Merge branch 'topic/asoc' into for-linus

1 files changed, 5 insertions, 5 deletions

diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index e8652b1ae326..54a30ef0ec8b 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -30,7 +30,6 @@
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
-#include <sound/soc-dapm.h>
 #include <sound/tlv.h>
 #include <sound/initval.h>
 
@@ -391,11 +390,12 @@ 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 +574,7 @@ 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;
 }