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

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46 KiB
C
Исходник Обычный вид История

/*
* 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/i2c.h>
#include <linux/gpio.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <linux/i2c/twl.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include "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
#define TWL6040_RAMP_NONE 0
#define TWL6040_RAMP_UP 1
#define TWL6040_RAMP_DOWN 2
#define TWL6040_HSL_VOL_MASK 0x0F
#define TWL6040_HSL_VOL_SHIFT 0
#define TWL6040_HSR_VOL_MASK 0xF0
#define TWL6040_HSR_VOL_SHIFT 4
#define TWL6040_HF_VOL_MASK 0x1F
#define TWL6040_HF_VOL_SHIFT 0
struct twl6040_output {
u16 active;
u16 left_vol;
u16 right_vol;
u16 left_step;
u16 right_step;
unsigned int step_delay;
u16 ramp;
u16 mute;
struct completion ramp_done;
};
struct twl6040_jack_data {
struct snd_soc_jack *jack;
int report;
};
/* codec private data */
struct twl6040_data {
int audpwron;
int naudint;
int codec_powered;
int pll;
int non_lp;
unsigned int sysclk;
struct snd_pcm_hw_constraint_list *sysclk_constraints;
struct completion ready;
struct twl6040_jack_data hs_jack;
struct snd_soc_codec *codec;
struct workqueue_struct *workqueue;
struct delayed_work delayed_work;
struct mutex mutex;
struct twl6040_output headset;
struct twl6040_output handsfree;
struct workqueue_struct *hf_workqueue;
struct workqueue_struct *hs_workqueue;
struct delayed_work hs_delayed_work;
struct delayed_work hf_delayed_work;
};
/*
* twl6040 register cache & default register settings
*/
static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = {
0x00, /* not used 0x00 */
0x4B, /* TWL6040_ASICID (ro) 0x01 */
0x00, /* TWL6040_ASICREV (ro) 0x02 */
0x00, /* TWL6040_INTID 0x03 */
0x00, /* TWL6040_INTMR 0x04 */
0x00, /* TWL6040_NCPCTRL 0x05 */
0x00, /* TWL6040_LDOCTL 0x06 */
0x60, /* TWL6040_HPPLLCTL 0x07 */
0x00, /* TWL6040_LPPLLCTL 0x08 */
0x4A, /* TWL6040_LPPLLDIV 0x09 */
0x00, /* TWL6040_AMICBCTL 0x0A */
0x00, /* TWL6040_DMICBCTL 0x0B */
0x18, /* TWL6040_MICLCTL 0x0C - No input selected on Left Mic */
0x18, /* TWL6040_MICRCTL 0x0D - No input selected on Right Mic */
0x00, /* TWL6040_MICGAIN 0x0E */
0x1B, /* TWL6040_LINEGAIN 0x0F */
0x00, /* TWL6040_HSLCTL 0x10 */
0x00, /* TWL6040_HSRCTL 0x11 */
0x00, /* TWL6040_HSGAIN 0x12 */
0x00, /* TWL6040_EARCTL 0x13 */
0x00, /* TWL6040_HFLCTL 0x14 */
0x00, /* TWL6040_HFLGAIN 0x15 */
0x00, /* TWL6040_HFRCTL 0x16 */
0x00, /* TWL6040_HFRGAIN 0x17 */
0x00, /* TWL6040_VIBCTLL 0x18 */
0x00, /* TWL6040_VIBDATL 0x19 */
0x00, /* TWL6040_VIBCTLR 0x1A */
0x00, /* TWL6040_VIBDATR 0x1B */
0x00, /* TWL6040_HKCTL1 0x1C */
0x00, /* TWL6040_HKCTL2 0x1D */
0x00, /* TWL6040_GPOCTL 0x1E */
0x00, /* TWL6040_ALB 0x1F */
0x00, /* TWL6040_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, /* TWL6040_TRIM1 0x28 */
0x00, /* TWL6040_TRIM2 0x29 */
0x00, /* TWL6040_TRIM3 0x2A */
0x00, /* TWL6040_HSOTRIM 0x2B */
0x00, /* TWL6040_HFOTRIM 0x2C */
0x09, /* TWL6040_ACCCTL 0x2D */
0x00, /* TWL6040_STATUS (ro) 0x2E */
};
/*
* twl6040 vio/gnd registers:
* registers under vio/gnd supply can be accessed
* before the power-up sequence, after NRESPWRON goes high
*/
static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = {
TWL6040_REG_ASICID,
TWL6040_REG_ASICREV,
TWL6040_REG_INTID,
TWL6040_REG_INTMR,
TWL6040_REG_NCPCTL,
TWL6040_REG_LDOCTL,
TWL6040_REG_AMICBCTL,
TWL6040_REG_DMICBCTL,
TWL6040_REG_HKCTL1,
TWL6040_REG_HKCTL2,
TWL6040_REG_GPOCTL,
TWL6040_REG_TRIM1,
TWL6040_REG_TRIM2,
TWL6040_REG_TRIM3,
TWL6040_REG_HSOTRIM,
TWL6040_REG_HFOTRIM,
TWL6040_REG_ACCCTL,
TWL6040_REG_STATUS,
};
/*
* twl6040 vdd/vss registers:
* registers under vdd/vss supplies can only be accessed
* after the power-up sequence
*/
static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = {
TWL6040_REG_HPPLLCTL,
TWL6040_REG_LPPLLCTL,
TWL6040_REG_LPPLLDIV,
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,
TWL6040_REG_VIBCTLL,
TWL6040_REG_VIBDATL,
TWL6040_REG_VIBCTLR,
TWL6040_REG_VIBDATR,
TWL6040_REG_ALB,
TWL6040_REG_DLB,
};
/*
* 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)
{
u8 value;
if (reg >= TWL6040_CACHEREGNUM)
return -EIO;
twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &value, reg);
twl6040_write_reg_cache(codec, reg, value);
return value;
}
/*
* write to the twl6040 register space
*/
static int twl6040_write(struct snd_soc_codec *codec,
unsigned int reg, unsigned int value)
{
if (reg >= TWL6040_CACHEREGNUM)
return -EIO;
twl6040_write_reg_cache(codec, reg, value);
return twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, value, reg);
}
static void twl6040_init_vio_regs(struct snd_soc_codec *codec)
{
u8 *cache = codec->reg_cache;
int reg, i;
/* allow registers to be accessed by i2c */
twl6040_write(codec, TWL6040_REG_ACCCTL, cache[TWL6040_REG_ACCCTL]);
for (i = 0; i < TWL6040_VIOREGNUM; i++) {
reg = twl6040_vio_reg[i];
/* skip read-only registers (ASICID, ASICREV, STATUS) */
switch (reg) {
case TWL6040_REG_ASICID:
case TWL6040_REG_ASICREV:
case TWL6040_REG_STATUS:
continue;
default:
break;
}
twl6040_write(codec, reg, cache[reg]);
}
}
static void twl6040_init_vdd_regs(struct snd_soc_codec *codec)
{
u8 *cache = codec->reg_cache;
int reg, i;
for (i = 0; i < TWL6040_VDDREGNUM; i++) {
reg = twl6040_vdd_reg[i];
twl6040_write(codec, reg, cache[reg]);
}
}
/*
* Ramp HS PGA volume to minimise pops at stream startup and shutdown.
*/
static inline int twl6040_hs_ramp_step(struct snd_soc_codec *codec,
unsigned int left_step, unsigned int right_step)
{
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
struct twl6040_output *headset = &priv->headset;
int left_complete = 0, right_complete = 0;
u8 reg, val;
/* left channel */
left_step = (left_step > 0xF) ? 0xF : left_step;
reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN);
val = (~reg & TWL6040_HSL_VOL_MASK);
if (headset->ramp == TWL6040_RAMP_UP) {
/* ramp step up */
if (val < headset->left_vol) {
val += left_step;
reg &= ~TWL6040_HSL_VOL_MASK;
twl6040_write(codec, TWL6040_REG_HSGAIN,
(reg | (~val & TWL6040_HSL_VOL_MASK)));
} else {
left_complete = 1;
}
} else if (headset->ramp == TWL6040_RAMP_DOWN) {
/* ramp step down */
if (val > 0x0) {
val -= left_step;
reg &= ~TWL6040_HSL_VOL_MASK;
twl6040_write(codec, TWL6040_REG_HSGAIN, reg |
(~val & TWL6040_HSL_VOL_MASK));
} else {
left_complete = 1;
}
}
/* right channel */
right_step = (right_step > 0xF) ? 0xF : right_step;
reg = twl6040_read_reg_cache(codec, TWL6040_REG_HSGAIN);
val = (~reg & TWL6040_HSR_VOL_MASK) >> TWL6040_HSR_VOL_SHIFT;
if (headset->ramp == TWL6040_RAMP_UP) {
/* ramp step up */
if (val < headset->right_vol) {
val += right_step;
reg &= ~TWL6040_HSR_VOL_MASK;
twl6040_write(codec, TWL6040_REG_HSGAIN,
(reg | (~val << TWL6040_HSR_VOL_SHIFT)));
} else {
right_complete = 1;
}
} else if (headset->ramp == TWL6040_RAMP_DOWN) {
/* ramp step down */
if (val > 0x0) {
val -= right_step;
reg &= ~TWL6040_HSR_VOL_MASK;
twl6040_write(codec, TWL6040_REG_HSGAIN,
reg | (~val << TWL6040_HSR_VOL_SHIFT));
} else {
right_complete = 1;
}
}
return left_complete & right_complete;
}
/*
* Ramp HF PGA volume to minimise pops at stream startup and shutdown.
*/
static inline int twl6040_hf_ramp_step(struct snd_soc_codec *codec,
unsigned int left_step, unsigned int right_step)
{
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
struct twl6040_output *handsfree = &priv->handsfree;
int left_complete = 0, right_complete = 0;
u16 reg, val;
/* left channel */
left_step = (left_step > 0x1D) ? 0x1D : left_step;
reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFLGAIN);
reg = 0x1D - reg;
val = (reg & TWL6040_HF_VOL_MASK);
if (handsfree->ramp == TWL6040_RAMP_UP) {
/* ramp step up */
if (val < handsfree->left_vol) {
val += left_step;
reg &= ~TWL6040_HF_VOL_MASK;
twl6040_write(codec, TWL6040_REG_HFLGAIN,
reg | (0x1D - val));
} else {
left_complete = 1;
}
} else if (handsfree->ramp == TWL6040_RAMP_DOWN) {
/* ramp step down */
if (val > 0) {
val -= left_step;
reg &= ~TWL6040_HF_VOL_MASK;
twl6040_write(codec, TWL6040_REG_HFLGAIN,
reg | (0x1D - val));
} else {
left_complete = 1;
}
}
/* right channel */
right_step = (right_step > 0x1D) ? 0x1D : right_step;
reg = twl6040_read_reg_cache(codec, TWL6040_REG_HFRGAIN);
reg = 0x1D - reg;
val = (reg & TWL6040_HF_VOL_MASK);
if (handsfree->ramp == TWL6040_RAMP_UP) {
/* ramp step up */
if (val < handsfree->right_vol) {
val += right_step;
reg &= ~TWL6040_HF_VOL_MASK;
twl6040_write(codec, TWL6040_REG_HFRGAIN,
reg | (0x1D - val));
} else {
right_complete = 1;
}
} else if (handsfree->ramp == TWL6040_RAMP_DOWN) {
/* ramp step down */
if (val > 0) {
val -= right_step;
reg &= ~TWL6040_HF_VOL_MASK;
twl6040_write(codec, TWL6040_REG_HFRGAIN,
reg | (0x1D - val));
}
}
return left_complete & right_complete;
}
/*
* This work ramps both output PGAs at stream start/stop time to
* minimise pop associated with DAPM power switching.
*/
static void twl6040_pga_hs_work(struct work_struct *work)
{
struct twl6040_data *priv =
container_of(work, struct twl6040_data, hs_delayed_work.work);
struct snd_soc_codec *codec = priv->codec;
struct twl6040_output *headset = &priv->headset;
unsigned int delay = headset->step_delay;
int i, headset_complete;
/* do we need to ramp at all ? */
if (headset->ramp == TWL6040_RAMP_NONE)
return;
/* HS PGA volumes have 4 bits of resolution to ramp */
for (i = 0; i <= 16; i++) {
headset_complete = 1;
if (headset->ramp != TWL6040_RAMP_NONE)
headset_complete = twl6040_hs_ramp_step(codec,
headset->left_step,
headset->right_step);
/* ramp finished ? */
if (headset_complete)
break;
/*
* TODO: tune: delay is longer over 0dB
* as increases are larger.
*/
if (i >= 8)
schedule_timeout_interruptible(msecs_to_jiffies(delay +
(delay >> 1)));
else
schedule_timeout_interruptible(msecs_to_jiffies(delay));
}
if (headset->ramp == TWL6040_RAMP_DOWN) {
headset->active = 0;
complete(&headset->ramp_done);
} else {
headset->active = 1;
}
headset->ramp = TWL6040_RAMP_NONE;
}
static void twl6040_pga_hf_work(struct work_struct *work)
{
struct twl6040_data *priv =
container_of(work, struct twl6040_data, hf_delayed_work.work);
struct snd_soc_codec *codec = priv->codec;
struct twl6040_output *handsfree = &priv->handsfree;
unsigned int delay = handsfree->step_delay;
int i, handsfree_complete;
/* do we need to ramp at all ? */
if (handsfree->ramp == TWL6040_RAMP_NONE)
return;
/* HF PGA volumes have 5 bits of resolution to ramp */
for (i = 0; i <= 32; i++) {
handsfree_complete = 1;
if (handsfree->ramp != TWL6040_RAMP_NONE)
handsfree_complete = twl6040_hf_ramp_step(codec,
handsfree->left_step,
handsfree->right_step);
/* ramp finished ? */
if (handsfree_complete)
break;
/*
* TODO: tune: delay is longer over 0dB
* as increases are larger.
*/
if (i >= 16)
schedule_timeout_interruptible(msecs_to_jiffies(delay +
(delay >> 1)));
else
schedule_timeout_interruptible(msecs_to_jiffies(delay));
}
if (handsfree->ramp == TWL6040_RAMP_DOWN) {
handsfree->active = 0;
complete(&handsfree->ramp_done);
} else
handsfree->active = 1;
handsfree->ramp = TWL6040_RAMP_NONE;
}
static int pga_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);
struct twl6040_output *out;
struct delayed_work *work;
struct workqueue_struct *queue;
switch (w->shift) {
case 2:
case 3:
out = &priv->headset;
work = &priv->hs_delayed_work;
queue = priv->hs_workqueue;
out->step_delay = 5; /* 5 ms between volume ramp steps */
break;
case 4:
out = &priv->handsfree;
work = &priv->hf_delayed_work;
queue = priv->hf_workqueue;
out->step_delay = 5; /* 5 ms between volume ramp steps */
if (SND_SOC_DAPM_EVENT_ON(event))
priv->non_lp++;
else
priv->non_lp--;
break;
default:
return -1;
}
switch (event) {
case SND_SOC_DAPM_POST_PMU:
if (out->active)
break;
/* don't use volume ramp for power-up */
out->left_step = out->left_vol;
out->right_step = out->right_vol;
if (!delayed_work_pending(work)) {
out->ramp = TWL6040_RAMP_UP;
queue_delayed_work(queue, work,
msecs_to_jiffies(1));
}
break;
case SND_SOC_DAPM_PRE_PMD:
if (!out->active)
break;
if (!delayed_work_pending(work)) {
/* use volume ramp for power-down */
out->left_step = 1;
out->right_step = 1;
out->ramp = TWL6040_RAMP_DOWN;
INIT_COMPLETION(out->ramp_done);
queue_delayed_work(queue, work,
msecs_to_jiffies(1));
wait_for_completion_timeout(&out->ramp_done,
msecs_to_jiffies(2000));
}
break;
}
return 0;
}
/* twl6040 codec manual power-up sequence */
static void twl6040_power_up(struct snd_soc_codec *codec)
{
u8 ncpctl, ldoctl, lppllctl, accctl;
ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL);
ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL);
lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL);
accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL);
/* enable reference system */
ldoctl |= TWL6040_REFENA;
twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
msleep(10);
/* enable internal oscillator */
ldoctl |= TWL6040_OSCENA;
twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
udelay(10);
/* enable high-side ldo */
ldoctl |= TWL6040_HSLDOENA;
twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
udelay(244);
/* enable negative charge pump */
ncpctl |= TWL6040_NCPENA | TWL6040_NCPOPEN;
twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl);
udelay(488);
/* enable low-side ldo */
ldoctl |= TWL6040_LSLDOENA;
twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
udelay(244);
/* enable low-power pll */
lppllctl |= TWL6040_LPLLENA;
twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
/* reset state machine */
accctl |= TWL6040_RESETSPLIT;
twl6040_write(codec, TWL6040_REG_ACCCTL, accctl);
mdelay(5);
accctl &= ~TWL6040_RESETSPLIT;
twl6040_write(codec, TWL6040_REG_ACCCTL, accctl);
/* disable internal oscillator */
ldoctl &= ~TWL6040_OSCENA;
twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
}
/* twl6040 codec manual power-down sequence */
static void twl6040_power_down(struct snd_soc_codec *codec)
{
u8 ncpctl, ldoctl, lppllctl, accctl;
ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL);
ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL);
lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL);
accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL);
/* enable internal oscillator */
ldoctl |= TWL6040_OSCENA;
twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
udelay(10);
/* disable low-power pll */
lppllctl &= ~TWL6040_LPLLENA;
twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
/* disable low-side ldo */
ldoctl &= ~TWL6040_LSLDOENA;
twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
udelay(244);
/* disable negative charge pump */
ncpctl &= ~(TWL6040_NCPENA | TWL6040_NCPOPEN);
twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl);
udelay(488);
/* disable high-side ldo */
ldoctl &= ~TWL6040_HSLDOENA;
twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
udelay(244);
/* disable internal oscillator */
ldoctl &= ~TWL6040_OSCENA;
twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
/* disable reference system */
ldoctl &= ~TWL6040_REFENA;
twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
msleep(10);
}
/* 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_HSDRVMODEL | TWL6040_HSDACMODEL;
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)
{
msleep(1);
return 0;
}
static int twl6040_power_mode_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);
if (SND_SOC_DAPM_EVENT_ON(event))
priv->non_lp++;
else
priv->non_lp--;
msleep(1);
return 0;
}
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, delayed_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_naudint_handler(int irq, void *data)
{
struct snd_soc_codec *codec = data;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
u8 intid;
twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID);
if (intid & TWL6040_THINT)
dev_alert(codec->dev, "die temp over-limit detection\n");
if ((intid & TWL6040_PLUGINT) || (intid & TWL6040_UNPLUGINT))
queue_delayed_work(priv->workqueue, &priv->delayed_work,
msecs_to_jiffies(200));
if (intid & TWL6040_HOOKINT)
dev_info(codec->dev, "hook detection\n");
if (intid & TWL6040_HFINT)
dev_alert(codec->dev, "hf drivers over current detection\n");
if (intid & TWL6040_VIBINT)
dev_alert(codec->dev, "vib drivers over current detection\n");
if (intid & TWL6040_READYINT)
complete(&priv->ready);
return IRQ_HANDLED;
}
static int twl6040_put_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec);
struct twl6040_output *out = NULL;
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
int ret;
unsigned int reg = mc->reg;
/* For HS and HF we shadow the values and only actually write
* them out when active in order to ensure the amplifier comes on
* as quietly as possible. */
switch (reg) {
case TWL6040_REG_HSGAIN:
out = &twl6040_priv->headset;
break;
default:
break;
}
if (out) {
out->left_vol = ucontrol->value.integer.value[0];
out->right_vol = ucontrol->value.integer.value[1];
if (!out->active)
return 1;
}
ret = snd_soc_put_volsw(kcontrol, ucontrol);
if (ret < 0)
return ret;
return 1;
}
static int twl6040_get_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec);
struct twl6040_output *out = &twl6040_priv->headset;
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int reg = mc->reg;
switch (reg) {
case TWL6040_REG_HSGAIN:
out = &twl6040_priv->headset;
ucontrol->value.integer.value[0] = out->left_vol;
ucontrol->value.integer.value[1] = out->right_vol;
return 0;
default:
break;
}
return snd_soc_get_volsw(kcontrol, ucontrol);
}
static int twl6040_put_volsw_2r_vu(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec);
struct twl6040_output *out = NULL;
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
int ret;
unsigned int reg = mc->reg;
/* For HS and HF we shadow the values and only actually write
* them out when active in order to ensure the amplifier comes on
* as quietly as possible. */
switch (reg) {
case TWL6040_REG_HFLGAIN:
case TWL6040_REG_HFRGAIN:
out = &twl6040_priv->handsfree;
break;
default:
break;
}
if (out) {
out->left_vol = ucontrol->value.integer.value[0];
out->right_vol = ucontrol->value.integer.value[1];
if (!out->active)
return 1;
}
ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
if (ret < 0)
return ret;
return 1;
}
static int twl6040_get_volsw_2r(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl6040_data *twl6040_priv = snd_soc_codec_get_drvdata(codec);
struct twl6040_output *out = &twl6040_priv->handsfree;
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int reg = mc->reg;
/* If these are cached registers use the cache */
switch (reg) {
case TWL6040_REG_HFLGAIN:
case TWL6040_REG_HFRGAIN:
out = &twl6040_priv->handsfree;
ucontrol->value.integer.value[0] = out->left_vol;
ucontrol->value.integer.value[1] = out->right_vol;
return 0;
default:
break;
}
return snd_soc_get_volsw_2r(kcontrol, ucontrol);
}
/* double control with volume update */
#define SOC_TWL6040_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax,\
xinvert, tlv_array)\
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw, .get = twl6040_get_volsw, \
.put = twl6040_put_volsw, \
.private_value = (unsigned long)&(struct soc_mixer_control) \
{.reg = xreg, .shift = shift_left, .rshift = shift_right,\
.max = xmax, .platform_max = xmax, .invert = xinvert} }
/* double control with volume update */
#define SOC_TWL6040_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax,\
xinvert, tlv_array)\
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
SNDRV_CTL_ELEM_ACCESS_READWRITE | \
SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw_2r, \
.get = twl6040_get_volsw_2r, .put = twl6040_put_volsw_2r_vu, \
.private_value = (unsigned long)&(struct soc_mixer_control) \
{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
.rshift = xshift, .max = xmax, .invert = xinvert}, }
/*
* 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 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_driver_switch_controls =
SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0);
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, 4, 0xF, 0, afm_amp_tlv),
/* Playback gains */
SOC_TWL6040_DOUBLE_TLV("Headset Playback Volume",
TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv),
SOC_TWL6040_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),
};
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"),
/* 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", "Left Front Capture",
TWL6040_REG_MICLCTL, 2, 0),
SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture",
TWL6040_REG_MICRCTL, 2, 0),
/* Microphone bias */
SND_SOC_DAPM_MICBIAS("Headset Mic Bias",
TWL6040_REG_AMICBCTL, 0, 0),
SND_SOC_DAPM_MICBIAS("Main Mic Bias",
TWL6040_REG_AMICBCTL, 4, 0),
SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias",
TWL6040_REG_DMICBCTL, 0, 0),
SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias",
TWL6040_REG_DMICBCTL, 4, 0),
/* DACs */
SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback",
TWL6040_REG_HSLCTL, 0, 0,
twl6040_hs_dac_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback",
TWL6040_REG_HSRCTL, 0, 0,
twl6040_hs_dac_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback",
TWL6040_REG_HFLCTL, 0, 0,
twl6040_power_mode_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback",
TWL6040_REG_HFRCTL, 0, 0,
twl6040_power_mode_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MUX("HF Left Playback",
SND_SOC_NOPM, 0, 0, &hfl_mux_controls),
SND_SOC_DAPM_MUX("HF Right Playback",
SND_SOC_NOPM, 0, 0, &hfr_mux_controls),
/* Analog playback Muxes */
SND_SOC_DAPM_MUX("HS Left Playback",
SND_SOC_NOPM, 0, 0, &hsl_mux_controls),
SND_SOC_DAPM_MUX("HS Right Playback",
SND_SOC_NOPM, 0, 0, &hsr_mux_controls),
/* Analog playback drivers */
SND_SOC_DAPM_OUT_DRV_E("Handsfree Left Driver",
TWL6040_REG_HFLCTL, 4, 0, NULL, 0,
pga_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_OUT_DRV_E("Handsfree Right Driver",
TWL6040_REG_HFRCTL, 4, 0, NULL, 0,
pga_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_OUT_DRV_E("Headset Left Driver",
TWL6040_REG_HSLCTL, 2, 0, NULL, 0,
pga_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_OUT_DRV_E("Headset Right Driver",
TWL6040_REG_HSRCTL, 2, 0, NULL, 0,
pga_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_SWITCH_E("Earphone Driver",
SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls,
twl6040_power_mode_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
/* Analog playback PGAs */
SND_SOC_DAPM_PGA("HFDAC Left PGA",
TWL6040_REG_HFLCTL, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("HFDAC Right PGA",
TWL6040_REG_HFRCTL, 1, 0, NULL, 0),
};
static const struct snd_soc_dapm_route intercon[] = {
/* 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"},
{"HS Left Playback", "HS DAC", "HSDAC Left"},
{"HS Left Playback", "Line-In amp", "AFMAmpL"},
{"HS Right Playback", "HS DAC", "HSDAC Right"},
{"HS Right Playback", "Line-In amp", "AFMAmpR"},
{"Headset Left Driver", "NULL", "HS Left Playback"},
{"Headset Right Driver", "NULL", "HS Right Playback"},
{"HSOL", NULL, "Headset Left Driver"},
{"HSOR", NULL, "Headset Right Driver"},
/* Earphone playback path */
{"Earphone Driver", "Switch", "HSDAC Left"},
{"EP", NULL, "Earphone Driver"},
{"HF Left Playback", "HF DAC", "HFDAC Left"},
{"HF Left Playback", "Line-In amp", "AFMAmpL"},
{"HF Right Playback", "HF DAC", "HFDAC Right"},
{"HF Right Playback", "Line-In amp", "AFMAmpR"},
{"HFDAC Left PGA", NULL, "HF Left Playback"},
{"HFDAC Right PGA", NULL, "HF Right Playback"},
{"Handsfree Left Driver", "Switch", "HFDAC Left PGA"},
{"Handsfree Right Driver", "Switch", "HFDAC Right PGA"},
{"HFL", NULL, "Handsfree Left Driver"},
{"HFR", NULL, "Handsfree Right Driver"},
};
static int twl6040_add_widgets(struct snd_soc_codec *codec)
{
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
struct snd_soc_dapm_context *dapm = &codec->dapm;
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
snd_soc_dapm_new_controls(dapm, twl6040_dapm_widgets,
ARRAY_SIZE(twl6040_dapm_widgets));
snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
snd_soc_dapm_new_widgets(dapm);
return 0;
}
static int twl6040_power_up_completion(struct snd_soc_codec *codec,
int naudint)
{
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int time_left;
u8 intid;
time_left = wait_for_completion_timeout(&priv->ready,
msecs_to_jiffies(144));
if (!time_left) {
twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &intid,
TWL6040_REG_INTID);
if (!(intid & TWL6040_READYINT)) {
dev_err(codec->dev, "timeout waiting for READYINT\n");
return -ETIMEDOUT;
}
}
priv->codec_powered = 1;
return 0;
}
static int twl6040_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int audpwron = priv->audpwron;
int naudint = priv->naudint;
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;
if (gpio_is_valid(audpwron)) {
/* use AUDPWRON line */
gpio_set_value(audpwron, 1);
/* wait for power-up completion */
ret = twl6040_power_up_completion(codec, naudint);
if (ret)
return ret;
/* sync registers updated during power-up sequence */
twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL);
twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL);
twl6040_read_reg_volatile(codec, TWL6040_REG_LPPLLCTL);
} else {
/* use manual power-up sequence */
twl6040_power_up(codec);
priv->codec_powered = 1;
}
/* initialize vdd/vss registers with reg_cache */
twl6040_init_vdd_regs(codec);
/* Set external boost GPO */
twl6040_write(codec, TWL6040_REG_GPOCTL, 0x02);
/* Set initial minimal gain values */
twl6040_write(codec, TWL6040_REG_HSGAIN, 0xFF);
twl6040_write(codec, TWL6040_REG_EARCTL, 0x1E);
twl6040_write(codec, TWL6040_REG_HFLGAIN, 0x1D);
twl6040_write(codec, TWL6040_REG_HFRGAIN, 0x1D);
break;
case SND_SOC_BIAS_OFF:
if (!priv->codec_powered)
break;
if (gpio_is_valid(audpwron)) {
/* use AUDPWRON line */
gpio_set_value(audpwron, 0);
/* power-down sequence latency */
udelay(500);
/* sync registers updated during power-down sequence */
twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL);
twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL);
twl6040_write_reg_cache(codec, TWL6040_REG_LPPLLCTL,
0x00);
} else {
/* use manual power-down sequence */
twl6040_power_down(codec);
}
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;
}
/* set of rates for each pll: low-power and high-performance */
static unsigned int lp_rates[] = {
88200,
96000,
};
static struct snd_pcm_hw_constraint_list lp_constraints = {
.count = ARRAY_SIZE(lp_rates),
.list = lp_rates,
};
static unsigned int hp_rates[] = {
96000,
};
static struct snd_pcm_hw_constraint_list hp_constraints = {
.count = ARRAY_SIZE(hp_rates),
.list = hp_rates,
};
static int twl6040_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
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 snd_soc_codec *codec = rtd->codec;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
priv->sysclk_constraints);
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_pcm_runtime *rtd = substream->private_data;
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 snd_soc_codec *codec = rtd->codec;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
u8 lppllctl;
int rate;
/* nothing to do for high-perf pll, it supports only 48 kHz */
if (priv->pll == TWL6040_HPPLL_ID)
return 0;
lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL);
rate = params_rate(params);
switch (rate) {
case 11250:
case 22500:
case 44100:
case 88200:
lppllctl |= TWL6040_LPLLFIN;
priv->sysclk = 17640000;
break;
case 8000:
case 16000:
case 32000:
case 48000:
case 96000:
lppllctl &= ~TWL6040_LPLLFIN;
priv->sysclk = 19200000;
break;
default:
dev_err(codec->dev, "unsupported rate %d\n", rate);
return -EINVAL;
}
twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
return 0;
}
static int twl6040_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
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 snd_soc_codec *codec = rtd->codec;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
if (!priv->sysclk) {
dev_err(codec->dev,
"no mclk configured, call set_sysclk() on init\n");
return -EINVAL;
}
/*
* capture is not supported at 17.64 MHz,
* it's reserved for headset low-power playback scenario
*/
if ((priv->sysclk == 17640000) &&
substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
dev_err(codec->dev,
"capture mode is not supported at %dHz\n",
priv->sysclk);
return -EINVAL;
}
if ((priv->sysclk == 17640000) && priv->non_lp) {
dev_err(codec->dev,
"some enabled paths aren't supported at %dHz\n",
priv->sysclk);
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);
u8 hppllctl, lppllctl;
hppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_HPPLLCTL);
lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL);
switch (clk_id) {
case TWL6040_SYSCLK_SEL_LPPLL:
switch (freq) {
case 32768:
/* headset dac and driver must be in low-power mode */
headset_power_mode(codec, 0);
/* clk32k input requires low-power pll */
lppllctl |= TWL6040_LPLLENA;
twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
mdelay(5);
lppllctl &= ~TWL6040_HPLLSEL;
twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
hppllctl &= ~TWL6040_HPLLENA;
twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl);
break;
default:
dev_err(codec->dev, "unknown mclk freq %d\n", freq);
return -EINVAL;
}
/* lppll divider */
switch (priv->sysclk) {
case 17640000:
lppllctl |= TWL6040_LPLLFIN;
break;
case 19200000:
lppllctl &= ~TWL6040_LPLLFIN;
break;
default:
/* sysclk not yet configured */
lppllctl &= ~TWL6040_LPLLFIN;
priv->sysclk = 19200000;
break;
}
twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
priv->pll = TWL6040_LPPLL_ID;
priv->sysclk_constraints = &lp_constraints;
break;
case TWL6040_SYSCLK_SEL_HPPLL:
hppllctl &= ~TWL6040_MCLK_MSK;
switch (freq) {
case 12000000:
/* mclk input, pll enabled */
hppllctl |= TWL6040_MCLK_12000KHZ |
TWL6040_HPLLSQRBP |
TWL6040_HPLLENA;
break;
case 19200000:
/* mclk input, pll disabled */
hppllctl |= TWL6040_MCLK_19200KHZ |
TWL6040_HPLLSQRENA |
TWL6040_HPLLBP;
break;
case 26000000:
/* mclk input, pll enabled */
hppllctl |= TWL6040_MCLK_26000KHZ |
TWL6040_HPLLSQRBP |
TWL6040_HPLLENA;
break;
case 38400000:
/* clk slicer, pll disabled */
hppllctl |= TWL6040_MCLK_38400KHZ |
TWL6040_HPLLSQRENA |
TWL6040_HPLLBP;
break;
default:
dev_err(codec->dev, "unknown mclk freq %d\n", freq);
return -EINVAL;
}
/* headset dac and driver must be in high-performance mode */
headset_power_mode(codec, 1);
twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl);
udelay(500);
lppllctl |= TWL6040_HPLLSEL;
twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
lppllctl &= ~TWL6040_LPLLENA;
twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
/* high-performance pll can provide only 19.2 MHz */
priv->pll = TWL6040_HPPLL_ID;
priv->sysclk = 19200000;
priv->sysclk_constraints = &hp_constraints;
break;
default:
dev_err(codec->dev, "unknown clk_id %d\n", clk_id);
return -EINVAL;
}
return 0;
}
static 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,
};
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_dai_driver twl6040_dai = {
.name = "twl6040-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 4,
.rates = TWL6040_RATES,
.formats = TWL6040_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = TWL6040_RATES,
.formats = TWL6040_FORMATS,
},
.ops = &twl6040_dai_ops,
};
#ifdef CONFIG_PM
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_suspend(struct snd_soc_codec *codec, pm_message_t state)
{
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)
{
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 twl4030_codec_data *twl_codec = codec->dev->platform_data;
struct twl6040_data *priv;
int audpwron, naudint;
int ret = 0;
u8 icrev, intmr = TWL6040_ALLINT_MSK;
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;
twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &icrev, TWL6040_REG_ASICREV);
if (twl_codec && (icrev > 0))
audpwron = twl_codec->audpwron_gpio;
else
audpwron = -EINVAL;
if (twl_codec)
naudint = twl_codec->naudint_irq;
else
naudint = 0;
priv->audpwron = audpwron;
priv->naudint = naudint;
priv->workqueue = create_singlethread_workqueue("twl6040-codec");
if (!priv->workqueue) {
ret = -ENOMEM;
goto work_err;
}
INIT_DELAYED_WORK(&priv->delayed_work, twl6040_accessory_work);
mutex_init(&priv->mutex);
init_completion(&priv->ready);
init_completion(&priv->headset.ramp_done);
init_completion(&priv->handsfree.ramp_done);
if (gpio_is_valid(audpwron)) {
ret = gpio_request(audpwron, "audpwron");
if (ret)
goto gpio1_err;
ret = gpio_direction_output(audpwron, 0);
if (ret)
goto gpio2_err;
priv->codec_powered = 0;
/* enable only codec ready interrupt */
intmr &= ~(TWL6040_READYMSK | TWL6040_PLUGMSK);
/* reset interrupt status to allow correct power up sequence */
twl6040_read_reg_volatile(codec, TWL6040_REG_INTID);
}
twl6040_write(codec, TWL6040_REG_INTMR, intmr);
if (naudint) {
/* audio interrupt */
ret = request_threaded_irq(naudint, NULL,
twl6040_naudint_handler,
IRQF_TRIGGER_LOW | IRQF_ONESHOT,
"twl6040_codec", codec);
if (ret)
goto gpio2_err;
}
/* init vio registers */
twl6040_init_vio_regs(codec);
priv->hf_workqueue = create_singlethread_workqueue("twl6040-hf");
if (priv->hf_workqueue == NULL) {
ret = -ENOMEM;
goto irq_err;
}
priv->hs_workqueue = create_singlethread_workqueue("twl6040-hs");
if (priv->hs_workqueue == NULL) {
ret = -ENOMEM;
goto wq_err;
}
INIT_DELAYED_WORK(&priv->hs_delayed_work, twl6040_pga_hs_work);
INIT_DELAYED_WORK(&priv->hf_delayed_work, twl6040_pga_hf_work);
/* power on device */
ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
if (ret)
goto bias_err;
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_add_controls(codec, twl6040_snd_controls,
ARRAY_SIZE(twl6040_snd_controls));
twl6040_add_widgets(codec);
return 0;
bias_err:
destroy_workqueue(priv->hs_workqueue);
wq_err:
destroy_workqueue(priv->hf_workqueue);
irq_err:
if (naudint)
free_irq(naudint, codec);
gpio2_err:
if (gpio_is_valid(audpwron))
gpio_free(audpwron);
gpio1_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);
int audpwron = priv->audpwron;
int naudint = priv->naudint;
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);
if (gpio_is_valid(audpwron))
gpio_free(audpwron);
if (naudint)
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
free_irq(naudint, codec);
destroy_workqueue(priv->workqueue);
destroy_workqueue(priv->hf_workqueue);
destroy_workqueue(priv->hs_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,
};
static int __devinit twl6040_codec_probe(struct platform_device *pdev)
{
return snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_twl6040, &twl6040_dai, 1);
}
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),
};
static int __init twl6040_codec_init(void)
{
return platform_driver_register(&twl6040_codec_driver);
}
module_init(twl6040_codec_init);
static void __exit twl6040_codec_exit(void)
{
platform_driver_unregister(&twl6040_codec_driver);
}
module_exit(twl6040_codec_exit);
MODULE_DESCRIPTION("ASoC TWL6040 codec driver");
MODULE_AUTHOR("Misael Lopez Cruz");
MODULE_LICENSE("GPL");