Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next

include/sound/soc-dapm.h

@@ -107,6 +107,10 @@ struct device;
 {	.id = snd_soc_dapm_mux, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.kcontrol_news = wcontrols, .num_kcontrols = 1}
+#define SND_SOC_DAPM_DEMUX(wname, wreg, wshift, winvert, wcontrols) \
+{	.id = snd_soc_dapm_demux, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = 1}
 
 /* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
 #define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\
@@ -444,11 +448,15 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
 struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm(
 	struct snd_kcontrol *kcontrol);
 
+int snd_soc_dapm_force_bias_level(struct snd_soc_dapm_context *dapm,
+	enum snd_soc_bias_level level);
+
 /* dapm widget types */
 enum snd_soc_dapm_type {
 	snd_soc_dapm_input = 0,		/* input pin */
 	snd_soc_dapm_output,		/* output pin */
 	snd_soc_dapm_mux,			/* selects 1 analog signal from many inputs */
+	snd_soc_dapm_demux,			/* connects the input to one of multiple outputs */
 	snd_soc_dapm_mixer,			/* mixes several analog signals together */
 	snd_soc_dapm_mixer_named_ctl,		/* mixer with named controls */
 	snd_soc_dapm_pga,			/* programmable gain/attenuation (volume) */
@@ -585,6 +593,10 @@ struct snd_soc_dapm_update {
 	int val;
 };
 
+struct snd_soc_dapm_wcache {
+	struct snd_soc_dapm_widget *widget;
+};
+
 /* DAPM context */
 struct snd_soc_dapm_context {
 	enum snd_soc_bias_level bias_level;
@@ -606,6 +618,9 @@ struct snd_soc_dapm_context {
 	int (*set_bias_level)(struct snd_soc_dapm_context *dapm,
 			      enum snd_soc_bias_level level);
 
+	struct snd_soc_dapm_wcache path_sink_cache;
+	struct snd_soc_dapm_wcache path_source_cache;
+
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_dapm;
 #endif
@@ -623,4 +638,35 @@ struct snd_soc_dapm_stats {
 	int neighbour_checks;
 };
 
+/**
+ * snd_soc_dapm_init_bias_level() - Initialize DAPM bias level
+ * @dapm: The DAPM context to initialize
+ * @level: The DAPM level to initialize to
+ *
+ * This function only sets the driver internal state of the DAPM level and will
+ * not modify the state of the device. Hence it should not be used during normal
+ * operation, but only to synchronize the internal state to the device state.
+ * E.g. during driver probe to set the DAPM level to the one corresponding with
+ * the power-on reset state of the device.
+ *
+ * To change the DAPM state of the device use snd_soc_dapm_set_bias_level().
+ */
+static inline void snd_soc_dapm_init_bias_level(
+	struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level)
+{
+	dapm->bias_level = level;
+}
+
+/**
+ * snd_soc_dapm_get_bias_level() - Get current DAPM bias level
+ * @dapm: The context for which to get the bias level
+ *
+ * Returns: The current bias level of the passed DAPM context.
+ */
+static inline enum snd_soc_bias_level snd_soc_dapm_get_bias_level(
+	struct snd_soc_dapm_context *dapm)
+{
+	return dapm->bias_level;
+}
+
 #endif

include/sound/soc.h

@@ -190,8 +190,12 @@
 #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \
 {	.reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
 	.mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues}
-#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \
-	SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues)
+#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xitems, xtexts, xvalues) \
+	SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xitems, xtexts, xvalues)
+#define SOC_VALUE_ENUM_SINGLE_AUTODISABLE(xreg, xshift, xmask, xitems, xtexts, xvalues) \
+{	.reg = xreg, .shift_l = xshift, .shift_r = xshift, \
+	.mask = xmask, .items = xitems, .texts = xtexts, \
+	.values = xvalues, .autodisable = 1}
 #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \
 	SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts)
 #define SOC_ENUM(xname, xenum) \
@@ -312,6 +316,11 @@
 							ARRAY_SIZE(xtexts), xtexts, xvalues)
 #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
 	SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
+
+#define SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
+	const struct soc_enum name = SOC_VALUE_ENUM_SINGLE_AUTODISABLE(xreg, \
+		xshift, xmask, ARRAY_SIZE(xtexts), xtexts, xvalues)
+
 #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \
 	const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts)
 
@@ -819,7 +828,7 @@ struct snd_soc_codec {
 	/* component */
 	struct snd_soc_component component;
 
-	/* dapm */
+	/* Don't access this directly, use snd_soc_codec_get_dapm() */
 	struct snd_soc_dapm_context dapm;
 
 #ifdef CONFIG_DEBUG_FS
@@ -1200,6 +1209,7 @@ struct soc_enum {
 	unsigned int mask;
 	const char * const *texts;
 	const unsigned int *values;
+	unsigned int autodisable:1;
 };
 
 /**
@@ -1281,6 +1291,58 @@ static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm(
 	return component->dapm_ptr;
 }
 
+/**
+ * snd_soc_codec_get_dapm() - Returns the DAPM context for the CODEC
+ * @codec: The CODEC for which to get the DAPM context
+ *
+ * Note: Use this function instead of directly accessing the CODEC's dapm field
+ */
+static inline struct snd_soc_dapm_context *snd_soc_codec_get_dapm(
+	struct snd_soc_codec *codec)
+{
+	return &codec->dapm;
+}
+
+/**
+ * snd_soc_dapm_init_bias_level() - Initialize CODEC DAPM bias level
+ * @dapm: The CODEC for which to initialize the DAPM bias level
+ * @level: The DAPM level to initialize to
+ *
+ * Initializes the CODEC DAPM bias level. See snd_soc_dapm_init_bias_level().
+ */
+static inline void snd_soc_codec_init_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+	snd_soc_dapm_init_bias_level(snd_soc_codec_get_dapm(codec), level);
+}
+
+/**
+ * snd_soc_dapm_get_bias_level() - Get current CODEC DAPM bias level
+ * @codec: The CODEC for which to get the DAPM bias level
+ *
+ * Returns: The current DAPM bias level of the CODEC.
+ */
+static inline enum snd_soc_bias_level snd_soc_codec_get_bias_level(
+	struct snd_soc_codec *codec)
+{
+	return snd_soc_dapm_get_bias_level(snd_soc_codec_get_dapm(codec));
+}
+
+/**
+ * snd_soc_codec_force_bias_level() - Set the CODEC DAPM bias level
+ * @codec: The CODEC for which to set the level
+ * @level: The level to set to
+ *
+ * Forces the CODEC bias level to a specific state. See
+ * snd_soc_dapm_force_bias_level().
+ */
+static inline int snd_soc_codec_force_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+	return snd_soc_dapm_force_bias_level(snd_soc_codec_get_dapm(codec),
+		level);
+}
+
 /**
  * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol
  * @kcontrol: The kcontrol

sound/soc/codecs/88pm860x-codec.c

@@ -1140,7 +1140,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Enable Audio PLL & Audio section */
 			data = AUDIO_PLL | AUDIO_SECTION_ON;
 			pm860x_reg_write(pm860x->i2c, REG_MISC2, data);
@@ -1156,7 +1156,6 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec,
 		pm860x_set_bits(pm860x->i2c, REG_MISC2, data, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/ab8500-codec.c

@@ -1209,6 +1209,7 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
 	struct device *dev = codec->dev;
 	bool apply_fir, apply_iir;
@@ -1234,15 +1235,14 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
 	apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR;
 	apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR;
 
-	status = snd_soc_dapm_force_enable_pin(&codec->dapm,
-					"ANC Configure Input");
+	status = snd_soc_dapm_force_enable_pin(dapm, "ANC Configure Input");
 	if (status < 0) {
 		dev_err(dev,
 			"%s: ERROR: Failed to enable power (status = %d)!\n",
 			__func__, status);
 		goto cleanup;
 	}
-	snd_soc_dapm_sync(&codec->dapm);
+	snd_soc_dapm_sync(dapm);
 
 	anc_configure(codec, apply_fir, apply_iir);
 
@@ -1259,8 +1259,8 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
 			drvdata->anc_status =  ANC_IIR_CONFIGURED;
 	}
 
-	status = snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
-	snd_soc_dapm_sync(&codec->dapm);
+	status = snd_soc_dapm_disable_pin(dapm, "ANC Configure Input");
+	snd_soc_dapm_sync(dapm);
 
 cleanup:
 	mutex_unlock(&drvdata->ctrl_lock);
@@ -1947,6 +1947,7 @@ static int ab8500_audio_init_audioblock(struct snd_soc_codec *codec)
 static int ab8500_audio_setup_mics(struct snd_soc_codec *codec,
 			struct amic_settings *amics)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	u8 value8;
 	unsigned int value;
 	int status;
@@ -1973,15 +1974,15 @@ static int ab8500_audio_setup_mics(struct snd_soc_codec *codec,
 	dev_dbg(codec->dev, "%s: Mic 1a regulator: %s\n", __func__,
 		amic_micbias_str(amics->mic1a_micbias));
 	route = &ab8500_dapm_routes_mic1a_vamicx[amics->mic1a_micbias];
-	status = snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	status = snd_soc_dapm_add_routes(dapm, route, 1);
 	dev_dbg(codec->dev, "%s: Mic 1b regulator: %s\n", __func__,
 		amic_micbias_str(amics->mic1b_micbias));
 	route = &ab8500_dapm_routes_mic1b_vamicx[amics->mic1b_micbias];
-	status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	status |= snd_soc_dapm_add_routes(dapm, route, 1);
 	dev_dbg(codec->dev, "%s: Mic 2 regulator: %s\n", __func__,
 		amic_micbias_str(amics->mic2_micbias));
 	route = &ab8500_dapm_routes_mic2_vamicx[amics->mic2_micbias];
-	status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	status |= snd_soc_dapm_add_routes(dapm, route, 1);
 	if (status < 0) {
 		dev_err(codec->dev,
 			"%s: Failed to add AMic-regulator DAPM-routes (%d).\n",
@@ -2461,6 +2462,7 @@ static void ab8500_codec_of_probe(struct device *dev, struct device_node *np,
 
 static int ab8500_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct device *dev = codec->dev;
 	struct device_node *np = dev->of_node;
 	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev);
@@ -2541,7 +2543,7 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec)
 		&ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value;
 	drvdata->sid_fir_values = (long *)fc->value;
 
-	(void)snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
+	snd_soc_dapm_disable_pin(dapm, "ANC Configure Input");
 
 	mutex_init(&drvdata->ctrl_lock);
 

sound/soc/codecs/adau1373.c

@@ -1444,7 +1444,6 @@ static int adau1373_set_bias_level(struct snd_soc_codec *codec,
 			ADAU1373_PWDN_CTRL3_PWR_EN, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/adau1701.c

@@ -565,7 +565,6 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/adau1761.c

@@ -466,7 +466,6 @@ static int adau1761_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/adau1781.c

@@ -339,7 +339,6 @@ static int adau1781_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/adau1977.c

@@ -493,12 +493,7 @@ static int adau1977_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	if (ret)
-		return ret;
-
-	codec->dapm.bias_level = level;
-
-	return 0;
+	return ret;
 }
 
 static int adau1977_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,

sound/soc/codecs/adav80x.c

@@ -714,7 +714,6 @@ static int adav80x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/ak4535.c

@@ -341,7 +341,6 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/ak4641.c

@@ -412,7 +412,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0x20);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			if (pdata && gpio_is_valid(pdata->gpio_power))
 				gpio_set_value(pdata->gpio_power, 1);
 			mdelay(1);
@@ -439,7 +439,6 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec,
 		regcache_mark_dirty(ak4641->regmap);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/ak4642.c

@@ -482,7 +482,6 @@ static int ak4642_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }

sound/soc/codecs/ak4671.c

@@ -577,7 +577,6 @@ static int ak4671_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/alc5623.c

@@ -826,7 +826,6 @@ static int alc5623_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -894,7 +893,7 @@ 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;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	alc5623_reset(codec);
 

sound/soc/codecs/alc5632.c

@@ -1000,7 +1000,6 @@ static int alc5632_set_bias_level(struct snd_soc_codec *codec,
 				ALC5632_PWR_MANAG_ADD1_MASK, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/arizona.c

@@ -208,11 +208,12 @@ static const struct snd_soc_dapm_widget arizona_spkr =
 
 int arizona_init_spk(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
 	struct arizona *arizona = priv->arizona;
 	int ret;
 
-	ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkl, 1);
+	ret = snd_soc_dapm_new_controls(dapm, &arizona_spkl, 1);
 	if (ret != 0)
 		return ret;
 
@@ -220,8 +221,7 @@ int arizona_init_spk(struct snd_soc_codec *codec)
 	case WM8997:
 		break;
 	default:
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
-						&arizona_spkr, 1);
+		ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1);
 		if (ret != 0)
 			return ret;
 		break;
@@ -258,13 +258,14 @@ static const struct snd_soc_dapm_route arizona_mono_routes[] = {
 
 int arizona_init_mono(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
 	struct arizona *arizona = priv->arizona;
 	int i;
 
 	for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) {
 		if (arizona->pdata.out_mono[i])
-			snd_soc_dapm_add_routes(&codec->dapm,
+			snd_soc_dapm_add_routes(dapm,
 						&arizona_mono_routes[i], 1);
 	}
 
@@ -274,6 +275,7 @@ EXPORT_SYMBOL_GPL(arizona_init_mono);
 
 int arizona_init_gpio(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
 	struct arizona *arizona = priv->arizona;
 	int i;
@@ -281,23 +283,21 @@ int arizona_init_gpio(struct snd_soc_codec *codec)
 	switch (arizona->type) {
 	case WM5110:
 	case WM8280:
-		snd_soc_dapm_disable_pin(&codec->dapm, "DRC2 Signal Activity");
+		snd_soc_dapm_disable_pin(dapm, "DRC2 Signal Activity");
 		break;
 	default:
 		break;
 	}
 
-	snd_soc_dapm_disable_pin(&codec->dapm, "DRC1 Signal Activity");
+	snd_soc_dapm_disable_pin(dapm, "DRC1 Signal Activity");
 
 	for (i = 0; i < ARRAY_SIZE(arizona->pdata.gpio_defaults); i++) {
 		switch (arizona->pdata.gpio_defaults[i] & ARIZONA_GPN_FN_MASK) {
 		case ARIZONA_GP_FN_DRC1_SIGNAL_DETECT:
-			snd_soc_dapm_enable_pin(&codec->dapm,
-						"DRC1 Signal Activity");
+			snd_soc_dapm_enable_pin(dapm, "DRC1 Signal Activity");
 			break;
 		case ARIZONA_GP_FN_DRC2_SIGNAL_DETECT:
-			snd_soc_dapm_enable_pin(&codec->dapm,
-						"DRC2 Signal Activity");
+			snd_soc_dapm_enable_pin(dapm, "DRC2 Signal Activity");
 			break;
 		default:
 			break;
@@ -1474,6 +1474,7 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai,
 				  int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = dai->codec;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
 	struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
 	struct snd_soc_dapm_route routes[2];
@@ -1504,15 +1505,15 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai,
 
 	routes[0].source = arizona_dai_clk_str(dai_priv->clk);
 	routes[1].source = arizona_dai_clk_str(dai_priv->clk);
-	snd_soc_dapm_del_routes(&codec->dapm, routes, ARRAY_SIZE(routes));
+	snd_soc_dapm_del_routes(dapm, routes, ARRAY_SIZE(routes));
 
 	routes[0].source = arizona_dai_clk_str(clk_id);
 	routes[1].source = arizona_dai_clk_str(clk_id);
-	snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes));
+	snd_soc_dapm_add_routes(dapm, routes, ARRAY_SIZE(routes));
 
 	dai_priv->clk = clk_id;
 
-	return snd_soc_dapm_sync(&codec->dapm);
+	return snd_soc_dapm_sync(dapm);
 }
 
 static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate)

sound/soc/codecs/cq93vc.c

@@ -92,7 +92,6 @@ static int cq93vc_set_bias_level(struct snd_soc_codec *codec,
 			     DAVINCI_VC_REG12_POWER_ALL_OFF);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }

sound/soc/codecs/cs4265.c

@@ -503,7 +503,6 @@ static int cs4265_set_bias_level(struct snd_soc_codec *codec,
 			CS4265_PWRCTL_PDN);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/cs42l52.c

@@ -897,7 +897,7 @@ static int cs42l52_set_bias_level(struct snd_soc_codec *codec,
 				    CS42L52_PWRCTL1_PDN_CODEC, 0);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_cache_only(cs42l52->regmap, false);
 			regcache_sync(cs42l52->regmap);
 		}
@@ -908,7 +908,6 @@ static int cs42l52_set_bias_level(struct snd_soc_codec *codec,
 		regcache_cache_only(cs42l52->regmap, true);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -956,7 +955,7 @@ static void cs42l52_beep_work(struct work_struct *work)
 	struct cs42l52_private *cs42l52 =
 		container_of(work, struct cs42l52_private, beep_work);
 	struct snd_soc_codec *codec = cs42l52->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	int i;
 	int val = 0;
 	int best = 0;

sound/soc/codecs/cs42l56.c

@@ -953,7 +953,7 @@ static int cs42l56_set_bias_level(struct snd_soc_codec *codec,
 				    CS42L56_PDN_ALL_MASK, 0);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_cache_only(cs42l56->regmap, false);
 			regcache_sync(cs42l56->regmap);
 			ret = regulator_bulk_enable(ARRAY_SIZE(cs42l56->supplies),
@@ -978,7 +978,6 @@ static int cs42l56_set_bias_level(struct snd_soc_codec *codec,
 						    cs42l56->supplies);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -1026,7 +1025,7 @@ static void cs42l56_beep_work(struct work_struct *work)
 	struct cs42l56_private *cs42l56 =
 		container_of(work, struct cs42l56_private, beep_work);
 	struct snd_soc_codec *codec = cs42l56->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	int i;
 	int val = 0;
 	int best = 0;

sound/soc/codecs/cs42l73.c

@@ -1208,7 +1208,7 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_cache_only(cs42l73->regmap, false);
 			regcache_sync(cs42l73->regmap);
 		}
@@ -1228,7 +1228,6 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, CS42L73_DMMCC, CS42L73_MCLKDIS, 1);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/cs42xx8.c

@@ -380,7 +380,7 @@ EXPORT_SYMBOL_GPL(cs42xx8_regmap_config);
 static int cs42xx8_codec_probe(struct snd_soc_codec *codec)
 {
 	struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	switch (cs42xx8->drvdata->num_adcs) {
 	case 3:

sound/soc/codecs/cx20442.c

@@ -333,7 +333,7 @@ static int cx20442_set_bias_level(struct snd_soc_codec *codec,
 
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
-		if (codec->dapm.bias_level != SND_SOC_BIAS_STANDBY)
+		if (snd_soc_codec_get_bias_level(codec) != SND_SOC_BIAS_STANDBY)
 			break;
 		if (IS_ERR(cx20442->por))
 			err = PTR_ERR(cx20442->por);
@@ -341,7 +341,7 @@ static int cx20442_set_bias_level(struct snd_soc_codec *codec,
 			err = regulator_enable(cx20442->por);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level != SND_SOC_BIAS_PREPARE)
+		if (snd_soc_codec_get_bias_level(codec) != SND_SOC_BIAS_PREPARE)
 			break;
 		if (IS_ERR(cx20442->por))
 			err = PTR_ERR(cx20442->por);
@@ -351,8 +351,6 @@ static int cx20442_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	if (!err)
-		codec->dapm.bias_level = level;
 
 	return err;
 }

sound/soc/codecs/da7213.c

@@ -1374,7 +1374,7 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Enable VMID reference & master bias */
 			snd_soc_update_bits(codec, DA7213_REFERENCES,
 					    DA7213_VMID_EN | DA7213_BIAS_EN,
@@ -1387,7 +1387,6 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec,
 				    DA7213_VMID_EN | DA7213_BIAS_EN, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/da732x.c

@@ -1432,7 +1432,7 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Init Codec */
 			snd_soc_write(codec, DA732X_REG_REF1,
 				      DA732X_VMID_FASTCHG);
@@ -1502,8 +1502,6 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/da9055.c

@@ -1364,7 +1364,7 @@ static int da9055_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Enable VMID reference & master bias */
 			snd_soc_update_bits(codec, DA9055_REFERENCES,
 					    DA9055_VMID_EN | DA9055_BIAS_EN,
@@ -1377,7 +1377,6 @@ static int da9055_set_bias_level(struct snd_soc_codec *codec,
 				    DA9055_VMID_EN | DA9055_BIAS_EN, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/es8328.c

@@ -536,7 +536,7 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			snd_soc_update_bits(codec, ES8328_CONTROL1,
 					ES8328_CONTROL1_VMIDSEL_MASK |
 					ES8328_CONTROL1_ENREF,
@@ -566,7 +566,6 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
 				0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/isabelle.c

@@ -909,8 +909,6 @@ static int isabelle_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/jz4740.c

@@ -258,7 +258,7 @@ 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->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			jz4740_codec_wakeup(regmap);
 
 		mask = JZ4740_CODEC_1_VREF_DISABLE |
@@ -281,8 +281,6 @@ static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/lm4857.c

@@ -89,8 +89,6 @@ static int lm4857_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/lm49453.c

@@ -1271,7 +1271,7 @@ static int lm49453_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			regcache_sync(lm49453->regmap);
 
 		snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG,
@@ -1284,8 +1284,6 @@ static int lm49453_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/max98088.c

@@ -1571,7 +1571,7 @@ static int max98088_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			regcache_sync(max98088->regmap);
 
 		snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN,
@@ -1584,7 +1584,6 @@ static int max98088_set_bias_level(struct snd_soc_codec *codec,
 		regcache_mark_dirty(max98088->regmap);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/max98090.c

@@ -1500,7 +1500,7 @@ static const struct snd_soc_dapm_route max98091_dapm_routes[] = {
 static int max98090_add_widgets(struct snd_soc_codec *codec)
 {
 	struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	snd_soc_add_codec_controls(codec, max98090_snd_controls,
 		ARRAY_SIZE(max98090_snd_controls));
@@ -1798,16 +1798,17 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
 		 * away from ON. Disable the clock in that case, otherwise
 		 * enable it.
 		 */
-		if (!IS_ERR(max98090->mclk)) {
-			if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
-				clk_disable_unprepare(max98090->mclk);
-			else
-				clk_prepare_enable(max98090->mclk);
-		}
+		if (IS_ERR(max98090->mclk))
+			break;
+
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON)
+			clk_disable_unprepare(max98090->mclk);
+		else
+			clk_prepare_enable(max98090->mclk);
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regcache_sync(max98090->regmap);
 			if (ret != 0) {
 				dev_err(codec->dev,
@@ -1824,7 +1825,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
 		regcache_mark_dirty(max98090->regmap);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -2187,7 +2187,6 @@ static void max98090_jack_work(struct work_struct *work)
 		struct max98090_priv,
 		jack_work.work);
 	struct snd_soc_codec *codec = max98090->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int status = 0;
 	int reg;
 
@@ -2266,8 +2265,6 @@ static void max98090_jack_work(struct work_struct *work)
 
 	snd_soc_jack_report(max98090->jack, status,
 			    SND_JACK_HEADSET | SND_JACK_BTN_0);
-
-	snd_soc_dapm_sync(dapm);
 }
 
 static irqreturn_t max98090_interrupt(int irq, void *data)

sound/soc/codecs/max98095.c

@@ -1650,16 +1650,17 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec,
 		 * away from ON. Disable the clock in that case, otherwise
 		 * enable it.
 		 */
-		if (!IS_ERR(max98095->mclk)) {
-			if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
-				clk_disable_unprepare(max98095->mclk);
-			else
-				clk_prepare_enable(max98095->mclk);
-		}
+		if (IS_ERR(max98095->mclk))
+			break;
+
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON)
+			clk_disable_unprepare(max98095->mclk);
+		else
+			clk_prepare_enable(max98095->mclk);
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regcache_sync(max98095->regmap);
 
 			if (ret != 0) {
@@ -1678,7 +1679,6 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec,
 		regcache_mark_dirty(max98095->regmap);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -2198,7 +2198,7 @@ static int max98095_suspend(struct snd_soc_codec *codec)
 	if (max98095->headphone_jack || max98095->mic_jack)
 		max98095_jack_detect_disable(codec);
 
-	max98095_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
 }
@@ -2208,7 +2208,7 @@ static int max98095_resume(struct snd_soc_codec *codec)
 	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
 	struct i2c_client *client = to_i2c_client(codec->dev);
 
-	max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	if (max98095->headphone_jack || max98095->mic_jack) {
 		max98095_jack_detect_enable(codec);

sound/soc/codecs/max9850.c

@@ -252,7 +252,7 @@ static int max9850_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regcache_sync(max9850->regmap);
 			if (ret) {
 				dev_err(codec->dev,
@@ -264,7 +264,6 @@ static int max9850_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_OFF:
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/ml26124.c

@@ -523,7 +523,7 @@ static int ml26124_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		/* VMID ON */
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
 					    ML26124_VMID, ML26124_VMID);
 			msleep(500);
@@ -536,7 +536,6 @@ static int ml26124_set_bias_level(struct snd_soc_codec *codec,
 				    ML26124_VMID, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/pcm512x.c

@@ -242,7 +242,7 @@ static int pcm512x_overclock_pll_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
 
-	switch (codec->dapm.bias_level) {
+	switch (snd_soc_codec_get_bias_level(codec)) {
 	case SND_SOC_BIAS_OFF:
 	case SND_SOC_BIAS_STANDBY:
 		break;
@@ -270,7 +270,7 @@ static int pcm512x_overclock_dsp_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
 
-	switch (codec->dapm.bias_level) {
+	switch (snd_soc_codec_get_bias_level(codec)) {
 	case SND_SOC_BIAS_OFF:
 	case SND_SOC_BIAS_STANDBY:
 		break;
@@ -298,7 +298,7 @@ static int pcm512x_overclock_dac_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
 
-	switch (codec->dapm.bias_level) {
+	switch (snd_soc_codec_get_bias_level(codec)) {
 	case SND_SOC_BIAS_OFF:
 	case SND_SOC_BIAS_STANDBY:
 		break;
@@ -641,8 +641,6 @@ static int pcm512x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/rt286.c

@@ -301,6 +301,7 @@ static int rt286_support_power_controls[] = {
 
 static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
 {
+	struct snd_soc_dapm_context *dapm;
 	unsigned int val, buf;
 
 	*hp = false;
@@ -308,6 +309,9 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
 
 	if (!rt286->codec)
 		return -EINVAL;
+
+	dapm = snd_soc_codec_get_dapm(rt286->codec);
+
 	if (rt286->pdata.cbj_en) {
 		regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf);
 		*hp = buf & 0x80000000;
@@ -316,14 +320,11 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
 			regmap_update_bits(rt286->regmap,
 				RT286_DC_GAIN, 0x200, 0x200);
 
-			snd_soc_dapm_force_enable_pin(&rt286->codec->dapm,
-							"HV");
-			snd_soc_dapm_force_enable_pin(&rt286->codec->dapm,
-							"VREF");
+			snd_soc_dapm_force_enable_pin(dapm, "HV");
+			snd_soc_dapm_force_enable_pin(dapm, "VREF");
 			/* power LDO1 */
-			snd_soc_dapm_force_enable_pin(&rt286->codec->dapm,
-							"LDO1");
-			snd_soc_dapm_sync(&rt286->codec->dapm);
+			snd_soc_dapm_force_enable_pin(dapm, "LDO1");
+			snd_soc_dapm_sync(dapm);
 
 			regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24);
 			msleep(50);
@@ -360,11 +361,11 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
 		*mic = buf & 0x80000000;
 	}
 
-	snd_soc_dapm_disable_pin(&rt286->codec->dapm, "HV");
-	snd_soc_dapm_disable_pin(&rt286->codec->dapm, "VREF");
+	snd_soc_dapm_disable_pin(dapm, "HV");
+	snd_soc_dapm_disable_pin(dapm, "VREF");
 	if (!*hp)
-		snd_soc_dapm_disable_pin(&rt286->codec->dapm, "LDO1");
-	snd_soc_dapm_sync(&rt286->codec->dapm);
+		snd_soc_dapm_disable_pin(dapm, "LDO1");
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
@@ -391,6 +392,7 @@ static void rt286_jack_detect_work(struct work_struct *work)
 
 int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
 
 	rt286->jack = jack;
@@ -398,7 +400,7 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
 	if (jack) {
 		/* enable IRQ */
 		if (rt286->jack->status & SND_JACK_HEADPHONE)
-			snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO1");
+			snd_soc_dapm_force_enable_pin(dapm, "LDO1");
 		regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2);
 		/* Send an initial empty report */
 		snd_soc_jack_report(rt286->jack, rt286->jack->status,
@@ -406,9 +408,9 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
 	} else {
 		/* disable IRQ */
 		regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x0);
-		snd_soc_dapm_disable_pin(&codec->dapm, "LDO1");
+		snd_soc_dapm_disable_pin(dapm, "LDO1");
 	}
-	snd_soc_dapm_sync(&codec->dapm);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
@@ -985,7 +987,7 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec,
 {
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
-		if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) {
+		if (SND_SOC_BIAS_STANDBY == snd_soc_codec_get_bias_level(codec)) {
 			snd_soc_write(codec,
 				RT286_SET_AUDIO_POWER, AC_PWRST_D0);
 			snd_soc_update_bits(codec,
@@ -1012,7 +1014,6 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }

sound/soc/codecs/rt5631.c

@@ -1546,7 +1546,7 @@ static int rt5631_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3,
 				RT5631_PWR_VREF | RT5631_PWR_MAIN_BIAS,
 				RT5631_PWR_VREF | RT5631_PWR_MAIN_BIAS);
@@ -1569,7 +1569,6 @@ static int rt5631_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -1615,7 +1614,7 @@ static int rt5631_probe(struct snd_soc_codec *codec)
 			RT5631_DMIC_R_CH_LATCH_RISING);
 	}
 
-	codec->dapm.bias_level = SND_SOC_BIAS_STANDBY;
+	snd_soc_codec_init_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
 }

sound/soc/codecs/rt5640.c

@@ -1870,7 +1870,7 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
 {
 	switch (level) {
 	case SND_SOC_BIAS_STANDBY:
-		if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) {
+		if (SND_SOC_BIAS_OFF == snd_soc_codec_get_bias_level(codec)) {
 			snd_soc_update_bits(codec, RT5640_PWR_ANLG1,
 				RT5640_PWR_VREF1 | RT5640_PWR_MB |
 				RT5640_PWR_BG | RT5640_PWR_VREF2,
@@ -1902,7 +1902,6 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -1935,11 +1934,12 @@ EXPORT_SYMBOL_GPL(rt5640_dmic_enable);
 
 static int rt5640_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
 
 	rt5640->codec = codec;
 
-	rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301);
 	snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030);
@@ -1951,18 +1951,18 @@ static int rt5640_probe(struct snd_soc_codec *codec)
 		snd_soc_add_codec_controls(codec,
 			rt5640_specific_snd_controls,
 			ARRAY_SIZE(rt5640_specific_snd_controls));
-		snd_soc_dapm_new_controls(&codec->dapm,
+		snd_soc_dapm_new_controls(dapm,
 			rt5640_specific_dapm_widgets,
 			ARRAY_SIZE(rt5640_specific_dapm_widgets));
-		snd_soc_dapm_add_routes(&codec->dapm,
+		snd_soc_dapm_add_routes(dapm,
 			rt5640_specific_dapm_routes,
 			ARRAY_SIZE(rt5640_specific_dapm_routes));
 		break;
 	case RT5640_ID_5639:
-		snd_soc_dapm_new_controls(&codec->dapm,
+		snd_soc_dapm_new_controls(dapm,
 			rt5639_specific_dapm_widgets,
 			ARRAY_SIZE(rt5639_specific_dapm_widgets));
-		snd_soc_dapm_add_routes(&codec->dapm,
+		snd_soc_dapm_add_routes(dapm,
 			rt5639_specific_dapm_routes,
 			ARRAY_SIZE(rt5639_specific_dapm_routes));
 		break;
@@ -1991,7 +1991,7 @@ static int rt5640_suspend(struct snd_soc_codec *codec)
 {
 	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
 
-	rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 	rt5640_reset(codec);
 	regcache_cache_only(rt5640->regmap, true);
 	regcache_mark_dirty(rt5640->regmap);

sound/soc/codecs/rt5645.c

@@ -2410,7 +2410,6 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -2521,7 +2520,7 @@ static int rt5645_probe(struct snd_soc_codec *codec)
 		break;
 	}
 
-	rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200);
 

sound/soc/codecs/rt5651.c

@@ -1571,7 +1571,7 @@ static int rt5651_set_bias_level(struct snd_soc_codec *codec,
 {
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
-		if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) {
+		if (SND_SOC_BIAS_STANDBY == snd_soc_codec_get_bias_level(codec)) {
 			snd_soc_update_bits(codec, RT5651_PWR_ANLG1,
 				RT5651_PWR_VREF1 | RT5651_PWR_MB |
 				RT5651_PWR_BG | RT5651_PWR_VREF2,
@@ -1604,7 +1604,6 @@ static int rt5651_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -1625,7 +1624,7 @@ static int rt5651_probe(struct snd_soc_codec *codec)
 		RT5651_PWR_FV1 | RT5651_PWR_FV2,
 		RT5651_PWR_FV1 | RT5651_PWR_FV2);
 
-	rt5651_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
 }

sound/soc/codecs/rt5670.c

@@ -416,12 +416,12 @@ static bool rt5670_readable_register(struct device *dev, unsigned int reg)
 static int rt5670_headset_detect(struct snd_soc_codec *codec, int jack_insert)
 {
 	int val;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
 
 	if (jack_insert) {
-		snd_soc_dapm_force_enable_pin(&codec->dapm,
-						       "Mic Det Power");
-		snd_soc_dapm_sync(&codec->dapm);
+		snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power");
+		snd_soc_dapm_sync(dapm);
 		snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x0);
 		snd_soc_update_bits(codec, RT5670_CJ_CTRL2,
 			RT5670_CBJ_DET_MODE | RT5670_CBJ_MN_JD,
@@ -447,15 +447,15 @@ static int rt5670_headset_detect(struct snd_soc_codec *codec, int jack_insert)
 		} else {
 			snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4);
 			rt5670->jack_type = SND_JACK_HEADPHONE;
-			snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power");
-			snd_soc_dapm_sync(&codec->dapm);
+			snd_soc_dapm_disable_pin(dapm, "Mic Det Power");
+			snd_soc_dapm_sync(dapm);
 		}
 	} else {
 		snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x0);
 		snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4);
 		rt5670->jack_type = 0;
-		snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power");
-		snd_soc_dapm_sync(&codec->dapm);
+		snd_soc_dapm_disable_pin(dapm, "Mic Det Power");
+		snd_soc_dapm_sync(dapm);
 	}
 
 	return rt5670->jack_type;
@@ -2603,7 +2603,7 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec,
 
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
-		if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) {
+		if (SND_SOC_BIAS_STANDBY == snd_soc_codec_get_bias_level(codec)) {
 			snd_soc_update_bits(codec, RT5670_PWR_ANLG1,
 				RT5670_PWR_VREF1 | RT5670_PWR_MB |
 				RT5670_PWR_BG | RT5670_PWR_VREF2,
@@ -2647,30 +2647,30 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
 
 static int rt5670_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
 
 	switch (snd_soc_read(codec, RT5670_RESET) & RT5670_ID_MASK) {
 	case RT5670_ID_5670:
 	case RT5670_ID_5671:
-		snd_soc_dapm_new_controls(&codec->dapm,
+		snd_soc_dapm_new_controls(dapm,
 			rt5670_specific_dapm_widgets,
 			ARRAY_SIZE(rt5670_specific_dapm_widgets));
-		snd_soc_dapm_add_routes(&codec->dapm,
+		snd_soc_dapm_add_routes(dapm,
 			rt5670_specific_dapm_routes,
 			ARRAY_SIZE(rt5670_specific_dapm_routes));
 		break;
 	case RT5670_ID_5672:
-		snd_soc_dapm_new_controls(&codec->dapm,
+		snd_soc_dapm_new_controls(dapm,
 			rt5672_specific_dapm_widgets,
 			ARRAY_SIZE(rt5672_specific_dapm_widgets));
-		snd_soc_dapm_add_routes(&codec->dapm,
+		snd_soc_dapm_add_routes(dapm,
 			rt5672_specific_dapm_routes,
 			ARRAY_SIZE(rt5672_specific_dapm_routes));
 		break;

sound/soc/codecs/rt5677.c

@@ -820,7 +820,7 @@ static int rt5677_dsp_vad_put(struct snd_kcontrol *kcontrol,
 
 	rt5677->dsp_vad_en = !!ucontrol->value.integer.value[0];
 
-	if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+	if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 		rt5677_set_dsp_vad(codec, rt5677->dsp_vad_en);
 
 	return 0;
@@ -2479,7 +2479,7 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
-		if (codec->dapm.bias_level != SND_SOC_BIAS_ON &&
+		if (snd_soc_codec_get_bias_level(codec) != SND_SOC_BIAS_ON &&
 			!rt5677->is_vref_slow) {
 			mdelay(20);
 			regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1,
@@ -4353,7 +4353,7 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) {
 			rt5677_set_dsp_vad(codec, false);
 
 			regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1,
@@ -4395,7 +4395,6 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -4606,22 +4605,23 @@ static void rt5677_free_gpio(struct i2c_client *i2c)
 
 static int rt5677_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
 	int i;
 
 	rt5677->codec = codec;
 
 	if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) {
-		snd_soc_dapm_add_routes(&codec->dapm,
+		snd_soc_dapm_add_routes(dapm,
 			rt5677_dmic2_clk_2,
 			ARRAY_SIZE(rt5677_dmic2_clk_2));
 	} else { /*use dmic1 clock by default*/
-		snd_soc_dapm_add_routes(&codec->dapm,
+		snd_soc_dapm_add_routes(dapm,
 			rt5677_dmic2_clk_1,
 			ARRAY_SIZE(rt5677_dmic2_clk_1));
 	}
 
-	rt5677_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	regmap_write(rt5677->regmap, RT5677_DIG_MISC, 0x0020);
 	regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x0c00);

sound/soc/codecs/sgtl5000.c

@@ -948,7 +948,7 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(
 						ARRAY_SIZE(sgtl5000->supplies),
 						sgtl5000->supplies);
@@ -979,7 +979,6 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/sirf-audio-codec.c

@@ -395,7 +395,7 @@ struct snd_soc_dai_driver sirf_audio_codec_dai = {
 
 static int sirf_audio_codec_probe(struct snd_soc_codec *codec)
 {
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	pm_runtime_enable(codec->dev);
 

sound/soc/codecs/sn95031.c

@@ -194,7 +194,7 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) {
 			pr_debug("vaud_bias powering up pll\n");
 			/* power up the pll */
 			snd_soc_write(codec, SN95031_AUDPLLCTRL, BIT(5));
@@ -205,17 +205,22 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		switch (snd_soc_codec_get_bias_level(codec)) {
+		case SND_SOC_BIAS_OFF:
 			pr_debug("vaud_bias power up rail\n");
 			/* power up the rail */
 			snd_soc_write(codec, SN95031_VAUD,
 					BIT(2)|BIT(1)|BIT(0));
 			msleep(1);
-		} else if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) {
+			break;
+		case SND_SOC_BIAS_PREPARE:
 			/* turn off pcm */
 			pr_debug("vaud_bias power dn pcm\n");
 			snd_soc_update_bits(codec, SN95031_PCM2C2, BIT(0), 0);
 			snd_soc_write(codec, SN95031_AUDPLLCTRL, 0);
+			break;
+		default:
+			break;
 		}
 		break;
 
@@ -226,7 +231,6 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/ssm2518.c

@@ -518,12 +518,7 @@ static int ssm2518_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	if (ret)
-		return ret;
-
-	codec->dapm.bias_level = level;
-
-	return 0;
+	return ret;
 }
 
 static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,

sound/soc/codecs/ssm2602.c

@@ -473,7 +473,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/ssm4567.c

@@ -361,12 +361,7 @@ static int ssm4567_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	if (ret)
-		return ret;
-
-	codec->dapm.bias_level = level;
-
-	return 0;
+	return ret;
 }
 
 static const struct snd_soc_dai_ops ssm4567_dai_ops = {

sound/soc/codecs/sta32x.c

@@ -819,7 +819,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies),
 						    sta32x->supplies);
 			if (ret != 0) {
@@ -854,7 +854,6 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec,
 				       sta32x->supplies);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -970,7 +969,7 @@ static int sta32x_probe(struct snd_soc_codec *codec)
 	if (sta32x->pdata->needs_esd_watchdog)
 		INIT_DELAYED_WORK(&sta32x->watchdog_work, sta32x_watchdog);
 
-	sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	/* Bias level configuration will have done an extra enable */
 	regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
 

sound/soc/codecs/sta350.c

@@ -853,7 +853,7 @@ static int sta350_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(
 				ARRAY_SIZE(sta350->supplies),
 				sta350->supplies);
@@ -890,7 +890,6 @@ static int sta350_set_bias_level(struct snd_soc_codec *codec,
 				       sta350->supplies);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -1037,7 +1036,7 @@ static int sta350_probe(struct snd_soc_codec *codec)
 	sta350->coef_shadow[60] = 0x400000;
 	sta350->coef_shadow[61] = 0x400000;
 
-	sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	/* Bias level configuration will have done an extra enable */
 	regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies);
 

sound/soc/codecs/sta529.c

@@ -165,7 +165,7 @@ static int sta529_set_bias_level(struct snd_soc_codec *codec, enum
 				FFX_CLK_ENB);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			regcache_sync(sta529->regmap);
 		snd_soc_update_bits(codec, STA529_FFXCFG0,
 					POWER_CNTLMSAK, POWER_STDBY);
@@ -179,12 +179,6 @@ static int sta529_set_bias_level(struct snd_soc_codec *codec, enum
 		break;
 	}
 
-	/*
-	 * store the label for powers down audio subsystem for suspend.This is
-	 * used by soc core layer
-	 */
-	codec->dapm.bias_level = level;
-
 	return 0;
 
 }

sound/soc/codecs/stac9766.c

@@ -236,7 +236,6 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec,
 		stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/tlv320aic23.c

@@ -506,7 +506,6 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/tlv320aic31xx.c

@@ -646,7 +646,7 @@ static int aic31xx_add_controls(struct snd_soc_codec *codec)
 
 static int aic31xx_add_widgets(struct snd_soc_codec *codec)
 {
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
 
@@ -1027,17 +1027,17 @@ static int aic31xx_set_bias_level(struct snd_soc_codec *codec,
 				  enum snd_soc_bias_level level)
 {
 	dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__,
-		codec->dapm.bias_level, level);
+		snd_soc_codec_get_bias_level(codec), level);
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 		break;
 	case SND_SOC_BIAS_PREPARE:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY)
 			aic31xx_clk_on(codec);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		switch (codec->dapm.bias_level) {
+		switch (snd_soc_codec_get_bias_level(codec)) {
 		case SND_SOC_BIAS_OFF:
 			aic31xx_power_on(codec);
 			break;
@@ -1049,11 +1049,10 @@ static int aic31xx_set_bias_level(struct snd_soc_codec *codec,
 		}
 		break;
 	case SND_SOC_BIAS_OFF:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY)
 			aic31xx_power_off(codec);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }

sound/soc/codecs/tlv320aic32x4.c

@@ -564,7 +564,6 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_OFF:
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/tlv320aic3x.c

@@ -147,6 +147,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 					struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
 	unsigned int reg = mc->reg;
@@ -179,7 +180,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 		update.mask = mask;
 		update.val = val;
 
-		snd_soc_dapm_mixer_update_power(&codec->dapm, kcontrol, connect,
+		snd_soc_dapm_mixer_update_power(dapm, kcontrol, connect,
 			&update);
 	}
 
@@ -979,7 +980,7 @@ 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;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	switch (aic3x->model) {
 	case AIC3X_MODEL_3X:
@@ -1384,7 +1385,7 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_ON:
 		break;
 	case SND_SOC_BIAS_PREPARE:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY &&
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY &&
 		    aic3x->master) {
 			/* enable pll */
 			snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG,
@@ -1394,7 +1395,7 @@ 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->dapm.bias_level == SND_SOC_BIAS_PREPARE &&
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_PREPARE &&
 		    aic3x->master) {
 			/* disable pll */
 			snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG,
@@ -1406,7 +1407,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
 			aic3x_set_power(codec, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }

sound/soc/codecs/tlv320dac33.c

@@ -633,7 +633,7 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Coming from OFF, switch on the codec */
 			ret = dac33_hard_power(codec, 1);
 			if (ret != 0)
@@ -644,14 +644,13 @@ 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->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			return 0;
 		ret = dac33_hard_power(codec, 0);
 		if (ret != 0)
 			return ret;
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }

sound/soc/codecs/twl4030.c

@@ -1588,14 +1588,13 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			twl4030_codec_enable(codec, 1);
 		break;
 	case SND_SOC_BIAS_OFF:
 		twl4030_codec_enable(codec, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }

sound/soc/codecs/twl6040.c

@@ -533,7 +533,7 @@ static int twl6040_pll_put_enum(struct snd_kcontrol *kcontrol,
 
 int twl6040_get_dl1_gain(struct snd_soc_codec *codec)
 {
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	if (snd_soc_dapm_get_pin_status(dapm, "EP"))
 		return -1; /* -1dB */
@@ -853,8 +853,6 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
@@ -1130,7 +1128,7 @@ static int twl6040_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	twl6040_init_chip(codec);
 
 	return 0;

sound/soc/codecs/uda134x.c

@@ -350,7 +350,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec,
 			pd->power(0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -478,6 +477,7 @@ static struct snd_soc_dai_driver uda134x_dai = {
 
 static int uda134x_soc_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct uda134x_priv *uda134x;
 	struct uda134x_platform_data *pd = codec->component.card->dev->platform_data;
 	const struct snd_soc_dapm_widget *widgets;
@@ -526,7 +526,7 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec)
 		num_widgets = ARRAY_SIZE(uda1340_dapm_widgets);
 	}
 
-	ret = snd_soc_dapm_new_controls(&codec->dapm, widgets, num_widgets);
+	ret = snd_soc_dapm_new_controls(dapm, widgets, num_widgets);
 	if (ret) {
 		printk(KERN_ERR "%s failed to register dapm controls: %d",
 			__func__, ret);

sound/soc/codecs/uda1380.c

@@ -590,9 +590,6 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec,
 	int reg;
 	struct uda1380_platform_data *pdata = codec->dev->platform_data;
 
-	if (codec->dapm.bias_level == level)
-		return 0;
-
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 	case SND_SOC_BIAS_PREPARE:
@@ -600,7 +597,7 @@ 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->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			if (gpio_is_valid(pdata->gpio_power)) {
 				gpio_set_value(pdata->gpio_power, 1);
 				mdelay(1);
@@ -623,7 +620,6 @@ 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->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm0010.c

@@ -751,13 +751,13 @@ static int wm0010_set_bias_level(struct snd_soc_codec *codec,
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_PREPARE)
 			wm0010_boot(codec);
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_PREPARE) {
 			mutex_lock(&wm0010->lock);
 			wm0010_halt(codec);
 			mutex_unlock(&wm0010->lock);
@@ -767,8 +767,6 @@ static int wm0010_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/wm1250-ev1.c

@@ -61,8 +61,6 @@ static int wm1250_ev1_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/wm5100.c

@@ -2101,7 +2101,7 @@ static void wm5100_micd_irq(struct wm5100_priv *wm5100)
 int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
 {
 	struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	if (jack) {
 		wm5100->jack = jack;
@@ -2336,6 +2336,7 @@ static void wm5100_free_gpio(struct i2c_client *i2c)
 
 static int wm5100_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct i2c_client *i2c = to_i2c_client(codec->dev);
 	struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec);
 	int ret, i;
@@ -2353,8 +2354,7 @@ static int wm5100_probe(struct snd_soc_codec *codec)
 	/* TODO: check if we're symmetric */
 
 	if (i2c->irq)
-		snd_soc_dapm_new_controls(&codec->dapm,
-					  wm5100_dapm_widgets_noirq,
+		snd_soc_dapm_new_controls(dapm, wm5100_dapm_widgets_noirq,
 					  ARRAY_SIZE(wm5100_dapm_widgets_noirq));
 
 	if (wm5100->pdata.hp_pol) {

sound/soc/codecs/wm5102.c

@@ -1827,6 +1827,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = {
 
 static int wm5102_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
@@ -1837,9 +1838,9 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec)
 	arizona_init_spk(codec);
 	arizona_init_gpio(codec);
 
-	snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
+	snd_soc_dapm_disable_pin(dapm, "HAPTICS");
 
-	priv->core.arizona->dapm = &codec->dapm;
+	priv->core.arizona->dapm = dapm;
 
 	return 0;
 }

sound/soc/codecs/wm5110.c

@@ -1598,10 +1598,11 @@ static struct snd_soc_dai_driver wm5110_dai[] = {
 
 static int wm5110_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	priv->core.arizona->dapm = &codec->dapm;
+	priv->core.arizona->dapm = dapm;
 
 	arizona_init_spk(codec);
 	arizona_init_gpio(codec);
@@ -1611,9 +1612,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
 	if (ret != 0)
 		return ret;
 
-	snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
-
-	priv->core.arizona->dapm = &codec->dapm;
+	snd_soc_dapm_disable_pin(dapm, "HAPTICS");
 
 	return 0;
 }

sound/soc/codecs/wm8350.c

@@ -1102,7 +1102,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies),
 						    priv->supplies);
 			if (ret != 0)
@@ -1235,7 +1235,6 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec,
 				       priv->supplies);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8400.c

@@ -1145,7 +1145,7 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(power),
 						    &power[0]);
 			if (ret != 0) {
@@ -1232,7 +1232,6 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8510.c

@@ -519,7 +519,7 @@ 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->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_sync(wm8510->regmap);
 
 			/* Initial cap charge at VMID 5k */
@@ -538,7 +538,6 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8523.c

@@ -308,7 +308,7 @@ static int wm8523_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies),
 						    wm8523->supplies);
 			if (ret != 0) {
@@ -344,7 +344,6 @@ static int wm8523_set_bias_level(struct snd_soc_codec *codec,
 				       wm8523->supplies);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8580.c

@@ -795,7 +795,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Power up and get individual control of the DACs */
 			snd_soc_update_bits(codec, WM8580_PWRDN1,
 					    WM8580_PWRDN1_PWDN |
@@ -812,7 +812,6 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 				    WM8580_PWRDN1_PWDN, WM8580_PWRDN1_PWDN);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8711.c

@@ -310,7 +310,7 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			regcache_sync(wm8711->regmap);
 
 		snd_soc_write(codec, WM8711_PWR, reg | 0x0040);
@@ -320,7 +320,6 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, WM8711_PWR, 0xffff);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8728.c

@@ -170,7 +170,7 @@ 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->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Power everything up... */
 			reg = snd_soc_read(codec, WM8728_DACCTL);
 			snd_soc_write(codec, WM8728_DACCTL, reg & ~0x4);
@@ -185,7 +185,6 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, WM8728_DACCTL, reg | 0x4);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8731.c

@@ -387,6 +387,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
 
 	switch (clk_id) {
@@ -421,7 +422,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 
 	wm8731->sysclk = freq;
 
-	snd_soc_dapm_sync(&codec->dapm);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
@@ -501,7 +502,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies),
 						    wm8731->supplies);
 			if (ret != 0)
@@ -523,7 +524,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 		regcache_mark_dirty(wm8731->regmap);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -599,7 +599,7 @@ static int wm8731_probe(struct snd_soc_codec *codec)
 		goto err_regulator_enable;
 	}
 
-	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Latch the update bits */
 	snd_soc_update_bits(codec, WM8731_LOUT1V, 0x100, 0);

sound/soc/codecs/wm8737.c

@@ -469,7 +469,7 @@ static int wm8737_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8737->supplies),
 						    wm8737->supplies);
 			if (ret != 0) {
@@ -512,7 +512,6 @@ static int wm8737_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -562,7 +561,7 @@ static int wm8737_probe(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, WM8737_RIGHT_PGA_VOLUME, WM8737_RVU,
 			    WM8737_RVU);
 
-	wm8737_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Bias level configuration will have done an extra enable */
 	regulator_bulk_disable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies);

sound/soc/codecs/wm8750.c

@@ -634,7 +634,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			snd_soc_cache_sync(codec);
 
 			/* Set VMID to 5k */
@@ -651,7 +651,6 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, WM8750_PWR1, 0x0001);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8753.c

@@ -1352,7 +1352,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
 		flush_delayed_work(&wm8753->charge_work);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* set vmid to 5k for quick power up */
 			snd_soc_write(codec, WM8753_PWR1, pwr_reg | 0x01c1);
 			schedule_delayed_work(&wm8753->charge_work,
@@ -1367,7 +1367,6 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, WM8753_PWR1, 0x0001);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8770.c

@@ -510,7 +510,7 @@ static int wm8770_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8770->supplies),
 						    wm8770->supplies);
 			if (ret) {
@@ -534,7 +534,6 @@ static int wm8770_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8776.c

@@ -344,7 +344,7 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_sync(wm8776->regmap);
 
 			/* Disable the global powerdown; DAPM does the rest */
@@ -357,7 +357,6 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8804.c

@@ -162,7 +162,7 @@ static int txsrc_put(struct snd_kcontrol *kcontrol,
 		     struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int val = ucontrol->value.enumerated.item[0] << e->shift_l;
 	unsigned int mask = 1 << e->shift_l;

sound/soc/codecs/wm8900.c

@@ -1049,7 +1049,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_STANDBY:
 		/* Charge capacitors if initial power up */
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* STARTUP_BIAS_ENA on */
 			snd_soc_write(codec, WM8900_REG_POWER1,
 				     WM8900_REG_POWER1_STARTUP_BIAS_ENA);
@@ -1117,7 +1117,6 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec,
 			     WM8900_REG_POWER2_SYSCLK_ENA);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -1138,7 +1137,7 @@ static int wm8900_suspend(struct snd_soc_codec *codec)
 	wm8900->fll_out = fll_out;
 	wm8900->fll_in = fll_in;
 
-	wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
 }
@@ -1156,7 +1155,7 @@ static int wm8900_resume(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Restart the FLL? */
 	if (wm8900->fll_out) {
@@ -1189,7 +1188,7 @@ static int wm8900_probe(struct snd_soc_codec *codec)
 	wm8900_reset(codec);
 
 	/* Turn the chip on */
-	wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Latch the volume update bits */
 	snd_soc_update_bits(codec, WM8900_REG_LINVOL, 0x100, 0x100);

sound/soc/codecs/wm8903.c

@@ -1105,7 +1105,7 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
 					    WM8903_POBCTRL | WM8903_ISEL_MASK |
 					    WM8903_STARTUP_BIAS_ENA |
@@ -1200,8 +1200,6 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/wm8904.c

@@ -1168,7 +1168,7 @@ 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;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	snd_soc_dapm_new_controls(dapm, wm8904_core_dapm_widgets,
 				  ARRAY_SIZE(wm8904_core_dapm_widgets));
@@ -1852,7 +1852,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies),
 						    wm8904->supplies);
 			if (ret != 0) {
@@ -1907,7 +1907,6 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
 		clk_disable_unprepare(wm8904->mclk);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8940.c

@@ -492,7 +492,7 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec,
 		ret = snd_soc_write(codec, WM8940_POWER1, pwr_reg | 0x1);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regcache_sync(wm8940->regmap);
 			if (ret < 0) {
 				dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
@@ -510,8 +510,6 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return ret;
 }
 
@@ -707,7 +705,7 @@ static int wm8940_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	ret = snd_soc_write(codec, WM8940_POWER1, 0x180);
 	if (ret < 0)

sound/soc/codecs/wm8955.c

@@ -785,7 +785,7 @@ static int wm8955_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8955->supplies),
 						    wm8955->supplies);
 			if (ret != 0) {
@@ -838,7 +838,6 @@ static int wm8955_set_bias_level(struct snd_soc_codec *codec,
 				       wm8955->supplies);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -929,7 +928,7 @@ static int wm8955_probe(struct snd_soc_codec *codec)
 					    WM8955_DMEN, WM8955_DMEN);
 	}
 
-	wm8955_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Bias level configuration will have done an extra enable */
 	regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), wm8955->supplies);

sound/soc/codecs/wm8960.c

@@ -445,7 +445,7 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec)
 {
 	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
 	struct wm8960_data *pdata = &wm8960->pdata;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct snd_soc_dapm_widget *w;
 
 	snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets,
@@ -476,7 +476,7 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec)
 	 * and save the result.
 	 */
 	list_for_each_entry(w, &codec->component.card->widgets, list) {
-		if (w->dapm != &codec->dapm)
+		if (w->dapm != dapm)
 			continue;
 		if (strcmp(w->name, "LOUT1 PGA") == 0)
 			wm8960->lout1 = w;
@@ -627,7 +627,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
-		switch (codec->dapm.bias_level) {
+		switch (snd_soc_codec_get_bias_level(codec)) {
 		case SND_SOC_BIAS_STANDBY:
 			if (!IS_ERR(wm8960->mclk)) {
 				ret = clk_prepare_enable(wm8960->mclk);
@@ -655,7 +655,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_sync(wm8960->regmap);
 
 			/* Enable anti-pop features */
@@ -691,8 +691,6 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
@@ -707,7 +705,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
-		switch (codec->dapm.bias_level) {
+		switch (snd_soc_codec_get_bias_level(codec)) {
 		case SND_SOC_BIAS_STANDBY:
 			/* Enable anti pop mode */
 			snd_soc_update_bits(codec, WM8960_APOP1,
@@ -778,7 +776,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		switch (codec->dapm.bias_level) {
+		switch (snd_soc_codec_get_bias_level(codec)) {
 		case SND_SOC_BIAS_PREPARE:
 			/* Disable HP discharge */
 			snd_soc_update_bits(codec, WM8960_APOP2,
@@ -802,8 +800,6 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/wm8961.c

@@ -758,7 +758,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) {
 			/* Enable bias generation */
 			reg = snd_soc_read(codec, WM8961_ANTI_POP);
 			reg |= WM8961_BUFIOEN | WM8961_BUFDCOPEN;
@@ -773,7 +773,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_PREPARE) {
 			/* VREF off */
 			reg = snd_soc_read(codec, WM8961_PWR_MGMT_1);
 			reg &= ~WM8961_VREF;
@@ -795,8 +795,6 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/wm8962.c

@@ -2361,7 +2361,7 @@ static int wm8962_add_widgets(struct snd_soc_codec *codec)
 {
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
 	struct wm8962_pdata *pdata = &wm8962->pdata;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	snd_soc_add_codec_controls(codec, wm8962_snd_controls,
 			     ARRAY_SIZE(wm8962_snd_controls));
@@ -2446,13 +2446,13 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec)
 	 * So we here provisionally enable it and then disable it afterward
 	 * if current bias_level hasn't reached SND_SOC_BIAS_ON.
 	 */
-	if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
+	if (snd_soc_codec_get_bias_level(codec) != SND_SOC_BIAS_ON)
 		snd_soc_update_bits(codec, WM8962_CLOCKING2,
 				WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA);
 
 	dspclk = snd_soc_read(codec, WM8962_CLOCKING1);
 
-	if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
+	if (snd_soc_codec_get_bias_level(codec) != SND_SOC_BIAS_ON)
 		snd_soc_update_bits(codec, WM8962_CLOCKING2,
 				WM8962_SYSCLK_ENA_MASK, 0);
 
@@ -2510,9 +2510,6 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec)
 static int wm8962_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	if (level == codec->dapm.bias_level)
-		return 0;
-
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 		break;
@@ -2530,7 +2527,7 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, WM8962_PWR_MGMT_1,
 				    WM8962_VMID_SEL_MASK, 0x100);
 
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			msleep(100);
 		break;
 
@@ -2538,7 +2535,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -2614,7 +2610,7 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream,
 	dev_dbg(codec->dev, "hw_params set BCLK %dHz LRCLK %dHz\n",
 		wm8962->bclk, wm8962->lrclk);
 
-	if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
+	if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON)
 		wm8962_configure_bclk(codec);
 
 	return 0;
@@ -3118,7 +3114,7 @@ static irqreturn_t wm8962_irq(int irq, void *data)
 int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
 {
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	int irq_mask, enable;
 
 	wm8962->jack = jack;
@@ -3164,7 +3160,7 @@ 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;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	int i;
 	int reg = 0;
 	int best = 0;
@@ -3415,6 +3411,7 @@ static void wm8962_free_gpio(struct snd_soc_codec *codec)
 
 static int wm8962_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	int ret;
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
 	int i;
@@ -3462,7 +3459,7 @@ static int wm8962_probe(struct snd_soc_codec *codec)
 	}
 	if (!dmicclk || !dmicdat) {
 		dev_dbg(codec->dev, "DMIC not in use, disabling\n");
-		snd_soc_dapm_nc_pin(&codec->dapm, "DMICDAT");
+		snd_soc_dapm_nc_pin(dapm, "DMICDAT");
 	}
 	if (dmicclk != dmicdat)
 		dev_warn(codec->dev, "DMIC GPIOs partially configured\n");

sound/soc/codecs/wm8971.c

@@ -577,7 +577,7 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec,
 		flush_delayed_work(&wm8971->charge_work);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			snd_soc_cache_sync(codec);
 			/* charge output caps - set vmid to 5k for quick power up */
 			snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x01c0);
@@ -594,7 +594,6 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, WM8971_PWR1, 0x0001);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8974.c

@@ -514,7 +514,7 @@ 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->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_sync(dev_get_regmap(codec->dev, NULL));
 
 			/* Initial cap charge at VMID 5k */
@@ -533,7 +533,6 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8978.c

@@ -868,7 +868,7 @@ 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->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Initial cap charge at VMID 5k */
 			snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1,
 				      power1 | 0x3);
@@ -888,7 +888,6 @@ static int wm8978_set_bias_level(struct snd_soc_codec *codec,
 
 	dev_dbg(codec->dev, "%s: %d, %x\n", __func__, level, power1);
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -928,7 +927,7 @@ static int wm8978_suspend(struct snd_soc_codec *codec)
 {
 	struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec);
 
-	wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 	/* Also switch PLL off */
 	snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, 0);
 
@@ -944,7 +943,7 @@ static int wm8978_resume(struct snd_soc_codec *codec)
 	/* Sync reg_cache with the hardware */
 	regcache_sync(wm8978->regmap);
 
-	wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	if (wm8978->f_pllout)
 		/* Switch PLL on */

sound/soc/codecs/wm8983.c

@@ -915,7 +915,7 @@ static int wm8983_set_bias_level(struct snd_soc_codec *codec,
 				    1 << WM8983_VMIDSEL_SHIFT);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regcache_sync(wm8983->regmap);
 			if (ret < 0) {
 				dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
@@ -963,7 +963,6 @@ static int wm8983_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8985.c

@@ -897,7 +897,7 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec,
 				    1 << WM8985_VMIDSEL_SHIFT);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8985->supplies),
 						    wm8985->supplies);
 			if (ret) {
@@ -957,7 +957,6 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8988.c

@@ -738,7 +738,7 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_sync(wm8988->regmap);
 
 			/* VREF, VMID=2x5k */
@@ -756,7 +756,6 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, WM8988_PWR1, 0x0000);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8990.c

@@ -1124,7 +1124,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regcache_sync(wm8990->regmap);
 			if (ret < 0) {
 				dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
@@ -1227,7 +1227,6 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -1281,7 +1280,7 @@ static int wm8990_probe(struct snd_soc_codec *codec)
 	wm8990_reset(codec);
 
 	/* charge output caps */
-	wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	snd_soc_update_bits(codec, WM8990_AUDIO_INTERFACE_4,
 			    WM8990_ALRCGPIO1, WM8990_ALRCGPIO1);

sound/soc/codecs/wm8991.c

@@ -1131,7 +1131,7 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_sync(wm8991->regmap);
 			/* Enable all output discharge bits */
 			snd_soc_write(codec, WM8991_ANTIPOP1, WM8991_DIS_LLINE |
@@ -1224,7 +1224,6 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/wm8993.c

@@ -992,7 +992,7 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(ARRAY_SIZE(wm8993->supplies),
 						    wm8993->supplies);
 			if (ret != 0)
@@ -1065,8 +1065,6 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
@@ -1485,7 +1483,7 @@ 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;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	wm8993->hubs_data.hp_startup_mode = 1;
 	wm8993->hubs_data.dcs_codes_l = -2;
@@ -1539,7 +1537,7 @@ static int wm8993_probe(struct snd_soc_codec *codec)
 	 * VMID as an output and can disable it.
 	 */
 	if (wm8993->pdata.lineout1_diff && wm8993->pdata.lineout2_diff)
-		codec->dapm.idle_bias_off = 1;
+		dapm->idle_bias_off = 1;
 
 	return 0;
 
@@ -1563,7 +1561,7 @@ static int wm8993_suspend(struct snd_soc_codec *codec)
 	wm8993->fll_fout = fll_fout;
 	wm8993->fll_fref = fll_fref;
 
-	wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
 }
@@ -1573,7 +1571,7 @@ static int wm8993_resume(struct snd_soc_codec *codec)
 	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Restart the FLL? */
 	if (wm8993->fll_fout) {