redonkable/alistair23-linux
redonkable/alistair23-linux
1
0
Fork 0
Code Releases Activity
alistair23-linux/sound/soc/codecs/ssm2602.c
Liam Girdwood f0fba2ad1b ASoC: multi-component - ASoC Multi-Component Support 
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.

 struct snd_soc_codec    --->  struct snd_soc_codec (device data)
                          +->  struct snd_soc_codec_driver (driver data)

 struct snd_soc_platform --->  struct snd_soc_platform (device data)
                          +->  struct snd_soc_platform_driver (driver data)

 struct snd_soc_dai      --->  struct snd_soc_dai (device data)
                          +->  struct snd_soc_dai_driver (driver data)

 struct snd_soc_device   --->  deleted

This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.

The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.

This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.

Other notable multi-component changes:-

 * Stream operations now de-reference less structures.
 * close_delayed work() now runs on a DAI basis rather than looping all DAIs
   in a card.
 * PM suspend()/resume() operations can now handle N CODECs and Platforms
   per sound card.
 * Added soc_bind_dai_link() to bind the component devices to the sound card.
 * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
   DAI link components.
 * sysfs entries can now be registered per component per card.
 * snd_soc_new_pcms() functionailty rolled into dai_link_probe().
 * snd_soc_register_codec() now does all the codec list and mutex init.

This patch changes the probe() and remove() of the CODEC drivers as follows:-

 o Make CODEC driver a platform driver
 o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
 o Removed all static codec pointers (drivers now support > 1 codec dev)
 o snd_soc_register_pcms() now done by core.
 o snd_soc_register_dai() folded into snd_soc_register_codec().

CS4270 portions:
Acked-by: Timur Tabi <timur@freescale.com>

Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>

TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>

Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>

MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>

i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>

J4740 platform fixes:-
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>

CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>

Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-08-12 14:00:00 +01:00

690 lines
18 KiB
C
Raw Blame History

/*
* 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 <linux/slab.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 SSM2602_VERSION "0.1"
/* codec private data */
struct ssm2602_priv {
unsigned int sysclk;
enum snd_soc_control_type control_type;
void *control_data;
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 Boost2 (+20dB)", SSM2602_APANA, 7, 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]),
};
/* 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));
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,
struct snd_soc_dai *dai)
{
u16 srate;
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 i2c_client *i2c = codec->control_data;
u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3;
int i = get_coeff(ssm2602->sysclk, params_rate(params));
if (substream == ssm2602->slave_substream) {
dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n");
return 0;
}
/*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_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 i2c_client *i2c = codec->control_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.
* TODO: the ssm2602 allows pairs of non-matching PB/REC rates
*/
if (ssm2602->master_substream) {
master_runtime = ssm2602->master_substream->runtime;
dev_dbg(&i2c->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 int ssm2602_pcm_prepare(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;
/* set active */
ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC);
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);
/* deactivate */
if (!codec->active)
ssm2602_write(codec, SSM2602_ACTIVE, 0);
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 = 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 = snd_soc_codec_get_drvdata(codec);
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 |= 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 */
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_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,
.prepare = ssm2602_pcm_prepare,
.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)
{
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);
return 0;
}
static int ssm2602_probe(struct snd_soc_codec *codec)
{
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
int ret = 0, reg;
pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION);
codec->bias_level = SND_SOC_BIAS_OFF,
codec->control_data = ssm2602->control_data;
ssm2602_reset(codec);
/*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_SELECT_DAC |
APANA_ENABLE_MIC_BOOST);
ssm2602_write(codec, SSM2602_PWR, 0);
snd_soc_add_controls(codec, ssm2602_snd_controls,
ARRAY_SIZE(ssm2602_snd_controls));
ssm2602_add_widgets(codec);
return ret;
}
/* 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 = ssm2602_probe,
.remove = ssm2602_remove,
.suspend = ssm2602_suspend,
.resume = ssm2602_resume,
.read = ssm2602_read_reg_cache,
.write = ssm2602_write,
.set_bias_level = ssm2602_set_bias_level,
.reg_cache_size = sizeof(ssm2602_reg),
.reg_word_size = sizeof(u16),
.reg_cache_default = ssm2602_reg,
};
#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 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_data = i2c;
ssm2602->control_type = SND_SOC_I2C;
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_ssm2602, &ssm2602_dai, 1);
if (ret < 0)
kfree(ssm2602);
return ret;
}
static int 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", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id);
/* corgi i2c codec control layer */
static struct i2c_driver ssm2602_i2c_driver = {
.driver = {
.name = "ssm2602-codec",
.owner = THIS_MODULE,
},
.probe = ssm2602_i2c_probe,
.remove = ssm2602_i2c_remove,
.id_table = ssm2602_i2c_id,
};
#endif
static int __init ssm2602_modinit(void)
{
int ret = 0;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&ssm2602_i2c_driver);
if (ret != 0) {
printk(KERN_ERR "Failed to register SSM2602 I2C driver: %d\n",
ret);
}
#endif
return ret;
}
module_init(ssm2602_modinit);
static void __exit ssm2602_exit(void)
{
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&ssm2602_i2c_driver);
#endif
}
module_exit(ssm2602_exit);
MODULE_DESCRIPTION("ASoC ssm2602 driver");
MODULE_AUTHOR("Cliff Cai");
MODULE_LICENSE("GPL");
View git blame Copy permalink