CHROMIUM: ASoC: Import entire upstream ASoC tree

sound/soc/codecs/tlv320aic23.c

Index: sound/soc/codecs/tlv320aic23.c
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index e8652b1ae32606b5c57bd0ff7d0a8b4a304fa9d5..54a30ef0ec8b7fec66bacd637c6cd0481cead799 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;
 }