TLV320AIC23_SIDETONE_0 0x100
#define TLV320AIC23_SIDETONE_6 0x000
#define TLV320AIC23_SIDETONE_9 0x040
#define TLV320AIC23_SIDETONE_12 0x080
#define TLV320AIC23_SIDETONE_18 0x0c0
extern struct snd_soc_dai tlv320aic23_dai;
extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23;
#endif /* _TLV320AIC23_H */
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