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

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C
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/*
* ALSA SoC TWL6040 codec driver
*
* Author: Misael Lopez Cruz <x0052729@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* version 2 as published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <linux/i2c/twl.h>
#include <linux/mfd/twl6040.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/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 plug_irq;
int codec_powered;
int pll;
int non_lp;
int pll_power_mode;
int hs_power_mode;
int hs_power_mode_locked;
unsigned int clk_in;
unsigned int sysclk;
u16 hs_left_step;
u16 hs_right_step;
u16 hf_left_step;
u16 hf_right_step;
struct twl6040_jack_data hs_jack;
struct snd_soc_codec *codec;
struct workqueue_struct *workqueue;
struct 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, /* REG_ASICID 0x01 (ro) */
0x00, /* REG_ASICREV 0x02 (ro) */
0x00, /* REG_INTID 0x03 */
0x00, /* REG_INTMR 0x04 */
0x00, /* REG_NCPCTRL 0x05 */
0x00, /* REG_LDOCTL 0x06 */
0x60, /* REG_HPPLLCTL 0x07 */
0x00, /* REG_LPPLLCTL 0x08 */
0x4A, /* REG_LPPLLDIV 0x09 */
0x00, /* REG_AMICBCTL 0x0A */
0x00, /* REG_DMICBCTL 0x0B */
0x00, /* REG_MICLCTL 0x0C */
0x00, /* REG_MICRCTL 0x0D */
0x00, /* REG_MICGAIN 0x0E */
0x1B, /* REG_LINEGAIN 0x0F */
0x00, /* REG_HSLCTL 0x10 */
0x00, /* REG_HSRCTL 0x11 */
0x00, /* REG_HSGAIN 0x12 */
0x00, /* REG_EARCTL 0x13 */
0x00, /* REG_HFLCTL 0x14 */
0x00, /* REG_HFLGAIN 0x15 */
0x00, /* REG_HFRCTL 0x16 */
0x00, /* REG_HFRGAIN 0x17 */
0x00, /* REG_VIBCTLL 0x18 */
0x00, /* REG_VIBDATL 0x19 */
0x00, /* REG_VIBCTLR 0x1A */
0x00, /* REG_VIBDATR 0x1B */
0x00, /* REG_HKCTL1 0x1C */
0x00, /* REG_HKCTL2 0x1D */
0x00, /* REG_GPOCTL 0x1E */
0x00, /* REG_ALB 0x1F */
0x00, /* REG_DLB 0x20 */
0x00, /* not used 0x21 */
0x00, /* not used 0x22 */
0x00, /* not used 0x23 */
0x00, /* not used 0x24 */
0x00, /* not used 0x25 */
0x00, /* not used 0x26 */
0x00, /* not used 0x27 */
0x00, /* REG_TRIM1 0x28 */
0x00, /* REG_TRIM2 0x29 */
0x00, /* REG_TRIM3 0x2A */
0x00, /* REG_HSOTRIM 0x2B */
0x00, /* REG_HFOTRIM 0x2C */
0x09, /* REG_ACCCTL 0x2D */
0x00, /* REG_STATUS 0x2E (ro) */
};
/* List of registers to be restored after power up */
static const int twl6040_restore_list[] = {
TWL6040_REG_MICLCTL,
TWL6040_REG_MICRCTL,
TWL6040_REG_MICGAIN,
TWL6040_REG_LINEGAIN,
TWL6040_REG_HSLCTL,
TWL6040_REG_HSRCTL,
TWL6040_REG_HSGAIN,
TWL6040_REG_EARCTL,
TWL6040_REG_HFLCTL,
TWL6040_REG_HFLGAIN,
TWL6040_REG_HFRCTL,
TWL6040_REG_HFRGAIN,
};
/* set of rates for each pll: low-power and high-performance */
static unsigned int lp_rates[] = {
8000,
11250,
16000,
22500,
32000,
44100,
48000,
88200,
96000,
};
static unsigned int hp_rates[] = {
8000,
16000,
32000,
48000,
96000,
};
static struct snd_pcm_hw_constraint_list sysclk_constraints[] = {
{ .count = ARRAY_SIZE(lp_rates), .list = lp_rates, },
{ .count = ARRAY_SIZE(hp_rates), .list = hp_rates, },
};
/*
* read twl6040 register cache
*/
static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec,
unsigned int reg)
{
u8 *cache = codec->reg_cache;
if (reg >= TWL6040_CACHEREGNUM)
return -EIO;
return cache[reg];
}
/*
* write twl6040 register cache
*/
static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec,
u8 reg, u8 value)
{
u8 *cache = codec->reg_cache;
if (reg >= TWL6040_CACHEREGNUM)
return;
cache[reg] = value;
}
/*
* read from twl6040 hardware register
*/
static int twl6040_read_reg_volatile(struct snd_soc_codec *codec,
unsigned int reg)
{
struct twl6040 *twl6040 = codec->control_data;
u8 value;
if (reg >= TWL6040_CACHEREGNUM)
return -EIO;
value = twl6040_reg_read(twl6040, 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)
{
struct twl6040 *twl6040 = codec->control_data;
if (reg >= TWL6040_CACHEREGNUM)
return -EIO;
twl6040_write_reg_cache(codec, reg, value);
return twl6040_reg_write(twl6040, reg, value);
}
static void twl6040_init_chip(struct snd_soc_codec *codec)
{
struct twl6040 *twl6040 = codec->control_data;
u8 val;
val = twl6040_get_revid(twl6040);
twl6040_write_reg_cache(codec, TWL6040_REG_ASICREV, val);
/* Change chip defaults */
/* No imput selected for microphone amplifiers */
twl6040_write_reg_cache(codec, TWL6040_REG_MICLCTL, 0x18);
twl6040_write_reg_cache(codec, TWL6040_REG_MICRCTL, 0x18);
/*
* We need to lower the default gain values, so the ramp code
* can work correctly for the first playback.
* This reduces the pop noise heard at the first playback.
*/
twl6040_write_reg_cache(codec, TWL6040_REG_HSGAIN, 0xff);
twl6040_write_reg_cache(codec, TWL6040_REG_EARCTL, 0x1e);
twl6040_write_reg_cache(codec, TWL6040_REG_HFLGAIN, 0x1d);
twl6040_write_reg_cache(codec, TWL6040_REG_HFRGAIN, 0x1d);
twl6040_write_reg_cache(codec, TWL6040_REG_LINEGAIN, 0);
}
static void twl6040_restore_regs(struct snd_soc_codec *codec)
{
u8 *cache = codec->reg_cache;
int reg, i;
for (i = 0; i < ARRAY_SIZE(twl6040_restore_list); i++) {
reg = twl6040_restore_list[i];
twl6040_write(codec, reg, cache[reg]);
}
}
/*
* 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) {
if (val + left_step > headset->left_vol)
val = headset->left_vol;
else
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) {
if ((int)val - (int)left_step < 0)
val = 0;
else
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) {
if (val + right_step > headset->right_vol)
val = headset->right_vol;
else
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) {
if ((int)val - (int)right_step < 0)
val = 0;
else
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) {
if (val + left_step > handsfree->left_vol)
val = handsfree->left_vol;
else
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) {
if ((int)val - (int)left_step < 0)
val = 0;
else
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) {
if (val + right_step > handsfree->right_vol)
val = handsfree->right_vol;
else
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) {
if ((int)val - (int)right_step < 0)
val = 0;
else
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 = 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 = 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->left_step = priv->hs_left_step;
out->right_step = priv->hs_right_step;
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->left_step = priv->hf_left_step;
out->right_step = priv->hf_right_step;
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->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;
}
/* 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);
int ret = 0;
if (SND_SOC_DAPM_EVENT_ON(event)) {
priv->non_lp++;
if (!strcmp(w->name, "Earphone Driver")) {
/* Earphone doesn't support low power mode */
priv->hs_power_mode_locked = 1;
ret = headset_power_mode(codec, 1);
}
} else {
priv->non_lp--;
if (!strcmp(w->name, "Earphone Driver")) {
priv->hs_power_mode_locked = 0;
ret = headset_power_mode(codec, priv->hs_power_mode);
}
}
msleep(1);
return ret;
}
static void twl6040_hs_jack_report(struct snd_soc_codec *codec,
struct snd_soc_jack *jack, int report)
{
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int status;
mutex_lock(&priv->mutex);
/* Sync status */
status = twl6040_read_reg_volatile(codec, TWL6040_REG_STATUS);
if (status & TWL6040_PLUGCOMP)
snd_soc_jack_report(jack, report, report);
else
snd_soc_jack_report(jack, 0, report);
mutex_unlock(&priv->mutex);
}
void twl6040_hs_jack_detect(struct snd_soc_codec *codec,
struct snd_soc_jack *jack, int report)
{
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
struct twl6040_jack_data *hs_jack = &priv->hs_jack;
hs_jack->jack = jack;
hs_jack->report = report;
twl6040_hs_jack_report(codec, hs_jack->jack, hs_jack->report);
}
EXPORT_SYMBOL_GPL(twl6040_hs_jack_detect);
static void twl6040_accessory_work(struct work_struct *work)
{
struct twl6040_data *priv = container_of(work,
struct twl6040_data, 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_audio_handler(int irq, void *data)
{
struct snd_soc_codec *codec = data;
struct twl6040 *twl6040 = codec->control_data;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
u8 intid;
intid = twl6040_reg_read(twl6040, TWL6040_REG_INTID);
if ((intid & TWL6040_PLUGINT) || (intid & TWL6040_UNPLUGINT))
queue_delayed_work(priv->workqueue, &priv->delayed_work,
msecs_to_jiffies(200));
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);
/* Headset power mode */
static const char *twl6040_power_mode_texts[] = {
"Low-Power", "High-Perfomance",
};
static const struct soc_enum twl6040_power_mode_enum =
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl6040_power_mode_texts),
twl6040_power_mode_texts);
static int twl6040_headset_power_get_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
ucontrol->value.enumerated.item[0] = priv->hs_power_mode;
return 0;
}
static int twl6040_headset_power_put_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int high_perf = ucontrol->value.enumerated.item[0];
int ret = 0;
if (!priv->hs_power_mode_locked)
ret = headset_power_mode(codec, high_perf);
if (!ret)
priv->hs_power_mode = high_perf;
return ret;
}
static int twl6040_pll_get_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
ucontrol->value.enumerated.item[0] = priv->pll_power_mode;
return 0;
}
static int twl6040_pll_put_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
priv->pll_power_mode = ucontrol->value.enumerated.item[0];
return 0;
}
int twl6040_get_clk_id(struct snd_soc_codec *codec)
{
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
return priv->pll_power_mode;
}
EXPORT_SYMBOL_GPL(twl6040_get_clk_id);
static const struct snd_kcontrol_new twl6040_snd_controls[] = {
/* Capture gains */
SOC_DOUBLE_TLV("Capture Preamplifier Volume",
TWL6040_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv),
SOC_DOUBLE_TLV("Capture Volume",
TWL6040_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv),
/* AFM gains */
SOC_DOUBLE_TLV("Aux FM Volume",
TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv),
/* Playback gains */
SOC_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),
SOC_ENUM_EXT("Headset Power Mode", twl6040_power_mode_enum,
twl6040_headset_power_get_enum,
twl6040_headset_power_put_enum),
SOC_ENUM_EXT("PLL Selection", twl6040_power_mode_enum,
twl6040_pll_get_enum, twl6040_pll_put_enum),
};
static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
/* Inputs */
SND_SOC_DAPM_INPUT("MAINMIC"),
SND_SOC_DAPM_INPUT("HSMIC"),
SND_SOC_DAPM_INPUT("SUBMIC"),
SND_SOC_DAPM_INPUT("AFML"),
SND_SOC_DAPM_INPUT("AFMR"),
/* Outputs */
SND_SOC_DAPM_OUTPUT("HSOL"),
SND_SOC_DAPM_OUTPUT("HSOR"),
SND_SOC_DAPM_OUTPUT("HFL"),
SND_SOC_DAPM_OUTPUT("HFR"),
SND_SOC_DAPM_OUTPUT("EP"),
/* 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_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
struct twl6040 *twl6040 = codec->control_data;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int ret;
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
if (priv->codec_powered)
break;
ret = twl6040_power(twl6040, 1);
if (ret)
return ret;
priv->codec_powered = 1;
twl6040_restore_regs(codec);
/* Set external boost GPO */
twl6040_write(codec, TWL6040_REG_GPOCTL, 0x02);
break;
case SND_SOC_BIAS_OFF:
if (!priv->codec_powered)
break;
twl6040_power(twl6040, 0);
priv->codec_powered = 0;
break;
}
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 16:53:46 +03:00
codec->dapm.bias_level = level;
return 0;
}
static int twl6040_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_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,
&sysclk_constraints[priv->pll_power_mode]);
return 0;
}
static int twl6040_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_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);
int rate;
rate = params_rate(params);
switch (rate) {
case 11250:
case 22500:
case 44100:
case 88200:
/* These rates are not supported when HPPLL is in use */
if (unlikely(priv->pll == TWL6040_SYSCLK_SEL_HPPLL)) {
dev_err(codec->dev, "HPPLL does not support rate %d\n",
rate);
return -EINVAL;
}
/* Capture is not supported with 17.64MHz sysclk */
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
dev_err(codec->dev,
"capture mode is not supported at %dHz\n",
rate);
return -EINVAL;
}
priv->sysclk = 17640000;
break;
case 8000:
case 16000:
case 32000:
case 48000:
case 96000:
priv->sysclk = 19200000;
break;
default:
dev_err(codec->dev, "unsupported rate %d\n", rate);
return -EINVAL;
}
return 0;
}
static int twl6040_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_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 *twl6040 = codec->control_data;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
int ret;
if (!priv->sysclk) {
dev_err(codec->dev,
"no mclk configured, call set_sysclk() on init\n");
return -EINVAL;
}
if ((priv->sysclk == 17640000) && priv->non_lp) {
dev_err(codec->dev,
"some enabled paths aren't supported at %dHz\n",
priv->sysclk);
return -EPERM;
}
ret = twl6040_set_pll(twl6040, priv->pll, priv->clk_in, priv->sysclk);
if (ret) {
dev_err(codec->dev, "Can not set PLL (%d)\n", ret);
return -EPERM;
}
return 0;
}
static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
switch (clk_id) {
case TWL6040_SYSCLK_SEL_LPPLL:
case TWL6040_SYSCLK_SEL_HPPLL:
priv->pll = clk_id;
priv->clk_in = freq;
break;
default:
dev_err(codec->dev, "unknown clk_id %d\n", clk_id);
return -EINVAL;
}
return 0;
}
static struct snd_soc_dai_ops twl6040_dai_ops = {
.startup = twl6040_startup,
.hw_params = twl6040_hw_params,
.prepare = twl6040_prepare,
.set_sysclk = twl6040_set_dai_sysclk,
};
static struct snd_soc_dai_driver twl6040_dai[] = {
{
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-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 5,
.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,
},
{
.name = "twl6040-ul",
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = TWL6040_RATES,
.formats = TWL6040_FORMATS,
},
.ops = &twl6040_dai_ops,
},
{
.name = "twl6040-dl1",
.playback = {
.stream_name = "Headset Playback",
.channels_min = 1,
.channels_max = 2,
.rates = TWL6040_RATES,
.formats = TWL6040_FORMATS,
},
.ops = &twl6040_dai_ops,
},
{
.name = "twl6040-dl2",
.playback = {
.stream_name = "Handsfree Playback",
.channels_min = 1,
.channels_max = 2,
.rates = TWL6040_RATES,
.formats = TWL6040_FORMATS,
},
.ops = &twl6040_dai_ops,
},
{
.name = "twl6040-vib",
.playback = {
.stream_name = "Vibra Playback",
.channels_min = 1,
.channels_max = 1,
.rates = SNDRV_PCM_RATE_CONTINUOUS,
.formats = TWL6040_FORMATS,
},
.ops = &twl6040_dai_ops,
},
};
#ifdef CONFIG_PM
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)
{
struct twl6040_data *priv;
struct twl4030_codec_data *pdata = dev_get_platdata(codec->dev);
struct platform_device *pdev = container_of(codec->dev,
struct platform_device, dev);
int ret = 0;
priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL);
if (priv == NULL)
return -ENOMEM;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
snd_soc_codec_set_drvdata(codec, priv);
priv->codec = codec;
codec->control_data = dev_get_drvdata(codec->dev->parent);
if (pdata && pdata->hs_left_step && pdata->hs_right_step) {
priv->hs_left_step = pdata->hs_left_step;
priv->hs_right_step = pdata->hs_right_step;
} else {
priv->hs_left_step = 1;
priv->hs_right_step = 1;
}
if (pdata && pdata->hf_left_step && pdata->hf_right_step) {
priv->hf_left_step = pdata->hf_left_step;
priv->hf_right_step = pdata->hf_right_step;
} else {
priv->hf_left_step = 1;
priv->hf_right_step = 1;
}
priv->plug_irq = platform_get_irq(pdev, 0);
if (priv->plug_irq < 0) {
dev_err(codec->dev, "invalid irq\n");
ret = -EINVAL;
goto work_err;
}
priv->workqueue = 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->headset.ramp_done);
init_completion(&priv->handsfree.ramp_done);
priv->hf_workqueue = create_singlethread_workqueue("twl6040-hf");
if (priv->hf_workqueue == NULL) {
ret = -ENOMEM;
goto hfwq_err;
}
priv->hs_workqueue = create_singlethread_workqueue("twl6040-hs");
if (priv->hs_workqueue == NULL) {
ret = -ENOMEM;
goto hswq_err;
}
INIT_DELAYED_WORK(&priv->hs_delayed_work, twl6040_pga_hs_work);
INIT_DELAYED_WORK(&priv->hf_delayed_work, twl6040_pga_hf_work);
ret = request_threaded_irq(priv->plug_irq, NULL, twl6040_audio_handler,
0, "twl6040_irq_plug", codec);
if (ret) {
dev_err(codec->dev, "PLUG IRQ request failed: %d\n", ret);
goto plugirq_err;
}
twl6040_init_chip(codec);
/* power on device */
ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
if (ret)
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:
free_irq(priv->plug_irq, codec);
plugirq_err:
destroy_workqueue(priv->hs_workqueue);
hswq_err:
destroy_workqueue(priv->hf_workqueue);
hfwq_err:
destroy_workqueue(priv->workqueue);
work_err:
kfree(priv);
return ret;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
static int twl6040_remove(struct snd_soc_codec *codec)
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF);
free_irq(priv->plug_irq, codec);
destroy_workqueue(priv->workqueue);
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, ARRAY_SIZE(twl6040_dai));
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
}
static int __devexit twl6040_codec_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static struct platform_driver twl6040_codec_driver = {
.driver = {
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
.name = "twl6040-codec",
.owner = THIS_MODULE,
},
.probe = twl6040_codec_probe,
.remove = __devexit_p(twl6040_codec_remove),
};
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");