summaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
-rw-r--r--include/sound/soc-dapm.h49
-rw-r--r--include/sound/soc.h9
-rw-r--r--sound/soc/atmel/playpaq_wm8510.c11
-rw-r--r--sound/soc/atmel/sam9g20_wm8731.c17
-rw-r--r--sound/soc/atmel/snd-soc-afeb9260.c13
-rw-r--r--sound/soc/codecs/88pm860x-codec.c9
-rw-r--r--sound/soc/codecs/ad1836.c5
-rw-r--r--sound/soc/codecs/ad193x.c5
-rw-r--r--sound/soc/codecs/ak4535.c9
-rw-r--r--sound/soc/codecs/ak4642.c2
-rw-r--r--sound/soc/codecs/ak4671.c9
-rw-r--r--sound/soc/codecs/alc5623.c23
-rw-r--r--sound/soc/codecs/cq93vc.c2
-rw-r--r--sound/soc/codecs/cs42l51.c5
-rw-r--r--sound/soc/codecs/cx20442.c15
-rw-r--r--sound/soc/codecs/da7210.c2
-rw-r--r--sound/soc/codecs/jz4740.c10
-rw-r--r--sound/soc/codecs/max98088.c12
-rw-r--r--sound/soc/codecs/ssm2602.c9
-rw-r--r--sound/soc/codecs/stac9766.c3
-rw-r--r--sound/soc/codecs/tlv320aic23.c9
-rw-r--r--sound/soc/codecs/tlv320aic3x.c22
-rw-r--r--sound/soc/codecs/tlv320dac33.c15
-rw-r--r--sound/soc/codecs/tpa6130a2.c5
-rw-r--r--sound/soc/codecs/twl4030.c13
-rw-r--r--sound/soc/codecs/twl6040.c12
-rw-r--r--sound/soc/codecs/uda134x.c2
-rw-r--r--sound/soc/codecs/uda1380.c13
-rw-r--r--sound/soc/codecs/wm2000.c5
-rw-r--r--sound/soc/codecs/wm8350.c28
-rw-r--r--sound/soc/codecs/wm8400.c11
-rw-r--r--sound/soc/codecs/wm8510.c11
-rw-r--r--sound/soc/codecs/wm8523.c11
-rw-r--r--sound/soc/codecs/wm8580.c11
-rw-r--r--sound/soc/codecs/wm8711.c9
-rw-r--r--sound/soc/codecs/wm8728.c11
-rw-r--r--sound/soc/codecs/wm8731.c13
-rw-r--r--sound/soc/codecs/wm8741.c7
-rw-r--r--sound/soc/codecs/wm8750.c11
-rw-r--r--sound/soc/codecs/wm8753.c29
-rw-r--r--sound/soc/codecs/wm8776.c9
-rw-r--r--sound/soc/codecs/wm8804.c6
-rw-r--r--sound/soc/codecs/wm8900.c11
-rw-r--r--sound/soc/codecs/wm8903.c11
-rw-r--r--sound/soc/codecs/wm8904.c33
-rw-r--r--sound/soc/codecs/wm8940.c5
-rw-r--r--sound/soc/codecs/wm8955.c11
-rw-r--r--sound/soc/codecs/wm8960.c25
-rw-r--r--sound/soc/codecs/wm8961.c11
-rw-r--r--sound/soc/codecs/wm8962.c30
-rw-r--r--sound/soc/codecs/wm8971.c29
-rw-r--r--sound/soc/codecs/wm8974.c11
-rw-r--r--sound/soc/codecs/wm8978.c11
-rw-r--r--sound/soc/codecs/wm8985.c11
-rw-r--r--sound/soc/codecs/wm8988.c9
-rw-r--r--sound/soc/codecs/wm8990.c11
-rw-r--r--sound/soc/codecs/wm8993.c9
-rw-r--r--sound/soc/codecs/wm8994.c13
-rw-r--r--sound/soc/codecs/wm9081.c9
-rw-r--r--sound/soc/codecs/wm9090.c17
-rw-r--r--sound/soc/codecs/wm9705.c6
-rw-r--r--sound/soc/codecs/wm9712.c9
-rw-r--r--sound/soc/codecs/wm9713.c8
-rw-r--r--sound/soc/codecs/wm_hubs.c18
-rw-r--r--sound/soc/davinci/davinci-evm.c21
-rw-r--r--sound/soc/ep93xx/snappercl15.c5
-rw-r--r--sound/soc/imx/wm1133-ev1.c7
-rw-r--r--sound/soc/jz4740/qi_lb60.c13
-rw-r--r--sound/soc/kirkwood/kirkwood-t5325.c13
-rw-r--r--sound/soc/omap/am3517evm.c13
-rw-r--r--sound/soc/omap/ams-delta.c82
-rw-r--r--sound/soc/omap/n810.c42
-rw-r--r--sound/soc/omap/omap3pandora.c44
-rw-r--r--sound/soc/omap/osk5912.c13
-rw-r--r--sound/soc/omap/rx51.c25
-rw-r--r--sound/soc/omap/sdp3430.c43
-rw-r--r--sound/soc/omap/sdp4430.c19
-rw-r--r--sound/soc/omap/zoom2.c35
-rw-r--r--sound/soc/pxa/corgi.c51
-rw-r--r--sound/soc/pxa/e740_wm9705.c29
-rw-r--r--sound/soc/pxa/e750_wm9705.c29
-rw-r--r--sound/soc/pxa/e800_wm9712.c7
-rw-r--r--sound/soc/pxa/magician.c35
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c17
-rw-r--r--sound/soc/pxa/palm27x.c33
-rw-r--r--sound/soc/pxa/poodle.c25
-rw-r--r--sound/soc/pxa/saarb.c17
-rw-r--r--sound/soc/pxa/spitz.c69
-rw-r--r--sound/soc/pxa/tavorevb3.c17
-rw-r--r--sound/soc/pxa/tosa.c37
-rw-r--r--sound/soc/pxa/z2.c15
-rw-r--r--sound/soc/pxa/zylonite.c11
-rw-r--r--sound/soc/s3c24xx/aquila_wm8994.c25
-rw-r--r--sound/soc/s3c24xx/goni_wm8994.c21
-rw-r--r--sound/soc/s3c24xx/jive_wm8750.c19
-rw-r--r--sound/soc/s3c24xx/neo1973_gta02_wm8753.c41
-rw-r--r--sound/soc/s3c24xx/neo1973_wm8753.c123
-rw-r--r--sound/soc/s3c24xx/rx1950_uda1380.c11
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec_hermes.c15
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c15
-rw-r--r--sound/soc/s3c24xx/smartq_wm8987.c21
-rw-r--r--sound/soc/s3c24xx/smdk64xx_wm8580.c16
-rw-r--r--sound/soc/s6000/s6105-ipcam.c40
-rw-r--r--sound/soc/sh/migor.c5
-rw-r--r--sound/soc/sh/sh7760-ac97.c2
-rw-r--r--sound/soc/soc-core.c29
-rw-r--r--sound/soc/soc-dapm.c355
-rw-r--r--sound/soc/soc-jack.c8
108 files changed, 1239 insertions, 1064 deletions
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 8fd3b41b763f..5881876e8f5b 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -297,6 +297,7 @@ enum snd_soc_dapm_type;
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);
@@ -324,16 +325,16 @@ int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol,
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 */
@@ -343,17 +344,21 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card);
/* 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 {
@@ -425,6 +430,7 @@ struct snd_soc_dapm_widget {
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 */
@@ -461,4 +467,21 @@ struct snd_soc_dapm_widget {
struct list_head power_list;
};
+/* DAPM context */
+struct snd_soc_dapm_context {
+ u32 pop_time;
+ 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 */
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_dapm;
+#endif
+};
+
#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index aaf34d7cd95e..b048e08e2cc7 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -238,6 +238,7 @@ struct soc_enum;
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;
@@ -436,7 +437,6 @@ struct snd_soc_codec {
/* 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 */
@@ -452,12 +452,7 @@ struct snd_soc_codec {
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;
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 5f4e59f4461c..aede7e74ec34 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -318,27 +318,28 @@ static const struct snd_soc_dapm_route intercon[] = {
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);
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 293569dfd0ed..da9c3037496f 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -140,6 +140,7 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
{
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
@@ -154,25 +155,25 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
}
/* 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;
}
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index e3d283561c19..92c709ed0965 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -105,19 +105,20 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 01d19e9f53f9..a15a3e974f0d 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -1172,7 +1172,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec,
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;
@@ -1185,7 +1185,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec,
pm860x_set_bits(codec->control_data, REG_MISC2, data, 0);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -1346,6 +1346,7 @@ EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect);
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;
@@ -1374,9 +1375,9 @@ static int pm860x_probe(struct snd_soc_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:
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index d272534c8f84..c71b05ddd752 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -220,6 +220,7 @@ static struct snd_soc_dai_driver ad1836_dai = {
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;
@@ -252,9 +253,9 @@ static int ad1836_probe(struct snd_soc_codec *codec)
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;
}
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index fa2834c91b9f..dc105d8aaa0f 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -353,6 +353,7 @@ static struct snd_soc_dai_driver ad193x_dai = {
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;
@@ -385,9 +386,9 @@ static int ad193x_probe(struct snd_soc_codec *codec)
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;
}
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index cd88c8f32a38..52abb93a7dce 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -290,10 +290,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -399,7 +400,7 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec,
ak4535_write(codec, AK4535_PM1, i & (~0x80));
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 90c90b7f4a2e..f00eba313dfd 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -26,7 +26,7 @@
#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>
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 24f5f49bb9d2..1d6573c38af4 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -437,10 +437,11 @@ static const struct snd_soc_dapm_route intercon[] = {
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;
}
@@ -602,7 +603,7 @@ static int ak4671_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index fac61744f8c7..5a45067b43ba 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -832,7 +832,7 @@ static int alc5623_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -888,10 +888,10 @@ static int alc5623_resume(struct snd_soc_codec *codec)
alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* charge alc5623 caps */
- if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+ if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- codec->bias_level = SND_SOC_BIAS_ON;
- alc5623_set_bias_level(codec, codec->bias_level);
+ codec->dapm.bias_level = SND_SOC_BIAS_ON;
+ alc5623_set_bias_level(codec, codec->dapm.bias_level);
}
return 0;
@@ -900,6 +900,7 @@ static int alc5623_resume(struct snd_soc_codec *codec)
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);
@@ -943,24 +944,24 @@ static int alc5623_probe(struct snd_soc_codec *codec)
snd_soc_add_controls(codec, alc5623_snd_controls,
ARRAY_SIZE(alc5623_snd_controls));
- snd_soc_dapm_new_controls(codec, alc5623_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
ARRAY_SIZE(alc5623_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));
switch (alc5623->id) {
default:
case 0x21:
case 0x22:
- snd_soc_dapm_new_controls(codec, alc5623_dapm_amp_widgets,
+ snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
ARRAY_SIZE(alc5623_dapm_amp_widgets));
- snd_soc_dapm_add_routes(codec, intercon_amp_spk,
- ARRAY_SIZE(intercon_amp_spk));
+ snd_soc_dapm_add_routes(dapm, intercon_amp_spk,
+ ARRAY_SIZE(intercon_amp_spk));
break;
case 0x23:
- snd_soc_dapm_add_routes(codec, intercon_spk,
- ARRAY_SIZE(intercon_spk));
+ snd_soc_dapm_add_routes(dapm, intercon_spk,
+ ARRAY_SIZE(intercon_spk));
break;
}
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 823643932dde..98b9e5294cbe 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -116,7 +116,7 @@ static int cq93vc_set_bias_level(struct snd_soc_codec *codec,
DAVINCI_VC_REG12_POWER_ALL_OFF);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index cb086eaf4e07..a7fdca36b490 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -519,6 +519,7 @@ static struct snd_soc_dai_driver cs42l51_dai = {
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;
@@ -550,9 +551,9 @@ static int cs42l51_probe(struct snd_soc_codec *codec)
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;
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index e8d27c8f9ba3..11beb1a77c4e 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -18,7 +18,7 @@
#include <sound/core.h>
#include <sound/initval.h>
-#include <sound/soc-dapm.h>
+#include <sound/soc.h>
#include "cx20442.h"
@@ -89,10 +89,11 @@ static const struct snd_soc_dapm_route cx20442_audio_map[] = {
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;
@@ -263,7 +264,7 @@ static void v253_close(struct tty_struct *tty)
/* Prevent the codec driver from further accessing the modem */
codec->hw_write = NULL;
cx20442->control_data = NULL;
- codec->pop_time = 0;
+ codec->dapm.pop_time = 0;
}
/* Line discipline .hangup() */
@@ -291,7 +292,7 @@ static void v253_receive(struct tty_struct *tty,
/* 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->dapm.pop_time = 1;
}
}
@@ -348,7 +349,7 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec)
cx20442->control_data = NULL;
codec->hw_write = NULL;
- codec->pop_time = 0;
+ codec->dapm.pop_time = 0;
return 0;
}
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 58bb9b994811..92fd9d7a9221 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -21,7 +21,7 @@
#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>
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index 16253ec9b022..8a45562a96d4 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -266,7 +266,7 @@ static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec,
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 |
@@ -288,23 +288,25 @@ static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec,
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);
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index bc22ee93a75d..ef06007d8895 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1224,15 +1224,17 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -1617,7 +1619,7 @@ static int max98088_set_bias_level(struct snd_soc_codec *codec,
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,
@@ -1630,7 +1632,7 @@ static int max98088_set_bias_level(struct snd_soc_codec *codec,
codec->cache_sync = 1;
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 6f38d619bf8a..adbc3e8dafc8 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -207,10 +207,11 @@ static const struct snd_soc_dapm_route audio_conn[] = {
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;
}
@@ -493,7 +494,7 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 00d67cc8e206..8aad3a2c4f3d 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -24,6 +24,7 @@
#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"
@@ -236,7 +237,7 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec,
stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index e8652b1ae326..d9d8e844d63f 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -391,11 +391,12 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
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;
}
@@ -574,7 +575,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index fc687790188b..6173c2b4c364 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -183,7 +183,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
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;
@@ -199,7 +199,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
}
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);
@@ -788,17 +788,19 @@ static const struct snd_soc_dapm_route intercon_3007[] = {
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;
@@ -1135,7 +1137,7 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
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);
@@ -1146,7 +1148,7 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
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);
@@ -1159,7 +1161,7 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
aic3x_set_power(codec, 0);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -1351,7 +1353,7 @@ static int aic3x_probe(struct snd_soc_codec *codec)
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) {
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index c5ab8c805771..7149c14b289e 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -628,11 +628,12 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -649,7 +650,7 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec,
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)
@@ -660,14 +661,14 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec,
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;
}
@@ -1415,7 +1416,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
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 */
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index ee4fb201de60..f9a92ea6b50a 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -388,16 +388,17 @@ static const struct snd_soc_dapm_route audio_map[] = {
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,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index cbebec6ba1ba..f4602e8b67cc 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1621,10 +1621,11 @@ static const struct snd_soc_dapm_route intercon[] = {
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;
}
@@ -1638,14 +1639,14 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
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;
}
@@ -2245,7 +2246,7 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec)
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);
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 10f6e5214511..0dd2d5397264 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -641,12 +641,12 @@ static const struct snd_soc_dapm_route intercon[] = {
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;
}
@@ -739,7 +739,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 7540a509a6f5..8ea81d48124a 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -389,7 +389,7 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec,
pd->power(0);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 0c6c725736c6..cd6dd19fa1aa 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -414,10 +414,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -603,7 +604,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec,
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) {
@@ -613,7 +614,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec,
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);
@@ -636,7 +637,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec,
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;
}
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 4bcd168794e1..9277d8d7474e 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -705,6 +705,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
/* 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) {
@@ -712,12 +713,12 @@ int wm2000_add_controls(struct snd_soc_codec *codec)
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;
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index f4f1fba38eb9..4c6c81e11544 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -230,8 +230,9 @@ static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec)
*/
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;
@@ -302,8 +303,8 @@ static int pga_event(struct snd_soc_dapm_widget *w,
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;
@@ -311,8 +312,8 @@ static int pga_event(struct snd_soc_dapm_widget *w,
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;
}
@@ -786,9 +787,10 @@ static const struct snd_soc_dapm_route audio_map[] = {
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) {
@@ -797,7 +799,7 @@ static int wm8350_add_widgets(struct snd_soc_codec *codec)
}
/* 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;
@@ -1184,7 +1186,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec,
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)
@@ -1317,7 +1319,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec,
priv->supplies);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -1550,7 +1552,7 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec)
/* 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);
@@ -1635,12 +1637,12 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec)
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();
}
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 850299786e02..96927a457a34 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -911,10 +911,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -1219,7 +1220,7 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec,
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) {
@@ -1306,7 +1307,7 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 8f107095760e..6b3833c7bdf3 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -216,10 +216,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -478,7 +479,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec,
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);
@@ -495,7 +496,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 712ef7c76f90..d3318886f43e 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -110,10 +110,11 @@ static const struct snd_soc_dapm_route intercon[] = {
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;
}
@@ -328,7 +329,7 @@ static int wm8523_set_bias_level(struct snd_soc_codec *codec,
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) {
@@ -367,7 +368,7 @@ static int wm8523_set_bias_level(struct snd_soc_codec *codec,
wm8523->supplies);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index a2e0ed59b376..dfd1dbd71f1d 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -302,10 +302,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -767,7 +768,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
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);
@@ -785,7 +786,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 54fbd76c8bca..ea2daf4da57c 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -93,10 +93,11 @@ static const struct snd_soc_dapm_route intercon[] = {
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;
}
@@ -318,7 +319,7 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, WM8711_PWR, 0xffff);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 075f35e4f4cb..23939976c3cc 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -73,10 +73,11 @@ static const struct snd_soc_dapm_route intercon[] = {
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;
}
@@ -180,7 +181,7 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec,
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);
@@ -197,7 +198,7 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, WM8728_DACCTL, reg | 0x4);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 631385802eb4..95ade3245056 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -165,10 +165,11 @@ static const struct snd_soc_dapm_route intercon[] = {
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;
}
@@ -319,7 +320,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(&codec->dapm);
return 0;
}
@@ -399,7 +400,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
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)
@@ -428,7 +429,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
wm8731->supplies);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 90e31e9aa6f7..43c49dfc9928 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -95,10 +95,11 @@ static const struct snd_soc_dapm_route intercon[] = {
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;
}
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 6c924cd2cfd4..178b967af73f 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -399,10 +399,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -615,7 +616,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec,
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);
@@ -630,7 +631,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, WM8750_PWR1, 0x0001);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 8f679a13f2bc..26096b47a493 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -670,10 +670,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -1292,7 +1293,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
wm8753_write(codec, WM8753_PWR1, 0x0001);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -1482,9 +1483,11 @@ static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
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)
@@ -1516,10 +1519,10 @@ static int wm8753_resume(struct snd_soc_codec *codec)
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));
}
@@ -1550,7 +1553,7 @@ static int wm8753_probe(struct snd_soc_codec *codec)
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) {
@@ -1569,7 +1572,7 @@ static int wm8753_probe(struct snd_soc_codec *codec)
/* 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 */
@@ -1604,7 +1607,7 @@ static int wm8753_probe(struct snd_soc_codec *codec)
/* 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;
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 04182c464e35..96474a40da8d 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -307,7 +307,7 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec,
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);
}
@@ -318,7 +318,7 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -405,6 +405,7 @@ static int wm8776_resume(struct snd_soc_codec *codec)
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);
@@ -428,9 +429,9 @@ static int wm8776_probe(struct snd_soc_codec *codec)
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;
}
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 4599e8e95aa2..031a0d421108 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -515,7 +515,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec,
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) {
@@ -537,7 +537,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -581,7 +581,7 @@ static int wm8804_probe(struct snd_soc_codec *codec)
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) {
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index aca4b1ea10bb..06ea9c0f863b 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -611,10 +611,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -1051,7 +1052,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec,
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);
@@ -1119,7 +1120,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec,
WM8900_REG_POWER2_SYSCLK_ENA);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 622b60238a82..4a6df4b69a04 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -923,10 +923,11 @@ static const struct snd_soc_dapm_route intercon[] = {
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;
}
@@ -946,7 +947,7 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
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);
@@ -991,7 +992,7 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 33be84e506ea..be90399c1cb4 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1428,10 +1428,11 @@ static const struct snd_soc_dapm_route wm8912_intercon[] = {
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) {
@@ -1443,20 +1444,20 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec)
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;
@@ -1464,17 +1465,17 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec)
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;
}
@@ -2139,7 +2140,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
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) {
@@ -2198,7 +2199,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
wm8904->supplies);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -2373,7 +2374,7 @@ static int wm8904_probe(struct snd_soc_codec *codec)
int ret, i;
codec->cache_sync = 1;
- codec->idle_bias_off = 1;
+ codec->dapm.idle_bias_off = 1;
switch (wm8904->devtype) {
case WM8904:
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 2cb16f895c46..c2def1b01ae0 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -291,13 +291,14 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index f89ad6c9a80b..df1940fdbf69 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -577,13 +577,14 @@ static const struct snd_soc_dapm_route wm8955_intercon[] = {
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;
@@ -786,7 +787,7 @@ static int wm8955_set_bias_level(struct snd_soc_codec *codec,
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) {
@@ -850,7 +851,7 @@ static int wm8955_set_bias_level(struct snd_soc_codec *codec,
wm8955->supplies);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 8d5efb333c33..0ea578815003 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -388,27 +388,28 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec)
{
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));
}
@@ -417,7 +418,7 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec)
* 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)
@@ -572,7 +573,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec,
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 |
@@ -610,7 +611,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -626,7 +627,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
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,
@@ -681,7 +682,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
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,
@@ -705,7 +706,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 4f326f604104..79b650945bb2 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -882,7 +882,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec,
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;
@@ -897,7 +897,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec,
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;
@@ -919,7 +919,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -959,6 +959,7 @@ static struct snd_soc_dai_driver wm8961_dai = {
static int wm8961_probe(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret = 0;
u16 reg;
@@ -1024,9 +1025,9 @@ static int wm8961_probe(struct snd_soc_codec *codec)
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;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 3fc63b43c6a1..80986105f52e 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2682,6 +2682,7 @@ static const struct snd_soc_dapm_route wm8962_spk_stereo_intercon[] = {
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));
@@ -2693,26 +2694,26 @@ static int wm8962_add_widgets(struct snd_soc_codec *codec)
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;
}
@@ -2819,7 +2820,7 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
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) {
@@ -2833,7 +2834,7 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
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) {
@@ -2883,7 +2884,7 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
wm8962->supplies);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -3441,6 +3442,7 @@ static void wm8962_beep_work(struct work_struct *work)
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;
@@ -3457,16 +3459,16 @@ static void wm8962_beep_work(struct work_struct *work)
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 -
@@ -3713,7 +3715,7 @@ static int wm8962_probe(struct snd_soc_codec *codec)
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) {
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 63f6dbf5d070..84b2dcb18aea 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -333,10 +333,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -553,7 +554,7 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, WM8971_PWR1, 0x0001);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -590,9 +591,11 @@ static struct snd_soc_dai_driver wm8971_dai = {
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)
@@ -620,11 +623,11 @@ static int wm8971_resume(struct snd_soc_codec *codec)
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));
}
@@ -643,7 +646,7 @@ static int wm8971_probe(struct snd_soc_codec *codec)
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;
@@ -653,8 +656,8 @@ static int wm8971_probe(struct snd_soc_codec *codec)
/* 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 */
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index b4363f6d19b3..d19bb14842d4 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -274,10 +274,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -530,7 +531,7 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec,
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);
@@ -547,7 +548,7 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 13b979a71a7c..ac43b6088e2e 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -355,11 +355,12 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -837,7 +838,7 @@ static int wm8978_set_bias_level(struct snd_soc_codec *codec,
/* 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);
@@ -857,7 +858,7 @@ static int wm8978_set_bias_level(struct snd_soc_codec *codec,
dev_dbg(codec->dev, "%s: %d, %x\n", __func__, level, power1);
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index fd2e7cca1228..c3c8fd23d503 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -533,10 +533,11 @@ static int eqmode_put(struct snd_kcontrol *kcontrol,
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;
}
@@ -879,7 +880,7 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec,
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) {
@@ -939,7 +940,7 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index d7f259711970..0bc2eb530c7a 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -677,7 +677,7 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec,
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);
@@ -693,7 +693,7 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, WM8988_PWR1, 0x0000);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -759,6 +759,7 @@ static int wm8988_resume(struct snd_soc_codec *codec)
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;
@@ -790,9 +791,9 @@ static int wm8988_probe(struct snd_soc_codec *codec)
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;
}
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 264828e4e67c..309664ea7dc3 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -914,11 +914,12 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -1170,7 +1171,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
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 |
@@ -1266,7 +1267,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 67fe5ccc6082..bcc54be572ce 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -970,7 +970,7 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec,
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)
@@ -1045,7 +1045,7 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -1424,6 +1424,7 @@ static struct snd_soc_dai_driver wm8993_dai = {
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;
@@ -1505,11 +1506,11 @@ static int wm8993_probe(struct snd_soc_codec *codec)
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);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index d81cac5b93b4..f7dea3d34a3e 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1835,7 +1835,7 @@ static int configure_clock(struct snd_soc_codec *codec)
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;
}
@@ -3108,7 +3108,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
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) {
@@ -3152,7 +3152,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
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 |
@@ -3187,7 +3187,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
}
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -3895,6 +3895,7 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data)
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);
@@ -4033,10 +4034,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
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;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index ecc7c37180c7..c03e2c3e24e1 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -805,7 +805,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
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;
@@ -865,7 +865,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
@@ -1228,6 +1228,7 @@ static struct snd_soc_dai_driver wm9081_dai = {
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;
@@ -1269,9 +1270,9 @@ static int wm9081_probe(struct snd_soc_codec *codec)
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;
}
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 99c046ba46bb..b5afa01aa383 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -443,31 +443,32 @@ static const struct snd_soc_dapm_route audio_map_in2_diff[] = {
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));
@@ -514,7 +515,7 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec,
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])
@@ -544,7 +545,7 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index a144acda751c..58d120824498 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -203,9 +203,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index d2f224d62744..3ca42a35e03a 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -432,10 +432,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -570,7 +571,7 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec,
ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 7da13b07a53d..87b236b16016 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -647,10 +647,12 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
@@ -1147,7 +1149,7 @@ static int wm9713_set_bias_level(struct snd_soc_codec *codec,
ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 008b1f27aea8..8aff0efe72f5 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -814,6 +814,8 @@ static const struct snd_soc_dapm_route lineout2_se_routes[] = {
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);
@@ -842,7 +844,7 @@ int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec)
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;
}
@@ -851,24 +853,26 @@ EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_controls);
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));
@@ -895,7 +899,7 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec,
* 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,
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 2b07b17a6b2d..a2cf64b221e5 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -132,26 +132,27 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c
index 28ab5ff772ac..f1c78516ccac 100644
--- a/sound/soc/ep93xx/snappercl15.c
+++ b/sound/soc/ep93xx/snappercl15.c
@@ -79,11 +79,12 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c
index 30fdb15065be..46fadf497243 100644
--- a/sound/soc/imx/wm1133-ev1.c
+++ b/sound/soc/imx/wm1133-ev1.c
@@ -213,11 +213,12 @@ static struct snd_soc_jack_pin mic_jack_pins[] = {
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 */
@@ -234,7 +235,7 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c
index ef1a99e6a3bd..70afbfada9fd 100644
--- a/sound/soc/jz4740/qi_lb60.c
+++ b/sound/soc/jz4740/qi_lb60.c
@@ -59,10 +59,11 @@ static int qi_lb60_codec_init(struct snd_soc_pcm_runtime *rtd)
{
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) {
@@ -70,9 +71,11 @@ static int qi_lb60_codec_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c
index 51b52e31cb0b..07b6ecaed2f2 100644
--- a/sound/soc/kirkwood/kirkwood-t5325.c
+++ b/sound/soc/kirkwood/kirkwood-t5325.c
@@ -69,17 +69,18 @@ static const struct snd_soc_dapm_route t5325_route[] = {
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(codec, t5325_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, t5325_dapm_widgets,
ARRAY_SIZE(t5325_dapm_widgets));
- snd_soc_dapm_add_routes(codec, t5325_route, ARRAY_SIZE(t5325_route));
+ snd_soc_dapm_add_routes(dapm, t5325_route, ARRAY_SIZE(t5325_route));
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Speaker");
+ 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(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
index 979dd508305f..668773def0dc 100644
--- a/sound/soc/omap/am3517evm.c
+++ b/sound/soc/omap/am3517evm.c
@@ -114,20 +114,21 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 438146addbb8..2101bdcee21f 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -26,7 +26,7 @@
#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>
@@ -94,6 +94,7 @@ static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
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;
@@ -112,48 +113,48 @@ static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
/* 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);
@@ -164,19 +165,20 @@ static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
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));
@@ -300,6 +302,7 @@ static int cx81801_open(struct tty_struct *tty)
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);
@@ -312,12 +315,12 @@ static void cx81801_close(struct tty_struct *tty)
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() */
@@ -432,16 +435,16 @@ static int ams_delta_set_bias_level(struct snd_soc_card *card,
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;
}
@@ -492,6 +495,7 @@ static void ams_delta_shutdown(struct snd_pcm_substream *substream)
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;
@@ -541,7 +545,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
}
/* 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,
@@ -550,7 +554,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
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,
@@ -560,13 +564,13 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
}
/* 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,
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index a3b6d897ad84..296cd9b7eecb 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -58,6 +58,7 @@ static int n810_dmic_func;
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) {
@@ -72,25 +73,25 @@ static void n810_ext_control(struct snd_soc_codec *codec)
}
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)
@@ -274,17 +275,18 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = {
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,
@@ -293,13 +295,13 @@ static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index dbd9d96b5f92..93e83c0f6660 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -170,51 +170,53 @@ static const struct snd_soc_dapm_route omap3pandora_in_map[] = {
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 = {
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index f0e662556428..c2a54204559d 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -116,19 +116,20 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 04b5723bf89b..62fc7a4f306b 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -58,19 +58,21 @@ static int rx51_jack_func;
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)
@@ -244,12 +246,13 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = {
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,
@@ -258,13 +261,13 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
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",
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index 07fbcf7d2411..a3dd07a39fec 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -191,39 +191,40 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c
index 4b4463db6ba0..3ce17318a291 100644
--- a/sound/soc/omap/sdp4430.c
+++ b/sound/soc/omap/sdp4430.c
@@ -129,6 +129,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
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 */
@@ -138,25 +139,25 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c
index 718031eeac34..cc5bc523b302 100644
--- a/sound/soc/omap/zoom2.c
+++ b/sound/soc/omap/zoom2.c
@@ -162,35 +162,36 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 97e9423615c9..810633cc3b6d 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -48,51 +48,53 @@ static int corgi_spk_func;
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)
@@ -274,10 +276,11 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
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,
@@ -286,13 +289,13 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index c82cedb602fd..38a84b821ff4 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -92,23 +92,24 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 4c143803a75e..2bc97e92446b 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -74,23 +74,24 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index d42e5fe832c5..eac846c7bd9c 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -75,12 +75,13 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index b8207ced4072..f1acdc57cfd8 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -44,27 +44,29 @@ static int magician_in_sel = MAGICIAN_MIC;
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)
@@ -395,15 +397,16 @@ static const struct snd_kcontrol_new uda1380_magician_controls[] = {
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,
@@ -412,13 +415,13 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index f284cc54bc80..f7a1e8f09f9a 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -130,13 +130,14 @@ static const struct snd_soc_dapm_route audio_map[] = {
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);
@@ -146,12 +147,12 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 13f6d485d571..530064dd06a9 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -77,37 +77,38 @@ static struct snd_soc_card palm27x_asoc;
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;
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index af84ee9c5e11..7353ee5034fe 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -46,6 +46,8 @@ static int poodle_spk_func;
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 */
@@ -53,23 +55,23 @@ static void poodle_ext_control(struct snd_soc_codec *codec)
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)
@@ -239,11 +241,12 @@ static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
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,
@@ -252,13 +255,13 @@ static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c
index d63cb474b4e1..ee06f9982c09 100644
--- a/sound/soc/pxa/saarb.c
+++ b/sound/soc/pxa/saarb.c
@@ -133,20 +133,21 @@ static struct snd_soc_card snd_soc_card_saarb = {
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;
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index f470f360f4dd..0680b11c2685 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -46,61 +46,63 @@ static int spitz_spk_func;
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)
@@ -276,16 +278,17 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
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,
@@ -294,13 +297,13 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c
index 248c283fc4df..18cbe0e7c223 100644
--- a/sound/soc/pxa/tavorevb3.c
+++ b/sound/soc/pxa/tavorevb3.c
@@ -133,20 +133,21 @@ static struct snd_soc_card snd_soc_card_evb3 = {
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;
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 73d0edd8ded9..0a9bd68ef749 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -49,31 +49,33 @@ static int tosa_spk_func;
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)
@@ -186,10 +188,11 @@ static const struct snd_kcontrol_new tosa_controls[] = {
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,
@@ -198,13 +201,13 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index 4cc841b44182..cacbcd4a55eb 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -140,22 +140,23 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index d27e05af7759..c74eac30ebff 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -73,21 +73,22 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c
index 235d1973f7d0..33bebdae08a7 100644
--- a/sound/soc/s3c24xx/aquila_wm8994.c
+++ b/sound/soc/s3c24xx/aquila_wm8994.c
@@ -93,27 +93,28 @@ static const struct snd_soc_dapm_route aquila_dapm_routes[] = {
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",
diff --git a/sound/soc/s3c24xx/goni_wm8994.c b/sound/soc/s3c24xx/goni_wm8994.c
index 694f702cc8e2..052729c6540d 100644
--- a/sound/soc/s3c24xx/goni_wm8994.c
+++ b/sound/soc/s3c24xx/goni_wm8994.c
@@ -97,25 +97,26 @@ static const struct snd_soc_dapm_route goni_dapm_routes[] = {
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",
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
index 49605cd83947..e3599e283568 100644
--- a/sound/soc/s3c24xx/jive_wm8750.c
+++ b/sound/soc/s3c24xx/jive_wm8750.c
@@ -111,18 +111,19 @@ static struct snd_soc_ops jive_ops = {
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",
@@ -130,8 +131,8 @@ static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
index e97bdf150a03..c3f63ef8ab12 100644
--- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
@@ -333,16 +333,17 @@ static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = {
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 */
@@ -353,25 +354,25 @@ static int neo1973_gta02_wm8753_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index f4f2ee731f01..e94ffe01a4a5 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -237,81 +237,83 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
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;
}
@@ -502,20 +504,21 @@ static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
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 */
@@ -528,10 +531,10 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/s3c24xx/rx1950_uda1380.c b/sound/soc/s3c24xx/rx1950_uda1380.c
index ffd5cf2fb0a9..105d177fa427 100644
--- a/sound/soc/s3c24xx/rx1950_uda1380.c
+++ b/sound/soc/s3c24xx/rx1950_uda1380.c
@@ -232,26 +232,27 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream,
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);
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
index f88453735ae2..05c793705d90 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
@@ -76,19 +76,20 @@ static const struct snd_soc_dapm_route base_map[] = {
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;
}
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
index c0967593510d..653dc7592e81 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
@@ -65,19 +65,20 @@ static const struct snd_soc_dapm_route base_map[] = {
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;
}
diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c
index dd20ca7f4681..1f6da1e27b1e 100644
--- a/sound/soc/s3c24xx/smartq_wm8987.c
+++ b/sound/soc/s3c24xx/smartq_wm8987.c
@@ -158,10 +158,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
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 */
@@ -172,20 +173,20 @@ static int smartq_wm8987_init(struct snd_soc_codec *codec)
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;
diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c
index 052e499b68d1..291939cf8483 100644
--- a/sound/soc/s3c24xx/smdk64xx_wm8580.c
+++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c
@@ -182,21 +182,22 @@ static const struct snd_soc_dapm_route audio_map_rx[] = {
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;
}
@@ -204,16 +205,17 @@ static int smdk64xx_wm8580_init_paiftx(struct snd_soc_pcm_runtime *rtd)
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;
}
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index 96c05e137538..db1803d9665a 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -107,6 +107,7 @@ static int output_type_put(struct snd_kcontrol *kcontrol,
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";
@@ -114,10 +115,10 @@ static int output_type_put(struct snd_kcontrol *kcontrol,
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;
}
@@ -137,35 +138,36 @@ static const struct snd_kcontrol_new audio_out_mux = {
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));
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index ac6c49ce6fdf..c61fc188394d 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -140,11 +140,12 @@ static const struct snd_soc_dapm_route audio_map[] = {
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;
}
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index f8e0ab82ef59..105d4112e3ba 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -23,7 +23,7 @@ extern struct snd_soc_platform_driver sh7760_soc_platform;
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;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2198936cfb68..3c7c884f212c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -255,18 +255,18 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644,
codec->debugfs_codec_root,
- &codec->pop_time);
+ &codec->dapm.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)
@@ -1017,7 +1017,7 @@ static int soc_suspend(struct device *dev)
/* 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++) {
@@ -1041,7 +1041,7 @@ static int soc_suspend(struct device *dev)
/* 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);
@@ -1110,7 +1110,7 @@ static void soc_resume_deferred(struct work_struct *work)
* 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);
@@ -1346,7 +1346,7 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num)
}
/* 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);
@@ -1470,8 +1470,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num)
}
/* 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;
@@ -3238,6 +3238,12 @@ int snd_soc_register_codec(struct device *dev,
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) {
@@ -3257,11 +3263,8 @@ int snd_soc_register_codec(struct device *dev,
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);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7d85c6496afa..b8f653eaffaa 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -42,6 +42,7 @@
#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>
@@ -120,35 +121,36 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget(
* 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;
@@ -241,7 +243,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
}
/* 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)
@@ -251,7 +253,7 @@ static int dapm_connect_mux(struct snd_soc_codec *codec,
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];
@@ -264,7 +266,7 @@ static int dapm_connect_mux(struct snd_soc_codec *codec,
}
/* 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)
{
@@ -273,7 +275,7 @@ static int dapm_connect_mixer(struct snd_soc_codec *codec,
/* 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;
@@ -290,6 +292,7 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
int change, power;
unsigned int old, new;
struct snd_soc_codec *codec = widget->codec;
+ struct snd_soc_dapm_context *dapm = widget->dapm;
/* check for valid widgets */
if (widget->reg < 0 || widget->id == snd_soc_dapm_input ||
@@ -309,10 +312,10 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
change = old != new;
if (change) {
- pop_dbg(codec->pop_time, "pop test %s : %s in %d ms\n",
+ pop_dbg(dapm->pop_time, "pop test %s : %s in %d ms\n",
widget->name, widget->power ? "on" : "off",
- codec->pop_time);
- pop_wait(codec->pop_time);
+ dapm->pop_time);
+ pop_wait(dapm->pop_time);
snd_soc_write(codec, widget->reg, new);
}
pr_debug("reg %x old %x new %x change %d\n", widget->reg,
@@ -321,12 +324,13 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
}
/* 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++) {
@@ -368,7 +372,7 @@ static int dapm_new_mixer(struct snd_soc_codec *codec,
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,
@@ -383,11 +387,12 @@ static int dapm_new_mixer(struct snd_soc_codec *codec,
}
/* 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) {
@@ -396,7 +401,8 @@ static int dapm_new_mux(struct snd_soc_codec *codec,
}
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;
@@ -411,7 +417,7 @@ 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)
@@ -421,11 +427,11 @@ static int dapm_new_pga(struct snd_soc_codec *codec,
}
/* 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;
}
@@ -435,7 +441,7 @@ static inline void dapm_clear_walk(struct snd_soc_codec *codec)
*/
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:
@@ -621,9 +627,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w)
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;
}
@@ -634,7 +640,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w)
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);
@@ -648,7 +654,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *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);
@@ -674,7 +680,7 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w)
}
}
- dapm_clear_walk(w->codec);
+ dapm_clear_walk(w->dapm);
return power;
}
@@ -710,7 +716,7 @@ static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget,
}
/* 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;
@@ -735,14 +741,14 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec,
if (power)
value |= cur_mask;
- pop_dbg(codec->pop_time,
+ pop_dbg(dapm->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(dapm->pop_time, "pop test : %s PRE_PMU\n",
w->name);
ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
if (ret < 0)
@@ -753,7 +759,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec,
/* 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(dapm->pop_time, "pop test : %s PRE_PMD\n",
w->name);
ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
if (ret < 0)
@@ -763,18 +769,18 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec,
}
if (reg >= 0) {
- pop_dbg(codec->pop_time,
+ pop_dbg(dapm->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, dapm->pop_time);
+ pop_wait(dapm->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(dapm->pop_time, "pop test : %s POST_PMU\n",
w->name);
ret = w->event(w,
NULL, SND_SOC_DAPM_POST_PMU);
@@ -786,7 +792,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec,
/* 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(dapm->pop_time, "pop test : %s POST_PMD\n",
w->name);
ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
if (ret < 0)
@@ -804,8 +810,8 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec,
* 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);
@@ -819,7 +825,7 @@ static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list,
/* 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;
@@ -877,7 +883,7 @@ static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list,
}
if (!list_empty(&pending))
- dapm_seq_run_coalesced(codec, &pending);
+ dapm_seq_run_coalesced(dapm, &pending);
}
/*
@@ -889,9 +895,9 @@ static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list,
* 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);
@@ -902,7 +908,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
/* 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);
@@ -938,7 +944,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
/* 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:
@@ -948,7 +954,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
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;
@@ -963,52 +969,52 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
}
}
- 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(dapm->pop_time, "DAPM sequencing finished, waiting %dms\n",
+ dapm->pop_time);
+ pop_wait(dapm->pop_time);
return 0;
}
@@ -1035,9 +1041,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
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);
@@ -1087,20 +1093,20 @@ static const struct file_operations dapm_widget_power_fops = {
.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
@@ -1109,7 +1115,7 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec)
}
}
#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
@@ -1130,7 +1136,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
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;
@@ -1146,7 +1152,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
}
if (found)
- dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+ dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP);
return 0;
}
@@ -1164,7 +1170,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
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;
@@ -1175,7 +1181,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
}
if (found)
- dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+ dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP);
return 0;
}
@@ -1191,7 +1197,7 @@ static ssize_t dapm_widget_show(struct device *dev,
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) {
@@ -1215,7 +1221,7 @@ static ssize_t dapm_widget_show(struct device *dev,
}
}
- switch (codec->bias_level) {
+ switch (codec->dapm.bias_level) {
case SND_SOC_BIAS_ON:
state = "On";
break;
@@ -1247,31 +1253,31 @@ static void snd_soc_dapm_sys_remove(struct device *dev)
}
/* 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)
@@ -1280,26 +1286,27 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec,
}
}
- 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;
@@ -1310,7 +1317,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
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;
@@ -1353,7 +1360,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
/* 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;
@@ -1374,14 +1381,14 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
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;
@@ -1389,7 +1396,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
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;
@@ -1397,7 +1404,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
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;
@@ -1414,7 +1421,7 @@ err:
/**
* snd_soc_dapm_add_routes - Add routes between DAPM widgets
- * @codec: codec
+ * @dapm: DAPM context
* @route: audio routes
* @num: number of routes
*
@@ -1425,13 +1432,13 @@ err:
* 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,
@@ -1447,17 +1454,17 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
/**
* 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;
@@ -1467,12 +1474,12 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
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:
@@ -1484,7 +1491,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
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:
@@ -1505,7 +1512,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
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);
@@ -1889,7 +1896,7 @@ int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol,
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);
@@ -1912,11 +1919,11 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
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);
@@ -1926,14 +1933,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch);
/**
* 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;
@@ -1941,11 +1948,12 @@ int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
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;
@@ -1955,7 +1963,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
/**
* snd_soc_dapm_new_controls - create new dapm controls
- * @codec: audio codec
+ * @dapm: DAPM context
* @widget: widget array
* @num: number of widgets
*
@@ -1963,14 +1971,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
*
* 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",
@@ -1983,29 +1991,12 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
}
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;
@@ -2028,7 +2019,30 @@ int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
}
}
- 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;
}
@@ -2036,7 +2050,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
/**
* 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
@@ -2044,15 +2058,15 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
* 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
@@ -2062,42 +2076,45 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
* 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
@@ -2109,26 +2126,27 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
* 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;
}
@@ -2139,7 +2157,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status);
/**
* 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
@@ -2148,11 +2166,12 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status);
* 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;
@@ -2170,20 +2189,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend);
*
* 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;
@@ -2195,9 +2214,9 @@ static void soc_dapm_shutdown_codec(struct snd_soc_codec *codec)
* 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);
}
}
@@ -2208,10 +2227,10 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
{
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 */
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 9f07551e155f..4d95abb40288 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -60,6 +60,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new);
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;
@@ -68,6 +69,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
return;
codec = jack->codec;
+ dapm = &codec->dapm;
mutex_lock(&codec->mutex);
@@ -88,15 +90,15 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
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);