ASoC: Decouple DAPM from CODECs

Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.

This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.

This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.

Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.

Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

59 files changed, 379 insertions, 312 deletions

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,