Merge branches 'topic/adsp' and 'topic/dapm' of git://git.kernel.org/…

…pub/scm/linux/kernel/git/broonie/sound into asoc-arizona

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)
 
@@ -807,7 +816,7 @@ 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
@@ -1188,6 +1197,7 @@ struct soc_enum {
 	unsigned int mask;
 	const char * const *texts;
 	const unsigned int *values;
+	unsigned int autodisable:1;
 };
 
 /**
@@ -1269,6 +1279,58 @@ 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);