WSL2-Linux-Kernel/sound/soc/codecs/twl6040.c

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C
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/*
* ALSA SoC TWL6040 codec driver
*
* Author: Misael Lopez Cruz <x0052729@ti.com>
*
* 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.
*
* 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, 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/platform_device.h>
#include <linux/slab.h>
#include <linux/mfd/twl6040.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 <sound/tlv.h>
#include "twl6040.h"
#define TWL6040_RATES SNDRV_PCM_RATE_8000_96000
#define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE)
#define TWL6040_OUTHS_0dB 0x00
#define TWL6040_OUTHS_M30dB 0x0F
#define TWL6040_OUTHF_0dB 0x03
#define TWL6040_OUTHF_M52dB 0x1D
/* Shadow register used by the driver */
#define TWL6040_REG_SW_SHADOW 0x2F
#define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1)
/* TWL6040_REG_SW_SHADOW (0x2F) fields */
#define TWL6040_EAR_PATH_ENABLE 0x01
struct twl6040_jack_data {
struct snd_soc_jack *jack;
struct delayed_work work;
int report;
};
/* codec private data */
struct twl6040_data {
int plug_irq;
int codec_powered;
int pll;
int pll_power_mode;
int hs_power_mode;
int hs_power_mode_locked;
unsigned int clk_in;
unsigned int sysclk;
u16 hs_left_step;
u16 hs_right_step;
u16 hf_left_step;
u16 hf_right_step;
struct twl6040_jack_data hs_jack;
struct snd_soc_codec *codec;
struct workqueue_struct *workqueue;
struct mutex mutex;
};
/*
* twl6040 register cache & default register settings
*/
static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = {
0x00, /* not used 0x00 */
0x4B, /* REG_ASICID 0x01 (ro) */
0x00, /* REG_ASICREV 0x02 (ro) */
0x00, /* REG_INTID 0x03 */
0x00, /* REG_INTMR 0x04 */
0x00, /* REG_NCPCTRL 0x05 */
0x00, /* REG_LDOCTL 0x06 */
0x60, /* REG_HPPLLCTL 0x07 */
0x00, /* REG_LPPLLCTL 0x08 */
0x4A, /* REG_LPPLLDIV 0x09 */
0x00, /* REG_AMICBCTL 0x0A */
0x00, /* REG_DMICBCTL 0x0B */
0x00, /* REG_MICLCTL 0x0C */
0x00, /* REG_MICRCTL 0x0D */
0x00, /* REG_MICGAIN 0x0E */
0x1B, /* REG_LINEGAIN 0x0F */
0x00, /* REG_HSLCTL 0x10 */
0x00, /* REG_HSRCTL 0x11 */
0x00, /* REG_HSGAIN 0x12 */
0x00, /* REG_EARCTL 0x13 */
0x00, /* REG_HFLCTL 0x14 */
0x00, /* REG_HFLGAIN 0x15 */
0x00, /* REG_HFRCTL 0x16 */
0x00, /* REG_HFRGAIN 0x17 */
0x00, /* REG_VIBCTLL 0x18 */
0x00, /* REG_VIBDATL 0x19 */
0x00, /* REG_VIBCTLR 0x1A */
0x00, /* REG_VIBDATR 0x1B */
0x00, /* REG_HKCTL1 0x1C */
0x00, /* REG_HKCTL2 0x1D */
0x00, /* REG_GPOCTL 0x1E */
0x00, /* REG_ALB 0x1F */
0x00, /* REG_DLB 0x20 */
0x00, /* not used 0x21 */
0x00, /* not used 0x22 */
0x00, /* not used 0x23 */
0x00, /* not used 0x24 */
0x00, /* not used 0x25 */
0x00, /* not used 0x26 */
0x00, /* not used 0x27 */
0x00, /* REG_TRIM1 0x28 */
0x00, /* REG_TRIM2 0x29 */
0x00, /* REG_TRIM3 0x2A */
0x00, /* REG_HSOTRIM 0x2B */
0x00, /* REG_HFOTRIM 0x2C */
0x09, /* REG_ACCCTL 0x2D */
0x00, /* REG_STATUS 0x2E (ro) */
0x00, /* REG_SW_SHADOW 0x2F - Shadow, non HW register */
};
/* List of registers to be restored after power up */
static const int twl6040_restore_list[] = {
TWL6040_REG_MICLCTL,
TWL6040_REG_MICRCTL,
TWL6040_REG_MICGAIN,
TWL6040_REG_LINEGAIN,
TWL6040_REG_HSLCTL,
TWL6040_REG_HSRCTL,
TWL6040_REG_HSGAIN,
TWL6040_REG_EARCTL,
TWL6040_REG_HFLCTL,
TWL6040_REG_HFLGAIN,
TWL6040_REG_HFRCTL,
TWL6040_REG_HFRGAIN,
};
/* set of rates for each pll: low-power and high-performance */
static unsigned int lp_rates[] = {
8000,
11250,
16000,
22500,
32000,
44100,
48000,
88200,
96000,
};
static unsigned int hp_rates[] = {
8000,
16000,
32000,
48000,
96000,
};
static struct snd_pcm_hw_constraint_list sysclk_constraints[] = {
{ .count = ARRAY_SIZE(lp_rates), .list = lp_rates, },
{ .count = ARRAY_SIZE(hp_rates), .list = hp_rates, },
};
/*
* read twl6040 register cache
*/
static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec,
unsigned int reg)
{
u8 *cache = codec->reg_cache;
if (reg >= TWL6040_CACHEREGNUM)
return -EIO;
return cache[reg];
}
/*
* write twl6040 register cache
*/
static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec,
u8 reg, u8 value)
{
u8 *cache = codec->reg_cache;
if (reg >= TWL6040_CACHEREGNUM)
return;
cache[reg] = value;
}
/*
* read from twl6040 hardware register
*/
static int twl6040_read_reg_volatile(struct snd_soc_codec *codec,
unsigned int reg)
{
struct twl6040 *twl6040 = codec->control_data;
u8 value;
if (reg >= TWL6040_CACHEREGNUM)
return -EIO;
if (likely(reg < TWL6040_REG_SW_SHADOW)) {
value = twl6040_reg_read(twl6040, reg);
twl6040_write_reg_cache(codec, reg, value);
} else {
value = twl6040_read_reg_cache(codec, reg);
}
return value;
}
/*
* write to the twl6040 register space
*/
static int twl6040_write(struct snd_soc_codec *codec,
unsigned int reg, unsigned int value)
{
struct twl6040 *twl6040 = codec->control_data;
if (reg >= TWL6040_CACHEREGNUM)
return -EIO;
twl6040_write_reg_cache(codec, reg, value);
if (likely(reg < TWL6040_REG_SW_SHADOW))
return twl6040_reg_write(twl6040, reg, value);
else
return 0;
}
static void twl6040_init_chip(struct snd_soc_codec *codec)
{
struct twl6040 *twl6040 = codec->control_data;
u8 val;
/* Update reg_cache: ASICREV, and TRIM values */
val = twl6040_get_revid(twl6040);
twl6040_write_reg_cache(codec, TWL6040_REG_ASICREV, val);
twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM1);
twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM2);
twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM3);
twl6040_read_reg_volatile(codec, TWL6040_REG_HSOTRIM);
twl6040_read_reg_volatile(codec, TWL6040_REG_HFOTRIM);
/* Change chip defaults */
/* No imput selected for microphone amplifiers */
twl6040_write_reg_cache(codec, TWL6040_REG_MICLCTL, 0x18);
twl6040_write_reg_cache(codec, TWL6040_REG_MICRCTL, 0x18);
/*
* We need to lower the default gain values, so the ramp code
* can work correctly for the first playback.
* This reduces the pop noise heard at the first playback.
*/
twl6040_write_reg_cache(codec, TWL6040_REG_HSGAIN, 0xff);
twl6040_write_reg_cache(codec, TWL6040_REG_EARCTL, 0x1e);
twl6040_write_reg_cache(codec, TWL6040_REG_HFLGAIN, 0x1d);
twl6040_write_reg_cache(codec, TWL6040_REG_HFRGAIN, 0x1d);
twl6040_write_reg_cache(codec, TWL6040_REG_LINEGAIN, 0);
}
static void twl6040_restore_regs(struct snd_soc_codec *codec)
{
u8 *cache = codec->reg_cache;
int reg, i;
for (i = 0; i < ARRAY_SIZE(twl6040_restore_list); i++) {
reg = twl6040_restore_list[i];
twl6040_write(codec, reg, cache[reg]);
}
}
/* set headset dac and driver power mode */
static int headset_power_mode(struct snd_soc_codec *codec, int high_perf)
{
int hslctl, hsrctl;
int mask = TWL6040_HSDRVMODE | TWL6040_HSDACMODE;
hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL);
hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL);
if (high_perf) {
hslctl &= ~mask;
hsrctl &= ~mask;
} else {
hslctl |= mask;
hsrctl |= mask;
}
twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl);
twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl);
return 0;
}
static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
u8 hslctl, hsrctl;
/*
* Workaround for Headset DC offset caused pop noise:
* Both HS DAC need to be turned on (before the HS driver) and off at
* the same time.
*/
hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL);
hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL);
if (SND_SOC_DAPM_EVENT_ON(event)) {
hslctl |= TWL6040_HSDACENA;
hsrctl |= TWL6040_HSDACENA;
} else {
hslctl &= ~TWL6040_HSDACENA;
hsrctl &= ~TWL6040_HSDACENA;
}
twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl);
twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl);
msleep(1);
return 0;
}
static int twl6040_ep_drv_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int ret = 0;
if (SND_SOC_DAPM_EVENT_ON(event)) {
/* Earphone doesn't support low power mode */
priv->hs_power_mode_locked = 1;
ret = headset_power_mode(codec, 1);
} else {
priv->hs_power_mode_locked = 0;
ret = headset_power_mode(codec, priv->hs_power_mode);
}
msleep(1);
return ret;
}
static void twl6040_hs_jack_report(struct snd_soc_codec *codec,
struct snd_soc_jack *jack, int report)
{
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int status;
mutex_lock(&priv->mutex);
/* Sync status */
status = twl6040_read_reg_volatile(codec, TWL6040_REG_STATUS);
if (status & TWL6040_PLUGCOMP)
snd_soc_jack_report(jack, report, report);
else
snd_soc_jack_report(jack, 0, report);
mutex_unlock(&priv->mutex);
}
void twl6040_hs_jack_detect(struct snd_soc_codec *codec,
struct snd_soc_jack *jack, int report)
{
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
struct twl6040_jack_data *hs_jack = &priv->hs_jack;
hs_jack->jack = jack;
hs_jack->report = report;
twl6040_hs_jack_report(codec, hs_jack->jack, hs_jack->report);
}
EXPORT_SYMBOL_GPL(twl6040_hs_jack_detect);
static void twl6040_accessory_work(struct work_struct *work)
{
struct twl6040_data *priv = container_of(work,
struct twl6040_data, hs_jack.work.work);
struct snd_soc_codec *codec = priv->codec;
struct twl6040_jack_data *hs_jack = &priv->hs_jack;
twl6040_hs_jack_report(codec, hs_jack->jack, hs_jack->report);
}
/* audio interrupt handler */
static irqreturn_t twl6040_audio_handler(int irq, void *data)
{
struct snd_soc_codec *codec = data;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
queue_delayed_work(priv->workqueue, &priv->hs_jack.work,
msecs_to_jiffies(200));
return IRQ_HANDLED;
}
static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget *widget = wlist->widgets[0];
struct snd_soc_codec *codec = widget->codec;
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
unsigned int val;
/* Do not allow changes while Input/FF efect is running */
val = twl6040_read_reg_volatile(codec, e->reg);
if (val & TWL6040_VIBENA && !(val & TWL6040_VIBSEL))
return -EBUSY;
return snd_soc_dapm_put_enum_double(kcontrol, ucontrol);
}
/*
* MICATT volume control:
* from -6 to 0 dB in 6 dB steps
*/
static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0);
/*
* MICGAIN volume control:
* from 6 to 30 dB in 6 dB steps
*/
static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0);
/*
* AFMGAIN volume control:
* from -18 to 24 dB in 6 dB steps
*/
static DECLARE_TLV_DB_SCALE(afm_amp_tlv, -1800, 600, 0);
/*
* HSGAIN volume control:
* from -30 to 0 dB in 2 dB steps
*/
static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0);
/*
* HFGAIN volume control:
* from -52 to 6 dB in 2 dB steps
*/
static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0);
/*
* EPGAIN volume control:
* from -24 to 6 dB in 2 dB steps
*/
static DECLARE_TLV_DB_SCALE(ep_tlv, -2400, 200, 0);
/* Left analog microphone selection */
static const char *twl6040_amicl_texts[] =
{"Headset Mic", "Main Mic", "Aux/FM Left", "Off"};
/* Right analog microphone selection */
static const char *twl6040_amicr_texts[] =
{"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"};
static const struct soc_enum twl6040_enum[] = {
SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 4, twl6040_amicl_texts),
SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 4, twl6040_amicr_texts),
};
static const char *twl6040_hs_texts[] = {
"Off", "HS DAC", "Line-In amp"
};
static const struct soc_enum twl6040_hs_enum[] = {
SOC_ENUM_SINGLE(TWL6040_REG_HSLCTL, 5, ARRAY_SIZE(twl6040_hs_texts),
twl6040_hs_texts),
SOC_ENUM_SINGLE(TWL6040_REG_HSRCTL, 5, ARRAY_SIZE(twl6040_hs_texts),
twl6040_hs_texts),
};
static const char *twl6040_hf_texts[] = {
"Off", "HF DAC", "Line-In amp"
};
static const struct soc_enum twl6040_hf_enum[] = {
SOC_ENUM_SINGLE(TWL6040_REG_HFLCTL, 2, ARRAY_SIZE(twl6040_hf_texts),
twl6040_hf_texts),
SOC_ENUM_SINGLE(TWL6040_REG_HFRCTL, 2, ARRAY_SIZE(twl6040_hf_texts),
twl6040_hf_texts),
};
static const char *twl6040_vibrapath_texts[] = {
"Input FF", "Audio PDM"
};
static const struct soc_enum twl6040_vibra_enum[] = {
SOC_ENUM_SINGLE(TWL6040_REG_VIBCTLL, 1,
ARRAY_SIZE(twl6040_vibrapath_texts),
twl6040_vibrapath_texts),
SOC_ENUM_SINGLE(TWL6040_REG_VIBCTLR, 1,
ARRAY_SIZE(twl6040_vibrapath_texts),
twl6040_vibrapath_texts),
};
static const struct snd_kcontrol_new amicl_control =
SOC_DAPM_ENUM("Route", twl6040_enum[0]);
static const struct snd_kcontrol_new amicr_control =
SOC_DAPM_ENUM("Route", twl6040_enum[1]);
/* Headset DAC playback switches */
static const struct snd_kcontrol_new hsl_mux_controls =
SOC_DAPM_ENUM("Route", twl6040_hs_enum[0]);
static const struct snd_kcontrol_new hsr_mux_controls =
SOC_DAPM_ENUM("Route", twl6040_hs_enum[1]);
/* Handsfree DAC playback switches */
static const struct snd_kcontrol_new hfl_mux_controls =
SOC_DAPM_ENUM("Route", twl6040_hf_enum[0]);
static const struct snd_kcontrol_new hfr_mux_controls =
SOC_DAPM_ENUM("Route", twl6040_hf_enum[1]);
static const struct snd_kcontrol_new ep_path_enable_control =
SOC_DAPM_SINGLE("Switch", TWL6040_REG_SW_SHADOW, 0, 1, 0);
static const struct snd_kcontrol_new auxl_switch_control =
SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 6, 1, 0);
static const struct snd_kcontrol_new auxr_switch_control =
SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 6, 1, 0);
/* Vibra playback switches */
static const struct snd_kcontrol_new vibral_mux_controls =
SOC_DAPM_ENUM_EXT("Route", twl6040_vibra_enum[0],
snd_soc_dapm_get_enum_double,
twl6040_soc_dapm_put_vibra_enum);
static const struct snd_kcontrol_new vibrar_mux_controls =
SOC_DAPM_ENUM_EXT("Route", twl6040_vibra_enum[1],
snd_soc_dapm_get_enum_double,
twl6040_soc_dapm_put_vibra_enum);
/* Headset power mode */
static const char *twl6040_power_mode_texts[] = {
"Low-Power", "High-Performance",
};
static const struct soc_enum twl6040_power_mode_enum =
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl6040_power_mode_texts),
twl6040_power_mode_texts);
static int twl6040_headset_power_get_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
ucontrol->value.enumerated.item[0] = priv->hs_power_mode;
return 0;
}
static int twl6040_headset_power_put_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int high_perf = ucontrol->value.enumerated.item[0];
int ret = 0;
if (!priv->hs_power_mode_locked)
ret = headset_power_mode(codec, high_perf);
if (!ret)
priv->hs_power_mode = high_perf;
return ret;
}
static int twl6040_pll_get_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
ucontrol->value.enumerated.item[0] = priv->pll_power_mode;
return 0;
}
static int twl6040_pll_put_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
priv->pll_power_mode = ucontrol->value.enumerated.item[0];
return 0;
}
int twl6040_get_dl1_gain(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
if (snd_soc_dapm_get_pin_status(dapm, "EP"))
return -1; /* -1dB */
if (snd_soc_dapm_get_pin_status(dapm, "HSOR") ||
snd_soc_dapm_get_pin_status(dapm, "HSOL")) {
u8 val = snd_soc_read(codec, TWL6040_REG_HSLCTL);
if (val & TWL6040_HSDACMODE)
/* HSDACL in LP mode */
return -8; /* -8dB */
else
/* HSDACL in HP mode */
return -1; /* -1dB */
}
return 0; /* 0dB */
}
EXPORT_SYMBOL_GPL(twl6040_get_dl1_gain);
int twl6040_get_clk_id(struct snd_soc_codec *codec)
{
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
return priv->pll_power_mode;
}
EXPORT_SYMBOL_GPL(twl6040_get_clk_id);
int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim)
{
if (unlikely(trim >= TWL6040_TRIM_INVAL))
return -EINVAL;
return twl6040_read_reg_cache(codec, TWL6040_REG_TRIM1 + trim);
}
EXPORT_SYMBOL_GPL(twl6040_get_trim_value);
int twl6040_get_hs_step_size(struct snd_soc_codec *codec)
{
struct twl6040 *twl6040 = codec->control_data;
if (twl6040_get_revid(twl6040) < TWL6040_REV_ES1_3)
/* For ES under ES_1.3 HS step is 2 mV */
return 2;
else
/* For ES_1.3 HS step is 1 mV */
return 1;
}
EXPORT_SYMBOL_GPL(twl6040_get_hs_step_size);
static const struct snd_kcontrol_new twl6040_snd_controls[] = {
/* Capture gains */
SOC_DOUBLE_TLV("Capture Preamplifier Volume",
TWL6040_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv),
SOC_DOUBLE_TLV("Capture Volume",
TWL6040_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv),
/* AFM gains */
SOC_DOUBLE_TLV("Aux FM Volume",
TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv),
/* Playback gains */
SOC_DOUBLE_TLV("Headset Playback Volume",
TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv),
SOC_DOUBLE_R_TLV("Handsfree Playback Volume",
TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv),
SOC_SINGLE_TLV("Earphone Playback Volume",
TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv),
SOC_ENUM_EXT("Headset Power Mode", twl6040_power_mode_enum,
twl6040_headset_power_get_enum,
twl6040_headset_power_put_enum),
SOC_ENUM_EXT("PLL Selection", twl6040_power_mode_enum,
twl6040_pll_get_enum, twl6040_pll_put_enum),
};
static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
/* Inputs */
SND_SOC_DAPM_INPUT("MAINMIC"),
SND_SOC_DAPM_INPUT("HSMIC"),
SND_SOC_DAPM_INPUT("SUBMIC"),
SND_SOC_DAPM_INPUT("AFML"),
SND_SOC_DAPM_INPUT("AFMR"),
/* Outputs */
SND_SOC_DAPM_OUTPUT("HSOL"),
SND_SOC_DAPM_OUTPUT("HSOR"),
SND_SOC_DAPM_OUTPUT("HFL"),
SND_SOC_DAPM_OUTPUT("HFR"),
SND_SOC_DAPM_OUTPUT("EP"),
SND_SOC_DAPM_OUTPUT("AUXL"),
SND_SOC_DAPM_OUTPUT("AUXR"),
SND_SOC_DAPM_OUTPUT("VIBRAL"),
SND_SOC_DAPM_OUTPUT("VIBRAR"),
/* Analog input muxes for the capture amplifiers */
SND_SOC_DAPM_MUX("Analog Left Capture Route",
SND_SOC_NOPM, 0, 0, &amicl_control),
SND_SOC_DAPM_MUX("Analog Right Capture Route",
SND_SOC_NOPM, 0, 0, &amicr_control),
/* Analog capture PGAs */
SND_SOC_DAPM_PGA("MicAmpL",
TWL6040_REG_MICLCTL, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("MicAmpR",
TWL6040_REG_MICRCTL, 0, 0, NULL, 0),
/* Auxiliary FM PGAs */
SND_SOC_DAPM_PGA("AFMAmpL",
TWL6040_REG_MICLCTL, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("AFMAmpR",
TWL6040_REG_MICRCTL, 1, 0, NULL, 0),
/* ADCs */
SND_SOC_DAPM_ADC("ADC Left", NULL, TWL6040_REG_MICLCTL, 2, 0),
SND_SOC_DAPM_ADC("ADC Right", NULL, TWL6040_REG_MICRCTL, 2, 0),
/* Microphone bias */
SND_SOC_DAPM_SUPPLY("Headset Mic Bias",
TWL6040_REG_AMICBCTL, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Main Mic Bias",
TWL6040_REG_AMICBCTL, 4, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Digital Mic1 Bias",
TWL6040_REG_DMICBCTL, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Digital Mic2 Bias",
TWL6040_REG_DMICBCTL, 4, 0, NULL, 0),
/* DACs */
SND_SOC_DAPM_DAC("HSDAC Left", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("HSDAC Right", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("HFDAC Left", NULL, TWL6040_REG_HFLCTL, 0, 0),
SND_SOC_DAPM_DAC("HFDAC Right", NULL, TWL6040_REG_HFRCTL, 0, 0),
/* Virtual DAC for vibra path (DL4 channel) */
SND_SOC_DAPM_DAC("VIBRA DAC", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("Handsfree Left Playback",
SND_SOC_NOPM, 0, 0, &hfl_mux_controls),
SND_SOC_DAPM_MUX("Handsfree Right Playback",
SND_SOC_NOPM, 0, 0, &hfr_mux_controls),
/* Analog playback Muxes */
SND_SOC_DAPM_MUX("Headset Left Playback",
SND_SOC_NOPM, 0, 0, &hsl_mux_controls),
SND_SOC_DAPM_MUX("Headset Right Playback",
SND_SOC_NOPM, 0, 0, &hsr_mux_controls),
SND_SOC_DAPM_MUX("Vibra Left Playback", SND_SOC_NOPM, 0, 0,
&vibral_mux_controls),
SND_SOC_DAPM_MUX("Vibra Right Playback", SND_SOC_NOPM, 0, 0,
&vibrar_mux_controls),
SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0,
&ep_path_enable_control),
SND_SOC_DAPM_SWITCH("AUXL Playback", SND_SOC_NOPM, 0, 0,
&auxl_switch_control),
SND_SOC_DAPM_SWITCH("AUXR Playback", SND_SOC_NOPM, 0, 0,
&auxr_switch_control),
/* Analog playback drivers */
SND_SOC_DAPM_OUT_DRV("HF Left Driver",
TWL6040_REG_HFLCTL, 4, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV("HF Right Driver",
TWL6040_REG_HFRCTL, 4, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV("HS Left Driver",
TWL6040_REG_HSLCTL, 2, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV("HS Right Driver",
TWL6040_REG_HSRCTL, 2, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV_E("Earphone Driver",
TWL6040_REG_EARCTL, 0, 0, NULL, 0,
twl6040_ep_drv_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_OUT_DRV("Vibra Left Driver",
TWL6040_REG_VIBCTLL, 0, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV("Vibra Right Driver",
TWL6040_REG_VIBCTLR, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Vibra Left Control", TWL6040_REG_VIBCTLL, 2, 0,
NULL, 0),
SND_SOC_DAPM_SUPPLY("Vibra Right Control", TWL6040_REG_VIBCTLR, 2, 0,
NULL, 0),
SND_SOC_DAPM_SUPPLY_S("HSDAC Power", 1, SND_SOC_NOPM, 0, 0,
twl6040_hs_dac_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
/* Analog playback PGAs */
SND_SOC_DAPM_PGA("HF Left PGA",
TWL6040_REG_HFLCTL, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("HF Right PGA",
TWL6040_REG_HFRCTL, 1, 0, NULL, 0),
};
static const struct snd_soc_dapm_route intercon[] = {
/* Stream -> DAC mapping */
{"HSDAC Left", NULL, "Legacy Playback"},
{"HSDAC Left", NULL, "Headset Playback"},
{"HSDAC Right", NULL, "Legacy Playback"},
{"HSDAC Right", NULL, "Headset Playback"},
{"HFDAC Left", NULL, "Legacy Playback"},
{"HFDAC Left", NULL, "Handsfree Playback"},
{"HFDAC Right", NULL, "Legacy Playback"},
{"HFDAC Right", NULL, "Handsfree Playback"},
{"VIBRA DAC", NULL, "Legacy Playback"},
{"VIBRA DAC", NULL, "Vibra Playback"},
/* ADC -> Stream mapping */
{"Legacy Capture" , NULL, "ADC Left"},
{"Capture", NULL, "ADC Left"},
{"Legacy Capture", NULL, "ADC Right"},
{"Capture" , NULL, "ADC Right"},
/* Capture path */
{"Analog Left Capture Route", "Headset Mic", "HSMIC"},
{"Analog Left Capture Route", "Main Mic", "MAINMIC"},
{"Analog Left Capture Route", "Aux/FM Left", "AFML"},
{"Analog Right Capture Route", "Headset Mic", "HSMIC"},
{"Analog Right Capture Route", "Sub Mic", "SUBMIC"},
{"Analog Right Capture Route", "Aux/FM Right", "AFMR"},
{"MicAmpL", NULL, "Analog Left Capture Route"},
{"MicAmpR", NULL, "Analog Right Capture Route"},
{"ADC Left", NULL, "MicAmpL"},
{"ADC Right", NULL, "MicAmpR"},
/* AFM path */
{"AFMAmpL", NULL, "AFML"},
{"AFMAmpR", NULL, "AFMR"},
{"HSDAC Left", NULL, "HSDAC Power"},
{"HSDAC Right", NULL, "HSDAC Power"},
{"Headset Left Playback", "HS DAC", "HSDAC Left"},
{"Headset Left Playback", "Line-In amp", "AFMAmpL"},
{"Headset Right Playback", "HS DAC", "HSDAC Right"},
{"Headset Right Playback", "Line-In amp", "AFMAmpR"},
{"HS Left Driver", NULL, "Headset Left Playback"},
{"HS Right Driver", NULL, "Headset Right Playback"},
{"HSOL", NULL, "HS Left Driver"},
{"HSOR", NULL, "HS Right Driver"},
/* Earphone playback path */
{"Earphone Playback", "Switch", "HSDAC Left"},
{"Earphone Driver", NULL, "Earphone Playback"},
{"EP", NULL, "Earphone Driver"},
{"Handsfree Left Playback", "HF DAC", "HFDAC Left"},
{"Handsfree Left Playback", "Line-In amp", "AFMAmpL"},
{"Handsfree Right Playback", "HF DAC", "HFDAC Right"},
{"Handsfree Right Playback", "Line-In amp", "AFMAmpR"},
{"HF Left PGA", NULL, "Handsfree Left Playback"},
{"HF Right PGA", NULL, "Handsfree Right Playback"},
{"HF Left Driver", NULL, "HF Left PGA"},
{"HF Right Driver", NULL, "HF Right PGA"},
{"HFL", NULL, "HF Left Driver"},
{"HFR", NULL, "HF Right Driver"},
{"AUXL Playback", "Switch", "HF Left PGA"},
{"AUXR Playback", "Switch", "HF Right PGA"},
{"AUXL", NULL, "AUXL Playback"},
{"AUXR", NULL, "AUXR Playback"},
/* Vibrator paths */
{"Vibra Left Playback", "Audio PDM", "VIBRA DAC"},
{"Vibra Right Playback", "Audio PDM", "VIBRA DAC"},
{"Vibra Left Driver", NULL, "Vibra Left Playback"},
{"Vibra Right Driver", NULL, "Vibra Right Playback"},
{"Vibra Left Driver", NULL, "Vibra Left Control"},
{"Vibra Right Driver", NULL, "Vibra Right Control"},
{"VIBRAL", NULL, "Vibra Left Driver"},
{"VIBRAR", NULL, "Vibra Right Driver"},
};
static int twl6040_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
struct twl6040 *twl6040 = codec->control_data;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int ret;
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
if (priv->codec_powered)
break;
ret = twl6040_power(twl6040, 1);
if (ret)
return ret;
priv->codec_powered = 1;
twl6040_restore_regs(codec);
/* Set external boost GPO */
twl6040_write(codec, TWL6040_REG_GPOCTL, 0x02);
break;
case SND_SOC_BIAS_OFF:
if (!priv->codec_powered)
break;
twl6040_power(twl6040, 0);
priv->codec_powered = 0;
break;
}
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 16:53:46 +03:00
codec->dapm.bias_level = level;
return 0;
}
static int twl6040_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&sysclk_constraints[priv->pll_power_mode]);
return 0;
}
static int twl6040_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 twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int rate;
rate = params_rate(params);
switch (rate) {
case 11250:
case 22500:
case 44100:
case 88200:
/* These rates are not supported when HPPLL is in use */
if (unlikely(priv->pll == TWL6040_SYSCLK_SEL_HPPLL)) {
dev_err(codec->dev, "HPPLL does not support rate %d\n",
rate);
return -EINVAL;
}
priv->sysclk = 17640000;
break;
case 8000:
case 16000:
case 32000:
case 48000:
case 96000:
priv->sysclk = 19200000;
break;
default:
dev_err(codec->dev, "unsupported rate %d\n", rate);
return -EINVAL;
}
return 0;
}
static int twl6040_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct twl6040 *twl6040 = codec->control_data;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int ret;
if (!priv->sysclk) {
dev_err(codec->dev,
"no mclk configured, call set_sysclk() on init\n");
return -EINVAL;
}
ret = twl6040_set_pll(twl6040, priv->pll, priv->clk_in, priv->sysclk);
if (ret) {
dev_err(codec->dev, "Can not set PLL (%d)\n", ret);
return -EPERM;
}
return 0;
}
static int twl6040_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 twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
switch (clk_id) {
case TWL6040_SYSCLK_SEL_LPPLL:
case TWL6040_SYSCLK_SEL_HPPLL:
priv->pll = clk_id;
priv->clk_in = freq;
break;
default:
dev_err(codec->dev, "unknown clk_id %d\n", clk_id);
return -EINVAL;
}
return 0;
}
static const struct snd_soc_dai_ops twl6040_dai_ops = {
.startup = twl6040_startup,
.hw_params = twl6040_hw_params,
.prepare = twl6040_prepare,
.set_sysclk = twl6040_set_dai_sysclk,
};
static struct snd_soc_dai_driver twl6040_dai[] = {
{
.name = "twl6040-legacy",
.playback = {
.stream_name = "Legacy Playback",
.channels_min = 1,
.channels_max = 5,
.rates = TWL6040_RATES,
.formats = TWL6040_FORMATS,
},
.capture = {
.stream_name = "Legacy Capture",
.channels_min = 1,
.channels_max = 2,
.rates = TWL6040_RATES,
.formats = TWL6040_FORMATS,
},
.ops = &twl6040_dai_ops,
},
{
.name = "twl6040-ul",
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = TWL6040_RATES,
.formats = TWL6040_FORMATS,
},
.ops = &twl6040_dai_ops,
},
{
.name = "twl6040-dl1",
.playback = {
.stream_name = "Headset Playback",
.channels_min = 1,
.channels_max = 2,
.rates = TWL6040_RATES,
.formats = TWL6040_FORMATS,
},
.ops = &twl6040_dai_ops,
},
{
.name = "twl6040-dl2",
.playback = {
.stream_name = "Handsfree Playback",
.channels_min = 1,
.channels_max = 2,
.rates = TWL6040_RATES,
.formats = TWL6040_FORMATS,
},
.ops = &twl6040_dai_ops,
},
{
.name = "twl6040-vib",
.playback = {
.stream_name = "Vibra Playback",
.channels_min = 1,
.channels_max = 1,
.rates = SNDRV_PCM_RATE_CONTINUOUS,
.formats = TWL6040_FORMATS,
},
.ops = &twl6040_dai_ops,
},
};
#ifdef CONFIG_PM
static int twl6040_suspend(struct snd_soc_codec *codec)
{
twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
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>
2010-03-17 23:15:21 +03:00
static int twl6040_resume(struct snd_soc_codec *codec)
{
twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
twl6040_set_bias_level(codec, codec->dapm.suspend_bias_level);
return 0;
}
#else
#define twl6040_suspend NULL
#define twl6040_resume NULL
#endif
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>
2010-03-17 23:15:21 +03:00
static int twl6040_probe(struct snd_soc_codec *codec)
{
struct twl6040_data *priv;
struct twl6040_codec_data *pdata = dev_get_platdata(codec->dev);
struct platform_device *pdev = container_of(codec->dev,
struct platform_device, dev);
int ret = 0;
priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL);
if (priv == NULL)
return -ENOMEM;
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>
2010-03-17 23:15:21 +03:00
snd_soc_codec_set_drvdata(codec, priv);
priv->codec = codec;
codec->control_data = dev_get_drvdata(codec->dev->parent);
if (pdata && pdata->hs_left_step && pdata->hs_right_step) {
priv->hs_left_step = pdata->hs_left_step;
priv->hs_right_step = pdata->hs_right_step;
} else {
priv->hs_left_step = 1;
priv->hs_right_step = 1;
}
if (pdata && pdata->hf_left_step && pdata->hf_right_step) {
priv->hf_left_step = pdata->hf_left_step;
priv->hf_right_step = pdata->hf_right_step;
} else {
priv->hf_left_step = 1;
priv->hf_right_step = 1;
}
priv->plug_irq = platform_get_irq(pdev, 0);
if (priv->plug_irq < 0) {
dev_err(codec->dev, "invalid irq\n");
ret = -EINVAL;
goto work_err;
}
priv->workqueue = alloc_workqueue("twl6040-codec", 0, 0);
if (!priv->workqueue) {
ret = -ENOMEM;
goto work_err;
}
INIT_DELAYED_WORK(&priv->hs_jack.work, twl6040_accessory_work);
mutex_init(&priv->mutex);
ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler,
0, "twl6040_irq_plug", codec);
if (ret) {
dev_err(codec->dev, "PLUG IRQ request failed: %d\n", ret);
goto plugirq_err;
}
twl6040_init_chip(codec);
/* power on device */
ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
if (!ret)
return 0;
/* Error path */
free_irq(priv->plug_irq, codec);
plugirq_err:
destroy_workqueue(priv->workqueue);
work_err:
kfree(priv);
return ret;
}
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>
2010-03-17 23:15:21 +03:00
static int twl6040_remove(struct snd_soc_codec *codec)
{
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>
2010-03-17 23:15:21 +03:00
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
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>
2010-03-17 23:15:21 +03:00
twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF);
free_irq(priv->plug_irq, codec);
destroy_workqueue(priv->workqueue);
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>
2010-03-17 23:15:21 +03:00
kfree(priv);
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>
2010-03-17 23:15:21 +03:00
return 0;
}
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>
2010-03-17 23:15:21 +03:00
static struct snd_soc_codec_driver soc_codec_dev_twl6040 = {
.probe = twl6040_probe,
.remove = twl6040_remove,
.suspend = twl6040_suspend,
.resume = twl6040_resume,
.read = twl6040_read_reg_cache,
.write = twl6040_write,
.set_bias_level = twl6040_set_bias_level,
.reg_cache_size = ARRAY_SIZE(twl6040_reg),
.reg_word_size = sizeof(u8),
.reg_cache_default = twl6040_reg,
.ignore_pmdown_time = true,
.controls = twl6040_snd_controls,
.num_controls = ARRAY_SIZE(twl6040_snd_controls),
.dapm_widgets = twl6040_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
.dapm_routes = intercon,
.num_dapm_routes = ARRAY_SIZE(intercon),
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>
2010-03-17 23:15:21 +03:00
};
static int __devinit twl6040_codec_probe(struct platform_device *pdev)
{
return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_twl6040,
twl6040_dai, ARRAY_SIZE(twl6040_dai));
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>
2010-03-17 23:15:21 +03:00
}
static int __devexit twl6040_codec_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static struct platform_driver twl6040_codec_driver = {
.driver = {
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>
2010-03-17 23:15:21 +03:00
.name = "twl6040-codec",
.owner = THIS_MODULE,
},
.probe = twl6040_codec_probe,
.remove = __devexit_p(twl6040_codec_remove),
};
module_platform_driver(twl6040_codec_driver);
MODULE_DESCRIPTION("ASoC TWL6040 codec driver");
MODULE_AUTHOR("Misael Lopez Cruz");
MODULE_LICENSE("GPL");