sound: ASoC codec: SSM2602 audio codec driver

[Some checkpatch fixups done by Mark Brown.]

Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This commit is contained in:
Cliff Cai 2008-09-05 18:09:57 +08:00 committed by Jaroslav Kysela
parent 41759c2eab
commit b7138212a8
4 changed files with 912 additions and 0 deletions

View file

@ -4,6 +4,7 @@ config SND_SOC_ALL_CODECS
select SPI
select SPI_MASTER
select SND_SOC_AK4535
select SND_SOC_SSM2602
select SND_SOC_UDA1380
select SND_SOC_WM8510
select SND_SOC_WM8580
@ -93,3 +94,6 @@ config SND_SOC_TLV320AIC26
config SND_SOC_TLV320AIC3X
tristate
depends on I2C
config SND_SOC_SSM2602
tristate

View file

@ -15,6 +15,7 @@ snd-soc-wm9713-objs := wm9713.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-ssm2602-objs := ssm2602.o
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
@ -33,3 +34,4 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o

776
sound/soc/codecs/ssm2602.c Normal file
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@ -0,0 +1,776 @@
/*
* 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/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/initval.h>
#include "ssm2602.h"
#define AUDIO_NAME "ssm2602"
#define SSM2602_VERSION "0.1"
struct snd_soc_codec_device soc_codec_dev_ssm2602;
/* codec private data */
struct ssm2602_priv {
unsigned int sysclk;
struct snd_pcm_substream *master_substream;
struct snd_pcm_substream *slave_substream;
};
/*
* 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] = {
0x0017, 0x0017, 0x0079, 0x0079,
0x0000, 0x0000, 0x0000, 0x000a,
0x0000, 0x0000
};
/*
* read ssm2602 register cache
*/
static inline unsigned int ssm2602_read_reg_cache(struct snd_soc_codec *codec,
unsigned int reg)
{
u16 *cache = codec->reg_cache;
if (reg == SSM2602_RESET)
return 0;
if (reg >= SSM2602_CACHEREGNUM)
return -1;
return cache[reg];
}
/*
* write ssm2602 register cache
*/
static inline void ssm2602_write_reg_cache(struct snd_soc_codec *codec,
u16 reg, unsigned int value)
{
u16 *cache = codec->reg_cache;
if (reg >= SSM2602_CACHEREGNUM)
return;
cache[reg] = value;
}
/*
* write to the ssm2602 register space
*/
static int ssm2602_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
u8 data[2];
/* data is
* D15..D9 ssm2602 register offset
* D8...D0 register data
*/
data[0] = (reg << 1) | ((value >> 8) & 0x0001);
data[1] = value & 0x00ff;
ssm2602_write_reg_cache(codec, reg, value);
if (codec->hw_write(codec->control_data, data, 2) == 2)
return 0;
else
return -EIO;
}
#define ssm2602_reset(c) ssm2602_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 struct snd_kcontrol_new ssm2602_snd_controls[] = {
SOC_DOUBLE_R("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V,
0, 127, 0),
SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V,
7, 1, 0),
SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0),
SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1),
SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0),
SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1),
SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 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("Capture Source", ssm2602_enum[0]),
SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]),
};
/* add non dapm controls */
static int ssm2602_add_controls(struct snd_soc_codec *codec)
{
int err, i;
for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) {
err = snd_ctl_add(codec->card,
snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL));
if (err < 0)
return err;
}
return 0;
}
/* Output Mixer */
static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0),
SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0),
SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 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 ssm2602_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1,
&ssm2602_output_mixer_controls[0],
ARRAY_SIZE(ssm2602_output_mixer_controls)),
SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1),
SND_SOC_DAPM_OUTPUT("LOUT"),
SND_SOC_DAPM_OUTPUT("LHPOUT"),
SND_SOC_DAPM_OUTPUT("ROUT"),
SND_SOC_DAPM_OUTPUT("RHPOUT"),
SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1),
SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls),
SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1),
SND_SOC_DAPM_INPUT("MICIN"),
SND_SOC_DAPM_INPUT("RLINEIN"),
SND_SOC_DAPM_INPUT("LLINEIN"),
};
static const struct snd_soc_dapm_route audio_conn[] = {
/* output mixer */
{"Output Mixer", "Line Bypass Switch", "Line Input"},
{"Output Mixer", "HiFi Playback Switch", "DAC"},
{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
/* outputs */
{"RHPOUT", NULL, "Output Mixer"},
{"ROUT", NULL, "Output Mixer"},
{"LHPOUT", NULL, "Output Mixer"},
{"LOUT", NULL, "Output Mixer"},
/* input mux */
{"Input Mux", "Line", "Line Input"},
{"Input Mux", "Mic", "Mic Bias"},
{"ADC", NULL, "Input Mux"},
/* inputs */
{"Line Input", NULL, "LLINEIN"},
{"Line Input", NULL, "RLINEIN"},
{"Mic Bias", NULL, "MICIN"},
};
static int ssm2602_add_widgets(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, ssm2602_dapm_widgets,
ARRAY_SIZE(ssm2602_dapm_widgets));
snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn));
snd_soc_dapm_new_widgets(codec);
return 0;
}
struct _coeff_div {
u32 mclk;
u32 rate;
u16 fs;
u8 sr:4;
u8 bosr:1;
u8 usb:1;
};
/* codec mclk clock divider coefficients */
static const struct _coeff_div coeff_div[] = {
/* 48k */
{12288000, 48000, 256, 0x0, 0x0, 0x0},
{18432000, 48000, 384, 0x0, 0x1, 0x0},
{12000000, 48000, 250, 0x0, 0x0, 0x1},
/* 32k */
{12288000, 32000, 384, 0x6, 0x0, 0x0},
{18432000, 32000, 576, 0x6, 0x1, 0x0},
{12000000, 32000, 375, 0x6, 0x0, 0x1},
/* 8k */
{12288000, 8000, 1536, 0x3, 0x0, 0x0},
{18432000, 8000, 2304, 0x3, 0x1, 0x0},
{11289600, 8000, 1408, 0xb, 0x0, 0x0},
{16934400, 8000, 2112, 0xb, 0x1, 0x0},
{12000000, 8000, 1500, 0x3, 0x0, 0x1},
/* 96k */
{12288000, 96000, 128, 0x7, 0x0, 0x0},
{18432000, 96000, 192, 0x7, 0x1, 0x0},
{12000000, 96000, 125, 0x7, 0x0, 0x1},
/* 44.1k */
{11289600, 44100, 256, 0x8, 0x0, 0x0},
{16934400, 44100, 384, 0x8, 0x1, 0x0},
{12000000, 44100, 272, 0x8, 0x1, 0x1},
/* 88.2k */
{11289600, 88200, 128, 0xf, 0x0, 0x0},
{16934400, 88200, 192, 0xf, 0x1, 0x0},
{12000000, 88200, 136, 0xf, 0x1, 0x1},
};
static inline int get_coeff(int mclk, int rate)
{
int i;
for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
return i;
}
return i;
}
static int ssm2602_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
u16 srate;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
struct ssm2602_priv *ssm2602 = codec->private_data;
u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3;
int i = get_coeff(ssm2602->sysclk, params_rate(params));
/*no match is found*/
if (i == ARRAY_SIZE(coeff_div))
return -EINVAL;
srate = (coeff_div[i].sr << 2) |
(coeff_div[i].bosr << 1) | coeff_div[i].usb;
ssm2602_write(codec, SSM2602_ACTIVE, 0);
ssm2602_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;
}
ssm2602_write(codec, SSM2602_IFACE, iface);
ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC);
return 0;
}
static int ssm2602_startup(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;
struct ssm2602_priv *ssm2602 = codec->private_data;
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.
*/
if (ssm2602->master_substream) {
master_runtime = ssm2602->master_substream->runtime;
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE,
master_runtime->rate,
master_runtime->rate);
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 int ssm2602_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 */
ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC);
return 0;
}
static void ssm2602_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)
ssm2602_write(codec, SSM2602_ACTIVE, 0);
}
static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 mute_reg = ssm2602_read_reg_cache(codec, SSM2602_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE;
if (mute)
ssm2602_write(codec, SSM2602_APDIGI,
mute_reg | APDIGI_ENABLE_DAC_MUTE);
else
ssm2602_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 = codec->private_data;
switch (freq) {
case 11289600:
case 12000000:
case 12288000:
case 16934400:
case 18432000:
ssm2602->sysclk = freq;
return 0;
}
return -EINVAL;
}
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 |= 0x0003;
break;
case SND_SOC_DAIFMT_DSP_B:
iface |= 0x0013;
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 */
ssm2602_write(codec, SSM2602_IFACE, iface);
return 0;
}
static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
u16 reg = ssm2602_read_reg_cache(codec, SSM2602_PWR) & 0xff7f;
switch (level) {
case SND_SOC_BIAS_ON:
/* vref/mid, osc on, dac unmute */
ssm2602_write(codec, SSM2602_PWR, reg);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
ssm2602_write(codec, SSM2602_PWR, reg | PWR_CLK_OUT_PDN);
break;
case SND_SOC_BIAS_OFF:
/* everything off, dac mute, inactive */
ssm2602_write(codec, SSM2602_ACTIVE, 0);
ssm2602_write(codec, SSM2602_PWR, 0xffff);
break;
}
codec->bias_level = level;
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)
struct snd_soc_dai ssm2602_dai = {
.name = "SSM2602",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = SSM2602_RATES,
.formats = SNDRV_PCM_FMTBIT_S32_LE,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SSM2602_RATES,
.formats = SNDRV_PCM_FMTBIT_S32_LE,},
.ops = {
.startup = ssm2602_startup,
.prepare = ssm2602_pcm_prepare,
.hw_params = ssm2602_hw_params,
.shutdown = ssm2602_shutdown,
},
.dai_ops = {
.digital_mute = ssm2602_mute,
.set_sysclk = ssm2602_set_dai_sysclk,
.set_fmt = ssm2602_set_dai_fmt,
}
};
EXPORT_SYMBOL_GPL(ssm2602_dai);
static int ssm2602_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;
ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int ssm2602_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
int i;
u8 data[2];
u16 *cache = codec->reg_cache;
/* Sync reg_cache with the hardware */
for (i = 0; i < ARRAY_SIZE(ssm2602_reg); i++) {
data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
data[1] = cache[i] & 0x00ff;
codec->hw_write(codec->control_data, data, 2);
}
ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
ssm2602_set_bias_level(codec, codec->suspend_bias_level);
return 0;
}
/*
* initialise the ssm2602 driver
* register the mixer and dsp interfaces with the kernel
*/
static int ssm2602_init(struct snd_soc_device *socdev)
{
struct snd_soc_codec *codec = socdev->codec;
int reg, ret = 0;
codec->name = "SSM2602";
codec->owner = THIS_MODULE;
codec->read = ssm2602_read_reg_cache;
codec->write = ssm2602_write;
codec->set_bias_level = ssm2602_set_bias_level;
codec->dai = &ssm2602_dai;
codec->num_dai = 1;
codec->reg_cache_size = sizeof(ssm2602_reg);
codec->reg_cache = kmemdup(ssm2602_reg, sizeof(ssm2602_reg),
GFP_KERNEL);
if (codec->reg_cache == NULL)
return -ENOMEM;
ssm2602_reset(codec);
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
pr_err("ssm2602: failed to create pcms\n");
goto pcm_err;
}
/*power on device*/
ssm2602_write(codec, SSM2602_ACTIVE, 0);
/* set the update bits */
reg = ssm2602_read_reg_cache(codec, SSM2602_LINVOL);
ssm2602_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH);
reg = ssm2602_read_reg_cache(codec, SSM2602_RINVOL);
ssm2602_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH);
reg = ssm2602_read_reg_cache(codec, SSM2602_LOUT1V);
ssm2602_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH);
reg = ssm2602_read_reg_cache(codec, SSM2602_ROUT1V);
ssm2602_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH);
/*select Line in as default input*/
ssm2602_write(codec, SSM2602_APANA,
APANA_ENABLE_MIC_BOOST2 | APANA_SELECT_DAC |
APANA_ENABLE_MIC_BOOST);
ssm2602_write(codec, SSM2602_PWR, 0);
ssm2602_add_controls(codec);
ssm2602_add_widgets(codec);
ret = snd_soc_register_card(socdev);
if (ret < 0) {
pr_err("ssm2602: 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 *ssm2602_socdev;
#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 ssm2602_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct snd_soc_device *socdev = ssm2602_socdev;
struct snd_soc_codec *codec = socdev->codec;
int ret;
i2c_set_clientdata(i2c, codec);
codec->control_data = i2c;
ret = ssm2602_init(socdev);
if (ret < 0)
pr_err("failed to initialise SSM2602\n");
return ret;
}
static int ssm2602_i2c_remove(struct i2c_client *client)
{
struct snd_soc_codec *codec = i2c_get_clientdata(client);
kfree(codec->reg_cache);
return 0;
}
static const struct i2c_device_id ssm2602_i2c_id[] = {
{ "ssm2602", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id);
/* corgi i2c codec control layer */
static struct i2c_driver ssm2602_i2c_driver = {
.driver = {
.name = "SSM2602 I2C Codec",
.owner = THIS_MODULE,
},
.probe = ssm2602_i2c_probe,
.remove = ssm2602_i2c_remove,
.id_table = ssm2602_i2c_id,
};
static int ssm2602_add_i2c_device(struct platform_device *pdev,
const struct ssm2602_setup_data *setup)
{
struct i2c_board_info info;
struct i2c_adapter *adapter;
struct i2c_client *client;
int ret;
ret = i2c_add_driver(&ssm2602_i2c_driver);
if (ret != 0) {
dev_err(&pdev->dev, "can't add i2c driver\n");
return ret;
}
memset(&info, 0, sizeof(struct i2c_board_info));
info.addr = setup->i2c_address;
strlcpy(info.type, "ssm2602", I2C_NAME_SIZE);
adapter = i2c_get_adapter(setup->i2c_bus);
if (!adapter) {
dev_err(&pdev->dev, "can't get i2c adapter %d\n",
setup->i2c_bus);
goto err_driver;
}
client = i2c_new_device(adapter, &info);
i2c_put_adapter(adapter);
if (!client) {
dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
(unsigned int)info.addr);
goto err_driver;
}
return 0;
err_driver:
i2c_del_driver(&ssm2602_i2c_driver);
return -ENODEV;
}
#endif
static int ssm2602_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct ssm2602_setup_data *setup;
struct snd_soc_codec *codec;
struct ssm2602_priv *ssm2602;
int ret = 0;
pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION);
setup = socdev->codec_data;
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (codec == NULL)
return -ENOMEM;
ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL);
if (ssm2602 == NULL) {
kfree(codec);
return -ENOMEM;
}
codec->private_data = ssm2602;
socdev->codec = codec;
mutex_init(&codec->mutex);
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
ssm2602_socdev = socdev;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
if (setup->i2c_address) {
codec->hw_write = (hw_write_t)i2c_master_send;
ret = ssm2602_add_i2c_device(pdev, setup);
}
#else
/* other interfaces */
#endif
return ret;
}
/* remove everything here */
static int ssm2602_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)
ssm2602_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_unregister_device(codec->control_data);
i2c_del_driver(&ssm2602_i2c_driver);
#endif
kfree(codec->private_data);
kfree(codec);
return 0;
}
struct snd_soc_codec_device soc_codec_dev_ssm2602 = {
.probe = ssm2602_probe,
.remove = ssm2602_remove,
.suspend = ssm2602_suspend,
.resume = ssm2602_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602);
MODULE_DESCRIPTION("ASoC ssm2602 driver");
MODULE_AUTHOR("Cliff Cai");
MODULE_LICENSE("GPL");

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/*
* File: sound/soc/codecs/ssm2602.h
* Author: Cliff Cai <Cliff.Cai@analog.com>
*
* Created: Tue June 06 2008
*
* 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
*/
#ifndef _SSM2602_H
#define _SSM2602_H
/* SSM2602 Codec Register definitions */
#define SSM2602_LINVOL 0x00
#define SSM2602_RINVOL 0x01
#define SSM2602_LOUT1V 0x02
#define SSM2602_ROUT1V 0x03
#define SSM2602_APANA 0x04
#define SSM2602_APDIGI 0x05
#define SSM2602_PWR 0x06
#define SSM2602_IFACE 0x07
#define SSM2602_SRATE 0x08
#define SSM2602_ACTIVE 0x09
#define SSM2602_RESET 0x0f
/*SSM2602 Codec Register Field definitions
*(Mask value to extract the corresponding Register field)
*/
/*Left ADC Volume Control (SSM2602_REG_LEFT_ADC_VOL)*/
#define LINVOL_LIN_VOL 0x01F /* Left Channel PGA Volume control */
#define LINVOL_LIN_ENABLE_MUTE 0x080 /* Left Channel Input Mute */
#define LINVOL_LRIN_BOTH 0x100 /* Left Channel Line Input Volume update */
/*Right ADC Volume Control (SSM2602_REG_RIGHT_ADC_VOL)*/
#define RINVOL_RIN_VOL 0x01F /* Right Channel PGA Volume control */
#define RINVOL_RIN_ENABLE_MUTE 0x080 /* Right Channel Input Mute */
#define RINVOL_RLIN_BOTH 0x100 /* Right Channel Line Input Volume update */
/*Left DAC Volume Control (SSM2602_REG_LEFT_DAC_VOL)*/
#define LOUT1V_LHP_VOL 0x07F /* Left Channel Headphone volume control */
#define LOUT1V_ENABLE_LZC 0x080 /* Left Channel Zero cross detect enable */
#define LOUT1V_LRHP_BOTH 0x100 /* Left Channel Headphone volume update */
/*Right DAC Volume Control (SSM2602_REG_RIGHT_DAC_VOL)*/
#define ROUT1V_RHP_VOL 0x07F /* Right Channel Headphone volume control */
#define ROUT1V_ENABLE_RZC 0x080 /* Right Channel Zero cross detect enable */
#define ROUT1V_RLHP_BOTH 0x100 /* Right Channel Headphone volume update */
/*Analogue Audio Path Control (SSM2602_REG_ANALOGUE_PATH)*/
#define APANA_ENABLE_MIC_BOOST 0x001 /* Primary Microphone Amplifier gain booster control */
#define APANA_ENABLE_MIC_MUTE 0x002 /* Microphone Mute Control */
#define APANA_ADC_IN_SELECT 0x004 /* Microphone/Line IN select to ADC (1=MIC, 0=Line In) */
#define APANA_ENABLE_BYPASS 0x008 /* Line input bypass to line output */
#define APANA_SELECT_DAC 0x010 /* Select DAC (1=Select DAC, 0=Don't Select DAC) */
#define APANA_ENABLE_SIDETONE 0x020 /* Enable/Disable Side Tone */
#define APANA_SIDETONE_ATTN 0x0C0 /* Side Tone Attenuation */
#define APANA_ENABLE_MIC_BOOST2 0x100 /* Secondary Microphone Amplifier gain booster control */
/*Digital Audio Path Control (SSM2602_REG_DIGITAL_PATH)*/
#define APDIGI_ENABLE_ADC_HPF 0x001 /* Enable/Disable ADC Highpass Filter */
#define APDIGI_DE_EMPHASIS 0x006 /* De-Emphasis Control */
#define APDIGI_ENABLE_DAC_MUTE 0x008 /* DAC Mute Control */
#define APDIGI_STORE_OFFSET 0x010 /* Store/Clear DC offset when HPF is disabled */
/*Power Down Control (SSM2602_REG_POWER)
*(1=Enable PowerDown, 0=Disable PowerDown)
*/
#define PWR_LINE_IN_PDN 0x001 /* Line Input Power Down */
#define PWR_MIC_PDN 0x002 /* Microphone Input & Bias Power Down */
#define PWR_ADC_PDN 0x004 /* ADC Power Down */
#define PWR_DAC_PDN 0x008 /* DAC Power Down */
#define PWR_OUT_PDN 0x010 /* Outputs Power Down */
#define PWR_OSC_PDN 0x020 /* Oscillator Power Down */
#define PWR_CLK_OUT_PDN 0x040 /* CLKOUT Power Down */
#define PWR_POWER_OFF 0x080 /* POWEROFF Mode */
/*Digital Audio Interface Format (SSM2602_REG_DIGITAL_IFACE)*/
#define IFACE_IFACE_FORMAT 0x003 /* Digital Audio input format control */
#define IFACE_AUDIO_DATA_LEN 0x00C /* Audio Data word length control */
#define IFACE_DAC_LR_POLARITY 0x010 /* Polarity Control for clocks in RJ,LJ and I2S modes */
#define IFACE_DAC_LR_SWAP 0x020 /* Swap DAC data control */
#define IFACE_ENABLE_MASTER 0x040 /* Enable/Disable Master Mode */
#define IFACE_BCLK_INVERT 0x080 /* Bit Clock Inversion control */
/*Sampling Control (SSM2602_REG_SAMPLING_CTRL)*/
#define SRATE_ENABLE_USB_MODE 0x001 /* Enable/Disable USB Mode */
#define SRATE_BOS_RATE 0x002 /* Base Over-Sampling rate */
#define SRATE_SAMPLE_RATE 0x03C /* Clock setting condition (Sampling rate control) */
#define SRATE_CORECLK_DIV2 0x040 /* Core Clock divider select */
#define SRATE_CLKOUT_DIV2 0x080 /* Clock Out divider select */
/*Active Control (SSM2602_REG_ACTIVE_CTRL)*/
#define ACTIVE_ACTIVATE_CODEC 0x001 /* Activate Codec Digital Audio Interface */
/*********************************************************************/
#define SSM2602_CACHEREGNUM 10
#define SSM2602_SYSCLK 0
#define SSM2602_DAI 0
struct ssm2602_setup_data {
int i2c_bus;
unsigned short i2c_address;
};
extern struct snd_soc_dai ssm2602_dai;
extern struct snd_soc_codec_device soc_codec_dev_ssm2602;
#endif