--- /dev/null
+#ifndef _INCLUDE_SOUND_ALC5623_H
+#define _INCLUDE_SOUND_ALC5623_H
+struct alc5623_platform_data {
+ /* configure : */
+ /* Lineout/Speaker Amps Vmid ratio control */
+ /* enable/disable adc/dac high pass filters */
+ unsigned int add_ctrl;
+ /* configure : */
+ /* output to enable when jack is low */
+ /* output to enable when jack is high */
+ /* jack detect (gpio/nc/jack detect [12] */
+ unsigned int jack_det_ctrl;
+};
+#endif
+
struct snd_soc_dapm_path;
struct snd_soc_dapm_pin;
struct snd_soc_dapm_route;
+struct snd_soc_dapm_context;
int dapm_reg_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event);
struct snd_ctl_elem_value *uncontrol);
int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *uncontrol);
-int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
+int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_widget *widget);
-int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
+int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_widget *widget,
int num);
/* dapm path setup */
-int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec);
-void snd_soc_dapm_free(struct snd_soc_codec *codec);
-int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
+int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
+void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
+int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num);
/* dapm events */
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
-void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec);
+void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm);
/* dapm audio pin control and status */
-int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin);
-int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin);
-int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin);
-int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin);
-int snd_soc_dapm_sync(struct snd_soc_codec *codec);
-int snd_soc_dapm_force_enable_pin(struct snd_soc_codec *codec,
+int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm,
+ const char *pin);
+int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm,
+ const char *pin);
+int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin);
+int snd_soc_dapm_get_pin_status(struct snd_soc_dapm_context *dapm,
+ const char *pin);
+int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm);
+int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
const char *pin);
-int snd_soc_dapm_ignore_suspend(struct snd_soc_codec *codec, const char *pin);
+int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
+ const char *pin);
/* dapm widget types */
enum snd_soc_dapm_type {
char *sname; /* stream name */
struct snd_soc_codec *codec;
struct list_head list;
+ struct snd_soc_dapm_context *dapm;
/* dapm control */
short reg; /* negative reg = no direct dapm */
struct list_head power_list;
};
+/* DAPM context */
+struct snd_soc_dapm_context {
+ struct list_head widgets;
+ struct list_head paths;
+ enum snd_soc_bias_level bias_level;
+ enum snd_soc_bias_level suspend_bias_level;
+ struct delayed_work delayed_work;
+ unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
+
+ struct device *dev; /* from parent - for debug */
+ struct snd_soc_codec *codec; /* parent codec */
+ struct snd_soc_card *card; /* parent card */
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_dapm;
+#endif
+};
+
#endif
struct snd_soc_ac97_ops;
struct snd_soc_jack;
struct snd_soc_jack_pin;
+#include <sound/soc-dapm.h>
#ifdef CONFIG_GPIOLIB
struct snd_soc_jack_gpio;
/* runtime */
struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */
unsigned int active;
- unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
unsigned int cache_only:1; /* Suppress writes to hardware */
unsigned int cache_sync:1; /* Cache needs to be synced to hardware */
unsigned int suspended:1; /* Codec is in suspend PM state */
void *reg_cache;
/* dapm */
- u32 pop_time;
- struct list_head dapm_widgets;
- struct list_head dapm_paths;
- enum snd_soc_bias_level bias_level;
- enum snd_soc_bias_level suspend_bias_level;
- struct delayed_work delayed_work;
+ struct snd_soc_dapm_context dapm;
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_codec_root;
struct dentry *debugfs_reg;
- struct dentry *debugfs_pop_time;
struct dentry *debugfs_dapm;
#endif
};
struct list_head codec_dev_list;
struct list_head platform_dev_list;
struct list_head dai_dev_list;
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_card_root;
+ struct dentry *debugfs_pop_time;
+#endif
+ u32 pop_time;
};
/* SoC machine DAI configuration, glues a codec and cpu DAI together */
};
/* codec IO */
-static inline unsigned int snd_soc_read(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- return codec->driver->read(codec, reg);
-}
-
-static inline unsigned int snd_soc_write(struct snd_soc_codec *codec,
- unsigned int reg, unsigned int val)
-{
- return codec->driver->write(codec, reg, val);
-}
+unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg);
+unsigned int snd_soc_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int val);
/* device driver data */
static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int i;
/*
* Add DAPM widgets
*/
for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++)
- snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]);
+ snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]);
/*
* Setup audio path interconnects
*/
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
/* always connected pins */
- snd_soc_dapm_enable_pin(codec, "Int Mic");
- snd_soc_dapm_enable_pin(codec, "Ext Spk");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_enable_pin(dapm, "Int Mic");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_sync(dapm);
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
printk(KERN_DEBUG
}
/* Add specific widgets */
- snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, at91sam9g20ek_dapm_widgets,
ARRAY_SIZE(at91sam9g20ek_dapm_widgets));
/* Set up specific audio path interconnects */
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
/* not connected */
- snd_soc_dapm_nc_pin(codec, "RLINEIN");
- snd_soc_dapm_nc_pin(codec, "LLINEIN");
+ snd_soc_dapm_nc_pin(dapm, "RLINEIN");
+ snd_soc_dapm_nc_pin(dapm, "LLINEIN");
#ifdef ENABLE_MIC_INPUT
- snd_soc_dapm_enable_pin(codec, "Int Mic");
+ snd_soc_dapm_enable_pin(dapm, "Int Mic");
#else
- snd_soc_dapm_nc_pin(codec, "Int Mic");
+ snd_soc_dapm_nc_pin(dapm, "Int Mic");
#endif
/* always connected */
- snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
/* Add afeb9260 specific widgets */
- snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* Set up afeb9260 specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Line In");
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Line In");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Enable Audio PLL & Audio section */
data = AUDIO_PLL | AUDIO_SECTION_RESET
| AUDIO_SECTION_ON;
pm860x_set_bits(codec->control_data, REG_MISC2, data, 0);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int pm860x_probe(struct snd_soc_codec *codec)
{
struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int i, ret;
pm860x->codec = codec;
snd_soc_add_controls(codec, pm860x_snd_controls,
ARRAY_SIZE(pm860x_snd_controls));
- snd_soc_dapm_new_controls(codec, pm860x_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, pm860x_dapm_widgets,
ARRAY_SIZE(pm860x_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
out_codec:
select SND_SOC_AK4535 if I2C
select SND_SOC_AK4642 if I2C
select SND_SOC_AK4671 if I2C
+ select SND_SOC_ALC5623 if I2C
select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
select SND_SOC_CS42L51 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_WM8741 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_WM8770 if SPI_MASTER
select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8900 if I2C
config SND_SOC_AK4671
tristate
+config SND_SOC_ALC5623
+ tristate
+
config SND_SOC_CQ0093VC
tristate
config SND_SOC_WM8753
tristate
+config SND_SOC_WM8770
+ tristate
+
config SND_SOC_WM8776
tristate
config SND_SOC_WM9090
tristate
+
snd-soc-l3-objs := l3.o
snd-soc-max98088-objs := max98088.o
snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-alc5623-objs := alc5623.o
snd-soc-spdif-objs := spdif_transciever.o
snd-soc-ssm2602-objs := ssm2602.o
snd-soc-stac9766-objs := stac9766.o
snd-soc-wm8741-objs := wm8741.o
snd-soc-wm8750-objs := wm8750.o
snd-soc-wm8753-objs := wm8753.o
+snd-soc-wm8770-objs := wm8770.o
snd-soc-wm8776-objs := wm8776.o
snd-soc-wm8804-objs := wm8804.o
snd-soc-wm8900-objs := wm8900.o
obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o
obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o
obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
obj-$(CONFIG_SND_SOC_WM8741) += snd-soc-wm8741.o
obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o
obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
+obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o
obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o
obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o
obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o
static int ad1836_probe(struct snd_soc_codec *codec)
{
struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret = 0;
codec->control_data = ad1836->control_data;
snd_soc_add_controls(codec, ad1836_snd_controls,
ARRAY_SIZE(ad1836_snd_controls));
- snd_soc_dapm_new_controls(codec, ad1836_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, ad1836_dapm_widgets,
ARRAY_SIZE(ad1836_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+ snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
return ret;
}
static int ad193x_probe(struct snd_soc_codec *codec)
{
struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
codec->control_data = ad193x->control_data;
snd_soc_add_controls(codec, ad193x_snd_controls,
ARRAY_SIZE(ad193x_snd_controls));
- snd_soc_dapm_new_controls(codec, ad193x_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, ad193x_dapm_widgets,
ARRAY_SIZE(ad193x_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+ snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
return ret;
}
static int ak4535_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, ak4535_dapm_widgets,
- ARRAY_SIZE(ak4535_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_controls(dapm, ak4535_dapm_widgets,
+ ARRAY_SIZE(ak4535_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
ak4535_write(codec, AK4535_PM1, i & (~0x80));
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
-#include <sound/soc-dapm.h>
+#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
static int ak4671_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets,
- ARRAY_SIZE(ak4671_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_new_controls(dapm, ak4671_dapm_widgets,
+ ARRAY_SIZE(ak4671_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
return 0;
}
snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
--- /dev/null
+/*
+ * alc5623.c -- alc562[123] ALSA Soc Audio driver
+ *
+ * Copyright 2008 Realtek Microelectronics
+ * Author: flove <flove@realtek.com> Ethan <eku@marvell.com>
+ *
+ * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ *
+ * Based on WM8753.c
+ *
+ * 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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/alc5623.h>
+
+#include "alc5623.h"
+
+static int caps_charge = 2000;
+module_param(caps_charge, int, 0);
+MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
+
+/* codec private data */
+struct alc5623_priv {
+ enum snd_soc_control_type control_type;
+ void *control_data;
+ struct mutex mutex;
+ u8 id;
+ unsigned int sysclk;
+ u16 reg_cache[ALC5623_VENDOR_ID2+2];
+ unsigned int add_ctrl;
+ unsigned int jack_det_ctrl;
+};
+
+static void alc5623_fill_cache(struct snd_soc_codec *codec)
+{
+ int i, step = codec->driver->reg_cache_step;
+ u16 *cache = codec->reg_cache;
+
+ /* not really efficient ... */
+ for (i = 0 ; i < codec->driver->reg_cache_size ; i += step)
+ cache[i] = codec->hw_read(codec, i);
+}
+
+static inline int alc5623_reset(struct snd_soc_codec *codec)
+{
+ return snd_soc_write(codec, ALC5623_RESET, 0);
+}
+
+static int amp_mixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ /* to power-on/off class-d amp generators/speaker */
+ /* need to write to 'index-46h' register : */
+ /* so write index num (here 0x46) to reg 0x6a */
+ /* and then 0xffff/0 to reg 0x6c */
+ snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0);
+ break;
+ }
+
+ return 0;
+}
+
+/*
+ * ALC5623 Controls
+ */
+
+static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
+static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0);
+static const unsigned int boost_tlv[] = {
+ TLV_DB_RANGE_HEAD(3),
+ 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+ 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
+ 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
+};
+static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = {
+ SOC_DOUBLE_TLV("Speaker Playback Volume",
+ ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+ SOC_DOUBLE("Speaker Playback Switch",
+ ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
+ SOC_DOUBLE_TLV("Headphone Playback Volume",
+ ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+ SOC_DOUBLE("Headphone Playback Switch",
+ ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
+};
+
+static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = {
+ SOC_DOUBLE_TLV("Speaker Playback Volume",
+ ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+ SOC_DOUBLE("Speaker Playback Switch",
+ ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
+ SOC_DOUBLE_TLV("Line Playback Volume",
+ ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+ SOC_DOUBLE("Line Playback Switch",
+ ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = {
+ SOC_DOUBLE_TLV("Line Playback Volume",
+ ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+ SOC_DOUBLE("Line Playback Switch",
+ ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
+ SOC_DOUBLE_TLV("Headphone Playback Volume",
+ ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+ SOC_DOUBLE("Headphone Playback Switch",
+ ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_snd_controls[] = {
+ SOC_DOUBLE_TLV("Auxout Playback Volume",
+ ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
+ SOC_DOUBLE("Auxout Playback Switch",
+ ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1),
+ SOC_DOUBLE_TLV("PCM Playback Volume",
+ ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv),
+ SOC_DOUBLE_TLV("AuxI Capture Volume",
+ ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv),
+ SOC_DOUBLE_TLV("LineIn Capture Volume",
+ ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
+ SOC_SINGLE_TLV("Mic1 Capture Volume",
+ ALC5623_MIC_VOL, 8, 31, 1, vol_tlv),
+ SOC_SINGLE_TLV("Mic2 Capture Volume",
+ ALC5623_MIC_VOL, 0, 31, 1, vol_tlv),
+ SOC_DOUBLE_TLV("Rec Capture Volume",
+ ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv),
+ SOC_SINGLE_TLV("Mic 1 Boost Volume",
+ ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv),
+ SOC_SINGLE_TLV("Mic 2 Boost Volume",
+ ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv),
+ SOC_SINGLE_TLV("Digital Boost Volume",
+ ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv),
+};
+
+/*
+ * DAPM Controls
+ */
+static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = {
+SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1),
+SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1),
+SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1),
+SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1),
+SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1),
+SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1),
+SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1),
+SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1),
+SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1),
+SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1),
+SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1),
+};
+
+static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = {
+SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1),
+SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1),
+SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1),
+SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1),
+SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1),
+};
+
+/* Left Record Mixer */
+static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = {
+SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1),
+SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1),
+SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1),
+SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1),
+SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1),
+SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1),
+SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1),
+};
+
+/* Right Record Mixer */
+static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = {
+SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1),
+SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1),
+SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1),
+SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1),
+SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1),
+SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1),
+SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1),
+};
+
+static const char *alc5623_spk_n_sour_sel[] = {
+ "RN/-R", "RP/+R", "LN/-R", "Vmid" };
+static const char *alc5623_hpl_out_input_sel[] = {
+ "Vmid", "HP Left Mix"};
+static const char *alc5623_hpr_out_input_sel[] = {
+ "Vmid", "HP Right Mix"};
+static const char *alc5623_spkout_input_sel[] = {
+ "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
+static const char *alc5623_aux_out_input_sel[] = {
+ "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
+
+/* auxout output mux */
+static const struct soc_enum alc5623_aux_out_input_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel);
+static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
+
+/* speaker output mux */
+static const struct soc_enum alc5623_spkout_input_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel);
+static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
+
+/* headphone left output mux */
+static const struct soc_enum alc5623_hpl_out_input_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel);
+static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
+
+/* headphone right output mux */
+static const struct soc_enum alc5623_hpr_out_input_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel);
+static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
+
+/* speaker output N select */
+static const struct soc_enum alc5623_spk_n_sour_enum =
+SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel);
+static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
+SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
+
+static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = {
+/* Muxes */
+SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
+ &alc5623_auxout_mux_controls),
+SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
+ &alc5623_spkout_mux_controls),
+SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
+ &alc5623_hpl_out_mux_controls),
+SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
+ &alc5623_hpr_out_mux_controls),
+SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
+ &alc5623_spkoutn_mux_controls),
+
+/* output mixers */
+SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
+ &alc5623_hp_mixer_controls[0],
+ ARRAY_SIZE(alc5623_hp_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0,
+ &alc5623_hpr_mixer_controls[0],
+ ARRAY_SIZE(alc5623_hpr_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0,
+ &alc5623_hpl_mixer_controls[0],
+ ARRAY_SIZE(alc5623_hpl_mixer_controls)),
+SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0,
+ &alc5623_mono_mixer_controls[0],
+ ARRAY_SIZE(alc5623_mono_mixer_controls)),
+SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0,
+ &alc5623_speaker_mixer_controls[0],
+ ARRAY_SIZE(alc5623_speaker_mixer_controls)),
+
+/* input mixers */
+SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0,
+ &alc5623_captureL_mixer_controls[0],
+ ARRAY_SIZE(alc5623_captureL_mixer_controls)),
+SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0,
+ &alc5623_captureR_mixer_controls[0],
+ ARRAY_SIZE(alc5623_captureR_mixer_controls)),
+
+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
+ ALC5623_PWR_MANAG_ADD2, 9, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
+ ALC5623_PWR_MANAG_ADD2, 8, 0),
+SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
+ ALC5623_PWR_MANAG_ADD2, 7, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
+ ALC5623_PWR_MANAG_ADD2, 6, 0),
+SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0),
+
+SND_SOC_DAPM_OUTPUT("AUXOUTL"),
+SND_SOC_DAPM_OUTPUT("AUXOUTR"),
+SND_SOC_DAPM_OUTPUT("HPL"),
+SND_SOC_DAPM_OUTPUT("HPR"),
+SND_SOC_DAPM_OUTPUT("SPKOUT"),
+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+SND_SOC_DAPM_INPUT("LINEINL"),
+SND_SOC_DAPM_INPUT("LINEINR"),
+SND_SOC_DAPM_INPUT("AUXINL"),
+SND_SOC_DAPM_INPUT("AUXINR"),
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2"),
+SND_SOC_DAPM_VMID("Vmid"),
+};
+
+static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
+static const struct soc_enum alc5623_amp_enum =
+ SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names);
+static const struct snd_kcontrol_new alc5623_amp_mux_controls =
+ SOC_DAPM_ENUM("Route", alc5623_amp_enum);
+
+static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = {
+SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0,
+ amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0),
+SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0,
+ &alc5623_amp_mux_controls),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ /* virtual mixer - mixes left & right channels */
+ {"I2S Mix", NULL, "Left DAC"},
+ {"I2S Mix", NULL, "Right DAC"},
+ {"Line Mix", NULL, "Right LineIn"},
+ {"Line Mix", NULL, "Left LineIn"},
+ {"AuxI Mix", NULL, "Left AuxI"},
+ {"AuxI Mix", NULL, "Right AuxI"},
+ {"AUXOUTL", NULL, "Left AuxOut"},
+ {"AUXOUTR", NULL, "Right AuxOut"},
+
+ /* HP mixer */
+ {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"},
+ {"HPL Mix", NULL, "HP Mix"},
+ {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"},
+ {"HPR Mix", NULL, "HP Mix"},
+ {"HP Mix", "LI2HP Playback Switch", "Line Mix"},
+ {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"},
+ {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"},
+ {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"},
+ {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"},
+
+ /* speaker mixer */
+ {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"},
+ {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"},
+ {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"},
+ {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"},
+ {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"},
+
+ /* mono mixer */
+ {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"},
+ {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"},
+ {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"},
+ {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"},
+ {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"},
+ {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"},
+ {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"},
+
+ /* Left record mixer */
+ {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"},
+ {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"},
+ {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
+ {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
+ {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"},
+ {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
+ {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
+
+ /*Right record mixer */
+ {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"},
+ {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"},
+ {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
+ {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
+ {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"},
+ {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
+ {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
+
+ /* headphone left mux */
+ {"Left Headphone Mux", "HP Left Mix", "HPL Mix"},
+ {"Left Headphone Mux", "Vmid", "Vmid"},
+
+ /* headphone right mux */
+ {"Right Headphone Mux", "HP Right Mix", "HPR Mix"},
+ {"Right Headphone Mux", "Vmid", "Vmid"},
+
+ /* speaker out mux */
+ {"SpeakerOut Mux", "Vmid", "Vmid"},
+ {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"},
+ {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"},
+ {"SpeakerOut Mux", "Mono Mix", "Mono Mix"},
+
+ /* Mono/Aux Out mux */
+ {"AuxOut Mux", "Vmid", "Vmid"},
+ {"AuxOut Mux", "HPOut Mix", "HPOut Mix"},
+ {"AuxOut Mux", "Speaker Mix", "Speaker Mix"},
+ {"AuxOut Mux", "Mono Mix", "Mono Mix"},
+
+ /* output pga */
+ {"HPL", NULL, "Left Headphone"},
+ {"Left Headphone", NULL, "Left Headphone Mux"},
+ {"HPR", NULL, "Right Headphone"},
+ {"Right Headphone", NULL, "Right Headphone Mux"},
+ {"Left AuxOut", NULL, "AuxOut Mux"},
+ {"Right AuxOut", NULL, "AuxOut Mux"},
+
+ /* input pga */
+ {"Left LineIn", NULL, "LINEINL"},
+ {"Right LineIn", NULL, "LINEINR"},
+ {"Left AuxI", NULL, "AUXINL"},
+ {"Right AuxI", NULL, "AUXINR"},
+ {"MIC1 Pre Amp", NULL, "MIC1"},
+ {"MIC2 Pre Amp", NULL, "MIC2"},
+ {"MIC1 PGA", NULL, "MIC1 Pre Amp"},
+ {"MIC2 PGA", NULL, "MIC2 Pre Amp"},
+
+ /* left ADC */
+ {"Left ADC", NULL, "Left Capture Mix"},
+
+ /* right ADC */
+ {"Right ADC", NULL, "Right Capture Mix"},
+
+ {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"},
+ {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"},
+ {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"},
+ {"SpeakerOut N Mux", "Vmid", "Vmid"},
+
+ {"SPKOUT", NULL, "SpeakerOut"},
+ {"SPKOUTN", NULL, "SpeakerOut N Mux"},
+};
+
+static const struct snd_soc_dapm_route intercon_spk[] = {
+ {"SpeakerOut", NULL, "SpeakerOut Mux"},
+};
+
+static const struct snd_soc_dapm_route intercon_amp_spk[] = {
+ {"AB Amp", NULL, "SpeakerOut Mux"},
+ {"D Amp", NULL, "SpeakerOut Mux"},
+ {"AB-D Amp Mux", "AB Amp", "AB Amp"},
+ {"AB-D Amp Mux", "D Amp", "D Amp"},
+ {"SpeakerOut", NULL, "AB-D Amp Mux"},
+};
+
+/* PLL divisors */
+struct _pll_div {
+ u32 pll_in;
+ u32 pll_out;
+ u16 regvalue;
+};
+
+/* Note : pll code from original alc5623 driver. Not sure of how good it is */
+/* usefull only for master mode */
+static const struct _pll_div codec_master_pll_div[] = {
+
+ { 2048000, 8192000, 0x0ea0},
+ { 3686400, 8192000, 0x4e27},
+ { 12000000, 8192000, 0x456b},
+ { 13000000, 8192000, 0x495f},
+ { 13100000, 8192000, 0x0320},
+ { 2048000, 11289600, 0xf637},
+ { 3686400, 11289600, 0x2f22},
+ { 12000000, 11289600, 0x3e2f},
+ { 13000000, 11289600, 0x4d5b},
+ { 13100000, 11289600, 0x363b},
+ { 2048000, 16384000, 0x1ea0},
+ { 3686400, 16384000, 0x9e27},
+ { 12000000, 16384000, 0x452b},
+ { 13000000, 16384000, 0x542f},
+ { 13100000, 16384000, 0x03a0},
+ { 2048000, 16934400, 0xe625},
+ { 3686400, 16934400, 0x9126},
+ { 12000000, 16934400, 0x4d2c},
+ { 13000000, 16934400, 0x742f},
+ { 13100000, 16934400, 0x3c27},
+ { 2048000, 22579200, 0x2aa0},
+ { 3686400, 22579200, 0x2f20},
+ { 12000000, 22579200, 0x7e2f},
+ { 13000000, 22579200, 0x742f},
+ { 13100000, 22579200, 0x3c27},
+ { 2048000, 24576000, 0x2ea0},
+ { 3686400, 24576000, 0xee27},
+ { 12000000, 24576000, 0x2915},
+ { 13000000, 24576000, 0x772e},
+ { 13100000, 24576000, 0x0d20},
+};
+
+static const struct _pll_div codec_slave_pll_div[] = {
+
+ { 1024000, 16384000, 0x3ea0},
+ { 1411200, 22579200, 0x3ea0},
+ { 1536000, 24576000, 0x3ea0},
+ { 2048000, 16384000, 0x1ea0},
+ { 2822400, 22579200, 0x1ea0},
+ { 3072000, 24576000, 0x1ea0},
+
+};
+
+static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int freq_in, unsigned int freq_out)
+{
+ int i;
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int gbl_clk = 0, pll_div = 0;
+ u16 reg;
+
+ if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK)
+ return -ENODEV;
+
+ /* Disable PLL power */
+ snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
+ ALC5623_PWR_ADD2_PLL,
+ 0);
+
+ /* pll is not used in slave mode */
+ reg = snd_soc_read(codec, ALC5623_DAI_CONTROL);
+ if (reg & ALC5623_DAI_SDP_SLAVE_MODE)
+ return 0;
+
+ if (!freq_in || !freq_out)
+ return 0;
+
+ switch (pll_id) {
+ case ALC5623_PLL_FR_MCLK:
+ for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) {
+ if (codec_master_pll_div[i].pll_in == freq_in
+ && codec_master_pll_div[i].pll_out == freq_out) {
+ /* PLL source from MCLK */
+ pll_div = codec_master_pll_div[i].regvalue;
+ break;
+ }
+ }
+ break;
+ case ALC5623_PLL_FR_BCK:
+ for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) {
+ if (codec_slave_pll_div[i].pll_in == freq_in
+ && codec_slave_pll_div[i].pll_out == freq_out) {
+ /* PLL source from Bitclk */
+ gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK;
+ pll_div = codec_slave_pll_div[i].regvalue;
+ break;
+ }
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (!pll_div)
+ return -EINVAL;
+
+ snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
+ snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div);
+ snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
+ ALC5623_PWR_ADD2_PLL,
+ ALC5623_PWR_ADD2_PLL);
+ gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL;
+ snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
+
+ return 0;
+}
+
+struct _coeff_div {
+ u16 fs;
+ u16 regvalue;
+};
+
+/* codec hifi mclk (after PLL) clock divider coefficients */
+/* values inspired from column BCLK=32Fs of Appendix A table */
+static const struct _coeff_div coeff_div[] = {
+ {256*8, 0x3a69},
+ {384*8, 0x3c6b},
+ {256*4, 0x2a69},
+ {384*4, 0x2c6b},
+ {256*2, 0x1a69},
+ {384*2, 0x1c6b},
+ {256*1, 0x0a69},
+ {384*1, 0x0c6b},
+};
+
+static int get_coeff(struct snd_soc_codec *codec, int rate)
+{
+ struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].fs * rate == alc5623->sysclk)
+ return i;
+ }
+ return -EINVAL;
+}
+
+/*
+ * Clock after PLL and dividers
+ */
+static int alc5623_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 alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+
+ switch (freq) {
+ case 8192000:
+ case 11289600:
+ case 12288000:
+ case 16384000:
+ case 16934400:
+ case 18432000:
+ case 22579200:
+ case 24576000:
+ alc5623->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface = ALC5623_DAI_SDP_MASTER_MODE;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ iface = ALC5623_DAI_SDP_SLAVE_MODE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= ALC5623_DAI_I2S_DF_I2S;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ iface |= ALC5623_DAI_I2S_DF_RIGHT;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= ALC5623_DAI_I2S_DF_LEFT;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= ALC5623_DAI_I2S_DF_PCM;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
+}
+
+static int alc5623_pcm_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;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+ int coeff, rate;
+ u16 iface;
+
+ iface = snd_soc_read(codec, ALC5623_DAI_CONTROL);
+ iface &= ~ALC5623_DAI_I2S_DL_MASK;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ iface |= ALC5623_DAI_I2S_DL_16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= ALC5623_DAI_I2S_DL_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= ALC5623_DAI_I2S_DL_24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= ALC5623_DAI_I2S_DL_32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set iface & srate */
+ snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
+ rate = params_rate(params);
+ coeff = get_coeff(codec, rate);
+ if (coeff < 0)
+ return -EINVAL;
+
+ coeff = coeff_div[coeff].regvalue;
+ dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n",
+ __func__, alc5623->sysclk, rate, coeff);
+ snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff);
+
+ return 0;
+}
+
+static int alc5623_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT;
+ u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute;
+
+ if (mute)
+ mute_reg |= hp_mute;
+
+ return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg);
+}
+
+#define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \
+ | ALC5623_PWR_ADD2_DAC_REF_CIR)
+
+#define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \
+ | ALC5623_PWR_ADD3_MIC1_BOOST_AD)
+
+#define ALC5623_ADD1_POWER_EN \
+ (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \
+ | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \
+ | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP)
+
+#define ALC5623_ADD1_POWER_EN_5622 \
+ (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \
+ | ALC5623_PWR_ADD1_HP_OUT_AMP)
+
+static void enable_power_depop(struct snd_soc_codec *codec)
+{
+ struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+
+ snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1,
+ ALC5623_PWR_ADD1_SOFTGEN_EN,
+ ALC5623_PWR_ADD1_SOFTGEN_EN);
+
+ snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN);
+
+ snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
+ ALC5623_MISC_HP_DEPOP_MODE2_EN,
+ ALC5623_MISC_HP_DEPOP_MODE2_EN);
+
+ msleep(500);
+
+ snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN);
+
+ /* avoid writing '1' into 5622 reserved bits */
+ if (alc5623->id == 0x22)
+ snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
+ ALC5623_ADD1_POWER_EN_5622);
+ else
+ snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
+ ALC5623_ADD1_POWER_EN);
+
+ /* disable HP Depop2 */
+ snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
+ ALC5623_MISC_HP_DEPOP_MODE2_EN,
+ 0);
+
+}
+
+static int alc5623_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ enable_power_depop(codec);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* everything off except vref/vmid, */
+ snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2,
+ ALC5623_PWR_ADD2_VREF);
+ snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3,
+ ALC5623_PWR_ADD3_MAIN_BIAS);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* everything off, dac mute, inactive */
+ snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0);
+ snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0);
+ snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
+ break;
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+#define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \
+ | SNDRV_PCM_FMTBIT_S24_LE \
+ | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops alc5623_dai_ops = {
+ .hw_params = alc5623_pcm_hw_params,
+ .digital_mute = alc5623_mute,
+ .set_fmt = alc5623_set_dai_fmt,
+ .set_sysclk = alc5623_set_dai_sysclk,
+ .set_pll = alc5623_set_dai_pll,
+};
+
+static struct snd_soc_dai_driver alc5623_dai = {
+ .name = "alc5623-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = ALC5623_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = ALC5623_FORMATS,},
+
+ .ops = &alc5623_dai_ops,
+};
+
+static int alc5623_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
+{
+ alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int alc5623_resume(struct snd_soc_codec *codec)
+{
+ int i, step = codec->driver->reg_cache_step;
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 2 ; i < codec->driver->reg_cache_size ; i += step)
+ snd_soc_write(codec, i, cache[i]);
+
+ alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* charge alc5623 caps */
+ if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
+ alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ codec->dapm.bias_level = SND_SOC_BIAS_ON;
+ alc5623_set_bias_level(codec, codec->dapm.bias_level);
+ }
+
+ return 0;
+}
+
+static int alc5623_probe(struct snd_soc_codec *codec)
+{
+ struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int ret;
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ alc5623_reset(codec);
+ alc5623_fill_cache(codec);
+
+ /* power on device */
+ alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ if (alc5623->add_ctrl) {
+ snd_soc_write(codec, ALC5623_ADD_CTRL_REG,
+ alc5623->add_ctrl);
+ }
+
+ if (alc5623->jack_det_ctrl) {
+ snd_soc_write(codec, ALC5623_JACK_DET_CTRL,
+ alc5623->jack_det_ctrl);
+ }
+
+ switch (alc5623->id) {
+ default:
+ case 0x21:
+ snd_soc_add_controls(codec, rt5621_vol_snd_controls,
+ ARRAY_SIZE(rt5621_vol_snd_controls));
+ break;
+ case 0x22:
+ snd_soc_add_controls(codec, rt5622_vol_snd_controls,
+ ARRAY_SIZE(rt5622_vol_snd_controls));
+ break;
+ case 0x23:
+ snd_soc_add_controls(codec, alc5623_vol_snd_controls,
+ ARRAY_SIZE(alc5623_vol_snd_controls));
+ break;
+ }
+
+ snd_soc_add_controls(codec, alc5623_snd_controls,
+ ARRAY_SIZE(alc5623_snd_controls));
+
+ snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
+ ARRAY_SIZE(alc5623_dapm_widgets));
+
+ /* set up audio path interconnects */
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
+
+ switch (alc5623->id) {
+ default:
+ case 0x21:
+ case 0x22:
+ snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
+ ARRAY_SIZE(alc5623_dapm_amp_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon_amp_spk,
+ ARRAY_SIZE(intercon_amp_spk));
+ break;
+ case 0x23:
+ snd_soc_dapm_add_routes(dapm, intercon_spk,
+ ARRAY_SIZE(intercon_spk));
+ break;
+ }
+
+ return ret;
+}
+
+/* power down chip */
+static int alc5623_remove(struct snd_soc_codec *codec)
+{
+ alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
+ .probe = alc5623_probe,
+ .remove = alc5623_remove,
+ .suspend = alc5623_suspend,
+ .resume = alc5623_resume,
+ .set_bias_level = alc5623_set_bias_level,
+ .reg_cache_size = ALC5623_VENDOR_ID2+2,
+ .reg_word_size = sizeof(u16),
+ .reg_cache_step = 2,
+};
+
+/*
+ * ALC5623 2 wire address is determined by A1 pin
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+static int alc5623_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct alc5623_platform_data *pdata;
+ struct alc5623_priv *alc5623;
+ int ret, vid1, vid2;
+
+ vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1);
+ if (vid1 < 0) {
+ dev_err(&client->dev, "failed to read I2C\n");
+ return -EIO;
+ }
+ vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
+
+ vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2);
+ if (vid2 < 0) {
+ dev_err(&client->dev, "failed to read I2C\n");
+ return -EIO;
+ }
+
+ if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
+ dev_err(&client->dev, "unknown or wrong codec\n");
+ dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n",
+ 0x10ec, id->driver_data,
+ vid1, vid2);
+ return -ENODEV;
+ }
+
+ dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2);
+
+ alc5623 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL);
+ if (alc5623 == NULL) {
+ ret = -ENOMEM;
+ goto err;
+ }
+
+ pdata = client->dev.platform_data;
+ if (pdata) {
+ alc5623->add_ctrl = pdata->add_ctrl;
+ alc5623->jack_det_ctrl = pdata->jack_det_ctrl;
+ }
+
+ alc5623->id = vid2;
+ switch (alc5623->id) {
+ case 0x21:
+ alc5623_dai.name = "alc5621-hifi";
+ break;
+ case 0x22:
+ alc5623_dai.name = "alc5622-hifi";
+ break;
+ default:
+ case 0x23:
+ alc5623_dai.name = "alc5623-hifi";
+ break;
+ }
+
+ i2c_set_clientdata(client, alc5623);
+ alc5623->control_data = client;
+ alc5623->control_type = SND_SOC_I2C;
+ mutex_init(&alc5623->mutex);
+
+ ret = snd_soc_register_codec(&client->dev,
+ &soc_codec_device_alc5623, &alc5623_dai, 1);
+ if (ret != 0) {
+ dev_err(&client->dev, "Failed to register codec: %d\n", ret);
+ goto err;
+ }
+
+ return 0;
+
+err:
+ return ret;
+}
+
+static int alc5623_i2c_remove(struct i2c_client *client)
+{
+ struct alc5623_priv *alc5623 = i2c_get_clientdata(client);
+
+ snd_soc_unregister_codec(&client->dev);
+ kfree(alc5623);
+ return 0;
+}
+
+static const struct i2c_device_id alc5623_i2c_table[] = {
+ {"alc5621", 0x21},
+ {"alc5622", 0x22},
+ {"alc5623", 0x23},
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table);
+
+/* i2c codec control layer */
+static struct i2c_driver alc5623_i2c_driver = {
+ .driver = {
+ .name = "alc562x-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = alc5623_i2c_probe,
+ .remove = __devexit_p(alc5623_i2c_remove),
+ .id_table = alc5623_i2c_table,
+};
+
+static int __init alc5623_modinit(void)
+{
+ int ret;
+
+ ret = i2c_add_driver(&alc5623_i2c_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "%s: can't add i2c driver", __func__);
+ return ret;
+ }
+
+ return ret;
+}
+module_init(alc5623_modinit);
+
+static void __exit alc5623_modexit(void)
+{
+ i2c_del_driver(&alc5623_i2c_driver);
+}
+module_exit(alc5623_modexit);
+
+MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
+MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
+MODULE_LICENSE("GPL");
--- /dev/null
+/*
+ * alc5623.h -- alc562[123] ALSA Soc Audio driver
+ *
+ * Copyright 2008 Realtek Microelectronics
+ * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * Author: flove <flove@realtek.com>
+ * Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * 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.
+ *
+ */
+
+#ifndef _ALC5623_H
+#define _ALC5623_H
+
+#define ALC5623_RESET 0x00
+/* 5621 5622 5623 */
+/* speaker output vol 2 2 */
+/* line output vol 4 2 */
+/* HP output vol 4 0 4 */
+#define ALC5623_SPK_OUT_VOL 0x02
+#define ALC5623_HP_OUT_VOL 0x04
+#define ALC5623_MONO_AUX_OUT_VOL 0x06
+#define ALC5623_AUXIN_VOL 0x08
+#define ALC5623_LINE_IN_VOL 0x0A
+#define ALC5623_STEREO_DAC_VOL 0x0C
+#define ALC5623_MIC_VOL 0x0E
+#define ALC5623_MIC_ROUTING_CTRL 0x10
+#define ALC5623_ADC_REC_GAIN 0x12
+#define ALC5623_ADC_REC_MIXER 0x14
+#define ALC5623_SOFT_VOL_CTRL_TIME 0x16
+/* ALC5623_OUTPUT_MIXER_CTRL : */
+/* same remark as for reg 2 line vs speaker */
+#define ALC5623_OUTPUT_MIXER_CTRL 0x1C
+#define ALC5623_MIC_CTRL 0x22
+
+#define ALC5623_DAI_CONTROL 0x34
+#define ALC5623_DAI_SDP_MASTER_MODE (0 << 15)
+#define ALC5623_DAI_SDP_SLAVE_MODE (1 << 15)
+#define ALC5623_DAI_I2S_PCM_MODE (1 << 14)
+#define ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL (1 << 7)
+#define ALC5623_DAI_ADC_DATA_L_R_SWAP (1 << 5)
+#define ALC5623_DAI_DAC_DATA_L_R_SWAP (1 << 4)
+#define ALC5623_DAI_I2S_DL_MASK (3 << 2)
+#define ALC5623_DAI_I2S_DL_32 (3 << 2)
+#define ALC5623_DAI_I2S_DL_24 (2 << 2)
+#define ALC5623_DAI_I2S_DL_20 (1 << 2)
+#define ALC5623_DAI_I2S_DL_16 (0 << 2)
+#define ALC5623_DAI_I2S_DF_PCM (3 << 0)
+#define ALC5623_DAI_I2S_DF_LEFT (2 << 0)
+#define ALC5623_DAI_I2S_DF_RIGHT (1 << 0)
+#define ALC5623_DAI_I2S_DF_I2S (0 << 0)
+
+#define ALC5623_STEREO_AD_DA_CLK_CTRL 0x36
+#define ALC5623_COMPANDING_CTRL 0x38
+
+#define ALC5623_PWR_MANAG_ADD1 0x3A
+#define ALC5623_PWR_ADD1_MAIN_I2S_EN (1 << 15)
+#define ALC5623_PWR_ADD1_ZC_DET_PD_EN (1 << 14)
+#define ALC5623_PWR_ADD1_MIC1_BIAS_EN (1 << 11)
+#define ALC5623_PWR_ADD1_SHORT_CURR_DET_EN (1 << 10)
+#define ALC5623_PWR_ADD1_SOFTGEN_EN (1 << 8) /* rsvd on 5622 */
+#define ALC5623_PWR_ADD1_DEPOP_BUF_HP (1 << 6) /* rsvd on 5622 */
+#define ALC5623_PWR_ADD1_HP_OUT_AMP (1 << 5)
+#define ALC5623_PWR_ADD1_HP_OUT_ENH_AMP (1 << 4) /* rsvd on 5622 */
+#define ALC5623_PWR_ADD1_DEPOP_BUF_AUX (1 << 2)
+#define ALC5623_PWR_ADD1_AUX_OUT_AMP (1 << 1)
+#define ALC5623_PWR_ADD1_AUX_OUT_ENH_AMP (1 << 0) /* rsvd on 5622 */
+
+#define ALC5623_PWR_MANAG_ADD2 0x3C
+#define ALC5623_PWR_ADD2_LINEOUT (1 << 15) /* rt5623 */
+#define ALC5623_PWR_ADD2_CLASS_AB (1 << 15) /* rt5621 */
+#define ALC5623_PWR_ADD2_CLASS_D (1 << 14) /* rt5621 */
+#define ALC5623_PWR_ADD2_VREF (1 << 13)
+#define ALC5623_PWR_ADD2_PLL (1 << 12)
+#define ALC5623_PWR_ADD2_DAC_REF_CIR (1 << 10)
+#define ALC5623_PWR_ADD2_L_DAC_CLK (1 << 9)
+#define ALC5623_PWR_ADD2_R_DAC_CLK (1 << 8)
+#define ALC5623_PWR_ADD2_L_ADC_CLK_GAIN (1 << 7)
+#define ALC5623_PWR_ADD2_R_ADC_CLK_GAIN (1 << 6)
+#define ALC5623_PWR_ADD2_L_HP_MIXER (1 << 5)
+#define ALC5623_PWR_ADD2_R_HP_MIXER (1 << 4)
+#define ALC5623_PWR_ADD2_SPK_MIXER (1 << 3)
+#define ALC5623_PWR_ADD2_MONO_MIXER (1 << 2)
+#define ALC5623_PWR_ADD2_L_ADC_REC_MIXER (1 << 1)
+#define ALC5623_PWR_ADD2_R_ADC_REC_MIXER (1 << 0)
+
+#define ALC5623_PWR_MANAG_ADD3 0x3E
+#define ALC5623_PWR_ADD3_MAIN_BIAS (1 << 15)
+#define ALC5623_PWR_ADD3_AUXOUT_L_VOL_AMP (1 << 14)
+#define ALC5623_PWR_ADD3_AUXOUT_R_VOL_AMP (1 << 13)
+#define ALC5623_PWR_ADD3_SPK_OUT (1 << 12)
+#define ALC5623_PWR_ADD3_HP_L_OUT_VOL (1 << 10)
+#define ALC5623_PWR_ADD3_HP_R_OUT_VOL (1 << 9)
+#define ALC5623_PWR_ADD3_LINEIN_L_VOL (1 << 7)
+#define ALC5623_PWR_ADD3_LINEIN_R_VOL (1 << 6)
+#define ALC5623_PWR_ADD3_AUXIN_L_VOL (1 << 5)
+#define ALC5623_PWR_ADD3_AUXIN_R_VOL (1 << 4)
+#define ALC5623_PWR_ADD3_MIC1_FUN_CTRL (1 << 3)
+#define ALC5623_PWR_ADD3_MIC2_FUN_CTRL (1 << 2)
+#define ALC5623_PWR_ADD3_MIC1_BOOST_AD (1 << 1)
+#define ALC5623_PWR_ADD3_MIC2_BOOST_AD (1 << 0)
+
+#define ALC5623_ADD_CTRL_REG 0x40
+
+#define ALC5623_GLOBAL_CLK_CTRL_REG 0x42
+#define ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL (1 << 15)
+#define ALC5623_GBL_CLK_SYS_SOUR_SEL_MCLK (0 << 15)
+#define ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK (1 << 14)
+#define ALC5623_GBL_CLK_PLL_SOUR_SEL_MCLK (0 << 14)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV8 (3 << 1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV4 (2 << 1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV2 (1 << 1)
+#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV1 (0 << 1)
+#define ALC5623_GBL_CLK_PLL_PRE_DIV2 (1 << 0)
+#define ALC5623_GBL_CLK_PLL_PRE_DIV1 (0 << 0)
+
+#define ALC5623_PLL_CTRL 0x44
+#define ALC5623_PLL_CTRL_N_VAL(n) (((n)&0xff) << 8)
+#define ALC5623_PLL_CTRL_K_VAL(k) (((k)&0x7) << 4)
+#define ALC5623_PLL_CTRL_M_VAL(m) ((m)&0xf)
+
+#define ALC5623_GPIO_OUTPUT_PIN_CTRL 0x4A
+#define ALC5623_GPIO_PIN_CONFIG 0x4C
+#define ALC5623_GPIO_PIN_POLARITY 0x4E
+#define ALC5623_GPIO_PIN_STICKY 0x50
+#define ALC5623_GPIO_PIN_WAKEUP 0x52
+#define ALC5623_GPIO_PIN_STATUS 0x54
+#define ALC5623_GPIO_PIN_SHARING 0x56
+#define ALC5623_OVER_CURR_STATUS 0x58
+#define ALC5623_JACK_DET_CTRL 0x5A
+
+#define ALC5623_MISC_CTRL 0x5E
+#define ALC5623_MISC_DISABLE_FAST_VREG (1 << 15)
+#define ALC5623_MISC_SPK_CLASS_AB_OC_PD (1 << 13) /* 5621 */
+#define ALC5623_MISC_SPK_CLASS_AB_OC_DET (1 << 12) /* 5621 */
+#define ALC5623_MISC_HP_DEPOP_MODE3_EN (1 << 10)
+#define ALC5623_MISC_HP_DEPOP_MODE2_EN (1 << 9)
+#define ALC5623_MISC_HP_DEPOP_MODE1_EN (1 << 8)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE3_EN (1 << 6)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE2_EN (1 << 5)
+#define ALC5623_MISC_AUXOUT_DEPOP_MODE1_EN (1 << 4)
+#define ALC5623_MISC_M_DAC_L_INPUT (1 << 3)
+#define ALC5623_MISC_M_DAC_R_INPUT (1 << 2)
+#define ALC5623_MISC_IRQOUT_INV_CTRL (1 << 0)
+
+#define ALC5623_PSEDUEO_SPATIAL_CTRL 0x60
+#define ALC5623_EQ_CTRL 0x62
+#define ALC5623_EQ_MODE_ENABLE 0x66
+#define ALC5623_AVC_CTRL 0x68
+#define ALC5623_HID_CTRL_INDEX 0x6A
+#define ALC5623_HID_CTRL_DATA 0x6C
+#define ALC5623_VENDOR_ID1 0x7C
+#define ALC5623_VENDOR_ID2 0x7E
+
+#define ALC5623_PLL_FR_MCLK 0
+#define ALC5623_PLL_FR_BCK 1
+#endif
DAVINCI_VC_REG12_POWER_ALL_OFF);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int cs42l51_probe(struct snd_soc_codec *codec)
{
struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret, reg;
codec->control_data = cs42l51->control_data;
snd_soc_add_controls(codec, cs42l51_snd_controls,
ARRAY_SIZE(cs42l51_snd_controls));
- snd_soc_dapm_new_controls(codec, cs42l51_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, cs42l51_dapm_widgets,
ARRAY_SIZE(cs42l51_dapm_widgets));
- snd_soc_dapm_add_routes(codec, cs42l51_routes,
+ snd_soc_dapm_add_routes(dapm, cs42l51_routes,
ARRAY_SIZE(cs42l51_routes));
return 0;
#include <sound/core.h>
#include <sound/initval.h>
-#include <sound/soc-dapm.h>
+#include <sound/soc.h>
#include "cx20442.h"
static int cx20442_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, cx20442_dapm_widgets,
- ARRAY_SIZE(cx20442_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, cx20442_audio_map,
+ snd_soc_dapm_new_controls(dapm, cx20442_dapm_widgets,
+ ARRAY_SIZE(cx20442_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, cx20442_audio_map,
ARRAY_SIZE(cx20442_audio_map));
return 0;
/* Prevent the codec driver from further accessing the modem */
codec->hw_write = NULL;
cx20442->control_data = NULL;
- codec->pop_time = 0;
+ codec->card->pop_time = 0;
}
/* Line discipline .hangup() */
/* Set up codec driver access to modem controls */
cx20442->control_data = tty;
codec->hw_write = (hw_write_t)tty->ops->write;
- codec->pop_time = 1;
+ codec->card->pop_time = 1;
}
}
cx20442->control_data = NULL;
codec->hw_write = NULL;
- codec->pop_time = 0;
+ codec->card->pop_time = 0;
return 0;
}
#include <linux/slab.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
-#include <sound/soc-dapm.h>
+#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
break;
case SND_SOC_BIAS_STANDBY:
/* The only way to clear the suspend flag is to reset the codec */
- if (codec->bias_level == SND_SOC_BIAS_OFF)
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
jz4740_codec_wakeup(codec);
mask = JZ4740_CODEC_1_VREF_DISABLE |
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int jz4740_codec_dev_probe(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE);
snd_soc_add_controls(codec, jz4740_codec_controls,
ARRAY_SIZE(jz4740_codec_controls));
- snd_soc_dapm_new_controls(codec, jz4740_codec_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, jz4740_codec_dapm_widgets,
ARRAY_SIZE(jz4740_codec_dapm_widgets));
- snd_soc_dapm_add_routes(codec, jz4740_codec_dapm_routes,
+ snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes,
ARRAY_SIZE(jz4740_codec_dapm_routes));
snd_soc_dapm_new_widgets(codec);
static int max98088_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, max98088_dapm_widgets,
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_new_controls(dapm, max98088_dapm_widgets,
ARRAY_SIZE(max98088_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
snd_soc_add_controls(codec, max98088_snd_controls,
ARRAY_SIZE(max98088_snd_controls));
- snd_soc_dapm_new_widgets(codec);
+ snd_soc_dapm_new_widgets(dapm);
return 0;
}
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF)
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
max98088_sync_cache(codec);
snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN,
codec->cache_sync = 1;
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int ssm2602_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, ssm2602_dapm_widgets,
- ARRAY_SIZE(ssm2602_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn));
+ snd_soc_dapm_new_controls(dapm, ssm2602_dapm_widgets,
+ ARRAY_SIZE(ssm2602_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_conn, ARRAY_SIZE(audio_conn));
return 0;
}
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
#include <sound/initval.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/soc-dapm.h>
#include <sound/tlv.h>
#include "stac9766.h"
stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
- ARRAY_SIZE(tlv320aic23_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
+ ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* set up audio path interconnects */
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
return 0;
}
tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) {
/* find dapm widget path assoc with kcontrol */
- list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+ list_for_each_entry(path, &widget->dapm->paths, list) {
if (path->kcontrol != kcontrol)
continue;
}
if (found)
- snd_soc_dapm_sync(widget->codec);
+ snd_soc_dapm_sync(widget->dapm);
}
ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
static int aic3x_add_widgets(struct snd_soc_codec *codec)
{
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
ARRAY_SIZE(aic3x_dapm_widgets));
/* set up audio path interconnects */
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
if (aic3x->model == AIC3X_MODEL_3007) {
- snd_soc_dapm_new_controls(codec, aic3007_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets,
ARRAY_SIZE(aic3007_dapm_widgets));
- snd_soc_dapm_add_routes(codec, intercon_3007, ARRAY_SIZE(intercon_3007));
+ snd_soc_dapm_add_routes(dapm, intercon_3007,
+ ARRAY_SIZE(intercon_3007));
}
return 0;
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
- if (codec->bias_level == SND_SOC_BIAS_STANDBY &&
+ if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY &&
aic3x->master) {
/* enable pll */
reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG);
case SND_SOC_BIAS_STANDBY:
if (!aic3x->power)
aic3x_set_power(codec, 1);
- if (codec->bias_level == SND_SOC_BIAS_PREPARE &&
+ if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE &&
aic3x->master) {
/* disable pll */
reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG);
aic3x_set_power(codec, 0);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
codec->control_data = aic3x->control_data;
aic3x->codec = codec;
- codec->idle_bias_off = 1;
+ codec->dapm.idle_bias_off = 1;
ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type);
if (ret != 0) {
static int dac33_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, dac33_dapm_widgets,
- ARRAY_SIZE(dac33_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ snd_soc_dapm_new_controls(dapm, dac33_dapm_widgets,
+ ARRAY_SIZE(dac33_dapm_widgets));
/* set up audio path interconnects */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Coming from OFF, switch on the codec */
ret = dac33_hard_power(codec, 1);
if (ret != 0)
break;
case SND_SOC_BIAS_OFF:
/* Do not power off, when the codec is already off */
- if (codec->bias_level == SND_SOC_BIAS_OFF)
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
return 0;
ret = dac33_hard_power(codec, 0);
if (ret != 0)
return ret;
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
codec->control_data = dac33->control_data;
codec->hw_write = (hw_write_t) i2c_master_send;
- codec->idle_bias_off = 1;
+ codec->dapm.idle_bias_off = 1;
dac33->codec = codec;
/* Read the tlv320dac33 ID registers */
int tpa6130a2_add_controls(struct snd_soc_codec *codec)
{
struct tpa6130a2_data *data;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
if (tpa6130a2_client == NULL)
return -ENODEV;
data = i2c_get_clientdata(tpa6130a2_client);
- snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, tpa6130a2_dapm_widgets,
ARRAY_SIZE(tpa6130a2_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
if (data->id == TPA6140A2)
return snd_soc_add_controls(codec, tpa6140a2_controls,
static int twl4030_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, twl4030_dapm_widgets,
- ARRAY_SIZE(twl4030_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_new_controls(dapm, twl4030_dapm_widgets,
+ ARRAY_SIZE(twl4030_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
return 0;
}
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF)
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
twl4030_codec_enable(codec, 1);
break;
case SND_SOC_BIAS_OFF:
twl4030_codec_enable(codec, 0);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
snd_soc_codec_set_drvdata(codec, twl4030);
/* Set the defaults, and power up the codec */
twl4030->sysclk = twl4030_codec_get_mclk() / 1000;
- codec->idle_bias_off = 1;
+ codec->dapm.idle_bias_off = 1;
twl4030_init_chip(codec);
static int twl6040_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, twl6040_dapm_widgets,
- ARRAY_SIZE(twl6040_dapm_widgets));
-
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_new_widgets(codec);
+ 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;
}
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
pd->power(0);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int uda1380_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
- ARRAY_SIZE(uda1380_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
+ ARRAY_SIZE(uda1380_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
int reg;
struct uda1380_platform_data *pdata = codec->dev->platform_data;
- if (codec->bias_level == level)
+ if (codec->dapm.bias_level == level)
return 0;
switch (level) {
uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
if (gpio_is_valid(pdata->gpio_power)) {
gpio_set_value(pdata->gpio_power, 1);
mdelay(1);
for (reg = UDA1380_MVOL; reg < UDA1380_CACHEREGNUM; reg++)
set_bit(reg - 0x10, &uda1380_cache_dirty);
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
/* Called from the machine driver */
int wm2000_add_controls(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
if (!wm2000_i2c) {
return -ENODEV;
}
- ret = snd_soc_dapm_new_controls(codec, wm2000_dapm_widgets,
+ ret = snd_soc_dapm_new_controls(dapm, wm2000_dapm_widgets,
ARRAY_SIZE(wm2000_dapm_widgets));
if (ret < 0)
return ret;
- ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
if (ret < 0)
return ret;
*/
static void wm8350_pga_work(struct work_struct *work)
{
- struct snd_soc_codec *codec =
- container_of(work, struct snd_soc_codec, delayed_work.work);
+ struct snd_soc_dapm_context *dapm =
+ container_of(work, struct snd_soc_dapm_context, delayed_work.work);
+ struct snd_soc_codec *codec = dapm->codec;
struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
struct wm8350_output *out1 = &wm8350_data->out1,
*out2 = &wm8350_data->out2;
out->ramp = WM8350_RAMP_UP;
out->active = 1;
- if (!delayed_work_pending(&codec->delayed_work))
- schedule_delayed_work(&codec->delayed_work,
+ if (!delayed_work_pending(&codec->dapm.delayed_work))
+ schedule_delayed_work(&codec->dapm.delayed_work,
msecs_to_jiffies(1));
break;
out->ramp = WM8350_RAMP_DOWN;
out->active = 0;
- if (!delayed_work_pending(&codec->delayed_work))
- schedule_delayed_work(&codec->delayed_work,
+ if (!delayed_work_pending(&codec->dapm.delayed_work))
+ schedule_delayed_work(&codec->dapm.delayed_work,
msecs_to_jiffies(1));
break;
}
static int wm8350_add_widgets(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
- ret = snd_soc_dapm_new_controls(codec,
+ ret = snd_soc_dapm_new_controls(dapm,
wm8350_dapm_widgets,
ARRAY_SIZE(wm8350_dapm_widgets));
if (ret != 0) {
}
/* set up audio paths */
- ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
if (ret != 0) {
dev_err(codec->dev, "DAPM route register failed\n");
return ret;
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies),
priv->supplies);
if (ret != 0)
priv->supplies);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
/* Put the codec into reset if it wasn't already */
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
- INIT_DELAYED_WORK(&codec->delayed_work, wm8350_pga_work);
+ INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8350_pga_work);
/* Enable the codec */
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
priv->mic.jack = NULL;
/* cancel any work waiting to be queued. */
- ret = cancel_delayed_work(&codec->delayed_work);
+ ret = cancel_delayed_work(&codec->dapm.delayed_work);
/* if there was any work waiting then we run it now and
* wait for its completion */
if (ret) {
- schedule_delayed_work(&codec->delayed_work, 0);
+ schedule_delayed_work(&codec->dapm.delayed_work, 0);
flush_scheduled_work();
}
static int wm8400_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8400_dapm_widgets,
- ARRAY_SIZE(wm8400_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_controls(dapm, wm8400_dapm_widgets,
+ ARRAY_SIZE(wm8400_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
ret = regulator_bulk_enable(ARRAY_SIZE(power),
&power[0]);
if (ret != 0) {
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8510_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8510_dapm_widgets,
- ARRAY_SIZE(wm8510_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_controls(dapm, wm8510_dapm_widgets,
+ ARRAY_SIZE(wm8510_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
case SND_SOC_BIAS_STANDBY:
power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN;
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Initial cap charge at VMID 5k */
snd_soc_write(codec, WM8510_POWER1, power1 | 0x3);
mdelay(100);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8523_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8523_dapm_widgets,
- ARRAY_SIZE(wm8523_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_new_controls(dapm, wm8523_dapm_widgets,
+ ARRAY_SIZE(wm8523_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
return 0;
}
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies),
wm8523->supplies);
if (ret != 0) {
wm8523->supplies);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8580_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets,
- ARRAY_SIZE(wm8580_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets,
+ ARRAY_SIZE(wm8580_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Power up and get individual control of the DACs */
reg = snd_soc_read(codec, WM8580_PWRDN1);
reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD);
snd_soc_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
.set_bias_level = wm8580_set_bias_level,
.reg_cache_size = ARRAY_SIZE(wm8580_reg),
.reg_word_size = sizeof(u16),
- .reg_cache_default = &wm8580_reg,
+ .reg_cache_default = wm8580_reg,
};
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static int wm8711_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8711_dapm_widgets,
- ARRAY_SIZE(wm8711_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_new_controls(dapm, wm8711_dapm_widgets,
+ ARRAY_SIZE(wm8711_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
return 0;
}
snd_soc_write(codec, WM8711_PWR, 0xffff);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8728_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8728_dapm_widgets,
- ARRAY_SIZE(wm8728_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_new_controls(dapm, wm8728_dapm_widgets,
+ ARRAY_SIZE(wm8728_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
return 0;
}
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Power everything up... */
reg = snd_soc_read(codec, WM8728_DACCTL);
snd_soc_write(codec, WM8728_DACCTL, reg & ~0x4);
snd_soc_write(codec, WM8728_DACCTL, reg | 0x4);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8731_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
- ARRAY_SIZE(wm8731_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_new_controls(dapm, wm8731_dapm_widgets,
+ ARRAY_SIZE(wm8731_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
return 0;
}
return -EINVAL;
}
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(&codec->dapm);
return 0;
}
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies),
wm8731->supplies);
if (ret != 0)
wm8731->supplies);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8741_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8741_dapm_widgets,
- ARRAY_SIZE(wm8741_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_new_controls(dapm, wm8741_dapm_widgets,
+ ARRAY_SIZE(wm8741_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
return 0;
}
.resume = wm8741_resume,
.reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults),
.reg_word_size = sizeof(u16),
- .reg_cache_default = &wm8741_reg_defaults,
+ .reg_cache_default = wm8741_reg_defaults,
};
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static int wm8750_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
- ARRAY_SIZE(wm8750_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets,
+ ARRAY_SIZE(wm8750_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Set VMID to 5k */
snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
snd_soc_write(codec, WM8750_PWR1, 0x0001);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8753_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
- ARRAY_SIZE(wm8753_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets,
+ ARRAY_SIZE(wm8753_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
wm8753_write(codec, WM8753_PWR1, 0x0001);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static void wm8753_work(struct work_struct *work)
{
- struct snd_soc_codec *codec =
- container_of(work, struct snd_soc_codec, delayed_work.work);
- wm8753_set_bias_level(codec, codec->bias_level);
+ struct snd_soc_dapm_context *dapm =
+ container_of(work, struct snd_soc_dapm_context,
+ delayed_work.work);
+ struct snd_soc_codec *codec = dapm->codec;
+ wm8753_set_bias_level(codec, dapm->bias_level);
}
static int wm8753_suspend(struct snd_soc_codec *codec, pm_message_t state)
wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* charge wm8753 caps */
- if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+ if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
- codec->bias_level = SND_SOC_BIAS_ON;
- schedule_delayed_work(&codec->delayed_work,
+ codec->dapm.bias_level = SND_SOC_BIAS_ON;
+ schedule_delayed_work(&codec->dapm.delayed_work,
msecs_to_jiffies(caps_charge));
}
struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
int ret = 0, reg;
- INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
+ INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8753_work);
ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8753->control_type);
if (ret < 0) {
/* charge output caps */
wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
- schedule_delayed_work(&codec->delayed_work,
+ schedule_delayed_work(&codec->dapm.delayed_work,
msecs_to_jiffies(caps_charge));
/* set the update bits */
/* power down chip */
static int wm8753_remove(struct snd_soc_codec *codec)
{
- run_delayed_work(&codec->delayed_work);
+ run_delayed_work(&codec->dapm.delayed_work);
wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
--- /dev/null
+/*
+ * wm8770.c -- WM8770 ALSA SoC Audio driver
+ *
+ * Copyright 2010 Wolfson Microelectronics plc
+ *
+ * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.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.
+ */
+
+#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/spi/spi.h>
+#include <linux/regulator/consumer.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8770.h"
+
+#define WM8770_NUM_SUPPLIES 3
+static const char *wm8770_supply_names[WM8770_NUM_SUPPLIES] = {
+ "AVDD1",
+ "AVDD2",
+ "DVDD"
+};
+
+static const u16 wm8770_reg_defs[WM8770_CACHEREGNUM] = {
+ 0x7f, 0x7f, 0x7f, 0x7f,
+ 0x7f, 0x7f, 0x7f, 0x7f,
+ 0x7f, 0xff, 0xff, 0xff,
+ 0xff, 0xff, 0xff, 0xff,
+ 0xff, 0xff, 0, 0x90, 0,
+ 0, 0x22, 0x22, 0x3e,
+ 0xc, 0xc, 0x100, 0x189,
+ 0x189, 0x8770
+};
+
+struct wm8770_priv {
+ enum snd_soc_control_type control_type;
+ struct regulator_bulk_data supplies[WM8770_NUM_SUPPLIES];
+ struct notifier_block disable_nb[WM8770_NUM_SUPPLIES];
+ struct snd_soc_codec *codec;
+ int sysclk;
+};
+
+static int vout12supply_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event);
+static int vout34supply_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event);
+
+/*
+ * We can't use the same notifier block for more than one supply and
+ * there's no way I can see to get from a callback to the caller
+ * except container_of().
+ */
+#define WM8770_REGULATOR_EVENT(n) \
+static int wm8770_regulator_event_##n(struct notifier_block *nb, \
+ unsigned long event, void *data) \
+{ \
+ struct wm8770_priv *wm8770 = container_of(nb, struct wm8770_priv, \
+ disable_nb[n]); \
+ if (event & REGULATOR_EVENT_DISABLE) { \
+ wm8770->codec->cache_sync = 1; \
+ } \
+ return 0; \
+}
+
+WM8770_REGULATOR_EVENT(0)
+WM8770_REGULATOR_EVENT(1)
+WM8770_REGULATOR_EVENT(2)
+
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 100, 0);
+static const DECLARE_TLV_DB_SCALE(dac_dig_tlv, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(dac_alg_tlv, -12700, 100, 1);
+
+static const char *dac_phase_text[][2] = {
+ { "DAC1 Normal", "DAC1 Inverted" },
+ { "DAC2 Normal", "DAC2 Inverted" },
+ { "DAC3 Normal", "DAC3 Inverted" },
+ { "DAC4 Normal", "DAC4 Inverted" },
+};
+
+static const struct soc_enum dac_phase[] = {
+ SOC_ENUM_DOUBLE(WM8770_DACPHASE, 0, 1, 2, dac_phase_text[0]),
+ SOC_ENUM_DOUBLE(WM8770_DACPHASE, 2, 3, 2, dac_phase_text[1]),
+ SOC_ENUM_DOUBLE(WM8770_DACPHASE, 4, 5, 2, dac_phase_text[2]),
+ SOC_ENUM_DOUBLE(WM8770_DACPHASE, 6, 7, 2, dac_phase_text[3]),
+};
+
+static const struct snd_kcontrol_new wm8770_snd_controls[] = {
+ /* global DAC playback controls */
+ SOC_SINGLE_TLV("DAC Playback Volume", WM8770_MSDIGVOL, 0, 255, 0,
+ dac_dig_tlv),
+ SOC_SINGLE("DAC Playback Switch", WM8770_DACMUTE, 4, 1, 1),
+ SOC_SINGLE("DAC Playback ZC Switch", WM8770_DACCTRL1, 0, 1, 0),
+
+ /* global VOUT playback controls */
+ SOC_SINGLE_TLV("VOUT Playback Volume", WM8770_MSALGVOL, 0, 127, 0,
+ dac_alg_tlv),
+ SOC_SINGLE("VOUT Playback ZC Switch", WM8770_MSALGVOL, 7, 1, 0),
+
+ /* VOUT1/2/3/4 specific controls */
+ SOC_DOUBLE_R_TLV("VOUT1 Playback Volume", WM8770_VOUT1LVOL,
+ WM8770_VOUT1RVOL, 0, 127, 0, dac_alg_tlv),
+ SOC_DOUBLE_R("VOUT1 Playback ZC Switch", WM8770_VOUT1LVOL,
+ WM8770_VOUT1RVOL, 7, 1, 0),
+ SOC_DOUBLE_R_TLV("VOUT2 Playback Volume", WM8770_VOUT2LVOL,
+ WM8770_VOUT2RVOL, 0, 127, 0, dac_alg_tlv),
+ SOC_DOUBLE_R("VOUT2 Playback ZC Switch", WM8770_VOUT2LVOL,
+ WM8770_VOUT2RVOL, 7, 1, 0),
+ SOC_DOUBLE_R_TLV("VOUT3 Playback Volume", WM8770_VOUT3LVOL,
+ WM8770_VOUT3RVOL, 0, 127, 0, dac_alg_tlv),
+ SOC_DOUBLE_R("VOUT3 Playback ZC Switch", WM8770_VOUT3LVOL,
+ WM8770_VOUT3RVOL, 7, 1, 0),
+ SOC_DOUBLE_R_TLV("VOUT4 Playback Volume", WM8770_VOUT4LVOL,
+ WM8770_VOUT4RVOL, 0, 127, 0, dac_alg_tlv),
+ SOC_DOUBLE_R("VOUT4 Playback ZC Switch", WM8770_VOUT4LVOL,
+ WM8770_VOUT4RVOL, 7, 1, 0),
+
+ /* DAC1/2/3/4 specific controls */
+ SOC_DOUBLE_R_TLV("DAC1 Playback Volume", WM8770_DAC1LVOL,
+ WM8770_DAC1RVOL, 0, 255, 0, dac_dig_tlv),
+ SOC_SINGLE("DAC1 Deemphasis Switch", WM8770_DACCTRL2, 0, 1, 0),
+ SOC_ENUM("DAC1 Phase", dac_phase[0]),
+ SOC_DOUBLE_R_TLV("DAC2 Playback Volume", WM8770_DAC2LVOL,
+ WM8770_DAC2RVOL, 0, 255, 0, dac_dig_tlv),
+ SOC_SINGLE("DAC2 Deemphasis Switch", WM8770_DACCTRL2, 1, 1, 0),
+ SOC_ENUM("DAC2 Phase", dac_phase[1]),
+ SOC_DOUBLE_R_TLV("DAC3 Playback Volume", WM8770_DAC3LVOL,
+ WM8770_DAC3RVOL, 0, 255, 0, dac_dig_tlv),
+ SOC_SINGLE("DAC3 Deemphasis Switch", WM8770_DACCTRL2, 2, 1, 0),
+ SOC_ENUM("DAC3 Phase", dac_phase[2]),
+ SOC_DOUBLE_R_TLV("DAC4 Playback Volume", WM8770_DAC4LVOL,
+ WM8770_DAC4RVOL, 0, 255, 0, dac_dig_tlv),
+ SOC_SINGLE("DAC4 Deemphasis Switch", WM8770_DACCTRL2, 3, 1, 0),
+ SOC_ENUM("DAC4 Phase", dac_phase[3]),
+
+ /* ADC specific controls */
+ SOC_DOUBLE_R_TLV("Capture Volume", WM8770_ADCLCTRL, WM8770_ADCRCTRL,
+ 0, 31, 0, adc_tlv),
+ SOC_DOUBLE_R("Capture Switch", WM8770_ADCLCTRL, WM8770_ADCRCTRL,
+ 5, 1, 1),
+
+ /* other controls */
+ SOC_SINGLE("ADC 128x Oversampling Switch", WM8770_MSTRCTRL, 3, 1, 0),
+ SOC_SINGLE("ADC Highpass Filter Switch", WM8770_IFACECTRL, 8, 1, 1)
+};
+
+static const char *ain_text[] = {
+ "AIN1", "AIN2", "AIN3", "AIN4",
+ "AIN5", "AIN6", "AIN7", "AIN8"
+};
+
+static const struct soc_enum ain_enum =
+ SOC_ENUM_DOUBLE(WM8770_ADCMUX, 0, 4, 8, ain_text);
+
+static const struct snd_kcontrol_new ain_mux =
+ SOC_DAPM_ENUM("Capture Mux", ain_enum);
+
+static const struct snd_kcontrol_new vout1_mix_controls[] = {
+ SOC_DAPM_SINGLE("DAC1 Switch", WM8770_OUTMUX1, 0, 1, 0),
+ SOC_DAPM_SINGLE("AUX1 Switch", WM8770_OUTMUX1, 1, 1, 0),
+ SOC_DAPM_SINGLE("Bypass Switch", WM8770_OUTMUX1, 2, 1, 0)
+};
+
+static const struct snd_kcontrol_new vout2_mix_controls[] = {
+ SOC_DAPM_SINGLE("DAC2 Switch", WM8770_OUTMUX1, 3, 1, 0),
+ SOC_DAPM_SINGLE("AUX2 Switch", WM8770_OUTMUX1, 4, 1, 0),
+ SOC_DAPM_SINGLE("Bypass Switch", WM8770_OUTMUX1, 5, 1, 0)
+};
+
+static const struct snd_kcontrol_new vout3_mix_controls[] = {
+ SOC_DAPM_SINGLE("DAC3 Switch", WM8770_OUTMUX2, 0, 1, 0),
+ SOC_DAPM_SINGLE("AUX3 Switch", WM8770_OUTMUX2, 1, 1, 0),
+ SOC_DAPM_SINGLE("Bypass Switch", WM8770_OUTMUX2, 2, 1, 0)
+};
+
+static const struct snd_kcontrol_new vout4_mix_controls[] = {
+ SOC_DAPM_SINGLE("DAC4 Switch", WM8770_OUTMUX2, 3, 1, 0),
+ SOC_DAPM_SINGLE("Bypass Switch", WM8770_OUTMUX2, 4, 1, 0)
+};
+
+static const struct snd_soc_dapm_widget wm8770_dapm_widgets[] = {
+ SND_SOC_DAPM_INPUT("AUX1"),
+ SND_SOC_DAPM_INPUT("AUX2"),
+ SND_SOC_DAPM_INPUT("AUX3"),
+
+ SND_SOC_DAPM_INPUT("AIN1"),
+ SND_SOC_DAPM_INPUT("AIN2"),
+ SND_SOC_DAPM_INPUT("AIN3"),
+ SND_SOC_DAPM_INPUT("AIN4"),
+ SND_SOC_DAPM_INPUT("AIN5"),
+ SND_SOC_DAPM_INPUT("AIN6"),
+ SND_SOC_DAPM_INPUT("AIN7"),
+ SND_SOC_DAPM_INPUT("AIN8"),
+
+ SND_SOC_DAPM_MUX("Capture Mux", WM8770_ADCMUX, 8, 1, &ain_mux),
+
+ SND_SOC_DAPM_ADC("ADC", "Capture", WM8770_PWDNCTRL, 1, 1),
+
+ SND_SOC_DAPM_DAC("DAC1", "Playback", WM8770_PWDNCTRL, 2, 1),
+ SND_SOC_DAPM_DAC("DAC2", "Playback", WM8770_PWDNCTRL, 3, 1),
+ SND_SOC_DAPM_DAC("DAC3", "Playback", WM8770_PWDNCTRL, 4, 1),
+ SND_SOC_DAPM_DAC("DAC4", "Playback", WM8770_PWDNCTRL, 5, 1),
+
+ SND_SOC_DAPM_SUPPLY("VOUT12 Supply", SND_SOC_NOPM, 0, 0,
+ vout12supply_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("VOUT34 Supply", SND_SOC_NOPM, 0, 0,
+ vout34supply_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MIXER("VOUT1 Mixer", SND_SOC_NOPM, 0, 0,
+ vout1_mix_controls, ARRAY_SIZE(vout1_mix_controls)),
+ SND_SOC_DAPM_MIXER("VOUT2 Mixer", SND_SOC_NOPM, 0, 0,
+ vout2_mix_controls, ARRAY_SIZE(vout2_mix_controls)),
+ SND_SOC_DAPM_MIXER("VOUT3 Mixer", SND_SOC_NOPM, 0, 0,
+ vout3_mix_controls, ARRAY_SIZE(vout3_mix_controls)),
+ SND_SOC_DAPM_MIXER("VOUT4 Mixer", SND_SOC_NOPM, 0, 0,
+ vout4_mix_controls, ARRAY_SIZE(vout4_mix_controls)),
+
+ SND_SOC_DAPM_OUTPUT("VOUT1"),
+ SND_SOC_DAPM_OUTPUT("VOUT2"),
+ SND_SOC_DAPM_OUTPUT("VOUT3"),
+ SND_SOC_DAPM_OUTPUT("VOUT4")
+};
+
+static const struct snd_soc_dapm_route wm8770_intercon[] = {
+ { "Capture Mux", "AIN1", "AIN1" },
+ { "Capture Mux", "AIN2", "AIN2" },
+ { "Capture Mux", "AIN3", "AIN3" },
+ { "Capture Mux", "AIN4", "AIN4" },
+ { "Capture Mux", "AIN5", "AIN5" },
+ { "Capture Mux", "AIN6", "AIN6" },
+ { "Capture Mux", "AIN7", "AIN7" },
+ { "Capture Mux", "AIN8", "AIN8" },
+
+ { "ADC", NULL, "Capture Mux" },
+
+ { "VOUT1 Mixer", NULL, "VOUT12 Supply" },
+ { "VOUT1 Mixer", "DAC1 Switch", "DAC1" },
+ { "VOUT1 Mixer", "AUX1 Switch", "AUX1" },
+ { "VOUT1 Mixer", "Bypass Switch", "Capture Mux" },
+
+ { "VOUT2 Mixer", NULL, "VOUT12 Supply" },
+ { "VOUT2 Mixer", "DAC2 Switch", "DAC2" },
+ { "VOUT2 Mixer", "AUX2 Switch", "AUX2" },
+ { "VOUT2 Mixer", "Bypass Switch", "Capture Mux" },
+
+ { "VOUT3 Mixer", NULL, "VOUT34 Supply" },
+ { "VOUT3 Mixer", "DAC3 Switch", "DAC3" },
+ { "VOUT3 Mixer", "AUX3 Switch", "AUX3" },
+ { "VOUT3 Mixer", "Bypass Switch", "Capture Mux" },
+
+ { "VOUT4 Mixer", NULL, "VOUT34 Supply" },
+ { "VOUT4 Mixer", "DAC4 Switch", "DAC4" },
+ { "VOUT4 Mixer", "Bypass Switch", "Capture Mux" },
+
+ { "VOUT1", NULL, "VOUT1 Mixer" },
+ { "VOUT2", NULL, "VOUT2 Mixer" },
+ { "VOUT3", NULL, "VOUT3 Mixer" },
+ { "VOUT4", NULL, "VOUT4 Mixer" }
+};
+
+static int vout12supply_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec;
+
+ codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec, WM8770_OUTMUX1, 0x180, 0);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, WM8770_OUTMUX1, 0x180, 0x180);
+ break;
+ }
+
+ return 0;
+}
+
+static int vout34supply_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec;
+
+ codec = w->codec;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec, WM8770_OUTMUX2, 0x180, 0);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, WM8770_OUTMUX2, 0x180, 0x180);
+ break;
+ }
+
+ return 0;
+}
+
+static int wm8770_reset(struct snd_soc_codec *codec)
+{
+ return snd_soc_write(codec, WM8770_RESET, 0);
+}
+
+static int wm8770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec;
+ int iface, master;
+
+ codec = dai->codec;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ master = 0x100;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ master = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ iface = 0;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x2;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0xc;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x8;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x4;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, WM8770_IFACECTRL, 0xf, iface);
+ snd_soc_update_bits(codec, WM8770_MSTRCTRL, 0x100, master);
+
+ return 0;
+}
+
+static const int mclk_ratios[] = {
+ 128,
+ 192,
+ 256,
+ 384,
+ 512,
+ 768
+};
+
+static int wm8770_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec;
+ struct wm8770_priv *wm8770;
+ int i;
+ int iface;
+ int shift;
+ int ratio;
+
+ codec = dai->codec;
+ wm8770 = snd_soc_codec_get_drvdata(codec);
+
+ iface = 0;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x10;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x20;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x30;
+ break;
+ }
+
+ switch (substream->stream) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ i = 0;
+ shift = 4;
+ break;
+ case SNDRV_PCM_STREAM_CAPTURE:
+ i = 2;
+ shift = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* Only need to set MCLK/LRCLK ratio if we're master */
+ if (snd_soc_read(codec, WM8770_MSTRCTRL) & 0x100) {
+ for (; i < ARRAY_SIZE(mclk_ratios); ++i) {
+ ratio = wm8770->sysclk / params_rate(params);
+ if (ratio == mclk_ratios[i])
+ break;
+ }
+
+ if (i == ARRAY_SIZE(mclk_ratios)) {
+ dev_err(codec->dev,
+ "Unable to configure MCLK ratio %d/%d\n",
+ wm8770->sysclk, params_rate(params));
+ return -EINVAL;
+ }
+
+ dev_dbg(codec->dev, "MCLK is %dfs\n", mclk_ratios[i]);
+
+ snd_soc_update_bits(codec, WM8770_MSTRCTRL, 0x7 << shift,
+ i << shift);
+ }
+
+ snd_soc_update_bits(codec, WM8770_IFACECTRL, 0x30, iface);
+
+ return 0;
+}
+
+static int wm8770_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec;
+
+ codec = dai->codec;
+ return snd_soc_update_bits(codec, WM8770_DACMUTE, 0x10,
+ !!mute << 4);
+}
+
+static int wm8770_set_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec;
+ struct wm8770_priv *wm8770;
+
+ codec = dai->codec;
+ wm8770 = snd_soc_codec_get_drvdata(codec);
+ wm8770->sysclk = freq;
+ return 0;
+}
+
+static void wm8770_sync_cache(struct snd_soc_codec *codec)
+{
+ int i;
+ u16 *cache;
+
+ if (!codec->cache_sync)
+ return;
+
+ codec->cache_only = 0;
+ cache = codec->reg_cache;
+ for (i = 0; i < codec->driver->reg_cache_size; i++) {
+ if (i == WM8770_RESET || cache[i] == wm8770_reg_defs[i])
+ continue;
+ snd_soc_write(codec, i, cache[i]);
+ }
+ codec->cache_sync = 0;
+}
+
+static int wm8770_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ int ret;
+ struct wm8770_priv *wm8770;
+
+ wm8770 = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ ret = regulator_bulk_enable(ARRAY_SIZE(wm8770->supplies),
+ wm8770->supplies);
+ if (ret) {
+ dev_err(codec->dev,
+ "Failed to enable supplies: %d\n",
+ ret);
+ return ret;
+ }
+ wm8770_sync_cache(codec);
+ /* global powerup */
+ snd_soc_write(codec, WM8770_PWDNCTRL, 0);
+ }
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* global powerdown */
+ snd_soc_write(codec, WM8770_PWDNCTRL, 1);
+ regulator_bulk_disable(ARRAY_SIZE(wm8770->supplies),
+ wm8770->supplies);
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+#define WM8770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops wm8770_dai_ops = {
+ .digital_mute = wm8770_mute,
+ .hw_params = wm8770_hw_params,
+ .set_fmt = wm8770_set_fmt,
+ .set_sysclk = wm8770_set_sysclk,
+};
+
+static struct snd_soc_dai_driver wm8770_dai = {
+ .name = "wm8770-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = WM8770_FORMATS
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = WM8770_FORMATS
+ },
+ .ops = &wm8770_dai_ops,
+ .symmetric_rates = 1
+};
+
+#ifdef CONFIG_PM
+static int wm8770_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ wm8770_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8770_resume(struct snd_soc_codec *codec)
+{
+ wm8770_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ return 0;
+}
+#else
+#define wm8770_suspend NULL
+#define wm8770_resume NULL
+#endif
+
+static int wm8770_probe(struct snd_soc_codec *codec)
+{
+ struct wm8770_priv *wm8770;
+ int ret;
+ int i;
+
+ wm8770 = snd_soc_codec_get_drvdata(codec);
+ wm8770->codec = codec;
+
+ codec->dapm.idle_bias_off = 1;
+
+ ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8770->control_type);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(wm8770->supplies); i++)
+ wm8770->supplies[i].supply = wm8770_supply_names[i];
+
+ ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8770->supplies),
+ wm8770->supplies);
+ if (ret) {
+ dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+ return ret;
+ }
+
+ wm8770->disable_nb[0].notifier_call = wm8770_regulator_event_0;
+ wm8770->disable_nb[1].notifier_call = wm8770_regulator_event_1;
+ wm8770->disable_nb[2].notifier_call = wm8770_regulator_event_2;
+
+ /* This should really be moved into the regulator core */
+ for (i = 0; i < ARRAY_SIZE(wm8770->supplies); i++) {
+ ret = regulator_register_notifier(wm8770->supplies[i].consumer,
+ &wm8770->disable_nb[i]);
+ if (ret) {
+ dev_err(codec->dev,
+ "Failed to register regulator notifier: %d\n",
+ ret);
+ }
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(wm8770->supplies),
+ wm8770->supplies);
+ if (ret) {
+ dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+ goto err_reg_get;
+ }
+
+ ret = wm8770_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
+ goto err_reg_enable;
+ }
+
+ wm8770_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* latch the volume update bits */
+ snd_soc_update_bits(codec, WM8770_MSDIGVOL, 0x100, 0x100);
+ snd_soc_update_bits(codec, WM8770_MSALGVOL, 0x100, 0x100);
+ snd_soc_update_bits(codec, WM8770_VOUT1RVOL, 0x100, 0x100);
+ snd_soc_update_bits(codec, WM8770_VOUT2RVOL, 0x100, 0x100);
+ snd_soc_update_bits(codec, WM8770_VOUT3RVOL, 0x100, 0x100);
+ snd_soc_update_bits(codec, WM8770_VOUT4RVOL, 0x100, 0x100);
+ snd_soc_update_bits(codec, WM8770_DAC1RVOL, 0x100, 0x100);
+ snd_soc_update_bits(codec, WM8770_DAC2RVOL, 0x100, 0x100);
+ snd_soc_update_bits(codec, WM8770_DAC3RVOL, 0x100, 0x100);
+ snd_soc_update_bits(codec, WM8770_DAC4RVOL, 0x100, 0x100);
+
+ /* mute all DACs */
+ snd_soc_update_bits(codec, WM8770_DACMUTE, 0x10, 0x10);
+
+ snd_soc_add_controls(codec, wm8770_snd_controls,
+ ARRAY_SIZE(wm8770_snd_controls));
+ snd_soc_dapm_new_controls(&codec->dapm, wm8770_dapm_widgets,
+ ARRAY_SIZE(wm8770_dapm_widgets));
+ snd_soc_dapm_add_routes(&codec->dapm, wm8770_intercon,
+ ARRAY_SIZE(wm8770_intercon));
+ return 0;
+
+err_reg_enable:
+ regulator_bulk_disable(ARRAY_SIZE(wm8770->supplies), wm8770->supplies);
+err_reg_get:
+ regulator_bulk_free(ARRAY_SIZE(wm8770->supplies), wm8770->supplies);
+ return ret;
+}
+
+static int wm8770_remove(struct snd_soc_codec *codec)
+{
+ struct wm8770_priv *wm8770;
+ int i;
+
+ wm8770 = snd_soc_codec_get_drvdata(codec);
+ wm8770_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ for (i = 0; i < ARRAY_SIZE(wm8770->supplies); ++i)
+ regulator_unregister_notifier(wm8770->supplies[i].consumer,
+ &wm8770->disable_nb[i]);
+ regulator_bulk_free(ARRAY_SIZE(wm8770->supplies), wm8770->supplies);
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_wm8770 = {
+ .probe = wm8770_probe,
+ .remove = wm8770_remove,
+ .suspend = wm8770_suspend,
+ .resume = wm8770_resume,
+ .set_bias_level = wm8770_set_bias_level,
+ .reg_cache_size = ARRAY_SIZE(wm8770_reg_defs),
+ .reg_word_size = sizeof (u16),
+ .reg_cache_default = wm8770_reg_defs
+};
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8770_spi_probe(struct spi_device *spi)
+{
+ struct wm8770_priv *wm8770;
+ int ret;
+
+ wm8770 = kzalloc(sizeof(struct wm8770_priv), GFP_KERNEL);
+ if (!wm8770)
+ return -ENOMEM;
+
+ wm8770->control_type = SND_SOC_SPI;
+ spi_set_drvdata(spi, wm8770);
+
+ ret = snd_soc_register_codec(&spi->dev,
+ &soc_codec_dev_wm8770, &wm8770_dai, 1);
+ if (ret < 0)
+ kfree(wm8770);
+ return ret;
+}
+
+static int __devexit wm8770_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ kfree(spi_get_drvdata(spi));
+ return 0;
+}
+
+static struct spi_driver wm8770_spi_driver = {
+ .driver = {
+ .name = "wm8770",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8770_spi_probe,
+ .remove = __devexit_p(wm8770_spi_remove)
+};
+#endif
+
+static int __init wm8770_modinit(void)
+{
+ int ret = 0;
+
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&wm8770_spi_driver);
+ if (ret) {
+ printk(KERN_ERR "Failed to register wm8770 SPI driver: %d\n",
+ ret);
+ }
+#endif
+ return ret;
+}
+module_init(wm8770_modinit);
+
+static void __exit wm8770_exit(void)
+{
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8770_spi_driver);
+#endif
+}
+module_exit(wm8770_exit);
+
+MODULE_DESCRIPTION("ASoC WM8770 driver");
+MODULE_AUTHOR("Dimitris Papastamos <dp@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
--- /dev/null
+/*
+ * wm8770.h -- WM8770 ASoC driver
+ *
+ * Copyright 2010 Wolfson Microelectronics plc
+ *
+ * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.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.
+ */
+
+#ifndef _WM8770_H
+#define _WM8770_H
+
+/* Registers */
+#define WM8770_VOUT1LVOL 0
+#define WM8770_VOUT1RVOL 0x1
+#define WM8770_VOUT2LVOL 0x2
+#define WM8770_VOUT2RVOL 0x3
+#define WM8770_VOUT3LVOL 0x4
+#define WM8770_VOUT3RVOL 0x5
+#define WM8770_VOUT4LVOL 0x6
+#define WM8770_VOUT4RVOL 0x7
+#define WM8770_MSALGVOL 0x8
+#define WM8770_DAC1LVOL 0x9
+#define WM8770_DAC1RVOL 0xa
+#define WM8770_DAC2LVOL 0xb
+#define WM8770_DAC2RVOL 0xc
+#define WM8770_DAC3LVOL 0xd
+#define WM8770_DAC3RVOL 0xe
+#define WM8770_DAC4LVOL 0xf
+#define WM8770_DAC4RVOL 0x10
+#define WM8770_MSDIGVOL 0x11
+#define WM8770_DACPHASE 0x12
+#define WM8770_DACCTRL1 0x13
+#define WM8770_DACMUTE 0x14
+#define WM8770_DACCTRL2 0x15
+#define WM8770_IFACECTRL 0x16
+#define WM8770_MSTRCTRL 0x17
+#define WM8770_PWDNCTRL 0x18
+#define WM8770_ADCLCTRL 0x19
+#define WM8770_ADCRCTRL 0x1a
+#define WM8770_ADCMUX 0x1b
+#define WM8770_OUTMUX1 0x1c
+#define WM8770_OUTMUX2 0x1d
+#define WM8770_RESET 0x31
+
+#define WM8770_CACHEREGNUM 0x20
+
+#endif
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Disable the global powerdown; DAPM does the rest */
snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0);
}
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8776_probe(struct snd_soc_codec *codec)
{
struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret = 0;
ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8776->control_type);
snd_soc_add_controls(codec, wm8776_snd_controls,
ARRAY_SIZE(wm8776_snd_controls));
- snd_soc_dapm_new_controls(codec, wm8776_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8776_dapm_widgets,
ARRAY_SIZE(wm8776_dapm_widgets));
- snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes));
+ snd_soc_dapm_add_routes(dapm, routes, ARRAY_SIZE(routes));
return ret;
}
snd_soc_update_bits(codec, WM8804_PWRDN, 0x9, 0);
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies),
wm8804->supplies);
if (ret) {
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
wm8804 = snd_soc_codec_get_drvdata(codec);
wm8804->codec = codec;
- codec->idle_bias_off = 1;
+ codec->dapm.idle_bias_off = 1;
ret = snd_soc_codec_set_cache_io(codec, 8, 8, wm8804->control_type);
if (ret < 0) {
static int wm8900_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8900_dapm_widgets,
- ARRAY_SIZE(wm8900_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_controls(dapm, wm8900_dapm_widgets,
+ ARRAY_SIZE(wm8900_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
case SND_SOC_BIAS_STANDBY:
/* Charge capacitors if initial power up */
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* STARTUP_BIAS_ENA on */
snd_soc_write(codec, WM8900_REG_POWER1,
WM8900_REG_POWER1_STARTUP_BIAS_ENA);
WM8900_REG_POWER2_SYSCLK_ENA);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8903_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8903_dapm_widgets,
- ARRAY_SIZE(wm8903_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_new_controls(dapm, wm8903_dapm_widgets,
+ ARRAY_SIZE(wm8903_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
return 0;
}
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
snd_soc_write(codec, WM8903_CLOCK_RATES_2,
WM8903_CLK_SYS_ENA);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8904_add_widgets(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_new_controls(codec, wm8904_core_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8904_core_dapm_widgets,
ARRAY_SIZE(wm8904_core_dapm_widgets));
- snd_soc_dapm_add_routes(codec, core_intercon,
+ snd_soc_dapm_add_routes(dapm, core_intercon,
ARRAY_SIZE(core_intercon));
switch (wm8904->devtype) {
snd_soc_add_controls(codec, wm8904_snd_controls,
ARRAY_SIZE(wm8904_snd_controls));
- snd_soc_dapm_new_controls(codec, wm8904_adc_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8904_adc_dapm_widgets,
ARRAY_SIZE(wm8904_adc_dapm_widgets));
- snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8904_dac_dapm_widgets,
ARRAY_SIZE(wm8904_dac_dapm_widgets));
- snd_soc_dapm_new_controls(codec, wm8904_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8904_dapm_widgets,
ARRAY_SIZE(wm8904_dapm_widgets));
- snd_soc_dapm_add_routes(codec, core_intercon,
+ snd_soc_dapm_add_routes(dapm, core_intercon,
ARRAY_SIZE(core_intercon));
- snd_soc_dapm_add_routes(codec, adc_intercon,
+ snd_soc_dapm_add_routes(dapm, adc_intercon,
ARRAY_SIZE(adc_intercon));
- snd_soc_dapm_add_routes(codec, dac_intercon,
+ snd_soc_dapm_add_routes(dapm, dac_intercon,
ARRAY_SIZE(dac_intercon));
- snd_soc_dapm_add_routes(codec, wm8904_intercon,
+ snd_soc_dapm_add_routes(dapm, wm8904_intercon,
ARRAY_SIZE(wm8904_intercon));
break;
snd_soc_add_controls(codec, wm8904_dac_snd_controls,
ARRAY_SIZE(wm8904_dac_snd_controls));
- snd_soc_dapm_new_controls(codec, wm8904_dac_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8904_dac_dapm_widgets,
ARRAY_SIZE(wm8904_dac_dapm_widgets));
- snd_soc_dapm_add_routes(codec, dac_intercon,
+ snd_soc_dapm_add_routes(dapm, dac_intercon,
ARRAY_SIZE(dac_intercon));
- snd_soc_dapm_add_routes(codec, wm8912_intercon,
+ snd_soc_dapm_add_routes(dapm, wm8912_intercon,
ARRAY_SIZE(wm8912_intercon));
break;
}
- snd_soc_dapm_new_widgets(codec);
+ snd_soc_dapm_new_widgets(dapm);
return 0;
}
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies),
wm8904->supplies);
if (ret != 0) {
wm8904->supplies);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
int ret, i;
codec->cache_sync = 1;
- codec->idle_bias_off = 1;
+ codec->dapm.idle_bias_off = 1;
switch (wm8904->devtype) {
case WM8904:
static int wm8940_add_widgets(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
- ret = snd_soc_dapm_new_controls(codec, wm8940_dapm_widgets,
+ ret = snd_soc_dapm_new_controls(dapm, wm8940_dapm_widgets,
ARRAY_SIZE(wm8940_dapm_widgets));
if (ret)
goto error_ret;
- ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
if (ret)
goto error_ret;
static int wm8955_add_widgets(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
snd_soc_add_controls(codec, wm8955_snd_controls,
ARRAY_SIZE(wm8955_snd_controls));
- snd_soc_dapm_new_controls(codec, wm8955_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8955_dapm_widgets,
ARRAY_SIZE(wm8955_dapm_widgets));
-
- snd_soc_dapm_add_routes(codec, wm8955_intercon,
+ snd_soc_dapm_add_routes(dapm, wm8955_intercon,
ARRAY_SIZE(wm8955_intercon));
return 0;
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
ret = regulator_bulk_enable(ARRAY_SIZE(wm8955->supplies),
wm8955->supplies);
if (ret != 0) {
wm8955->supplies);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
{
struct wm8960_data *pdata = codec->dev->platform_data;
struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
struct snd_soc_dapm_widget *w;
- snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets,
ARRAY_SIZE(wm8960_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+ snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
/* In capless mode OUT3 is used to provide VMID for the
* headphone outputs, otherwise it is used as a mono mixer.
*/
if (pdata && pdata->capless) {
- snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets_capless,
+ snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets_capless,
ARRAY_SIZE(wm8960_dapm_widgets_capless));
- snd_soc_dapm_add_routes(codec, audio_paths_capless,
+ snd_soc_dapm_add_routes(dapm, audio_paths_capless,
ARRAY_SIZE(audio_paths_capless));
} else {
- snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets_out3,
+ snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets_out3,
ARRAY_SIZE(wm8960_dapm_widgets_out3));
- snd_soc_dapm_add_routes(codec, audio_paths_out3,
+ snd_soc_dapm_add_routes(dapm, audio_paths_out3,
ARRAY_SIZE(audio_paths_out3));
}
* list each time to find the desired power state do so now
* and save the result.
*/
- list_for_each_entry(w, &codec->dapm_widgets, list) {
+ list_for_each_entry(w, &codec->dapm.widgets, list) {
if (strcmp(w->name, "LOUT1 PGA") == 0)
wm8960->lout1 = w;
if (strcmp(w->name, "ROUT1 PGA") == 0)
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Enable anti-pop features */
snd_soc_write(codec, WM8960_APOP1,
WM8960_POBCTRL | WM8960_SOFT_ST |
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
break;
case SND_SOC_BIAS_PREPARE:
- switch (codec->bias_level) {
+ switch (codec->dapm.bias_level) {
case SND_SOC_BIAS_STANDBY:
/* Enable anti pop mode */
snd_soc_update_bits(codec, WM8960_APOP1,
break;
case SND_SOC_BIAS_STANDBY:
- switch (codec->bias_level) {
+ switch (codec->dapm.bias_level) {
case SND_SOC_BIAS_PREPARE:
/* Disable HP discharge */
snd_soc_update_bits(codec, WM8960_APOP2,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
break;
case SND_SOC_BIAS_PREPARE:
- if (codec->bias_level == SND_SOC_BIAS_STANDBY) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
/* Enable bias generation */
reg = snd_soc_read(codec, WM8961_ANTI_POP);
reg |= WM8961_BUFIOEN | WM8961_BUFDCOPEN;
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_PREPARE) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) {
/* VREF off */
reg = snd_soc_read(codec, WM8961_PWR_MGMT_1);
reg &= ~WM8961_VREF;
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8961_probe(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret = 0;
u16 reg;
snd_soc_add_controls(codec, wm8961_snd_controls,
ARRAY_SIZE(wm8961_snd_controls));
- snd_soc_dapm_new_controls(codec, wm8961_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8961_dapm_widgets,
ARRAY_SIZE(wm8961_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+ snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
return 0;
}
static int wm8962_add_widgets(struct snd_soc_codec *codec)
{
struct wm8962_pdata *pdata = dev_get_platdata(codec->dev);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
snd_soc_add_controls(codec, wm8962_snd_controls,
ARRAY_SIZE(wm8962_snd_controls));
ARRAY_SIZE(wm8962_spk_stereo_controls));
- snd_soc_dapm_new_controls(codec, wm8962_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8962_dapm_widgets,
ARRAY_SIZE(wm8962_dapm_widgets));
if (pdata && pdata->spk_mono)
- snd_soc_dapm_new_controls(codec, wm8962_dapm_spk_mono_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8962_dapm_spk_mono_widgets,
ARRAY_SIZE(wm8962_dapm_spk_mono_widgets));
else
- snd_soc_dapm_new_controls(codec, wm8962_dapm_spk_stereo_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8962_dapm_spk_stereo_widgets,
ARRAY_SIZE(wm8962_dapm_spk_stereo_widgets));
- snd_soc_dapm_add_routes(codec, wm8962_intercon,
+ snd_soc_dapm_add_routes(dapm, wm8962_intercon,
ARRAY_SIZE(wm8962_intercon));
if (pdata && pdata->spk_mono)
- snd_soc_dapm_add_routes(codec, wm8962_spk_mono_intercon,
+ snd_soc_dapm_add_routes(dapm, wm8962_spk_mono_intercon,
ARRAY_SIZE(wm8962_spk_mono_intercon));
else
- snd_soc_dapm_add_routes(codec, wm8962_spk_stereo_intercon,
+ snd_soc_dapm_add_routes(dapm, wm8962_spk_stereo_intercon,
ARRAY_SIZE(wm8962_spk_stereo_intercon));
- snd_soc_dapm_disable_pin(codec, "Beep");
+ snd_soc_dapm_disable_pin(dapm, "Beep");
return 0;
}
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
int ret;
- if (level == codec->bias_level)
+ if (level == codec->dapm.bias_level)
return 0;
switch (level) {
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies),
wm8962->supplies);
if (ret != 0) {
wm8962->supplies);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
if (active & (WM8962_MICSCD_EINT | WM8962_MICD_EINT)) {
dev_dbg(codec->dev, "Microphone event detected\n");
+ pm_wakeup_event(codec->dev, 300);
+
schedule_delayed_work(&wm8962->mic_work,
msecs_to_jiffies(250));
}
struct wm8962_priv *wm8962 =
container_of(work, struct wm8962_priv, beep_work);
struct snd_soc_codec *codec = wm8962->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int i;
int reg = 0;
int best = 0;
reg = WM8962_BEEP_ENA | (best << WM8962_BEEP_RATE_SHIFT);
- snd_soc_dapm_enable_pin(codec, "Beep");
+ snd_soc_dapm_enable_pin(dapm, "Beep");
} else {
dev_dbg(codec->dev, "Disabling beep\n");
- snd_soc_dapm_disable_pin(codec, "Beep");
+ snd_soc_dapm_disable_pin(dapm, "Beep");
}
snd_soc_update_bits(codec, WM8962_BEEP_GENERATOR_1,
WM8962_BEEP_ENA | WM8962_BEEP_RATE_MASK, reg);
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
}
/* For usability define a way of injecting beep events for the device -
INIT_DELAYED_WORK(&wm8962->mic_work, wm8962_mic_work);
codec->cache_sync = 1;
- codec->idle_bias_off = 1;
+ codec->dapm.idle_bias_off = 1;
ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C);
if (ret != 0) {
static int wm8971_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8971_dapm_widgets,
- ARRAY_SIZE(wm8971_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_controls(dapm, wm8971_dapm_widgets,
+ ARRAY_SIZE(wm8971_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
snd_soc_write(codec, WM8971_PWR1, 0x0001);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static void wm8971_work(struct work_struct *work)
{
- struct snd_soc_codec *codec =
- container_of(work, struct snd_soc_codec, delayed_work.work);
- wm8971_set_bias_level(codec, codec->bias_level);
+ struct snd_soc_dapm_context *dapm =
+ container_of(work, struct snd_soc_dapm_context,
+ delayed_work.work);
+ struct snd_soc_codec *codec = dapm->codec;
+ wm8971_set_bias_level(codec, codec->dapm.bias_level);
}
static int wm8971_suspend(struct snd_soc_codec *codec, pm_message_t state)
wm8971_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* charge wm8971 caps */
- if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+ if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e;
snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0);
- codec->bias_level = SND_SOC_BIAS_ON;
- queue_delayed_work(wm8971_workq, &codec->delayed_work,
+ codec->dapm.bias_level = SND_SOC_BIAS_ON;
+ queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work,
msecs_to_jiffies(1000));
}
return ret;
}
- INIT_DELAYED_WORK(&codec->delayed_work, wm8971_work);
+ INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work);
wm8971_workq = create_workqueue("wm8971");
if (wm8971_workq == NULL)
return -ENOMEM;
/* charge output caps - set vmid to 5k for quick power up */
reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e;
snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0);
- codec->bias_level = SND_SOC_BIAS_STANDBY;
- queue_delayed_work(wm8971_workq, &codec->delayed_work,
+ codec->dapm.bias_level = SND_SOC_BIAS_STANDBY;
+ queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work,
msecs_to_jiffies(1000));
/* set the update bits */
static int wm8974_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8974_dapm_widgets,
- ARRAY_SIZE(wm8974_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_controls(dapm, wm8974_dapm_widgets,
+ ARRAY_SIZE(wm8974_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
case SND_SOC_BIAS_STANDBY:
power1 |= WM8974_POWER1_BIASEN | WM8974_POWER1_BUFIOEN;
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Initial cap charge at VMID 5k */
snd_soc_write(codec, WM8974_POWER1, power1 | 0x3);
mdelay(100);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8978_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8978_dapm_widgets,
- ARRAY_SIZE(wm8978_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ snd_soc_dapm_new_controls(dapm, wm8978_dapm_widgets,
+ ARRAY_SIZE(wm8978_dapm_widgets));
/* set up the WM8978 audio map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
/* bit 3: enable bias, bit 2: enable I/O tie off buffer */
power1 |= 0xc;
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Initial cap charge at VMID 5k */
snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1,
power1 | 0x3);
dev_dbg(codec->dev, "%s: %d, %x\n", __func__, level, power1);
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8985_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8985_dapm_widgets,
- ARRAY_SIZE(wm8985_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map,
+ snd_soc_dapm_new_controls(dapm, wm8985_dapm_widgets,
+ ARRAY_SIZE(wm8985_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map,
ARRAY_SIZE(audio_map));
return 0;
}
1 << WM8985_VMIDSEL_SHIFT);
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
ret = regulator_bulk_enable(ARRAY_SIZE(wm8985->supplies),
wm8985->supplies);
if (ret) {
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* VREF, VMID=2x5k */
snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1);
snd_soc_write(codec, WM8988_PWR1, 0x0000);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8988_probe(struct snd_soc_codec *codec)
{
struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret = 0;
u16 reg;
snd_soc_add_controls(codec, wm8988_snd_controls,
ARRAY_SIZE(wm8988_snd_controls));
- snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8988_dapm_widgets,
ARRAY_SIZE(wm8988_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
static int wm8990_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8990_dapm_widgets,
- ARRAY_SIZE(wm8990_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ snd_soc_dapm_new_controls(dapm, wm8990_dapm_widgets,
+ ARRAY_SIZE(wm8990_dapm_widgets));
/* set up the WM8990 audio map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Enable all output discharge bits */
snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
WM8990_DIS_RLINE | WM8990_DIS_OUT3 |
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
0);
}
wm8993->class_w_users++;
+ wm8993->hubs_data.class_w = true;
}
/* Implement the change */
WM8993_CP_DYN_V);
}
wm8993->class_w_users--;
+ wm8993->hubs_data.class_w = false;
}
dev_dbg(codec->dev, "Indirect DAC use count now %d\n",
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
ret = regulator_bulk_enable(ARRAY_SIZE(wm8993->supplies),
wm8993->supplies);
if (ret != 0)
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8993_probe(struct snd_soc_codec *codec)
{
struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret, i, val;
wm8993->hubs_data.hp_startup_mode = 1;
ARRAY_SIZE(wm8993_eq_controls));
}
- snd_soc_dapm_new_controls(codec, wm8993_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8993_dapm_widgets,
ARRAY_SIZE(wm8993_dapm_widgets));
wm_hubs_add_analogue_controls(codec);
- snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes));
+ snd_soc_dapm_add_routes(dapm, routes, ARRAY_SIZE(routes));
wm_hubs_add_analogue_routes(codec, wm8993->pdata.lineout1_diff,
wm8993->pdata.lineout2_diff);
if (!wm8994_volatile(reg))
wm8994->reg_cache[reg] = value;
- dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
return wm8994_reg_write(codec->control_data, reg, value);
}
snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8994_SYSCLK_SRC, new);
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(&codec->dapm);
return 0;
}
static void wm8994_update_class_w(struct snd_soc_codec *codec)
{
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int enable = 1;
int source = 0; /* GCC flow analysis can't track enable */
int reg, reg_r;
WM8994_CP_DYN_PWR |
WM8994_CP_DYN_SRC_SEL_MASK,
source | WM8994_CP_DYN_PWR);
+ wm8994->hubs.class_w = true;
} else {
dev_dbg(codec->dev, "Class W disabled\n");
snd_soc_update_bits(codec, WM8994_CLASS_W_1,
WM8994_CP_DYN_PWR, 0);
+ wm8994->hubs.class_w = false;
}
}
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Tweak DC servo and DSP configuration for
* improved performance. */
if (wm8994->revision < 4) {
break;
case SND_SOC_BIAS_OFF:
- if (codec->bias_level == SND_SOC_BIAS_STANDBY) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
/* Switch over to startup biases */
snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
WM8994_BIAS_SRC |
}
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm8994_codec_probe(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret, i;
codec->control_data = dev_get_drvdata(codec->dev->parent);
wm_hubs_add_analogue_controls(codec);
snd_soc_add_controls(codec, wm8994_snd_controls,
ARRAY_SIZE(wm8994_snd_controls));
- snd_soc_dapm_new_controls(codec, wm8994_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8994_dapm_widgets,
ARRAY_SIZE(wm8994_dapm_widgets));
wm_hubs_add_analogue_routes(codec, 0, 0);
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
return 0;
case SND_SOC_BIAS_STANDBY:
/* Initial cold start */
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Disable LINEOUT discharge */
reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL);
reg &= ~WM9081_LINEOUT_DISCH;
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm9081_probe(struct snd_soc_codec *codec)
{
struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
u16 reg;
ARRAY_SIZE(wm9081_eq_controls));
}
- snd_soc_dapm_new_controls(codec, wm9081_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm9081_dapm_widgets,
ARRAY_SIZE(wm9081_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+ snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
return ret;
}
static int wm9090_add_controls(struct snd_soc_codec *codec)
{
struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int i;
- snd_soc_dapm_new_controls(codec, wm9090_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm9090_dapm_widgets,
ARRAY_SIZE(wm9090_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
snd_soc_add_controls(codec, wm9090_controls,
ARRAY_SIZE(wm9090_controls));
if (wm9090->pdata.lin1_diff) {
- snd_soc_dapm_add_routes(codec, audio_map_in1_diff,
+ snd_soc_dapm_add_routes(dapm, audio_map_in1_diff,
ARRAY_SIZE(audio_map_in1_diff));
} else {
- snd_soc_dapm_add_routes(codec, audio_map_in1_se,
+ snd_soc_dapm_add_routes(dapm, audio_map_in1_se,
ARRAY_SIZE(audio_map_in1_se));
snd_soc_add_controls(codec, wm9090_in1_se_controls,
ARRAY_SIZE(wm9090_in1_se_controls));
}
if (wm9090->pdata.lin2_diff) {
- snd_soc_dapm_add_routes(codec, audio_map_in2_diff,
+ snd_soc_dapm_add_routes(dapm, audio_map_in2_diff,
ARRAY_SIZE(audio_map_in2_diff));
} else {
- snd_soc_dapm_add_routes(codec, audio_map_in2_se,
+ snd_soc_dapm_add_routes(dapm, audio_map_in2_se,
ARRAY_SIZE(audio_map_in2_se));
snd_soc_add_controls(codec, wm9090_in2_se_controls,
ARRAY_SIZE(wm9090_in2_se_controls));
break;
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
/* Restore the register cache */
for (i = 1; i < codec->driver->reg_cache_size; i++) {
if (reg_cache[i] == wm9090_reg_defaults[i])
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm9705_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets,
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_new_controls(dapm, wm9705_dapm_widgets,
ARRAY_SIZE(wm9705_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
static int wm9712_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets,
- ARRAY_SIZE(wm9712_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_controls(dapm, wm9712_dapm_widgets,
+ ARRAY_SIZE(wm9712_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int wm9713_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets,
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_new_controls(dapm, wm9713_dapm_widgets,
ARRAY_SIZE(wm9713_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
u16 reg, reg_l, reg_r, dcs_cfg;
+ /* If we're using a digital only path and have a previously
+ * callibrated DC servo offset stored then use that. */
+ if (hubs->class_w && hubs->class_w_dcs) {
+ dev_dbg(codec->dev, "Using cached DC servo offset %x\n",
+ hubs->class_w_dcs);
+ snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs);
+ wait_for_dc_servo(codec,
+ WM8993_DCS_TRIG_DAC_WR_0 |
+ WM8993_DCS_TRIG_DAC_WR_1);
+ return;
+ }
+
/* Set for 32 series updates */
snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
WM8993_DCS_SERIES_NO_01_MASK,
wait_for_dc_servo(codec,
WM8993_DCS_TRIG_SERIES_0 | WM8993_DCS_TRIG_SERIES_1);
+ /* Different chips in the family support different readback
+ * methods.
+ */
+ switch (hubs->dcs_readback_mode) {
+ case 0:
+ reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
+ & WM8993_DCS_INTEG_CHAN_0_MASK;;
+ reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
+ & WM8993_DCS_INTEG_CHAN_1_MASK;
+ break;
+ case 1:
+ reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
+ reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
+ >> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+ reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
+ break;
+ default:
+ WARN(1, "Unknown DCS readback method\n");
+ break;
+ }
+
+ dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
+
/* Apply correction to DC servo result */
if (hubs->dcs_codes) {
dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
hubs->dcs_codes);
- /* Different chips in the family support different
- * readback methods.
- */
- switch (hubs->dcs_readback_mode) {
- case 0:
- reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
- & WM8993_DCS_INTEG_CHAN_0_MASK;;
- reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
- & WM8993_DCS_INTEG_CHAN_1_MASK;
- break;
- case 1:
- reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
- reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
- >> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
- reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
- break;
- default:
- WARN(1, "Unknown DCS readback method\n");
- break;
- }
-
- dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
-
/* HPOUT1L */
if (reg_l + hubs->dcs_codes > 0 &&
reg_l + hubs->dcs_codes < 0xff)
wait_for_dc_servo(codec,
WM8993_DCS_TRIG_DAC_WR_0 |
WM8993_DCS_TRIG_DAC_WR_1);
+ } else {
+ dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+ dcs_cfg |= reg_r;
}
+
+ /* Save the callibrated offset if we're in class W mode and
+ * therefore don't have any analogue signal mixed in. */
+ if (hubs->class_w)
+ hubs->class_w_dcs = dcs_cfg;
}
/*
ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
+ /* Updating the analogue gains invalidates the DC servo cache */
+ hubs->class_w_dcs = 0;
+
/* If we're applying an offset correction then updating the
* callibration would be likely to introduce further offsets. */
if (hubs->dcs_codes)
int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
/* Latch volume update bits & default ZC on */
snd_soc_update_bits(codec, WM8993_LEFT_LINE_INPUT_1_2_VOLUME,
WM8993_IN1_VU, WM8993_IN1_VU);
snd_soc_add_controls(codec, analogue_snd_controls,
ARRAY_SIZE(analogue_snd_controls));
- snd_soc_dapm_new_controls(codec, analogue_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, analogue_dapm_widgets,
ARRAY_SIZE(analogue_dapm_widgets));
return 0;
}
int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec,
int lineout1_diff, int lineout2_diff)
{
- snd_soc_dapm_add_routes(codec, analogue_routes,
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_add_routes(dapm, analogue_routes,
ARRAY_SIZE(analogue_routes));
if (lineout1_diff)
- snd_soc_dapm_add_routes(codec,
+ snd_soc_dapm_add_routes(dapm,
lineout1_diff_routes,
ARRAY_SIZE(lineout1_diff_routes));
else
- snd_soc_dapm_add_routes(codec,
+ snd_soc_dapm_add_routes(dapm,
lineout1_se_routes,
ARRAY_SIZE(lineout1_se_routes));
if (lineout2_diff)
- snd_soc_dapm_add_routes(codec,
+ snd_soc_dapm_add_routes(dapm,
lineout2_diff_routes,
ARRAY_SIZE(lineout2_diff_routes));
else
- snd_soc_dapm_add_routes(codec,
+ snd_soc_dapm_add_routes(dapm,
lineout2_se_routes,
ARRAY_SIZE(lineout2_se_routes));
* VMID as an output and can disable it.
*/
if (lineout1_diff && lineout2_diff)
- codec->idle_bias_off = 1;
+ codec->dapm.idle_bias_off = 1;
if (lineout1fb)
snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
int dcs_codes;
int dcs_readback_mode;
int hp_startup_mode;
+
+ bool class_w;
+ u16 class_w_dcs;
};
extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *);
static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
/* Add davinci-evm specific widgets */
- snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
ARRAY_SIZE(aic3x_dapm_widgets));
/* Set up davinci-evm specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* not connected */
- snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
- snd_soc_dapm_disable_pin(codec, "HPLCOM");
- snd_soc_dapm_disable_pin(codec, "HPRCOM");
+ snd_soc_dapm_disable_pin(dapm, "MONO_LOUT");
+ snd_soc_dapm_disable_pin(dapm, "HPLCOM");
+ snd_soc_dapm_disable_pin(dapm, "HPRCOM");
/* always connected */
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Line Out");
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
- snd_soc_dapm_enable_pin(codec, "Line In");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Line Out");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin(dapm, "Line In");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int snappercl15_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
static struct snd_soc_dai_driver imx_ssi_dai = {
.probe = imx_ssi_dai_probe,
.playback = {
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_96000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_new_controls(codec, wm1133_ev1_widgets,
+ snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets,
ARRAY_SIZE(wm1133_ev1_widgets));
- snd_soc_dapm_add_routes(codec, wm1133_ev1_map,
+ snd_soc_dapm_add_routes(dapm, wm1133_ev1_map,
ARRAY_SIZE(wm1133_ev1_map));
/* Headphone jack detection */
wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
SND_JACK_BTN_0);
- snd_soc_dapm_force_enable_pin(codec, "Mic Bias");
+ snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
return 0;
}
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
- snd_soc_dapm_nc_pin(codec, "LIN");
- snd_soc_dapm_nc_pin(codec, "RIN");
+ snd_soc_dapm_nc_pin(dapm, "LIN");
+ snd_soc_dapm_nc_pin(dapm, "RIN");
ret = snd_soc_dai_set_fmt(cpu_dai, QI_LB60_DAIFMT);
if (ret < 0) {
return ret;
}
- snd_soc_dapm_new_controls(codec, qi_lb60_widgets, ARRAY_SIZE(qi_lb60_widgets));
- snd_soc_dapm_add_routes(codec, qi_lb60_routes, ARRAY_SIZE(qi_lb60_routes));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_new_controls(dapm, qi_lb60_widgets,
+ ARRAY_SIZE(qi_lb60_widgets));
+ snd_soc_dapm_add_routes(dapm, qi_lb60_routes,
+ ARRAY_SIZE(qi_lb60_routes));
+ snd_soc_dapm_sync(dapm);
return 0;
}
Say Y if you want to add support for SoC audio on
Openrd Client.
+config SND_KIRKWOOD_SOC_T5325
+ tristate "SoC Audio support for HP t5325"
+ depends on SND_KIRKWOOD_SOC && MACH_T5325
+ select SND_KIRKWOOD_SOC_I2S
+ select SND_SOC_ALC5623
+ help
+ Say Y if you want to add support for SoC audio on
+ the HP t5325 thin client.
+
obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o
snd-soc-openrd-objs := kirkwood-openrd.o
+snd-soc-t5325-objs := kirkwood-t5325.o
obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o
+obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o
--- /dev/null
+/*
+ * kirkwood-t5325.c
+ *
+ * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <mach/kirkwood.h>
+#include <plat/audio.h>
+#include <asm/mach-types.h>
+#include "../codecs/alc5623.h"
+
+static int t5325_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret;
+ unsigned int freq, fmt;
+
+ fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
+ ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+ if (ret < 0)
+ return ret;
+
+ freq = params_rate(params) * 256;
+
+ return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
+
+}
+
+static struct snd_soc_ops t5325_ops = {
+ .hw_params = t5325_hw_params,
+};
+
+static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route t5325_route[] = {
+ { "Headphone Jack", NULL, "HPL" },
+ { "Headphone Jack", NULL, "HPR" },
+
+ {"Speaker", NULL, "SPKOUT"},
+ {"Speaker", NULL, "SPKOUTN"},
+
+ { "MIC1", NULL, "Mic Jack" },
+ { "MIC2", NULL, "Mic Jack" },
+};
+
+static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_new_controls(dapm, t5325_dapm_widgets,
+ ARRAY_SIZE(t5325_dapm_widgets));
+
+ snd_soc_dapm_add_routes(dapm, t5325_route, ARRAY_SIZE(t5325_route));
+
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Speaker");
+
+ snd_soc_dapm_sync(dapm);
+
+ return 0;
+}
+
+static struct snd_soc_dai_link t5325_dai[] = {
+{
+ .name = "ALC5621",
+ .stream_name = "ALC5621 HiFi",
+ .cpu_dai_name = "kirkwood-i2s",
+ .platform_name = "kirkwood-pcm-audio",
+ .codec_dai_name = "alc5621-hifi",
+ .codec_name = "alc562x-codec.0-001a",
+ .ops = &t5325_ops,
+ .init = t5325_dai_init,
+},
+};
+
+
+static struct snd_soc_card t5325 = {
+ .name = "t5325",
+ .dai_link = t5325_dai,
+ .num_links = ARRAY_SIZE(t5325_dai),
+};
+
+static struct platform_device *t5325_snd_device;
+
+static int __init t5325_init(void)
+{
+ int ret;
+
+ if (!machine_is_t5325())
+ return 0;
+
+ t5325_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!t5325_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(t5325_snd_device,
+ &t5325);
+
+ ret = platform_device_add(t5325_snd_device);
+ if (ret) {
+ printk(KERN_ERR "%s: platform_device_add failed\n", __func__);
+ platform_device_put(t5325_snd_device);
+ }
+
+ return ret;
+}
+module_init(t5325_init);
+
+static void __exit t5325_exit(void)
+{
+ platform_device_unregister(t5325_snd_device);
+}
+module_exit(t5325_exit);
+
+MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
+MODULE_DESCRIPTION("ALSA SoC t5325 audio client");
+MODULE_LICENSE("GPL");
static int am3517evm_aic23_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
/* Add am3517-evm specific widgets */
- snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* Set up davinci-evm specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* always connected */
- snd_soc_dapm_enable_pin(codec, "Line Out");
- snd_soc_dapm_enable_pin(codec, "Line In");
- snd_soc_dapm_enable_pin(codec, "Mic In");
+ snd_soc_dapm_enable_pin(dapm, "Line Out");
+ snd_soc_dapm_enable_pin(dapm, "Line In");
+ snd_soc_dapm_enable_pin(dapm, "Mic In");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
#include <linux/spinlock.h>
#include <linux/tty.h>
-#include <sound/soc-dapm.h>
+#include <sound/soc.h>
#include <sound/jack.h>
#include <asm/mach-types.h>
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
unsigned short pins;
int pin, changed = 0;
/* Setup pins after corresponding bits if changed */
pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
- if (pin != snd_soc_dapm_get_pin_status(codec, "Mouthpiece")) {
+ if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(codec, "Mouthpiece");
+ snd_soc_dapm_enable_pin(dapm, "Mouthpiece");
else
- snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+ snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
}
pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
- if (pin != snd_soc_dapm_get_pin_status(codec, "Earpiece")) {
+ if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(codec, "Earpiece");
+ snd_soc_dapm_enable_pin(dapm, "Earpiece");
else
- snd_soc_dapm_disable_pin(codec, "Earpiece");
+ snd_soc_dapm_disable_pin(dapm, "Earpiece");
}
pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
- if (pin != snd_soc_dapm_get_pin_status(codec, "Microphone")) {
+ if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(codec, "Microphone");
+ snd_soc_dapm_enable_pin(dapm, "Microphone");
else
- snd_soc_dapm_disable_pin(codec, "Microphone");
+ snd_soc_dapm_disable_pin(dapm, "Microphone");
}
pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
- if (pin != snd_soc_dapm_get_pin_status(codec, "Speaker")) {
+ if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(codec, "Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Speaker");
else
- snd_soc_dapm_disable_pin(codec, "Speaker");
+ snd_soc_dapm_disable_pin(dapm, "Speaker");
}
pin = !!(pins & (1 << AMS_DELTA_AGC));
if (pin != ams_delta_audio_agc) {
ams_delta_audio_agc = pin;
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(codec, "AGCIN");
+ snd_soc_dapm_enable_pin(dapm, "AGCIN");
else
- snd_soc_dapm_disable_pin(codec, "AGCIN");
+ snd_soc_dapm_disable_pin(dapm, "AGCIN");
}
if (changed)
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
mutex_unlock(&codec->mutex);
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
unsigned short pins, mode;
- pins = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") <<
+ pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") <<
AMS_DELTA_MOUTHPIECE) |
- (snd_soc_dapm_get_pin_status(codec, "Earpiece") <<
+ (snd_soc_dapm_get_pin_status(dapm, "Earpiece") <<
AMS_DELTA_EARPIECE));
if (pins)
- pins |= (snd_soc_dapm_get_pin_status(codec, "Microphone") <<
+ pins |= (snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
AMS_DELTA_MICROPHONE);
else
- pins = ((snd_soc_dapm_get_pin_status(codec, "Microphone") <<
+ pins = ((snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
AMS_DELTA_MICROPHONE) |
- (snd_soc_dapm_get_pin_status(codec, "Speaker") <<
+ (snd_soc_dapm_get_pin_status(dapm, "Speaker") <<
AMS_DELTA_SPEAKER) |
(ams_delta_audio_agc << AMS_DELTA_AGC));
static void cx81801_close(struct tty_struct *tty)
{
struct snd_soc_codec *codec = tty->disc_data;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
del_timer_sync(&cx81801_timer);
v253_ops.close(tty);
/* Revert back to default audio input/output constellation */
- snd_soc_dapm_disable_pin(codec, "Mouthpiece");
- snd_soc_dapm_enable_pin(codec, "Earpiece");
- snd_soc_dapm_enable_pin(codec, "Microphone");
- snd_soc_dapm_disable_pin(codec, "Speaker");
- snd_soc_dapm_disable_pin(codec, "AGCIN");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
+ snd_soc_dapm_enable_pin(dapm, "Earpiece");
+ snd_soc_dapm_enable_pin(dapm, "Microphone");
+ snd_soc_dapm_disable_pin(dapm, "Speaker");
+ snd_soc_dapm_disable_pin(dapm, "AGCIN");
+ snd_soc_dapm_sync(dapm);
}
/* Line discipline .hangup() */
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
case SND_SOC_BIAS_STANDBY:
- if (codec->bias_level == SND_SOC_BIAS_OFF)
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
AMS_DELTA_LATCH2_MODEM_NRESET);
break;
case SND_SOC_BIAS_OFF:
- if (codec->bias_level != SND_SOC_BIAS_OFF)
+ if (codec->dapm.bias_level != SND_SOC_BIAS_OFF)
ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
0);
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_card *card = rtd->card;
int ret;
}
/* Add board specific DAPM widgets and routes */
- ret = snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets,
+ ret = snd_soc_dapm_new_controls(dapm, ams_delta_dapm_widgets,
ARRAY_SIZE(ams_delta_dapm_widgets));
if (ret) {
dev_warn(card->dev,
return 0;
}
- ret = snd_soc_dapm_add_routes(codec, ams_delta_audio_map,
+ ret = snd_soc_dapm_add_routes(dapm, ams_delta_audio_map,
ARRAY_SIZE(ams_delta_audio_map));
if (ret) {
dev_warn(card->dev,
}
/* Set up initial pin constellation */
- snd_soc_dapm_disable_pin(codec, "Mouthpiece");
- snd_soc_dapm_enable_pin(codec, "Earpiece");
- snd_soc_dapm_enable_pin(codec, "Microphone");
- snd_soc_dapm_disable_pin(codec, "Speaker");
- snd_soc_dapm_disable_pin(codec, "AGCIN");
- snd_soc_dapm_disable_pin(codec, "AGCOUT");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
+ snd_soc_dapm_enable_pin(dapm, "Earpiece");
+ snd_soc_dapm_enable_pin(dapm, "Microphone");
+ snd_soc_dapm_disable_pin(dapm, "Speaker");
+ snd_soc_dapm_disable_pin(dapm, "AGCIN");
+ snd_soc_dapm_disable_pin(dapm, "AGCOUT");
+ snd_soc_dapm_sync(dapm);
/* Add virtual switch */
ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
static void n810_ext_control(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int hp = 0, line1l = 0;
switch (n810_jack_func) {
}
if (n810_spk_func)
- snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(codec, "Ext Spk");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk");
if (hp)
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
else
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
if (line1l)
- snd_soc_dapm_enable_pin(codec, "LINE1L");
+ snd_soc_dapm_enable_pin(dapm, "LINE1L");
else
- snd_soc_dapm_disable_pin(codec, "LINE1L");
+ snd_soc_dapm_disable_pin(dapm, "LINE1L");
if (n810_dmic_func)
- snd_soc_dapm_enable_pin(codec, "DMic");
+ snd_soc_dapm_enable_pin(dapm, "DMic");
else
- snd_soc_dapm_disable_pin(codec, "DMic");
+ snd_soc_dapm_disable_pin(dapm, "DMic");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
}
static int n810_startup(struct snd_pcm_substream *substream)
static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* Not connected */
- snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
- snd_soc_dapm_nc_pin(codec, "HPLCOM");
- snd_soc_dapm_nc_pin(codec, "HPRCOM");
- snd_soc_dapm_nc_pin(codec, "MIC3L");
- snd_soc_dapm_nc_pin(codec, "MIC3R");
- snd_soc_dapm_nc_pin(codec, "LINE1R");
- snd_soc_dapm_nc_pin(codec, "LINE2L");
- snd_soc_dapm_nc_pin(codec, "LINE2R");
+ snd_soc_dapm_nc_pin(dapm, "MONO_LOUT");
+ snd_soc_dapm_nc_pin(dapm, "HPLCOM");
+ snd_soc_dapm_nc_pin(dapm, "HPRCOM");
+ snd_soc_dapm_nc_pin(dapm, "MIC3L");
+ snd_soc_dapm_nc_pin(dapm, "MIC3R");
+ snd_soc_dapm_nc_pin(dapm, "LINE1R");
+ snd_soc_dapm_nc_pin(dapm, "LINE2L");
+ snd_soc_dapm_nc_pin(dapm, "LINE2R");
/* Add N810 specific controls */
err = snd_soc_add_controls(codec, aic33_n810_controls,
return err;
/* Add N810 specific widgets */
- snd_soc_dapm_new_controls(codec, aic33_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, aic33_dapm_widgets,
ARRAY_SIZE(aic33_dapm_widgets));
/* Set up N810 specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
/* All TWL4030 output pins are floating */
- snd_soc_dapm_nc_pin(codec, "EARPIECE");
- snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
- snd_soc_dapm_nc_pin(codec, "PREDRIVER");
- snd_soc_dapm_nc_pin(codec, "HSOL");
- snd_soc_dapm_nc_pin(codec, "HSOR");
- snd_soc_dapm_nc_pin(codec, "CARKITL");
- snd_soc_dapm_nc_pin(codec, "CARKITR");
- snd_soc_dapm_nc_pin(codec, "HFL");
- snd_soc_dapm_nc_pin(codec, "HFR");
- snd_soc_dapm_nc_pin(codec, "VIBRA");
-
- ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
+ snd_soc_dapm_nc_pin(dapm, "EARPIECE");
+ snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
+ snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
+ snd_soc_dapm_nc_pin(dapm, "HSOL");
+ snd_soc_dapm_nc_pin(dapm, "HSOR");
+ snd_soc_dapm_nc_pin(dapm, "CARKITL");
+ snd_soc_dapm_nc_pin(dapm, "CARKITR");
+ snd_soc_dapm_nc_pin(dapm, "HFL");
+ snd_soc_dapm_nc_pin(dapm, "HFR");
+ snd_soc_dapm_nc_pin(dapm, "VIBRA");
+
+ ret = snd_soc_dapm_new_controls(dapm, omap3pandora_out_dapm_widgets,
ARRAY_SIZE(omap3pandora_out_dapm_widgets));
if (ret < 0)
return ret;
- snd_soc_dapm_add_routes(codec, omap3pandora_out_map,
+ snd_soc_dapm_add_routes(dapm, omap3pandora_out_map,
ARRAY_SIZE(omap3pandora_out_map));
- return snd_soc_dapm_sync(codec);
+ return snd_soc_dapm_sync(dapm);
}
static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
/* Not comnnected */
- snd_soc_dapm_nc_pin(codec, "HSMIC");
- snd_soc_dapm_nc_pin(codec, "CARKITMIC");
- snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
- snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
+ snd_soc_dapm_nc_pin(dapm, "HSMIC");
+ snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
+ snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
+ snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
- ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets,
+ ret = snd_soc_dapm_new_controls(dapm, omap3pandora_in_dapm_widgets,
ARRAY_SIZE(omap3pandora_in_dapm_widgets));
if (ret < 0)
return ret;
- snd_soc_dapm_add_routes(codec, omap3pandora_in_map,
+ snd_soc_dapm_add_routes(dapm, omap3pandora_in_map,
ARRAY_SIZE(omap3pandora_in_map));
- return snd_soc_dapm_sync(codec);
+ return snd_soc_dapm_sync(dapm);
}
static struct snd_soc_ops omap3pandora_ops = {
static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
/* Add osk5912 specific widgets */
- snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* Set up osk5912 specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Line In");
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Line In");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static void rx51_ext_control(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
if (rx51_spk_func)
- snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(codec, "Ext Spk");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk");
if (rx51_dmic_func)
- snd_soc_dapm_enable_pin(codec, "DMic");
+ snd_soc_dapm_enable_pin(dapm, "DMic");
else
- snd_soc_dapm_disable_pin(codec, "DMic");
+ snd_soc_dapm_disable_pin(dapm, "DMic");
gpio_set_value(RX51_TVOUT_SEL_GPIO,
rx51_jack_func == RX51_JACK_TVOUT);
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
}
static int rx51_startup(struct snd_pcm_substream *substream)
static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* Set up NC codec pins */
- snd_soc_dapm_nc_pin(codec, "MIC3L");
- snd_soc_dapm_nc_pin(codec, "MIC3R");
- snd_soc_dapm_nc_pin(codec, "LINE1R");
+ snd_soc_dapm_nc_pin(dapm, "MIC3L");
+ snd_soc_dapm_nc_pin(dapm, "MIC3R");
+ snd_soc_dapm_nc_pin(dapm, "LINE1R");
/* Add RX-51 specific controls */
err = snd_soc_add_controls(codec, aic34_rx51_controls,
return err;
/* Add RX-51 specific widgets */
- snd_soc_dapm_new_controls(codec, aic34_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, aic34_dapm_widgets,
ARRAY_SIZE(aic34_dapm_widgets));
/* Set up RX-51 specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
/* AV jack detection */
err = snd_soc_jack_new(codec, "AV Jack",
static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
/* Add SDP3430 specific widgets */
- ret = snd_soc_dapm_new_controls(codec, sdp3430_twl4030_dapm_widgets,
+ ret = snd_soc_dapm_new_controls(dapm, sdp3430_twl4030_dapm_widgets,
ARRAY_SIZE(sdp3430_twl4030_dapm_widgets));
if (ret)
return ret;
/* Set up SDP3430 specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* SDP3430 connected pins */
- snd_soc_dapm_enable_pin(codec, "Ext Mic");
- snd_soc_dapm_enable_pin(codec, "Ext Spk");
- snd_soc_dapm_disable_pin(codec, "Headset Mic");
- snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+ snd_soc_dapm_enable_pin(dapm, "Ext Mic");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
/* TWL4030 not connected pins */
- snd_soc_dapm_nc_pin(codec, "AUXL");
- snd_soc_dapm_nc_pin(codec, "AUXR");
- snd_soc_dapm_nc_pin(codec, "CARKITMIC");
- snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
- snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
-
- snd_soc_dapm_nc_pin(codec, "OUTL");
- snd_soc_dapm_nc_pin(codec, "OUTR");
- snd_soc_dapm_nc_pin(codec, "EARPIECE");
- snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
- snd_soc_dapm_nc_pin(codec, "PREDRIVER");
- snd_soc_dapm_nc_pin(codec, "CARKITL");
- snd_soc_dapm_nc_pin(codec, "CARKITR");
-
- ret = snd_soc_dapm_sync(codec);
+ snd_soc_dapm_nc_pin(dapm, "AUXL");
+ snd_soc_dapm_nc_pin(dapm, "AUXR");
+ snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
+ snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
+ snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
+
+ snd_soc_dapm_nc_pin(dapm, "OUTL");
+ snd_soc_dapm_nc_pin(dapm, "OUTR");
+ snd_soc_dapm_nc_pin(dapm, "EARPIECE");
+ snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
+ snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
+ snd_soc_dapm_nc_pin(dapm, "CARKITL");
+ snd_soc_dapm_nc_pin(dapm, "CARKITR");
+
+ ret = snd_soc_dapm_sync(dapm);
if (ret)
return ret;
static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
/* Add SDP4430 specific controls */
return ret;
/* Add SDP4430 specific widgets */
- ret = snd_soc_dapm_new_controls(codec, sdp4430_twl6040_dapm_widgets,
+ ret = snd_soc_dapm_new_controls(dapm, sdp4430_twl6040_dapm_widgets,
ARRAY_SIZE(sdp4430_twl6040_dapm_widgets));
if (ret)
return ret;
/* Set up SDP4430 specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* SDP4430 connected pins */
- snd_soc_dapm_enable_pin(codec, "Ext Mic");
- snd_soc_dapm_enable_pin(codec, "Ext Spk");
- snd_soc_dapm_enable_pin(codec, "Headset Mic");
- snd_soc_dapm_enable_pin(codec, "Headset Stereophone");
+ snd_soc_dapm_enable_pin(dapm, "Ext Mic");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin(dapm, "Headset Stereophone");
/* TWL6040 not connected pins */
- snd_soc_dapm_nc_pin(codec, "AFML");
- snd_soc_dapm_nc_pin(codec, "AFMR");
+ snd_soc_dapm_nc_pin(dapm, "AFML");
+ snd_soc_dapm_nc_pin(dapm, "AFMR");
- ret = snd_soc_dapm_sync(codec);
+ ret = snd_soc_dapm_sync(dapm);
return ret;
}
static int zoom2_twl4030_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
/* Add Zoom2 specific widgets */
- ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets,
+ ret = snd_soc_dapm_new_controls(dapm, zoom2_twl4030_dapm_widgets,
ARRAY_SIZE(zoom2_twl4030_dapm_widgets));
if (ret)
return ret;
/* Set up Zoom2 specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* Zoom2 connected pins */
- snd_soc_dapm_enable_pin(codec, "Ext Mic");
- snd_soc_dapm_enable_pin(codec, "Ext Spk");
- snd_soc_dapm_enable_pin(codec, "Headset Mic");
- snd_soc_dapm_enable_pin(codec, "Headset Stereophone");
- snd_soc_dapm_enable_pin(codec, "Aux In");
+ snd_soc_dapm_enable_pin(dapm, "Ext Mic");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin(dapm, "Headset Stereophone");
+ snd_soc_dapm_enable_pin(dapm, "Aux In");
/* TWL4030 not connected pins */
- snd_soc_dapm_nc_pin(codec, "CARKITMIC");
- snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
- snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
- snd_soc_dapm_nc_pin(codec, "EARPIECE");
- snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
- snd_soc_dapm_nc_pin(codec, "PREDRIVER");
- snd_soc_dapm_nc_pin(codec, "CARKITL");
- snd_soc_dapm_nc_pin(codec, "CARKITR");
-
- ret = snd_soc_dapm_sync(codec);
+ snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
+ snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
+ snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
+ snd_soc_dapm_nc_pin(dapm, "EARPIECE");
+ snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
+ snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
+ snd_soc_dapm_nc_pin(dapm, "CARKITL");
+ snd_soc_dapm_nc_pin(dapm, "CARKITR");
+
+ ret = snd_soc_dapm_sync(dapm);
return ret;
}
static void corgi_ext_control(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
/* set up jack connection */
switch (corgi_jack_func) {
case CORGI_HP:
/* set = unmute headphone */
gpio_set_value(CORGI_GPIO_MUTE_L, 1);
gpio_set_value(CORGI_GPIO_MUTE_R, 1);
- snd_soc_dapm_disable_pin(codec, "Mic Jack");
- snd_soc_dapm_disable_pin(codec, "Line Jack");
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headset Jack");
break;
case CORGI_MIC:
/* reset = mute headphone */
gpio_set_value(CORGI_GPIO_MUTE_L, 0);
gpio_set_value(CORGI_GPIO_MUTE_R, 0);
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
- snd_soc_dapm_disable_pin(codec, "Line Jack");
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
- snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headset Jack");
break;
case CORGI_LINE:
gpio_set_value(CORGI_GPIO_MUTE_L, 0);
gpio_set_value(CORGI_GPIO_MUTE_R, 0);
- snd_soc_dapm_disable_pin(codec, "Mic Jack");
- snd_soc_dapm_enable_pin(codec, "Line Jack");
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
- snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headset Jack");
break;
case CORGI_HEADSET:
gpio_set_value(CORGI_GPIO_MUTE_L, 0);
gpio_set_value(CORGI_GPIO_MUTE_R, 1);
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
- snd_soc_dapm_disable_pin(codec, "Line Jack");
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Headset Jack");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headset Jack");
break;
}
if (corgi_spk_func == CORGI_SPK_ON)
- snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(codec, "Ext Spk");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk");
/* signal a DAPM event */
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
}
static int corgi_startup(struct snd_pcm_substream *substream)
static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
- snd_soc_dapm_nc_pin(codec, "LLINEIN");
- snd_soc_dapm_nc_pin(codec, "RLINEIN");
+ snd_soc_dapm_nc_pin(dapm, "LLINEIN");
+ snd_soc_dapm_nc_pin(dapm, "RLINEIN");
/* Add corgi specific controls */
err = snd_soc_add_controls(codec, wm8731_corgi_controls,
return err;
/* Add corgi specific widgets */
- snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8731_dapm_widgets,
ARRAY_SIZE(wm8731_dapm_widgets));
/* Set up corgi specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int e740_ac97_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
-
- snd_soc_dapm_nc_pin(codec, "HPOUTL");
- snd_soc_dapm_nc_pin(codec, "HPOUTR");
- snd_soc_dapm_nc_pin(codec, "PHONE");
- snd_soc_dapm_nc_pin(codec, "LINEINL");
- snd_soc_dapm_nc_pin(codec, "LINEINR");
- snd_soc_dapm_nc_pin(codec, "CDINL");
- snd_soc_dapm_nc_pin(codec, "CDINR");
- snd_soc_dapm_nc_pin(codec, "PCBEEP");
- snd_soc_dapm_nc_pin(codec, "MIC2");
-
- snd_soc_dapm_new_controls(codec, e740_dapm_widgets,
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_nc_pin(dapm, "HPOUTL");
+ snd_soc_dapm_nc_pin(dapm, "HPOUTR");
+ snd_soc_dapm_nc_pin(dapm, "PHONE");
+ snd_soc_dapm_nc_pin(dapm, "LINEINL");
+ snd_soc_dapm_nc_pin(dapm, "LINEINR");
+ snd_soc_dapm_nc_pin(dapm, "CDINL");
+ snd_soc_dapm_nc_pin(dapm, "CDINR");
+ snd_soc_dapm_nc_pin(dapm, "PCBEEP");
+ snd_soc_dapm_nc_pin(dapm, "MIC2");
+
+ snd_soc_dapm_new_controls(dapm, e740_dapm_widgets,
ARRAY_SIZE(e740_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int e750_ac97_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
-
- snd_soc_dapm_nc_pin(codec, "LOUT");
- snd_soc_dapm_nc_pin(codec, "ROUT");
- snd_soc_dapm_nc_pin(codec, "PHONE");
- snd_soc_dapm_nc_pin(codec, "LINEINL");
- snd_soc_dapm_nc_pin(codec, "LINEINR");
- snd_soc_dapm_nc_pin(codec, "CDINL");
- snd_soc_dapm_nc_pin(codec, "CDINR");
- snd_soc_dapm_nc_pin(codec, "PCBEEP");
- snd_soc_dapm_nc_pin(codec, "MIC2");
-
- snd_soc_dapm_new_controls(codec, e750_dapm_widgets,
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_nc_pin(dapm, "LOUT");
+ snd_soc_dapm_nc_pin(dapm, "ROUT");
+ snd_soc_dapm_nc_pin(dapm, "PHONE");
+ snd_soc_dapm_nc_pin(dapm, "LINEINL");
+ snd_soc_dapm_nc_pin(dapm, "LINEINR");
+ snd_soc_dapm_nc_pin(dapm, "CDINL");
+ snd_soc_dapm_nc_pin(dapm, "CDINR");
+ snd_soc_dapm_nc_pin(dapm, "PCBEEP");
+ snd_soc_dapm_nc_pin(dapm, "MIC2");
+
+ snd_soc_dapm_new_controls(dapm, e750_dapm_widgets,
ARRAY_SIZE(e750_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int e800_ac97_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_new_controls(codec, e800_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, e800_dapm_widgets,
ARRAY_SIZE(e800_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_sync(dapm);
return 0;
}
static void magician_ext_control(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
if (magician_spk_switch)
- snd_soc_dapm_enable_pin(codec, "Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Speaker");
else
- snd_soc_dapm_disable_pin(codec, "Speaker");
+ snd_soc_dapm_disable_pin(dapm, "Speaker");
if (magician_hp_switch)
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
else
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
switch (magician_in_sel) {
case MAGICIAN_MIC:
- snd_soc_dapm_disable_pin(codec, "Headset Mic");
- snd_soc_dapm_enable_pin(codec, "Call Mic");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin(dapm, "Call Mic");
break;
case MAGICIAN_MIC_EXT:
- snd_soc_dapm_disable_pin(codec, "Call Mic");
- snd_soc_dapm_enable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Call Mic");
+ snd_soc_dapm_enable_pin(dapm, "Headset Mic");
break;
}
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
}
static int magician_startup(struct snd_pcm_substream *substream)
static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* NC codec pins */
- snd_soc_dapm_nc_pin(codec, "VOUTLHP");
- snd_soc_dapm_nc_pin(codec, "VOUTRHP");
+ snd_soc_dapm_nc_pin(dapm, "VOUTLHP");
+ snd_soc_dapm_nc_pin(dapm, "VOUTRHP");
/* FIXME: is anything connected here? */
- snd_soc_dapm_nc_pin(codec, "VINL");
- snd_soc_dapm_nc_pin(codec, "VINR");
+ snd_soc_dapm_nc_pin(dapm, "VINL");
+ snd_soc_dapm_nc_pin(dapm, "VINR");
/* Add magician specific controls */
err = snd_soc_add_controls(codec, uda1380_magician_controls,
return err;
/* Add magician specific widgets */
- snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
ARRAY_SIZE(uda1380_dapm_widgets));
/* Set up magician specific audio path interconnects */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
unsigned short reg;
/* Add mioa701 specific widgets */
- snd_soc_dapm_new_controls(codec, ARRAY_AND_SIZE(mioa701_dapm_widgets));
+ snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets));
/* Set up mioa701 specific audio path audio_mapnects */
- snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map));
/* Prepare GPIO8 for rear speaker amplifier */
reg = codec->driver->read(codec, AC97_GPIO_CFG);
reg = codec->driver->read(codec, AC97_3D_CONTROL);
codec->driver->write(codec, AC97_3D_CONTROL, reg | 0xc000);
- snd_soc_dapm_enable_pin(codec, "Front Speaker");
- snd_soc_dapm_enable_pin(codec, "Rear Speaker");
- snd_soc_dapm_enable_pin(codec, "Front Mic");
- snd_soc_dapm_enable_pin(codec, "GSM Line In");
- snd_soc_dapm_enable_pin(codec, "GSM Line Out");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_enable_pin(dapm, "Front Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Rear Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Front Mic");
+ snd_soc_dapm_enable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* add palm27x specific widgets */
- err = snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
+ err = snd_soc_dapm_new_controls(dapm, palm27x_dapm_widgets,
ARRAY_SIZE(palm27x_dapm_widgets));
if (err)
return err;
/* set up palm27x specific audio path audio_map */
- err = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ err = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
if (err)
return err;
/* connected pins */
if (machine_is_palmld())
- snd_soc_dapm_enable_pin(codec, "MIC1");
- snd_soc_dapm_enable_pin(codec, "HPOUTL");
- snd_soc_dapm_enable_pin(codec, "HPOUTR");
- snd_soc_dapm_enable_pin(codec, "LOUT2");
- snd_soc_dapm_enable_pin(codec, "ROUT2");
+ snd_soc_dapm_enable_pin(dapm, "MIC1");
+ snd_soc_dapm_enable_pin(dapm, "HPOUTL");
+ snd_soc_dapm_enable_pin(dapm, "HPOUTR");
+ snd_soc_dapm_enable_pin(dapm, "LOUT2");
+ snd_soc_dapm_enable_pin(dapm, "ROUT2");
/* not connected pins */
- snd_soc_dapm_nc_pin(codec, "OUT3");
- snd_soc_dapm_nc_pin(codec, "MONOOUT");
- snd_soc_dapm_nc_pin(codec, "LINEINL");
- snd_soc_dapm_nc_pin(codec, "LINEINR");
- snd_soc_dapm_nc_pin(codec, "PCBEEP");
- snd_soc_dapm_nc_pin(codec, "PHONE");
- snd_soc_dapm_nc_pin(codec, "MIC2");
-
- err = snd_soc_dapm_sync(codec);
+ snd_soc_dapm_nc_pin(dapm, "OUT3");
+ snd_soc_dapm_nc_pin(dapm, "MONOOUT");
+ snd_soc_dapm_nc_pin(dapm, "LINEINL");
+ snd_soc_dapm_nc_pin(dapm, "LINEINR");
+ snd_soc_dapm_nc_pin(dapm, "PCBEEP");
+ snd_soc_dapm_nc_pin(dapm, "PHONE");
+ snd_soc_dapm_nc_pin(dapm, "MIC2");
+
+ err = snd_soc_dapm_sync(dapm);
if (err)
return err;
static void poodle_ext_control(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
/* set up jack connection */
if (poodle_jack_func == POODLE_HP) {
/* set = unmute headphone */
POODLE_LOCOMO_GPIO_MUTE_L, 1);
locomo_gpio_write(&poodle_locomo_device.dev,
POODLE_LOCOMO_GPIO_MUTE_R, 1);
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
} else {
locomo_gpio_write(&poodle_locomo_device.dev,
POODLE_LOCOMO_GPIO_MUTE_L, 0);
locomo_gpio_write(&poodle_locomo_device.dev,
POODLE_LOCOMO_GPIO_MUTE_R, 0);
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
}
/* set the enpoints to their new connetion states */
if (poodle_spk_func == POODLE_SPK_ON)
- snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(codec, "Ext Spk");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk");
/* signal a DAPM event */
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
}
static int poodle_startup(struct snd_pcm_substream *substream)
static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
- snd_soc_dapm_nc_pin(codec, "LLINEIN");
- snd_soc_dapm_nc_pin(codec, "RLINEIN");
- snd_soc_dapm_enable_pin(codec, "MICIN");
+ snd_soc_dapm_nc_pin(dapm, "LLINEIN");
+ snd_soc_dapm_nc_pin(dapm, "RLINEIN");
+ snd_soc_dapm_enable_pin(dapm, "MICIN");
/* Add poodle specific controls */
err = snd_soc_add_controls(codec, wm8731_poodle_controls,
return err;
/* Add poodle specific widgets */
- snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8731_dapm_widgets,
ARRAY_SIZE(wm8731_dapm_widgets));
/* Set up poodle specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
- snd_soc_dapm_new_controls(codec, saarb_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, saarb_dapm_widgets,
ARRAY_SIZE(saarb_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* connected pins */
- snd_soc_dapm_enable_pin(codec, "Ext Speaker");
- snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
- snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
- snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
- snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+ snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
+ snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
+ snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
- ret = snd_soc_dapm_sync(codec);
+ ret = snd_soc_dapm_sync(dapm);
if (ret)
return ret;
static void spitz_ext_control(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
if (spitz_spk_func == SPITZ_SPK_ON)
- snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(codec, "Ext Spk");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk");
/* set up jack connection */
switch (spitz_jack_func) {
case SPITZ_HP:
/* enable and unmute hp jack, disable mic bias */
- snd_soc_dapm_disable_pin(codec, "Headset Jack");
- snd_soc_dapm_disable_pin(codec, "Mic Jack");
- snd_soc_dapm_disable_pin(codec, "Line Jack");
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 1);
gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
break;
case SPITZ_MIC:
/* enable mic jack and bias, mute hp */
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
- snd_soc_dapm_disable_pin(codec, "Headset Jack");
- snd_soc_dapm_disable_pin(codec, "Line Jack");
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
case SPITZ_LINE:
/* enable line jack, disable mic bias and mute hp */
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
- snd_soc_dapm_disable_pin(codec, "Headset Jack");
- snd_soc_dapm_disable_pin(codec, "Mic Jack");
- snd_soc_dapm_enable_pin(codec, "Line Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin(dapm, "Line Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
case SPITZ_HEADSET:
/* enable and unmute headset jack enable mic bias, mute L hp */
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
- snd_soc_dapm_disable_pin(codec, "Line Jack");
- snd_soc_dapm_enable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headset Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
break;
case SPITZ_HP_OFF:
/* jack removed, everything off */
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
- snd_soc_dapm_disable_pin(codec, "Headset Jack");
- snd_soc_dapm_disable_pin(codec, "Mic Jack");
- snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin(dapm, "Line Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
}
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
}
static int spitz_startup(struct snd_pcm_substream *substream)
static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* NC codec pins */
- snd_soc_dapm_nc_pin(codec, "RINPUT1");
- snd_soc_dapm_nc_pin(codec, "LINPUT2");
- snd_soc_dapm_nc_pin(codec, "RINPUT2");
- snd_soc_dapm_nc_pin(codec, "LINPUT3");
- snd_soc_dapm_nc_pin(codec, "RINPUT3");
- snd_soc_dapm_nc_pin(codec, "OUT3");
- snd_soc_dapm_nc_pin(codec, "MONO1");
+ snd_soc_dapm_nc_pin(dapm, "RINPUT1");
+ snd_soc_dapm_nc_pin(dapm, "LINPUT2");
+ snd_soc_dapm_nc_pin(dapm, "RINPUT2");
+ snd_soc_dapm_nc_pin(dapm, "LINPUT3");
+ snd_soc_dapm_nc_pin(dapm, "RINPUT3");
+ snd_soc_dapm_nc_pin(dapm, "OUT3");
+ snd_soc_dapm_nc_pin(dapm, "MONO1");
/* Add spitz specific controls */
err = snd_soc_add_controls(codec, wm8750_spitz_controls,
return err;
/* Add spitz specific widgets */
- snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets,
ARRAY_SIZE(wm8750_dapm_widgets));
/* Set up spitz specific audio paths */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
- snd_soc_dapm_new_controls(codec, evb3_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, evb3_dapm_widgets,
ARRAY_SIZE(evb3_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* connected pins */
- snd_soc_dapm_enable_pin(codec, "Ext Speaker");
- snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
- snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
- snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
- snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+ snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
+ snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
+ snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
- ret = snd_soc_dapm_sync(codec);
+ ret = snd_soc_dapm_sync(dapm);
if (ret)
return ret;
static void tosa_ext_control(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
/* set up jack connection */
switch (tosa_jack_func) {
case TOSA_HP:
- snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headset Jack");
break;
case TOSA_MIC_INT:
- snd_soc_dapm_enable_pin(codec, "Mic (Internal)");
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
- snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_enable_pin(dapm, "Mic (Internal)");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headset Jack");
break;
case TOSA_HEADSET:
- snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headset Jack");
break;
}
if (tosa_spk_func == TOSA_SPK_ON)
- snd_soc_dapm_enable_pin(codec, "Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Speaker");
else
- snd_soc_dapm_disable_pin(codec, "Speaker");
+ snd_soc_dapm_disable_pin(dapm, "Speaker");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
}
static int tosa_startup(struct snd_pcm_substream *substream)
static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
- snd_soc_dapm_nc_pin(codec, "OUT3");
- snd_soc_dapm_nc_pin(codec, "MONOOUT");
+ snd_soc_dapm_nc_pin(dapm, "OUT3");
+ snd_soc_dapm_nc_pin(dapm, "MONOOUT");
/* add tosa specific controls */
err = snd_soc_add_controls(codec, tosa_controls,
return err;
/* add tosa specific widgets */
- snd_soc_dapm_new_controls(codec, tosa_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, tosa_dapm_widgets,
ARRAY_SIZE(tosa_dapm_widgets));
/* set up tosa specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
/* NC codec pins */
- snd_soc_dapm_disable_pin(codec, "LINPUT3");
- snd_soc_dapm_disable_pin(codec, "RINPUT3");
- snd_soc_dapm_disable_pin(codec, "OUT3");
- snd_soc_dapm_disable_pin(codec, "MONO");
+ snd_soc_dapm_disable_pin(dapm, "LINPUT3");
+ snd_soc_dapm_disable_pin(dapm, "RINPUT3");
+ snd_soc_dapm_disable_pin(dapm, "OUT3");
+ snd_soc_dapm_disable_pin(dapm, "MONO");
/* Add z2 specific widgets */
- snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets,
ARRAY_SIZE(wm8750_dapm_widgets));
/* Set up z2 specific audio paths */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- ret = snd_soc_dapm_sync(codec);
+ ret = snd_soc_dapm_sync(dapm);
if (ret)
goto err;
static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
if (clk_pout)
snd_soc_dai_set_pll(rtd->codec_dai, 0, 0,
clk_get_rate(pout), 0);
- snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, zylonite_dapm_widgets,
ARRAY_SIZE(zylonite_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* Static setup for now */
- snd_soc_dapm_enable_pin(codec, "Headphone");
- snd_soc_dapm_enable_pin(codec, "Headset Earpiece");
+ snd_soc_dapm_enable_pin(dapm, "Headphone");
+ snd_soc_dapm_enable_pin(dapm, "Headset Earpiece");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
/* add aquila specific widgets */
- snd_soc_dapm_new_controls(codec, aquila_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, aquila_dapm_widgets,
ARRAY_SIZE(aquila_dapm_widgets));
/* set up aquila specific audio routes */
- snd_soc_dapm_add_routes(codec, aquila_dapm_routes,
+ snd_soc_dapm_add_routes(dapm, aquila_dapm_routes,
ARRAY_SIZE(aquila_dapm_routes));
/* set endpoints to not connected */
- snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN");
- snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP");
- snd_soc_dapm_nc_pin(codec, "LINEOUT1N");
- snd_soc_dapm_nc_pin(codec, "LINEOUT1P");
- snd_soc_dapm_nc_pin(codec, "LINEOUT2N");
- snd_soc_dapm_nc_pin(codec, "LINEOUT2P");
- snd_soc_dapm_nc_pin(codec, "SPKOUTRN");
- snd_soc_dapm_nc_pin(codec, "SPKOUTRP");
-
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
+ snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
+ snd_soc_dapm_nc_pin(dapm, "SPKOUTRN");
+ snd_soc_dapm_nc_pin(dapm, "SPKOUTRP");
+
+ snd_soc_dapm_sync(dapm);
/* Headset jack detection */
ret = snd_soc_jack_new(&aquila, "Headset Jack",
static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
/* add goni specific widgets */
- snd_soc_dapm_new_controls(codec, goni_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, goni_dapm_widgets,
ARRAY_SIZE(goni_dapm_widgets));
/* set up goni specific audio routes */
- snd_soc_dapm_add_routes(codec, goni_dapm_routes,
+ snd_soc_dapm_add_routes(dapm, goni_dapm_routes,
ARRAY_SIZE(goni_dapm_routes));
/* set endpoints to not connected */
- snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN");
- snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP");
- snd_soc_dapm_nc_pin(codec, "LINEOUT1N");
- snd_soc_dapm_nc_pin(codec, "LINEOUT1P");
- snd_soc_dapm_nc_pin(codec, "LINEOUT2N");
- snd_soc_dapm_nc_pin(codec, "LINEOUT2P");
-
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
+ snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
+
+ snd_soc_dapm_sync(dapm);
/* Headset jack detection */
ret = snd_soc_jack_new(&goni, "Headset Jack",
static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* These endpoints are not being used. */
- snd_soc_dapm_nc_pin(codec, "LINPUT2");
- snd_soc_dapm_nc_pin(codec, "RINPUT2");
- snd_soc_dapm_nc_pin(codec, "LINPUT3");
- snd_soc_dapm_nc_pin(codec, "RINPUT3");
- snd_soc_dapm_nc_pin(codec, "OUT3");
- snd_soc_dapm_nc_pin(codec, "MONO");
+ snd_soc_dapm_nc_pin(dapm, "LINPUT2");
+ snd_soc_dapm_nc_pin(dapm, "RINPUT2");
+ snd_soc_dapm_nc_pin(dapm, "LINPUT3");
+ snd_soc_dapm_nc_pin(dapm, "RINPUT3");
+ snd_soc_dapm_nc_pin(dapm, "OUT3");
+ snd_soc_dapm_nc_pin(dapm, "MONO");
/* Add jive specific widgets */
- err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+ err = snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets,
ARRAY_SIZE(wm8750_dapm_widgets));
if (err) {
printk(KERN_ERR "%s: failed to add widgets (%d)\n",
return err;
}
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int neo1973_gta02_wm8753_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* set up NC codec pins */
- snd_soc_dapm_nc_pin(codec, "OUT3");
- snd_soc_dapm_nc_pin(codec, "OUT4");
- snd_soc_dapm_nc_pin(codec, "LINE1");
- snd_soc_dapm_nc_pin(codec, "LINE2");
+ snd_soc_dapm_nc_pin(dapm, "OUT3");
+ snd_soc_dapm_nc_pin(dapm, "OUT4");
+ snd_soc_dapm_nc_pin(dapm, "LINE1");
+ snd_soc_dapm_nc_pin(dapm, "LINE2");
/* Add neo1973 gta02 specific widgets */
- snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets,
ARRAY_SIZE(wm8753_dapm_widgets));
/* add neo1973 gta02 specific controls */
return err;
/* set up neo1973 gta02 specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* set endpoints to default off mode */
- snd_soc_dapm_disable_pin(codec, "Stereo Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line In");
- snd_soc_dapm_disable_pin(codec, "Headset Mic");
- snd_soc_dapm_disable_pin(codec, "Handset Mic");
- snd_soc_dapm_disable_pin(codec, "Handset Spk");
+ snd_soc_dapm_disable_pin(dapm, "Stereo Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Handset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Handset Spk");
/* allow audio paths from the GSM modem to run during suspend */
- snd_soc_dapm_ignore_suspend(codec, "Stereo Out");
- snd_soc_dapm_ignore_suspend(codec, "GSM Line Out");
- snd_soc_dapm_ignore_suspend(codec, "GSM Line In");
- snd_soc_dapm_ignore_suspend(codec, "Headset Mic");
- snd_soc_dapm_ignore_suspend(codec, "Handset Mic");
- snd_soc_dapm_ignore_suspend(codec, "Handset Spk");
-
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
+ snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out");
+ snd_soc_dapm_ignore_suspend(dapm, "GSM Line In");
+ snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
+ snd_soc_dapm_ignore_suspend(dapm, "Handset Mic");
+ snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
+
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
pr_debug("Entered %s\n", __func__);
switch (neo1973_scenario) {
case NEO_AUDIO_OFF:
- snd_soc_dapm_disable_pin(codec, "Audio Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line In");
- snd_soc_dapm_disable_pin(codec, "Headset Mic");
- snd_soc_dapm_disable_pin(codec, "Call Mic");
+ snd_soc_dapm_disable_pin(dapm, "Audio Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Call Mic");
break;
case NEO_GSM_CALL_AUDIO_HANDSET:
- snd_soc_dapm_enable_pin(codec, "Audio Out");
- snd_soc_dapm_enable_pin(codec, "GSM Line Out");
- snd_soc_dapm_enable_pin(codec, "GSM Line In");
- snd_soc_dapm_disable_pin(codec, "Headset Mic");
- snd_soc_dapm_enable_pin(codec, "Call Mic");
+ snd_soc_dapm_enable_pin(dapm, "Audio Out");
+ snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_enable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin(dapm, "Call Mic");
break;
case NEO_GSM_CALL_AUDIO_HEADSET:
- snd_soc_dapm_enable_pin(codec, "Audio Out");
- snd_soc_dapm_enable_pin(codec, "GSM Line Out");
- snd_soc_dapm_enable_pin(codec, "GSM Line In");
- snd_soc_dapm_enable_pin(codec, "Headset Mic");
- snd_soc_dapm_disable_pin(codec, "Call Mic");
+ snd_soc_dapm_enable_pin(dapm, "Audio Out");
+ snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_enable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Call Mic");
break;
case NEO_GSM_CALL_AUDIO_BLUETOOTH:
- snd_soc_dapm_disable_pin(codec, "Audio Out");
- snd_soc_dapm_enable_pin(codec, "GSM Line Out");
- snd_soc_dapm_enable_pin(codec, "GSM Line In");
- snd_soc_dapm_disable_pin(codec, "Headset Mic");
- snd_soc_dapm_disable_pin(codec, "Call Mic");
+ snd_soc_dapm_disable_pin(dapm, "Audio Out");
+ snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_enable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Call Mic");
break;
case NEO_STEREO_TO_SPEAKERS:
- snd_soc_dapm_enable_pin(codec, "Audio Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line In");
- snd_soc_dapm_disable_pin(codec, "Headset Mic");
- snd_soc_dapm_disable_pin(codec, "Call Mic");
+ snd_soc_dapm_enable_pin(dapm, "Audio Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Call Mic");
break;
case NEO_STEREO_TO_HEADPHONES:
- snd_soc_dapm_enable_pin(codec, "Audio Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line In");
- snd_soc_dapm_disable_pin(codec, "Headset Mic");
- snd_soc_dapm_disable_pin(codec, "Call Mic");
+ snd_soc_dapm_enable_pin(dapm, "Audio Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Call Mic");
break;
case NEO_CAPTURE_HANDSET:
- snd_soc_dapm_disable_pin(codec, "Audio Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line In");
- snd_soc_dapm_disable_pin(codec, "Headset Mic");
- snd_soc_dapm_enable_pin(codec, "Call Mic");
+ snd_soc_dapm_disable_pin(dapm, "Audio Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin(dapm, "Call Mic");
break;
case NEO_CAPTURE_HEADSET:
- snd_soc_dapm_disable_pin(codec, "Audio Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line In");
- snd_soc_dapm_enable_pin(codec, "Headset Mic");
- snd_soc_dapm_disable_pin(codec, "Call Mic");
+ snd_soc_dapm_disable_pin(dapm, "Audio Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Call Mic");
break;
case NEO_CAPTURE_BLUETOOTH:
- snd_soc_dapm_disable_pin(codec, "Audio Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line In");
- snd_soc_dapm_disable_pin(codec, "Headset Mic");
- snd_soc_dapm_disable_pin(codec, "Call Mic");
+ snd_soc_dapm_disable_pin(dapm, "Audio Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Call Mic");
break;
default:
- snd_soc_dapm_disable_pin(codec, "Audio Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line Out");
- snd_soc_dapm_disable_pin(codec, "GSM Line In");
- snd_soc_dapm_disable_pin(codec, "Headset Mic");
- snd_soc_dapm_disable_pin(codec, "Call Mic");
+ snd_soc_dapm_disable_pin(dapm, "Audio Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Call Mic");
}
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
pr_debug("Entered %s\n", __func__);
/* set up NC codec pins */
- snd_soc_dapm_nc_pin(codec, "LOUT2");
- snd_soc_dapm_nc_pin(codec, "ROUT2");
- snd_soc_dapm_nc_pin(codec, "OUT3");
- snd_soc_dapm_nc_pin(codec, "OUT4");
- snd_soc_dapm_nc_pin(codec, "LINE1");
- snd_soc_dapm_nc_pin(codec, "LINE2");
+ snd_soc_dapm_nc_pin(dapm, "LOUT2");
+ snd_soc_dapm_nc_pin(dapm, "ROUT2");
+ snd_soc_dapm_nc_pin(dapm, "OUT3");
+ snd_soc_dapm_nc_pin(dapm, "OUT4");
+ snd_soc_dapm_nc_pin(dapm, "LINE1");
+ snd_soc_dapm_nc_pin(dapm, "LINE2");
/* Add neo1973 specific widgets */
- snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets,
ARRAY_SIZE(wm8753_dapm_widgets));
/* set endpoints to default mode */
return err;
/* set up neo1973 specific audio routes */
- err = snd_soc_dapm_add_routes(codec, dapm_routes,
+ err = snd_soc_dapm_add_routes(dapm, dapm_routes,
ARRAY_SIZE(dapm_routes));
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* Add rx1950 specific widgets */
- err = snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+ err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
ARRAY_SIZE(uda1380_dapm_widgets));
if (err)
return err;
/* Set up rx1950 specific audio path audio_mapnects */
- err = snd_soc_dapm_add_routes(codec, audio_map,
+ err = snd_soc_dapm_add_routes(dapm, audio_map,
ARRAY_SIZE(audio_map));
if (err)
return err;
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Speaker");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
&hp_jack);
static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_new_controls(codec, dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, dapm_widgets,
ARRAY_SIZE(dapm_widgets));
- snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map));
+ snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map));
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Line In");
- snd_soc_dapm_enable_pin(codec, "Line Out");
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Line In");
+ snd_soc_dapm_enable_pin(dapm, "Line Out");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
simtec_audio_init(rtd);
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_new_controls(codec, dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, dapm_widgets,
ARRAY_SIZE(dapm_widgets));
- snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map));
+ snd_soc_dapm_add_routes(dapm, base_map, ARRAY_SIZE(base_map));
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Line In");
- snd_soc_dapm_enable_pin(codec, "Line Out");
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Line In");
+ snd_soc_dapm_enable_pin(dapm, "Line Out");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
simtec_audio_init(rtd);
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int smartq_wm8987_init(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err = 0;
/* Add SmartQ specific widgets */
- snd_soc_dapm_new_controls(codec, wm8987_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, wm8987_dapm_widgets,
ARRAY_SIZE(wm8987_dapm_widgets));
/* add SmartQ specific controls */
return err;
/* setup SmartQ specific audio path */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* set endpoints to not connected */
- snd_soc_dapm_nc_pin(codec, "LINPUT1");
- snd_soc_dapm_nc_pin(codec, "RINPUT1");
- snd_soc_dapm_nc_pin(codec, "OUT3");
- snd_soc_dapm_nc_pin(codec, "ROUT1");
+ snd_soc_dapm_nc_pin(dapm, "LINPUT1");
+ snd_soc_dapm_nc_pin(dapm, "RINPUT1");
+ snd_soc_dapm_nc_pin(dapm, "OUT3");
+ snd_soc_dapm_nc_pin(dapm, "ROUT1");
/* set endpoints to default off mode */
- snd_soc_dapm_enable_pin(codec, "Internal Speaker");
- snd_soc_dapm_enable_pin(codec, "Internal Mic");
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Internal Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Internal Mic");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- err = snd_soc_dapm_sync(codec);
+ err = snd_soc_dapm_sync(dapm);
if (err)
return err;
static int smdk64xx_wm8580_init_paiftx(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
/* Add smdk64xx specific Capture widgets */
- snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt,
+ snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets_cpt,
ARRAY_SIZE(wm8580_dapm_widgets_cpt));
/* Set up PAIFTX audio path */
- snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx));
+ snd_soc_dapm_add_routes(dapm, audio_map_tx, ARRAY_SIZE(audio_map_tx));
/* Enabling the microphone requires the fitting of a 0R
* resistor to connect the line from the microphone jack.
*/
- snd_soc_dapm_disable_pin(codec, "MicIn");
+ snd_soc_dapm_disable_pin(dapm, "MicIn");
/* signal a DAPM event */
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
static int smdk64xx_wm8580_init_paifrx(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
/* Add smdk64xx specific Playback widgets */
- snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk,
+ snd_soc_dapm_new_controls(dapm, wm8580_dapm_widgets_pbk,
ARRAY_SIZE(wm8580_dapm_widgets_pbk));
/* Set up PAIFRX audio path */
- snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx));
+ snd_soc_dapm_add_routes(dapm, audio_map_rx, ARRAY_SIZE(audio_map_rx));
/* signal a DAPM event */
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = kcontrol->private_data;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
unsigned int val = (ucontrol->value.enumerated.item[0] != 0);
char *differential = "Audio Out Differential";
char *stereo = "Audio Out Stereo";
if (kcontrol->private_value == val)
return 0;
kcontrol->private_value = val;
- snd_soc_dapm_disable_pin(codec, val ? differential : stereo);
- snd_soc_dapm_sync(codec);
- snd_soc_dapm_enable_pin(codec, val ? stereo : differential);
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_disable_pin(dapm, val ? differential : stereo);
+ snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_enable_pin(dapm, val ? stereo : differential);
+ snd_soc_dapm_sync(dapm);
return 1;
}
static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
/* Add s6105 specific widgets */
- snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
ARRAY_SIZE(aic3x_dapm_widgets));
/* Set up s6105 specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
/* not present */
- snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
- snd_soc_dapm_nc_pin(codec, "LINE2L");
- snd_soc_dapm_nc_pin(codec, "LINE2R");
+ snd_soc_dapm_nc_pin(dapm, "MONO_LOUT");
+ snd_soc_dapm_nc_pin(dapm, "LINE2L");
+ snd_soc_dapm_nc_pin(dapm, "LINE2R");
/* not connected */
- snd_soc_dapm_nc_pin(codec, "MIC3L"); /* LINE2L on this chip */
- snd_soc_dapm_nc_pin(codec, "MIC3R"); /* LINE2R on this chip */
- snd_soc_dapm_nc_pin(codec, "LLOUT");
- snd_soc_dapm_nc_pin(codec, "RLOUT");
- snd_soc_dapm_nc_pin(codec, "HPRCOM");
+ snd_soc_dapm_nc_pin(dapm, "MIC3L"); /* LINE2L on this chip */
+ snd_soc_dapm_nc_pin(dapm, "MIC3R"); /* LINE2R on this chip */
+ snd_soc_dapm_nc_pin(dapm, "LLOUT");
+ snd_soc_dapm_nc_pin(dapm, "RLOUT");
+ snd_soc_dapm_nc_pin(dapm, "HPRCOM");
/* always connected */
- snd_soc_dapm_enable_pin(codec, "Audio In");
+ snd_soc_dapm_enable_pin(dapm, "Audio In");
/* must correspond to audio_out_mux.private_value initializer */
- snd_soc_dapm_disable_pin(codec, "Audio Out Differential");
- snd_soc_dapm_sync(codec);
- snd_soc_dapm_enable_pin(codec, "Audio Out Stereo");
+ snd_soc_dapm_disable_pin(dapm, "Audio Out Differential");
+ snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_enable_pin(dapm, "Audio Out Stereo");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
snd_ctl_add(codec->snd_card, snd_ctl_new1(&audio_out_mux, codec));
static int migor_dai_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_new_controls(codec, migor_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, migor_dapm_widgets,
ARRAY_SIZE(migor_dapm_widgets));
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
static int machine_init(struct snd_soc_pcm_runtime *rtd)
{
- snd_soc_dapm_sync(rtd->codec);
+ snd_soc_dapm_sync(&rtd->codec->dapm);
return 0;
}
if (codec->cache_only)
return -1;
+ BUG_ON(!codec->hw_read);
return codec->hw_read(codec, reg);
}
return 0;
}
- dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
ret = codec->hw_write(codec->control_data, data, 2);
if (ret == 2)
return 0;
if (codec->cache_only)
return -1;
+ BUG_ON(!codec->hw_read);
return codec->hw_read(codec, reg);
}
return 0;
}
- dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
ret = codec->hw_write(codec->control_data, data, 2);
if (ret == 2)
return 0;
return 0;
}
- dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
if (codec->hw_write(codec->control_data, data, 2) == 2)
return 0;
else
if (codec->cache_only)
return -1;
+ BUG_ON(!codec->hw_read);
return codec->hw_read(codec, reg);
}
return 0;
}
- dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
if (codec->hw_write(codec->control_data, data, 3) == 3)
return 0;
else
if (codec->cache_only)
return -1;
+ BUG_ON(!codec->hw_read);
return codec->hw_read(codec, reg);
} else {
return cache[reg];
if (codec->cache_only)
return -1;
+ BUG_ON(!codec->hw_read);
return codec->hw_read(codec, reg);
}
return 0;
}
- dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
ret = codec->hw_write(codec->control_data, data, 3);
if (ret == 3)
return 0;
if (codec->cache_only)
return -1;
+ BUG_ON(!codec->hw_read);
return codec->hw_read(codec, reg);
}
return 0;
}
- dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
-
ret = codec->hw_write(codec->control_data, data, 4);
if (ret == 4)
return 0;
static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
{
- codec->debugfs_codec_root = debugfs_create_dir(codec->name ,
- debugfs_root);
+ struct dentry *debugfs_card_root = codec->card->debugfs_card_root;
+
+ codec->debugfs_codec_root = debugfs_create_dir(codec->name,
+ debugfs_card_root);
if (!codec->debugfs_codec_root) {
printk(KERN_WARNING
"ASoC: Failed to create codec debugfs directory\n");
printk(KERN_WARNING
"ASoC: Failed to create codec register debugfs file\n");
- codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644,
- codec->debugfs_codec_root,
- &codec->pop_time);
- if (!codec->debugfs_pop_time)
- printk(KERN_WARNING
- "Failed to create pop time debugfs file\n");
-
- codec->debugfs_dapm = debugfs_create_dir("dapm",
+ codec->dapm.debugfs_dapm = debugfs_create_dir("dapm",
codec->debugfs_codec_root);
- if (!codec->debugfs_dapm)
+ if (!codec->dapm.debugfs_dapm)
printk(KERN_WARNING
"Failed to create DAPM debugfs directory\n");
- snd_soc_dapm_debugfs_init(codec);
+ snd_soc_dapm_debugfs_init(&codec->dapm);
}
static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
.llseek = default_llseek,/* read accesses f_pos */
};
+static void soc_init_card_debugfs(struct snd_soc_card *card)
+{
+ card->debugfs_card_root = debugfs_create_dir(card->name,
+ debugfs_root);
+ if (!card->debugfs_card_root) {
+ dev_warn(card->dev,
+ "ASoC: Failed to create codec debugfs directory\n");
+ return;
+ }
+
+ card->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644,
+ card->debugfs_card_root,
+ &card->pop_time);
+ if (!card->debugfs_pop_time)
+ dev_warn(card->dev,
+ "Failed to create pop time debugfs file\n");
+}
+
+static void soc_cleanup_card_debugfs(struct snd_soc_card *card)
+{
+ debugfs_remove_recursive(card->debugfs_card_root);
+}
+
#else
static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
/* close any waiting streams and save state */
for (i = 0; i < card->num_rtd; i++) {
run_delayed_work(&card->rtd[i].delayed_work);
- card->rtd[i].codec->suspend_bias_level = card->rtd[i].codec->bias_level;
+ card->rtd[i].codec->dapm.suspend_bias_level = card->rtd[i].codec->dapm.bias_level;
}
for (i = 0; i < card->num_rtd; i++) {
/* If there are paths active then the CODEC will be held with
* bias _ON and should not be suspended. */
if (!codec->suspended && codec->driver->suspend) {
- switch (codec->bias_level) {
+ switch (codec->dapm.bias_level) {
case SND_SOC_BIAS_STANDBY:
case SND_SOC_BIAS_OFF:
codec->driver->suspend(codec, PMSG_SUSPEND);
* resume. Otherwise the suspend was suppressed.
*/
if (codec->driver->resume && codec->suspended) {
- switch (codec->bias_level) {
+ switch (codec->dapm.bias_level) {
case SND_SOC_BIAS_STANDBY:
case SND_SOC_BIAS_OFF:
codec->driver->resume(codec);
}
/* Make sure all DAPM widgets are freed */
- snd_soc_dapm_free(codec);
+ snd_soc_dapm_free(&codec->dapm);
soc_cleanup_codec_debugfs(codec);
device_remove_file(&rtd->dev, &dev_attr_codec_reg);
/* probe the CODEC */
if (!codec->probed) {
+ codec->dapm.card = card;
if (codec->driver->probe) {
ret = codec->driver->probe(codec);
if (ret < 0) {
}
/* Make sure all DAPM widgets are instantiated */
- snd_soc_dapm_new_widgets(codec);
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_new_widgets(&codec->dapm);
+ snd_soc_dapm_sync(&codec->dapm);
/* register the rtd device */
rtd->dev.release = rtd_release;
INIT_LIST_HEAD(&card->codec_dev_list);
INIT_LIST_HEAD(&card->platform_dev_list);
+ soc_init_card_debugfs(card);
+
ret = snd_soc_register_card(card);
if (ret != 0) {
dev_err(&pdev->dev, "Failed to register card\n");
for (i = 0; i < card->num_rtd; i++)
soc_remove_dai_link(card, i);
+ soc_cleanup_card_debugfs(card);
+
/* remove the card */
if (card->remove)
card->remove(pdev);
}
EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec);
+unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+ unsigned int ret;
+
+ ret = codec->driver->read(codec, reg);
+ dev_dbg(codec->dev, "read %x => %x\n", reg, ret);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_read);
+
+unsigned int snd_soc_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int val)
+{
+ dev_dbg(codec->dev, "write %x = %x\n", reg, val);
+ return codec->driver->write(codec, reg, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_write);
+
/**
* snd_soc_update_bits - update codec register bits
* @codec: audio codec
return -ENOMEM;
}
+ INIT_LIST_HEAD(&codec->dapm.widgets);
+ INIT_LIST_HEAD(&codec->dapm.paths);
+ codec->dapm.bias_level = SND_SOC_BIAS_OFF;
+ codec->dapm.dev = dev;
+ codec->dapm.codec = codec;
+
/* allocate CODEC register cache */
if (codec_drv->reg_cache_size && codec_drv->reg_word_size) {
codec->dev = dev;
codec->driver = codec_drv;
- codec->bias_level = SND_SOC_BIAS_OFF;
codec->num_dai = num_dai;
mutex_init(&codec->mutex);
- INIT_LIST_HEAD(&codec->dapm_widgets);
- INIT_LIST_HEAD(&codec->dapm_paths);
for (i = 0; i < num_dai; i++) {
fixup_codec_formats(&dai_drv[i].playback);
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
+#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
* Returns 0 for success else error.
*/
static int snd_soc_dapm_set_bias_level(struct snd_soc_card *card,
- struct snd_soc_codec *codec, enum snd_soc_bias_level level)
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
{
int ret = 0;
switch (level) {
case SND_SOC_BIAS_ON:
- dev_dbg(codec->dev, "Setting full bias\n");
+ dev_dbg(dapm->dev, "Setting full bias\n");
break;
case SND_SOC_BIAS_PREPARE:
- dev_dbg(codec->dev, "Setting bias prepare\n");
+ dev_dbg(dapm->dev, "Setting bias prepare\n");
break;
case SND_SOC_BIAS_STANDBY:
- dev_dbg(codec->dev, "Setting standby bias\n");
+ dev_dbg(dapm->dev, "Setting standby bias\n");
break;
case SND_SOC_BIAS_OFF:
- dev_dbg(codec->dev, "Setting bias off\n");
+ dev_dbg(dapm->dev, "Setting bias off\n");
break;
default:
- dev_err(codec->dev, "Setting invalid bias %d\n", level);
+ dev_err(dapm->dev, "Setting invalid bias %d\n", level);
return -EINVAL;
}
if (card && card->set_bias_level)
ret = card->set_bias_level(card, level);
if (ret == 0) {
- if (codec->driver->set_bias_level)
- ret = codec->driver->set_bias_level(codec, level);
+ if (dapm->codec && dapm->codec->driver->set_bias_level)
+ ret = dapm->codec->driver->set_bias_level(dapm->codec, level);
else
- codec->bias_level = level;
+ dapm->bias_level = level;
}
return ret;
}
/* connect mux widget to its interconnecting audio paths */
-static int dapm_connect_mux(struct snd_soc_codec *codec,
+static int dapm_connect_mux(struct snd_soc_dapm_context *dapm,
struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
struct snd_soc_dapm_path *path, const char *control_name,
const struct snd_kcontrol_new *kcontrol)
for (i = 0; i < e->max; i++) {
if (!(strcmp(control_name, e->texts[i]))) {
- list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list, &dapm->paths);
list_add(&path->list_sink, &dest->sources);
list_add(&path->list_source, &src->sinks);
path->name = (char*)e->texts[i];
}
/* connect mixer widget to its interconnecting audio paths */
-static int dapm_connect_mixer(struct snd_soc_codec *codec,
+static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm,
struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
struct snd_soc_dapm_path *path, const char *control_name)
{
/* search for mixer kcontrol */
for (i = 0; i < dest->num_kcontrols; i++) {
if (!strcmp(control_name, dest->kcontrols[i].name)) {
- list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list, &dapm->paths);
list_add(&path->list_sink, &dest->sources);
list_add(&path->list_source, &src->sinks);
path->name = dest->kcontrols[i].name;
int change, power;
unsigned int old, new;
struct snd_soc_codec *codec = widget->codec;
+ struct snd_soc_dapm_context *dapm = widget->dapm;
+ struct snd_soc_card *card = dapm->card;
/* check for valid widgets */
if (widget->reg < 0 || widget->id == snd_soc_dapm_input ||
change = old != new;
if (change) {
- pop_dbg(codec->pop_time, "pop test %s : %s in %d ms\n",
+ pop_dbg(card->pop_time, "pop test %s : %s in %d ms\n",
widget->name, widget->power ? "on" : "off",
- codec->pop_time);
- pop_wait(codec->pop_time);
+ card->pop_time);
+ pop_wait(card->pop_time);
snd_soc_write(codec, widget->reg, new);
}
pr_debug("reg %x old %x new %x change %d\n", widget->reg,
}
/* create new dapm mixer control */
-static int dapm_new_mixer(struct snd_soc_codec *codec,
+static int dapm_new_mixer(struct snd_soc_dapm_context *dapm,
struct snd_soc_dapm_widget *w)
{
int i, ret = 0;
size_t name_len;
struct snd_soc_dapm_path *path;
+ struct snd_card *card = dapm->codec->card->snd_card;
/* add kcontrol */
for (i = 0; i < w->num_kcontrols; i++) {
path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
path->long_name);
- ret = snd_ctl_add(codec->card->snd_card, path->kcontrol);
+ ret = snd_ctl_add(card, path->kcontrol);
if (ret < 0) {
printk(KERN_ERR "asoc: failed to add dapm kcontrol %s: %d\n",
path->long_name,
}
/* create new dapm mux control */
-static int dapm_new_mux(struct snd_soc_codec *codec,
+static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
struct snd_soc_dapm_widget *w)
{
struct snd_soc_dapm_path *path = NULL;
struct snd_kcontrol *kcontrol;
+ struct snd_card *card = dapm->codec->card->snd_card;
int ret = 0;
if (!w->num_kcontrols) {
}
kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
- ret = snd_ctl_add(codec->card->snd_card, kcontrol);
+ ret = snd_ctl_add(card, kcontrol);
+
if (ret < 0)
goto err;
}
/* create new dapm volume control */
-static int dapm_new_pga(struct snd_soc_codec *codec,
+static int dapm_new_pga(struct snd_soc_dapm_context *dapm,
struct snd_soc_dapm_widget *w)
{
if (w->num_kcontrols)
}
/* reset 'walked' bit for each dapm path */
-static inline void dapm_clear_walk(struct snd_soc_codec *codec)
+static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_dapm_path *p;
- list_for_each_entry(p, &codec->dapm_paths, list)
+ list_for_each_entry(p, &dapm->paths, list)
p->walked = 0;
}
*/
static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget)
{
- int level = snd_power_get_state(widget->codec->card->snd_card);
+ int level = snd_power_get_state(widget->dapm->codec->card->snd_card);
switch (level) {
case SNDRV_CTL_POWER_D3hot:
int in, out;
in = is_connected_input_ep(w);
- dapm_clear_walk(w->codec);
+ dapm_clear_walk(w->dapm);
out = is_connected_output_ep(w);
- dapm_clear_walk(w->codec);
+ dapm_clear_walk(w->dapm);
return out != 0 && in != 0;
}
if (w->active) {
in = is_connected_input_ep(w);
- dapm_clear_walk(w->codec);
+ dapm_clear_walk(w->dapm);
return in != 0;
} else {
return dapm_generic_check_power(w);
if (w->active) {
out = is_connected_output_ep(w);
- dapm_clear_walk(w->codec);
+ dapm_clear_walk(w->dapm);
return out != 0;
} else {
return dapm_generic_check_power(w);
}
}
- dapm_clear_walk(w->codec);
+ dapm_clear_walk(w->dapm);
return power;
}
}
/* Apply the coalesced changes from a DAPM sequence */
-static void dapm_seq_run_coalesced(struct snd_soc_codec *codec,
+static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm,
struct list_head *pending)
{
struct snd_soc_dapm_widget *w;
+ struct snd_soc_card *card = dapm->card;
int reg, power, ret;
unsigned int value = 0;
unsigned int mask = 0;
if (power)
value |= cur_mask;
- pop_dbg(codec->pop_time,
+ pop_dbg(card->pop_time,
"pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n",
w->name, reg, value, mask);
/* power up pre event */
if (w->power && w->event &&
(w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
- pop_dbg(codec->pop_time, "pop test : %s PRE_PMU\n",
+ pop_dbg(card->pop_time, "pop test : %s PRE_PMU\n",
w->name);
ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
if (ret < 0)
/* power down pre event */
if (!w->power && w->event &&
(w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
- pop_dbg(codec->pop_time, "pop test : %s PRE_PMD\n",
+ pop_dbg(card->pop_time, "pop test : %s PRE_PMD\n",
w->name);
ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
if (ret < 0)
}
if (reg >= 0) {
- pop_dbg(codec->pop_time,
+ pop_dbg(card->pop_time,
"pop test : Applying 0x%x/0x%x to %x in %dms\n",
- value, mask, reg, codec->pop_time);
- pop_wait(codec->pop_time);
- snd_soc_update_bits(codec, reg, mask, value);
+ value, mask, reg, card->pop_time);
+ pop_wait(card->pop_time);
+ snd_soc_update_bits(dapm->codec, reg, mask, value);
}
list_for_each_entry(w, pending, power_list) {
/* power up post event */
if (w->power && w->event &&
(w->event_flags & SND_SOC_DAPM_POST_PMU)) {
- pop_dbg(codec->pop_time, "pop test : %s POST_PMU\n",
+ pop_dbg(card->pop_time, "pop test : %s POST_PMU\n",
w->name);
ret = w->event(w,
NULL, SND_SOC_DAPM_POST_PMU);
/* power down post event */
if (!w->power && w->event &&
(w->event_flags & SND_SOC_DAPM_POST_PMD)) {
- pop_dbg(codec->pop_time, "pop test : %s POST_PMD\n",
+ pop_dbg(card->pop_time, "pop test : %s POST_PMD\n",
w->name);
ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
if (ret < 0)
* Currently anything that requires more than a single write is not
* handled.
*/
-static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list,
- int event, int sort[])
+static void dapm_seq_run(struct snd_soc_dapm_context *dapm,
+ struct list_head *list, int event, int sort[])
{
struct snd_soc_dapm_widget *w, *n;
LIST_HEAD(pending);
/* Do we need to apply any queued changes? */
if (sort[w->id] != cur_sort || w->reg != cur_reg) {
if (!list_empty(&pending))
- dapm_seq_run_coalesced(codec, &pending);
+ dapm_seq_run_coalesced(dapm, &pending);
INIT_LIST_HEAD(&pending);
cur_sort = -1;
}
if (!list_empty(&pending))
- dapm_seq_run_coalesced(codec, &pending);
+ dapm_seq_run_coalesced(dapm, &pending);
}
/*
* o Input pin to Output pin (bypass, sidetone)
* o DAC to ADC (loopback).
*/
-static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
+static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
{
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_card *card = dapm->codec->card;
struct snd_soc_dapm_widget *w;
LIST_HEAD(up_list);
LIST_HEAD(down_list);
/* Check which widgets we need to power and store them in
* lists indicating if they should be powered up or down.
*/
- list_for_each_entry(w, &codec->dapm_widgets, list) {
+ list_for_each_entry(w, &dapm->widgets, list) {
switch (w->id) {
case snd_soc_dapm_pre:
dapm_seq_insert(w, &down_list, dapm_down_seq);
/* If there are no DAPM widgets then try to figure out power from the
* event type.
*/
- if (list_empty(&codec->dapm_widgets)) {
+ if (list_empty(&dapm->widgets)) {
switch (event) {
case SND_SOC_DAPM_STREAM_START:
case SND_SOC_DAPM_STREAM_RESUME:
sys_power = 0;
break;
case SND_SOC_DAPM_STREAM_NOP:
- switch (codec->bias_level) {
+ switch (dapm->bias_level) {
case SND_SOC_BIAS_STANDBY:
case SND_SOC_BIAS_OFF:
sys_power = 0;
}
}
- if (sys_power && codec->bias_level == SND_SOC_BIAS_OFF) {
- ret = snd_soc_dapm_set_bias_level(card, codec,
+ if (sys_power && dapm->bias_level == SND_SOC_BIAS_OFF) {
+ ret = snd_soc_dapm_set_bias_level(card, dapm,
SND_SOC_BIAS_STANDBY);
if (ret != 0)
pr_err("Failed to turn on bias: %d\n", ret);
}
/* If we're changing to all on or all off then prepare */
- if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) ||
- (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) {
- ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_PREPARE);
+ if ((sys_power && dapm->bias_level == SND_SOC_BIAS_STANDBY) ||
+ (!sys_power && dapm->bias_level == SND_SOC_BIAS_ON)) {
+ ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_PREPARE);
if (ret != 0)
pr_err("Failed to prepare bias: %d\n", ret);
}
/* Power down widgets first; try to avoid amplifying pops. */
- dapm_seq_run(codec, &down_list, event, dapm_down_seq);
+ dapm_seq_run(dapm, &down_list, event, dapm_down_seq);
/* Now power up. */
- dapm_seq_run(codec, &up_list, event, dapm_up_seq);
+ dapm_seq_run(dapm, &up_list, event, dapm_up_seq);
/* If we just powered the last thing off drop to standby bias */
- if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) {
- ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_STANDBY);
+ if (dapm->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) {
+ ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
if (ret != 0)
pr_err("Failed to apply standby bias: %d\n", ret);
}
/* If we're in standby and can support bias off then do that */
- if (codec->bias_level == SND_SOC_BIAS_STANDBY &&
- codec->idle_bias_off) {
- ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_OFF);
+ if (dapm->bias_level == SND_SOC_BIAS_STANDBY &&
+ dapm->idle_bias_off) {
+ ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_OFF);
if (ret != 0)
pr_err("Failed to turn off bias: %d\n", ret);
}
/* If we just powered up then move to active bias */
- if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) {
- ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_ON);
+ if (dapm->bias_level == SND_SOC_BIAS_PREPARE && sys_power) {
+ ret = snd_soc_dapm_set_bias_level(card, dapm, SND_SOC_BIAS_ON);
if (ret != 0)
pr_err("Failed to apply active bias: %d\n", ret);
}
- pop_dbg(codec->pop_time, "DAPM sequencing finished, waiting %dms\n",
- codec->pop_time);
- pop_wait(codec->pop_time);
+ pop_dbg(card->pop_time, "DAPM sequencing finished, waiting %dms\n",
+ card->pop_time);
+ pop_wait(card->pop_time);
return 0;
}
return -ENOMEM;
in = is_connected_input_ep(w);
- dapm_clear_walk(w->codec);
+ dapm_clear_walk(w->dapm);
out = is_connected_output_ep(w);
- dapm_clear_walk(w->codec);
+ dapm_clear_walk(w->dapm);
ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d",
w->name, w->power ? "On" : "Off", in, out);
.llseek = default_llseek,
};
-void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec)
+void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_dapm_widget *w;
struct dentry *d;
- if (!codec->debugfs_dapm)
+ if (!dapm->debugfs_dapm)
return;
- list_for_each_entry(w, &codec->dapm_widgets, list) {
+ list_for_each_entry(w, &dapm->widgets, list) {
if (!w->name)
continue;
d = debugfs_create_file(w->name, 0444,
- codec->debugfs_dapm, w,
+ dapm->debugfs_dapm, w,
&dapm_widget_power_fops);
if (!d)
printk(KERN_WARNING
}
}
#else
-void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec)
+void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm)
{
}
#endif
return 0;
/* find dapm widget path assoc with kcontrol */
- list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+ list_for_each_entry(path, &widget->dapm->paths, list) {
if (path->kcontrol != kcontrol)
continue;
}
if (found)
- dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+ dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP);
return 0;
}
return -ENODEV;
/* find dapm widget path assoc with kcontrol */
- list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+ list_for_each_entry(path, &widget->dapm->paths, list) {
if (path->kcontrol != kcontrol)
continue;
}
if (found)
- dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+ dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP);
return 0;
}
int count = 0;
char *state = "not set";
- list_for_each_entry(w, &codec->dapm_widgets, list) {
+ list_for_each_entry(w, &codec->dapm.widgets, list) {
/* only display widgets that burnm power */
switch (w->id) {
}
}
- switch (codec->bias_level) {
+ switch (codec->dapm.bias_level) {
case SND_SOC_BIAS_ON:
state = "On";
break;
}
/* free all dapm widgets and resources */
-static void dapm_free_widgets(struct snd_soc_codec *codec)
+static void dapm_free_widgets(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_dapm_widget *w, *next_w;
struct snd_soc_dapm_path *p, *next_p;
- list_for_each_entry_safe(w, next_w, &codec->dapm_widgets, list) {
+ list_for_each_entry_safe(w, next_w, &dapm->widgets, list) {
list_del(&w->list);
kfree(w);
}
- list_for_each_entry_safe(p, next_p, &codec->dapm_paths, list) {
+ list_for_each_entry_safe(p, next_p, &dapm->paths, list) {
list_del(&p->list);
kfree(p->long_name);
kfree(p);
}
}
-static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec,
+static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm,
const char *pin, int status)
{
struct snd_soc_dapm_widget *w;
- list_for_each_entry(w, &codec->dapm_widgets, list) {
+ list_for_each_entry(w, &dapm->widgets, list) {
if (!strcmp(w->name, pin)) {
- pr_debug("dapm: %s: pin %s\n", codec->name, pin);
+ pr_debug("dapm: %s: pin %s\n", dapm->codec->name, pin);
w->connected = status;
/* Allow disabling of forced pins */
if (status == 0)
}
}
- pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin);
+ pr_err("dapm: %s: configuring unknown pin %s\n",
+ dapm->codec->name, pin);
return -EINVAL;
}
/**
* snd_soc_dapm_sync - scan and power dapm paths
- * @codec: audio codec
+ * @dapm: DAPM context
*
* Walks all dapm audio paths and powers widgets according to their
* stream or path usage.
*
* Returns 0 for success.
*/
-int snd_soc_dapm_sync(struct snd_soc_codec *codec)
+int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm)
{
- return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
+ return dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP);
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_sync);
-static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
+static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route)
{
struct snd_soc_dapm_path *path;
int ret = 0;
/* find src and dest widgets */
- list_for_each_entry(w, &codec->dapm_widgets, list) {
+ list_for_each_entry(w, &dapm->widgets, list) {
if (!wsink && !(strcmp(w->name, sink))) {
wsink = w;
/* connect static paths */
if (control == NULL) {
- list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list, &dapm->paths);
list_add(&path->list_sink, &wsink->sources);
list_add(&path->list_source, &wsource->sinks);
path->connect = 1;
case snd_soc_dapm_supply:
case snd_soc_dapm_aif_in:
case snd_soc_dapm_aif_out:
- list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list, &dapm->paths);
list_add(&path->list_sink, &wsink->sources);
list_add(&path->list_source, &wsource->sinks);
path->connect = 1;
return 0;
case snd_soc_dapm_mux:
case snd_soc_dapm_value_mux:
- ret = dapm_connect_mux(codec, wsource, wsink, path, control,
+ ret = dapm_connect_mux(dapm, wsource, wsink, path, control,
&wsink->kcontrols[0]);
if (ret != 0)
goto err;
case snd_soc_dapm_switch:
case snd_soc_dapm_mixer:
case snd_soc_dapm_mixer_named_ctl:
- ret = dapm_connect_mixer(codec, wsource, wsink, path, control);
+ ret = dapm_connect_mixer(dapm, wsource, wsink, path, control);
if (ret != 0)
goto err;
break;
case snd_soc_dapm_mic:
case snd_soc_dapm_line:
case snd_soc_dapm_spk:
- list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list, &dapm->paths);
list_add(&path->list_sink, &wsink->sources);
list_add(&path->list_source, &wsource->sinks);
path->connect = 0;
/**
* snd_soc_dapm_add_routes - Add routes between DAPM widgets
- * @codec: codec
+ * @dapm: DAPM context
* @route: audio routes
* @num: number of routes
*
* Returns 0 for success else error. On error all resources can be freed
* with a call to snd_soc_card_free().
*/
-int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
+int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num)
{
int i, ret;
for (i = 0; i < num; i++) {
- ret = snd_soc_dapm_add_route(codec, route);
+ ret = snd_soc_dapm_add_route(dapm, route);
if (ret < 0) {
printk(KERN_ERR "Failed to add route %s->%s\n",
route->source,
/**
* snd_soc_dapm_new_widgets - add new dapm widgets
- * @codec: audio codec
+ * @dapm: DAPM context
*
* Checks the codec for any new dapm widgets and creates them if found.
*
* Returns 0 for success.
*/
-int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
+int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_dapm_widget *w;
- list_for_each_entry(w, &codec->dapm_widgets, list)
+ list_for_each_entry(w, &dapm->widgets, list)
{
if (w->new)
continue;
case snd_soc_dapm_mixer:
case snd_soc_dapm_mixer_named_ctl:
w->power_check = dapm_generic_check_power;
- dapm_new_mixer(codec, w);
+ dapm_new_mixer(dapm, w);
break;
case snd_soc_dapm_mux:
case snd_soc_dapm_value_mux:
w->power_check = dapm_generic_check_power;
- dapm_new_mux(codec, w);
+ dapm_new_mux(dapm, w);
break;
case snd_soc_dapm_adc:
case snd_soc_dapm_aif_out:
break;
case snd_soc_dapm_pga:
w->power_check = dapm_generic_check_power;
- dapm_new_pga(codec, w);
+ dapm_new_pga(dapm, w);
break;
case snd_soc_dapm_input:
case snd_soc_dapm_output:
w->new = 1;
}
- dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
+ dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets);
mutex_lock(&codec->mutex);
ucontrol->value.integer.value[0] =
- snd_soc_dapm_get_pin_status(codec, pin);
+ snd_soc_dapm_get_pin_status(&codec->dapm, pin);
mutex_unlock(&codec->mutex);
mutex_lock(&codec->mutex);
if (ucontrol->value.integer.value[0])
- snd_soc_dapm_enable_pin(codec, pin);
+ snd_soc_dapm_enable_pin(&codec->dapm, pin);
else
- snd_soc_dapm_disable_pin(codec, pin);
+ snd_soc_dapm_disable_pin(&codec->dapm, pin);
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(&codec->dapm);
mutex_unlock(&codec->mutex);
/**
* snd_soc_dapm_new_control - create new dapm control
- * @codec: audio codec
+ * @dapm: DAPM context
* @widget: widget template
*
* Creates a new dapm control based upon the template.
*
* Returns 0 for success else error.
*/
-int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
+int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_widget *widget)
{
struct snd_soc_dapm_widget *w;
if ((w = dapm_cnew_widget(widget)) == NULL)
return -ENOMEM;
- w->codec = codec;
+ w->dapm = dapm;
+ w->codec = dapm->codec;
INIT_LIST_HEAD(&w->sources);
INIT_LIST_HEAD(&w->sinks);
INIT_LIST_HEAD(&w->list);
- list_add(&w->list, &codec->dapm_widgets);
+ list_add(&w->list, &dapm->widgets);
/* machine layer set ups unconnected pins and insertions */
w->connected = 1;
/**
* snd_soc_dapm_new_controls - create new dapm controls
- * @codec: audio codec
+ * @dapm: DAPM context
* @widget: widget array
* @num: number of widgets
*
*
* Returns 0 for success else error.
*/
-int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
+int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_widget *widget,
int num)
{
int i, ret;
for (i = 0; i < num; i++) {
- ret = snd_soc_dapm_new_control(codec, widget);
+ ret = snd_soc_dapm_new_control(dapm, widget);
if (ret < 0) {
printk(KERN_ERR
"ASoC: Failed to create DAPM control %s: %d\n",
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
-
-/**
- * snd_soc_dapm_stream_event - send a stream event to the dapm core
- * @codec: audio codec
- * @stream: stream name
- * @event: stream event
- *
- * Sends a stream event to the dapm core. The core then makes any
- * necessary widget power changes.
- *
- * Returns 0 for success else error.
- */
-int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
+static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm,
const char *stream, int event)
{
- struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_widget *w;
- if (stream == NULL)
- return 0;
-
- mutex_lock(&codec->mutex);
- list_for_each_entry(w, &codec->dapm_widgets, list)
+ list_for_each_entry(w, &dapm->widgets, list)
{
if (!w->sname)
continue;
}
}
- dapm_power_widgets(codec, event);
+ dapm_power_widgets(dapm, event);
+}
+
+/**
+ * snd_soc_dapm_stream_event - send a stream event to the dapm core
+ * @rtd: PCM runtime data
+ * @stream: stream name
+ * @event: stream event
+ *
+ * Sends a stream event to the dapm core. The core then makes any
+ * necessary widget power changes.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
+ const char *stream, int event)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+
+ if (stream == NULL)
+ return 0;
+
+ mutex_lock(&codec->mutex);
+ soc_dapm_stream_event(&codec->dapm, stream, event);
mutex_unlock(&codec->mutex);
return 0;
}
/**
* snd_soc_dapm_enable_pin - enable pin.
- * @codec: SoC codec
+ * @dapm: DAPM context
* @pin: pin name
*
* Enables input/output pin and its parents or children widgets iff there is
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
*/
-int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin)
+int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm, const char *pin)
{
- return snd_soc_dapm_set_pin(codec, pin, 1);
+ return snd_soc_dapm_set_pin(dapm, pin, 1);
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
/**
* snd_soc_dapm_force_enable_pin - force a pin to be enabled
- * @codec: SoC codec
+ * @dapm: DAPM context
* @pin: pin name
*
* Enables input/output pin regardless of any other state. This is
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
*/
-int snd_soc_dapm_force_enable_pin(struct snd_soc_codec *codec, const char *pin)
+int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
+ const char *pin)
{
struct snd_soc_dapm_widget *w;
- list_for_each_entry(w, &codec->dapm_widgets, list) {
+ list_for_each_entry(w, &dapm->widgets, list) {
if (!strcmp(w->name, pin)) {
- pr_debug("dapm: %s: pin %s\n", codec->name, pin);
+ pr_debug("dapm: %s: pin %s\n", dapm->codec->name, pin);
w->connected = 1;
w->force = 1;
return 0;
}
}
- pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin);
+ pr_err("dapm: %s: configuring unknown pin %s\n",
+ dapm->codec->name, pin);
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin);
/**
* snd_soc_dapm_disable_pin - disable pin.
- * @codec: SoC codec
+ * @dapm: DAPM context
* @pin: pin name
*
* Disables input/output pin and its parents or children widgets.
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
*/
-int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin)
+int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm,
+ const char *pin)
{
- return snd_soc_dapm_set_pin(codec, pin, 0);
+ return snd_soc_dapm_set_pin(dapm, pin, 0);
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
/**
* snd_soc_dapm_nc_pin - permanently disable pin.
- * @codec: SoC codec
+ * @dapm: DAPM context
* @pin: pin name
*
* Marks the specified pin as being not connected, disabling it along
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
*/
-int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin)
+int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin)
{
- return snd_soc_dapm_set_pin(codec, pin, 0);
+ return snd_soc_dapm_set_pin(dapm, pin, 0);
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
/**
* snd_soc_dapm_get_pin_status - get audio pin status
- * @codec: audio codec
+ * @dapm: DAPM context
* @pin: audio signal pin endpoint (or start point)
*
* Get audio pin status - connected or disconnected.
*
* Returns 1 for connected otherwise 0.
*/
-int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin)
+int snd_soc_dapm_get_pin_status(struct snd_soc_dapm_context *dapm,
+ const char *pin)
{
struct snd_soc_dapm_widget *w;
- list_for_each_entry(w, &codec->dapm_widgets, list) {
+ list_for_each_entry(w, &dapm->widgets, list) {
if (!strcmp(w->name, pin))
return w->connected;
}
/**
* snd_soc_dapm_ignore_suspend - ignore suspend status for DAPM endpoint
- * @codec: audio codec
+ * @dapm: DAPM context
* @pin: audio signal pin endpoint (or start point)
*
* Mark the given endpoint or pin as ignoring suspend. When the
* normal means at suspend time, it will not be turned on if it was not
* already enabled.
*/
-int snd_soc_dapm_ignore_suspend(struct snd_soc_codec *codec, const char *pin)
+int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
+ const char *pin)
{
struct snd_soc_dapm_widget *w;
- list_for_each_entry(w, &codec->dapm_widgets, list) {
+ list_for_each_entry(w, &dapm->widgets, list) {
if (!strcmp(w->name, pin)) {
w->ignore_suspend = 1;
return 0;
*
* Free all dapm widgets and resources.
*/
-void snd_soc_dapm_free(struct snd_soc_codec *codec)
+void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm)
{
- snd_soc_dapm_sys_remove(codec->dev);
- dapm_free_widgets(codec);
+ snd_soc_dapm_sys_remove(dapm->dev);
+ dapm_free_widgets(dapm);
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
-static void soc_dapm_shutdown_codec(struct snd_soc_codec *codec)
+static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_dapm_widget *w;
LIST_HEAD(down_list);
int powerdown = 0;
- list_for_each_entry(w, &codec->dapm_widgets, list) {
+ list_for_each_entry(w, &dapm->widgets, list) {
if (w->power) {
dapm_seq_insert(w, &down_list, dapm_down_seq);
w->power = 0;
* standby.
*/
if (powerdown) {
- snd_soc_dapm_set_bias_level(NULL, codec, SND_SOC_BIAS_PREPARE);
- dapm_seq_run(codec, &down_list, 0, dapm_down_seq);
- snd_soc_dapm_set_bias_level(NULL, codec, SND_SOC_BIAS_STANDBY);
+ snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_PREPARE);
+ dapm_seq_run(dapm, &down_list, 0, dapm_down_seq);
+ snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_STANDBY);
}
}
{
struct snd_soc_codec *codec;
- list_for_each_entry(codec, &card->codec_dev_list, list)
- soc_dapm_shutdown_codec(codec);
-
- snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_OFF);
+ list_for_each_entry(codec, &card->codec_dev_list, list) {
+ soc_dapm_shutdown_codec(&codec->dapm);
+ snd_soc_dapm_set_bias_level(card, &codec->dapm, SND_SOC_BIAS_OFF);
+ }
}
/* Module information */
void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
{
struct snd_soc_codec *codec;
+ struct snd_soc_dapm_context *dapm;
struct snd_soc_jack_pin *pin;
int enable;
int oldstatus;
return;
codec = jack->codec;
+ dapm = &codec->dapm;
mutex_lock(&codec->mutex);
enable = !enable;
if (enable)
- snd_soc_dapm_enable_pin(codec, pin->pin);
+ snd_soc_dapm_enable_pin(dapm, pin->pin);
else
- snd_soc_dapm_disable_pin(codec, pin->pin);
+ snd_soc_dapm_disable_pin(dapm, pin->pin);
}
/* Report before the DAPM sync to help users updating micbias status */
blocking_notifier_call_chain(&jack->notifier, status, NULL);
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
snd_jack_report(jack->jack, status);
INIT_DELAYED_WORK(&gpios[i].work, gpio_work);
gpios[i].jack = jack;
- ret = request_irq(gpio_to_irq(gpios[i].gpio),
- gpio_handler,
- IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
- jack->codec->dev->driver->name,
- &gpios[i]);
+ ret = request_any_context_irq(gpio_to_irq(gpios[i].gpio),
+ gpio_handler,
+ IRQF_TRIGGER_RISING |
+ IRQF_TRIGGER_FALLING,
+ jack->codec->dev->driver->name,
+ &gpios[i]);
if (ret)
goto err;
projects / deliverable / linux.git / commitdiff
