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 01d19e9f53f..a15a3e974f0 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 d272534c8f8..c71b05ddd75 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 fa2834c91b9..dc105d8aaa0 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 cd88c8f32a3..52abb93a7dc 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 90c90b7f4a2..f00eba313df 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 24f5f49bb9d..1d6573c38af 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 fac61744f8c..5a45067b43b 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 823643932dd..98b9e5294cb 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 cb086eaf4e0..a7fdca36b49 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 e8d27c8f9ba..11beb1a77c4 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 58bb9b99481..92fd9d7a922 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 16253ec9b02..8a45562a96d 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 bc22ee93a75..ef06007d889 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 6f38d619bf8..adbc3e8dafc 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 00d67cc8e20..8aad3a2c4f3 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 e8652b1ae32..d9d8e844d63 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 fc687790188..6173c2b4c36 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 s