From fa880775ab0d5a8d540972f7b6800fad1af16b75 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 27 Apr 2015 22:13:23 +0200
Subject: ASoC: Add helper functions bias level management

Currently drivers are responsible for managing the bias_level field of
their DAPM context. The DAPM state itself is managed by the DAPM core
though and the core has certain expectations on how and when the bias_level
field should be updated. If drivers don't adhere to these undefined
behavior can occur.

This patch adds a few helper functions for manipulating the DAPM context
state, each function with a description on when it should be used and what
its effects are. This will also help us to move more of the bias_level
management from drivers to the DAPM core.

For convenience also add snd_soc_codec_* wrappers around these helpers.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 35 +++++++++++++++++++++++++++++++----
 1 file changed, 31 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index defe0f0082b5..b24782b50809 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -525,6 +525,35 @@ static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm)
 		snd_soc_component_async_complete(dapm->component);
 }
 
+/**
+ * snd_soc_dapm_force_bias_level() - Sets the DAPM bias level
+ * @dapm: The DAPM context for which to set the level
+ * @level: The level to set
+ *
+ * Forces the DAPM bias level to a specific state. It will call the bias level
+ * callback of DAPM context with the specified level. This will even happen if
+ * the context is already at the same level. Furthermore it will not go through
+ * the normal bias level sequencing, meaning any intermediate states between the
+ * current and the target state will not be entered.
+ *
+ * Note that the change in bias level is only temporary and the next time
+ * snd_soc_dapm_sync() is called the state will be set to the level as
+ * determined by the DAPM core. The function is mainly intended to be used to
+ * used during probe or resume from suspend to power up the device so
+ * initialization can be done, before the DAPM core takes over.
+ */
+int snd_soc_dapm_force_bias_level(struct snd_soc_dapm_context *dapm,
+	enum snd_soc_bias_level level)
+{
+	int ret = 0;
+
+	if (dapm->set_bias_level)
+		ret = dapm->set_bias_level(dapm, level);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_force_bias_level);
+
 /**
  * snd_soc_dapm_set_bias_level - set the bias level for the system
  * @dapm: DAPM context
@@ -547,10 +576,8 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
 	if (ret != 0)
 		goto out;
 
-	if (dapm->set_bias_level)
-		ret = dapm->set_bias_level(dapm, level);
-	else if (!card || dapm != &card->dapm)
-		dapm->bias_level = level;
+	if (!card || dapm != &card->dapm)
+		ret = snd_soc_dapm_force_bias_level(dapm, level);
 
 	if (ret != 0)
 		goto out;
-- 
cgit v1.2.3


From bd1204cb51f15d202f95222e873a94ed5d07b784 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 27 Apr 2015 22:13:24 +0200
Subject: ASoC: Route all bias level updates through the core

Use the new snd_soc_codec_force_bias_level() helper function to invoke the
bias_level callback of a driver instead of calling the callback by hand.
Currently the effect of this is the same, but having all bias level updates
go through a central place will allow us to move more of the bias level
management into the DAPM core.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98095.c | 4 ++--
 sound/soc/codecs/rt5640.c   | 4 ++--
 sound/soc/codecs/rt5645.c   | 2 +-
 sound/soc/codecs/rt5651.c   | 2 +-
 sound/soc/codecs/rt5677.c   | 2 +-
 sound/soc/codecs/sta32x.c   | 2 +-
 sound/soc/codecs/sta350.c   | 2 +-
 sound/soc/codecs/twl6040.c  | 2 +-
 sound/soc/codecs/wm8731.c   | 2 +-
 sound/soc/codecs/wm8737.c   | 2 +-
 sound/soc/codecs/wm8900.c   | 6 +++---
 sound/soc/codecs/wm8940.c   | 2 +-
 sound/soc/codecs/wm8955.c   | 2 +-
 sound/soc/codecs/wm8978.c   | 4 ++--
 sound/soc/codecs/wm8990.c   | 2 +-
 sound/soc/codecs/wm8993.c   | 4 ++--
 sound/soc/codecs/wm8994.c   | 2 +-
 sound/soc/codecs/wm9712.c   | 2 +-
 sound/soc/codecs/wm9713.c   | 2 +-
 19 files changed, 25 insertions(+), 25 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 8fba0c3db798..d6e80a932ec3 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -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);
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 178e55d4d481..d39b25cd62ef 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1939,7 +1939,7 @@ static int rt5640_probe(struct snd_soc_codec *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);
@@ -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);
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 69528ae5410c..b1e681a3e8db 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -2520,7 +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);
 
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 9f4c7be6d798..35c972505948 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -1625,7 +1625,7 @@ 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;
 }
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index af182586712d..ba408ad23457 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -4618,7 +4618,7 @@ static int rt5677_probe(struct snd_soc_codec *codec)
 			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);
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 007a0e3bc273..686ec765ea0e 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -970,7 +970,7 @@ 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);
 
diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c
index 669e3228241e..46fc07a94fcd 100644
--- a/sound/soc/codecs/sta350.c
+++ b/sound/soc/codecs/sta350.c
@@ -1037,7 +1037,7 @@ 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);
 
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index aeec27b6f1af..9bd887ed7f44 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -1130,7 +1130,7 @@ 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;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 2245b6a32f3d..00898d9d977d 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -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);
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index ada9ac1ba2c6..40e6acb4f3f4 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -560,7 +560,7 @@ 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);
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 2eb986c19b88..065da37bbf21 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1138,7 +1138,7 @@ 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 +1156,7 @@ 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 +1189,7 @@ 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);
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index e4142b4309eb..4b4b9973c740 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -707,7 +707,7 @@ 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)
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 00bec915d652..8080eabf63bd 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -929,7 +929,7 @@ 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);
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index cf7032911721..572b1bf07d6c 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -928,7 +928,7 @@ 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 +944,7 @@ 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 */
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c93bffcb3cfb..c642b3abba5d 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1281,7 +1281,7 @@ 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);
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 2e70a270eb28..b8385ac26b90 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1563,7 +1563,7 @@ 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 +1573,7 @@ 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) {
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 4fbc7689339a..fedf48d8e7ae 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3163,7 +3163,7 @@ static int wm8994_codec_suspend(struct snd_soc_codec *codec)
 				 i + 1, ret);
 	}
 
-	wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 98c9525bd751..9119a779f728 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -646,7 +646,7 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec)
 	if (ret < 0)
 		return ret;
 
-	wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	if (ret == 0) {
 		/* Sync reg_cache with the hardware after cold reset */
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 79552953e1bd..39c3e717c577 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1201,7 +1201,7 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec)
 	if (ret < 0)
 		return ret;
 
-	wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* do we need to re-start the PLL ? */
 	if (wm9713->pll_in)
-- 
cgit v1.2.3


From f4bf8d770b58862c2af9d17adc2fee05bef8f2c0 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 27 Apr 2015 22:13:25 +0200
Subject: ASoC: Move bias level update to the core

All drivers have the same line at the end of the set_bias_level callback to
update the bias_level state. Move this update into
snd_soc_dapm_force_bias_level() and remove them from the drivers.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/88pm860x-codec.c | 1 -
 sound/soc/codecs/adau1373.c       | 1 -
 sound/soc/codecs/adau1701.c       | 1 -
 sound/soc/codecs/adau1761.c       | 1 -
 sound/soc/codecs/adau1781.c       | 1 -
 sound/soc/codecs/adau1977.c       | 2 --
 sound/soc/codecs/adav80x.c        | 1 -
 sound/soc/codecs/ak4535.c         | 1 -
 sound/soc/codecs/ak4641.c         | 1 -
 sound/soc/codecs/ak4642.c         | 1 -
 sound/soc/codecs/ak4671.c         | 1 -
 sound/soc/codecs/alc5623.c        | 1 -
 sound/soc/codecs/alc5632.c        | 1 -
 sound/soc/codecs/cq93vc.c         | 1 -
 sound/soc/codecs/cs4265.c         | 1 -
 sound/soc/codecs/cs42l52.c        | 1 -
 sound/soc/codecs/cs42l56.c        | 1 -
 sound/soc/codecs/cs42l73.c        | 1 -
 sound/soc/codecs/cx20442.c        | 2 --
 sound/soc/codecs/da7213.c         | 1 -
 sound/soc/codecs/da732x.c         | 2 --
 sound/soc/codecs/da9055.c         | 1 -
 sound/soc/codecs/es8328.c         | 1 -
 sound/soc/codecs/isabelle.c       | 2 --
 sound/soc/codecs/jz4740.c         | 2 --
 sound/soc/codecs/lm4857.c         | 2 --
 sound/soc/codecs/lm49453.c        | 2 --
 sound/soc/codecs/max98088.c       | 1 -
 sound/soc/codecs/max98090.c       | 1 -
 sound/soc/codecs/max98095.c       | 1 -
 sound/soc/codecs/max9850.c        | 1 -
 sound/soc/codecs/ml26124.c        | 1 -
 sound/soc/codecs/pcm512x.c        | 2 --
 sound/soc/codecs/rt286.c          | 1 -
 sound/soc/codecs/rt5631.c         | 1 -
 sound/soc/codecs/rt5640.c         | 1 -
 sound/soc/codecs/rt5645.c         | 1 -
 sound/soc/codecs/rt5651.c         | 1 -
 sound/soc/codecs/rt5670.c         | 1 -
 sound/soc/codecs/rt5677.c         | 1 -
 sound/soc/codecs/sgtl5000.c       | 1 -
 sound/soc/codecs/sn95031.c        | 1 -
 sound/soc/codecs/ssm2518.c        | 2 --
 sound/soc/codecs/ssm2602.c        | 1 -
 sound/soc/codecs/ssm4567.c        | 7 +------
 sound/soc/codecs/sta32x.c         | 1 -
 sound/soc/codecs/sta350.c         | 1 -
 sound/soc/codecs/sta529.c         | 6 ------
 sound/soc/codecs/stac9766.c       | 1 -
 sound/soc/codecs/tlv320aic23.c    | 1 -
 sound/soc/codecs/tlv320aic31xx.c  | 1 -
 sound/soc/codecs/tlv320aic32x4.c  | 1 -
 sound/soc/codecs/tlv320aic3x.c    | 1 -
 sound/soc/codecs/tlv320dac33.c    | 1 -
 sound/soc/codecs/twl4030.c        | 1 -
 sound/soc/codecs/twl6040.c        | 2 --
 sound/soc/codecs/uda134x.c        | 1 -
 sound/soc/codecs/uda1380.c        | 1 -
 sound/soc/codecs/wm0010.c         | 2 --
 sound/soc/codecs/wm1250-ev1.c     | 2 --
 sound/soc/codecs/wm8350.c         | 1 -
 sound/soc/codecs/wm8400.c         | 1 -
 sound/soc/codecs/wm8510.c         | 1 -
 sound/soc/codecs/wm8523.c         | 1 -
 sound/soc/codecs/wm8580.c         | 1 -
 sound/soc/codecs/wm8711.c         | 1 -
 sound/soc/codecs/wm8728.c         | 1 -
 sound/soc/codecs/wm8731.c         | 1 -
 sound/soc/codecs/wm8737.c         | 1 -
 sound/soc/codecs/wm8750.c         | 1 -
 sound/soc/codecs/wm8753.c         | 1 -
 sound/soc/codecs/wm8770.c         | 1 -
 sound/soc/codecs/wm8776.c         | 1 -
 sound/soc/codecs/wm8900.c         | 1 -
 sound/soc/codecs/wm8903.c         | 2 --
 sound/soc/codecs/wm8904.c         | 1 -
 sound/soc/codecs/wm8940.c         | 2 --
 sound/soc/codecs/wm8955.c         | 1 -
 sound/soc/codecs/wm8960.c         | 4 ----
 sound/soc/codecs/wm8961.c         | 2 --
 sound/soc/codecs/wm8962.c         | 1 -
 sound/soc/codecs/wm8971.c         | 1 -
 sound/soc/codecs/wm8974.c         | 1 -
 sound/soc/codecs/wm8978.c         | 1 -
 sound/soc/codecs/wm8983.c         | 1 -
 sound/soc/codecs/wm8985.c         | 1 -
 sound/soc/codecs/wm8988.c         | 1 -
 sound/soc/codecs/wm8990.c         | 1 -
 sound/soc/codecs/wm8991.c         | 1 -
 sound/soc/codecs/wm8993.c         | 2 --
 sound/soc/codecs/wm8994.c         | 2 --
 sound/soc/codecs/wm8995.c         | 1 -
 sound/soc/codecs/wm8996.c         | 2 --
 sound/soc/codecs/wm9081.c         | 2 --
 sound/soc/codecs/wm9090.c         | 2 --
 sound/soc/codecs/wm9712.c         | 1 -
 sound/soc/codecs/wm9713.c         | 1 -
 sound/soc/soc-dapm.c              | 3 +++
 98 files changed, 4 insertions(+), 130 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index a0f265327fdf..c0b2686a6aac 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 783dcb57043a..a43160254929 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index d4e219b6b98f..808b964086e3 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c
index a1baeee160f4..5ba24618b576 100644
--- a/sound/soc/codecs/adau1761.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c
index 35581f43fa6d..9c01ef0de0c0 100644
--- a/sound/soc/codecs/adau1781.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c
index 7ad8e156e2df..c5b1b8e4e7fc 100644
--- a/sound/soc/codecs/adau1977.c
+++ b/sound/soc/codecs/adau1977.c
@@ -496,8 +496,6 @@ static int adau1977_set_bias_level(struct snd_soc_codec *codec,
 	if (ret)
 		return ret;
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 4373ada95648..260a652e4a43 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 9130d916f2f4..8670861e5bec 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 81b54a270bd8..3b22b587a820 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 13585e88f597..7c0f6552c229 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/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;
 }
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 2a58b1dccd2f..0e59063aeb6f 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 0e357996864b..e92b5ae3cab2 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index db3283abbe18..607a63b9705f 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index d6dedd4eab29..1c895a53001d 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/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;
 }
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index cac48ddf3ba6..d7ec4756e45b 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 1589e7a881d8..3c49a756b89b 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -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;
 }
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index cbc654fe48c7..a7638c52b509 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -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;
 }
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 8ecedba79606..156ec938f441 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 0f334bc1b63c..13041ccf1010 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -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;
 }
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 9ec577f0edb4..925dd3c16d6c 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index 911c26c705fc..06519057bdff 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -1502,8 +1502,6 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index ad19cc56702b..3bdc95a70112 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index c5f35a07e8e4..996e3f4e7343 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -566,7 +566,6 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
 				0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index 3a89ce66d51d..ebd90283c960 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index 933f4476d76c..8425d262e566 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -281,8 +281,6 @@ static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index a924bb9d7886..79ad4cbdcdd4 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index c4dfde9bdf1c..166fd4c88ddb 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -1284,8 +1284,6 @@ static int lm49453_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 805b3f8cd39d..3200aa80f1f2 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 3e33ef2acf3c..c5736b2f7c76 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -1824,7 +1824,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
 		regcache_mark_dirty(max98090->regmap);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index d6e80a932ec3..66c7ca431a2e 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1678,7 +1678,6 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec,
 		regcache_mark_dirty(max98095->regmap);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 10f8e47ce2c2..f6b616b6ffca 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 711f55039522..f1d5778e6599 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index e12764d15431..c305b2871c59 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -641,8 +641,6 @@ static int pcm512x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 0fcda35a3a93..dbdbb9e8d4ba 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -1012,7 +1012,6 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 2c10d77727af..e285d8ad260a 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1569,7 +1569,6 @@ static int rt5631_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index d39b25cd62ef..7d488d8b03d6 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1902,7 +1902,6 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index b1e681a3e8db..ea583675fa00 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -2409,7 +2409,6 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 35c972505948..f03c6fc1a7e9 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -1604,7 +1604,6 @@ static int rt5651_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index cc7f84a150a7..9235711e86c2 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -2647,7 +2647,6 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index ba408ad23457..696ba587969e 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -4392,7 +4392,6 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 3593a1496056..b01c985a2307 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -979,7 +979,6 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 7947c0ebb1ed..e4743684cc1d 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -226,7 +226,6 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index 67ea55adb307..40b22b3fd5f6 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -521,8 +521,6 @@ static int ssm2518_set_bias_level(struct snd_soc_codec *codec,
 	if (ret)
 		return ret;
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 314eaece1b7d..296a140b8c35 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c
index a984485108cd..643bcff4a919 100644
--- a/sound/soc/codecs/ssm4567.c
+++ b/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 = {
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 686ec765ea0e..033b7d9f45f7 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c
index 46fc07a94fcd..50d8bbf90ce2 100644
--- a/sound/soc/codecs/sta350.c
+++ b/sound/soc/codecs/sta350.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index b0f436d10125..c3217af1ca29 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -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;
 
 }
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 6464caf72b21..2341e8e6bfc1 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index cc17e7e5126e..cd8c02b6e4de 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index c86dd9aae157..e629273019d0 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -1053,7 +1053,6 @@ static int aic31xx_set_bias_level(struct snd_soc_codec *codec,
 			aic31xx_power_off(codec);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 015467ed606b..ad6cb90e5f9b 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 51c4713ac6e3..57d709075746 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1406,7 +1406,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
 			aic3x_set_power(codec, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 4e3e607dec13..33e93f62de30 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -651,7 +651,6 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec,
 			return ret;
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index d04693e9cf9f..e725e13a7f59 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1595,7 +1595,6 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 		twl4030_codec_enable(codec, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 9bd887ed7f44..b8ecce206af8 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -853,8 +853,6 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index f883308c00de..dbecbc05cf7b 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index dc7778b6dd7f..cc5b1769958a 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -623,7 +623,6 @@ 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;
 }
 
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index f37989ec7cba..3358dd6811fa 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -767,8 +767,6 @@ static int wm0010_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index 8011f75fb6cb..048f00568260 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/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;
 }
 
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index c65e5a75fc1a..dd0d0248e641 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index b0d84e552fca..adbfebe04c77 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1232,7 +1232,6 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 8736ad094b24..a380c10e867b 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -538,7 +538,6 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index b1cc94f5fc4b..34ebe95d93f1 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 0a887c5ec83a..5951d88e3dc9 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 121e46d53779..a4aab6e7f5cc 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 55c7fb4fc786..a737068d5576 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 00898d9d977d..a13a20ac47af 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -523,7 +523,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 		regcache_mark_dirty(wm8731->regmap);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index 40e6acb4f3f4..4a9407dadae3 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -510,7 +510,6 @@ static int wm8737_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index eb0a1644ba11..d6ff25a9d5af 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index c50a5959345f..b7d38f7ba636 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index 53e977da2f86..c24db8037201 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -534,7 +534,6 @@ static int wm8770_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index c13050b77931..b0e3c3bbd440 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -357,7 +357,6 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 065da37bbf21..e7d2ecd150cf 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 04b04f8e147c..5e0bef62d974 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1200,8 +1200,6 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 215e93c1ddf0..a7a8fa0567b1 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 4b4b9973c740..f2d6a490713f 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -510,8 +510,6 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 8080eabf63bd..f400d5c7234c 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 3035d9856415..6fa832b6365b 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -691,8 +691,6 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
@@ -802,8 +800,6 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 95e2c1bfc809..6f95d7044aac 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -795,8 +795,6 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 118b0034ba23..00793b7b0a83 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2538,7 +2538,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index f9cbabdc6238..94eb27ec572f 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index ff0e4646b934..d2180c83a5cc 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -533,7 +533,6 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 572b1bf07d6c..e2363b9a38a0 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
index 5d1cf08a72b8..f9245715cebd 100644
--- a/sound/soc/codecs/wm8983.c
+++ b/sound/soc/codecs/wm8983.c
@@ -963,7 +963,6 @@ static int wm8983_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index 0b3b54c9971d..4e6901b5c819 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -957,7 +957,6 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 24968aa8618a..92680c6d247e 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -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;
 }
 
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c642b3abba5d..ff377cab5775 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1227,7 +1227,6 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 49df0dc607e6..abd439fb0820 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -1224,7 +1224,6 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index b8385ac26b90..52ec4fe03b23 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1065,8 +1065,6 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index fedf48d8e7ae..2d32b542f103 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2546,8 +2546,6 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 66103c2b012e..47af27fb339a 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1990,7 +1990,6 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 308748a022c5..3dce50751469 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1628,8 +1628,6 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 13a3f335ea5b..02d9a5012c1b 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -898,8 +898,6 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 60d243c904f5..03bca8581bc7 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -515,8 +515,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 9119a779f728..1fda104dfc45 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -610,7 +610,6 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec,
 		ac97_write(codec, AC97_POWERDOWN, 0xffff);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 39c3e717c577..9d18a0ec4280 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1171,7 +1171,6 @@ static int wm9713_set_bias_level(struct snd_soc_codec *codec,
 		ac97_write(codec, AC97_POWERDOWN, 0xffff);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index b24782b50809..79b947820231 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -550,6 +550,9 @@ int snd_soc_dapm_force_bias_level(struct snd_soc_dapm_context *dapm,
 	if (dapm->set_bias_level)
 		ret = dapm->set_bias_level(dapm, level);
 
+	if (ret == 0)
+		dapm->bias_level = level;
+
 	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_force_bias_level);
-- 
cgit v1.2.3


From fb75ee66c2707f56397eb29c01decf36254e3d46 Mon Sep 17 00:00:00 2001
From: kbuild test robot <fengguang.wu@intel.com>
Date: Tue, 28 Apr 2015 12:20:32 +0800
Subject: ASoC: adau1977: fix simple_return.cocci warnings

sound/soc/codecs/adau1977.c:496:5-8: WARNING: end returns can be simpified

 Simplify a trivial if-return sequence.  Possibly combine with a
 preceding function call.
Generated by: scripts/coccinelle/misc/simple_return.cocci

Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/adau1977.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c
index c5b1b8e4e7fc..3fb09c165055 100644
--- a/sound/soc/codecs/adau1977.c
+++ b/sound/soc/codecs/adau1977.c
@@ -493,10 +493,7 @@ static int adau1977_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	if (ret)
-		return ret;
-
-	return 0;
+	return ret;
 }
 
 static int adau1977_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
-- 
cgit v1.2.3


From beb9969b8a644991dbfdaf18b9f1161a39a91df8 Mon Sep 17 00:00:00 2001
From: kbuild test robot <fengguang.wu@intel.com>
Date: Tue, 28 Apr 2015 12:20:32 +0800
Subject: ASoC: ssm2518: fix simple_return.cocci warnings

sound/soc/codecs/ssm2518.c:521:5-8: WARNING: end returns can be simpified

 Simplify a trivial if-return sequence.  Possibly combine with a
 preceding function call.
Generated by: scripts/coccinelle/misc/simple_return.cocci

Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ssm2518.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index 40b22b3fd5f6..13c6ab0f7af0 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -518,10 +518,7 @@ static int ssm2518_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	if (ret)
-		return ret;
-
-	return 0;
+	return ret;
 }
 
 static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
-- 
cgit v1.2.3


From 40b7bea10ae09595da5d66228d93e3920306790d Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Fri, 1 May 2015 12:37:24 +0100
Subject: ASoC: dapm: Remove local OOM error message

The memory subsystem is pretty chatty on failure no need to have local
OOM messages as well.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 79b947820231..beb48b608142 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -310,12 +310,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 	struct soc_mixer_control *mc;
 
 	data = kzalloc(sizeof(*data), GFP_KERNEL);
-	if (!data) {
-		dev_err(widget->dapm->dev,
-				"ASoC: can't allocate kcontrol data for %s\n",
-				widget->name);
+	if (!data)
 		return -ENOMEM;
-	}
 
 	INIT_LIST_HEAD(&data->paths);
 
-- 
cgit v1.2.3


From 773da9b358bfbef1b7a862425fea0d9d9d3443f8 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Fri, 1 May 2015 12:37:25 +0100
Subject: ASoC: dapm: Append "Autodisable" to autodisable widget names

This makes it a little easier to follow what is happening in debugfs.
Additionally is also useful in facilitating work to add autodisable
muxes because the control name is already used for the mux widget and
thus shouldn't be reused for the autodisable widget.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 23 ++++++++++++++++++++---
 1 file changed, 20 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index beb48b608142..a0d97f89eb75 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -308,6 +308,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 {
 	struct dapm_kcontrol_data *data;
 	struct soc_mixer_control *mc;
+	const char *name;
+	int ret;
 
 	data = kzalloc(sizeof(*data), GFP_KERNEL);
 	if (!data)
@@ -324,6 +326,13 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 		if (mc->autodisable) {
 			struct snd_soc_dapm_widget template;
 
+			name = kasprintf(GFP_KERNEL, "%s %s", kcontrol->id.name,
+					 "Autodisable");
+			if (!name) {
+				ret = -ENOMEM;
+				goto err_data;
+			}
+
 			memset(&template, 0, sizeof(template));
 			template.reg = mc->reg;
 			template.mask = (1 << fls(mc->max)) - 1;
@@ -334,15 +343,15 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 				template.off_val = 0;
 			template.on_val = template.off_val;
 			template.id = snd_soc_dapm_kcontrol;
-			template.name = kcontrol->id.name;
+			template.name = name;
 
 			data->value = template.on_val;
 
 			data->widget = snd_soc_dapm_new_control(widget->dapm,
 				&template);
 			if (!data->widget) {
-				kfree(data);
-				return -ENOMEM;
+				ret = -ENOMEM;
+				goto err_name;
 			}
 		}
 		break;
@@ -353,11 +362,19 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 	kcontrol->private_data = data;
 
 	return 0;
+
+err_name:
+	kfree(name);
+err_data:
+	kfree(data);
+	return ret;
 }
 
 static void dapm_kcontrol_free(struct snd_kcontrol *kctl)
 {
 	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl);
+	if (data->widget)
+		kfree(data->widget->name);
 	kfree(data->wlist);
 	kfree(data);
 }
-- 
cgit v1.2.3


From 561ed680b764b288feeb74a24e1d9fb3da98ec7b Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Fri, 1 May 2015 12:37:26 +0100
Subject: ASoC: dapm: Add support for autodisable mux controls

Commit 57295073b6ac ("ASoC: dapm: Implement mixer input auto-disable")
added support for autodisable controls, controls whose values are only
written to the hardware when their respective widgets are powered up.
But it only added support for controls based on the mixer abstraction.

This patch add support for mux controls (DAPM controls based on the
enum abstraction) to be auto-disabled as well. As each mux can only have
a single control, there is no need to tie the autodisable widget to the
inputs (as is done for the mixer controls) it can be tided directly to
the mux widget itself.

Note that it is assumed that the first entry in a autodisable mux
control will always represent the off state for the mux and is what the
mux will be set to whilst it is disabled.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  | 10 +++++++
 sound/soc/soc-dapm.c | 78 +++++++++++++++++++++++++++++++++++++++-------------
 2 files changed, 69 insertions(+), 19 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index b257a09a98d1..2f2e59e1513e 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -192,6 +192,10 @@
 	.mask = xmask, .items = xitems, .texts = xtexts, .values = 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)
 
@@ -1188,6 +1197,7 @@ struct soc_enum {
 	unsigned int mask;
 	const char * const *texts;
 	const unsigned int *values;
+	unsigned int autodisable:1;
 };
 
 /**
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a0d97f89eb75..79e6cf4b7de1 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -308,6 +308,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 {
 	struct dapm_kcontrol_data *data;
 	struct soc_mixer_control *mc;
+	struct soc_enum *e;
 	const char *name;
 	int ret;
 
@@ -355,6 +356,41 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 			}
 		}
 		break;
+	case snd_soc_dapm_mux:
+		e = (struct soc_enum *)kcontrol->private_value;
+
+		if (e->autodisable) {
+			struct snd_soc_dapm_widget template;
+
+			name = kasprintf(GFP_KERNEL, "%s %s", kcontrol->id.name,
+					 "Autodisable");
+			if (!name) {
+				ret = -ENOMEM;
+				goto err_data;
+			}
+
+			memset(&template, 0, sizeof(template));
+			template.reg = e->reg;
+			template.mask = e->mask << e->shift_l;
+			template.shift = e->shift_l;
+			template.off_val = snd_soc_enum_item_to_val(e, 0);
+			template.on_val = template.off_val;
+			template.id = snd_soc_dapm_kcontrol;
+			template.name = name;
+
+			data->value = template.on_val;
+
+			data->widget = snd_soc_dapm_new_control(widget->dapm,
+					&template);
+			if (!data->widget) {
+				ret = -ENOMEM;
+				goto err_name;
+			}
+
+			snd_soc_dapm_add_path(widget->dapm, data->widget,
+					      widget, NULL, NULL);
+		}
+		break;
 	default:
 		break;
 	}
@@ -418,11 +454,6 @@ static void dapm_kcontrol_add_path(const struct snd_kcontrol *kcontrol,
 	struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol);
 
 	list_add_tail(&path->list_kcontrol, &data->paths);
-
-	if (data->widget) {
-		snd_soc_dapm_add_path(data->widget->dapm, data->widget,
-		    path->source, NULL, NULL);
-	}
 }
 
 static bool dapm_kcontrol_is_powered(const struct snd_kcontrol *kcontrol)
@@ -820,6 +851,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
 {
 	int i, ret;
 	struct snd_soc_dapm_path *path;
+	struct dapm_kcontrol_data *data;
 
 	/* add kcontrol */
 	for (i = 0; i < w->num_kcontrols; i++) {
@@ -829,16 +861,20 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
 			if (path->name != (char *)w->kcontrol_news[i].name)
 				continue;
 
-			if (w->kcontrols[i]) {
-				dapm_kcontrol_add_path(w->kcontrols[i], path);
-				continue;
+			if (!w->kcontrols[i]) {
+				ret = dapm_create_or_share_mixmux_kcontrol(w, i);
+				if (ret < 0)
+					return ret;
 			}
 
-			ret = dapm_create_or_share_mixmux_kcontrol(w, i);
-			if (ret < 0)
-				return ret;
-
 			dapm_kcontrol_add_path(w->kcontrols[i], path);
+
+			data = snd_kcontrol_chip(w->kcontrols[i]);
+			if (data->widget)
+				snd_soc_dapm_add_path(data->widget->dapm,
+						      data->widget,
+						      path->source,
+						      NULL, NULL);
 		}
 	}
 
@@ -2945,16 +2981,19 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
+	struct snd_soc_card *card = dapm->card;
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int reg_val, val;
 
-	if (e->reg != SND_SOC_NOPM) {
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+	if (e->reg != SND_SOC_NOPM && dapm_kcontrol_is_powered(kcontrol)) {
 		int ret = soc_dapm_read(dapm, e->reg, &reg_val);
 		if (ret)
 			return ret;
 	} else {
 		reg_val = dapm_kcontrol_get_value(kcontrol);
 	}
+	mutex_unlock(&card->dapm_mutex);
 
 	val = (reg_val >> e->shift_l) & e->mask;
 	ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val);
@@ -2984,7 +3023,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_soc_card *card = dapm->card;
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int *item = ucontrol->value.enumerated.item;
-	unsigned int val, change;
+	unsigned int val, change, reg_change = 0;
 	unsigned int mask;
 	struct snd_soc_dapm_update update;
 	int ret = 0;
@@ -3003,19 +3042,20 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
+	change = dapm_kcontrol_set_value(kcontrol, val);
+
 	if (e->reg != SND_SOC_NOPM)
-		change = soc_dapm_test_bits(dapm, e->reg, mask, val);
-	else
-		change = dapm_kcontrol_set_value(kcontrol, val);
+		reg_change = soc_dapm_test_bits(dapm, e->reg, mask, val);
 
-	if (change) {
-		if (e->reg != SND_SOC_NOPM) {
+	if (change || reg_change) {
+		if (reg_change) {
 			update.kcontrol = kcontrol;
 			update.reg = e->reg;
 			update.mask = mask;
 			update.val = val;
 			card->update = &update;
 		}
+		change |= reg_change;
 
 		ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e);
 
-- 
cgit v1.2.3


From 92fa12426741d52b39ec92ad77c9843d3fc2b3d6 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 1 May 2015 18:02:42 +0200
Subject: ASoC: dapm: Add new widgets to the end of the widget list

Currently new widgets are appended to the beginning of the cards widget
list. This has the effect that widgets that are created while iterating
over the widget list in snd_soc_dapm_new_widgets() (like e.g. the
auto-disable widgets) are not covered during that invocation of the
function. If no further invocations of snd_soc_dapm_new_widgets() happen
these widgets will not be fully initialized and e.g. no debugfs entries are
created for them.

By adding new widgets to the end of the widget list we make sure that
widgets that are created in snd_soc_dapm_new_widgets() will still be
handled during the same snd_soc_dapm_new_widgets() invocation and are
always fully initialized.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 79e6cf4b7de1..5c159f4f8097 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3252,7 +3252,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 	INIT_LIST_HEAD(&w->sinks);
 	INIT_LIST_HEAD(&w->list);
 	INIT_LIST_HEAD(&w->dirty);
-	list_add(&w->list, &dapm->card->widgets);
+	list_add_tail(&w->list, &dapm->card->widgets);
 
 	w->inputs = -1;
 	w->outputs = -1;
-- 
cgit v1.2.3


From d714f97c5b8c4c5da56b89a7289acb3f12ef7abb Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 1 May 2015 18:02:43 +0200
Subject: ASoC: dapm: Add demux support

A demux is conceptually similar to a mux. Where a mux has multiple input
and one output and selects one of the inputs to be connected to the output,
the demux has one input and multiple outputs and selects one of the outputs
to which the input gets connected.

This similarity makes it straight forward to support them in DAPM using the
existing mux support, we only need to swap sinks and sources when initially
setting up the paths.

The only slightly tricky part is that there can only be one control per
path. Since mixers/muxes are at the sink of a path and a demux is at the
source and both types want a control it is not possible to directly connect
a demux output to a mixer/mux input. The patch adds some sanity checks to
make sure that this does not happen.

Drivers who want to model hardware which directly connects a demux output
to a mixer/mux input can do this by inserting a dummy widget between the
two. E.g.:

	{ "Dummy", "Demux Control", "Demux" },
	{ "Mixer", "Mixer Control", "Dummy" },

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dapm.h |   5 +++
 sound/soc/soc-dapm.c     | 112 ++++++++++++++++++++++++++++++++++++++++-------
 2 files changed, 102 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 70216d20e897..96c5e0ec81d1 100644
--- a/include/sound/soc-dapm.h
+++ b/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,\
@@ -452,6 +456,7 @@ 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) */
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 5c159f4f8097..a2e5f2278caa 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -70,6 +70,7 @@ static int dapm_up_seq[] = {
 	[snd_soc_dapm_aif_out] = 4,
 	[snd_soc_dapm_mic] = 5,
 	[snd_soc_dapm_mux] = 6,
+	[snd_soc_dapm_demux] = 6,
 	[snd_soc_dapm_dac] = 7,
 	[snd_soc_dapm_switch] = 8,
 	[snd_soc_dapm_mixer] = 8,
@@ -100,6 +101,7 @@ static int dapm_down_seq[] = {
 	[snd_soc_dapm_mic] = 7,
 	[snd_soc_dapm_micbias] = 8,
 	[snd_soc_dapm_mux] = 9,
+	[snd_soc_dapm_demux] = 9,
 	[snd_soc_dapm_aif_in] = 10,
 	[snd_soc_dapm_aif_out] = 10,
 	[snd_soc_dapm_dai_in] = 10,
@@ -356,6 +358,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 			}
 		}
 		break;
+	case snd_soc_dapm_demux:
 	case snd_soc_dapm_mux:
 		e = (struct soc_enum *)kcontrol->private_value;
 
@@ -639,9 +642,10 @@ out:
 
 /* connect mux widget to its interconnecting audio paths */
 static int dapm_connect_mux(struct snd_soc_dapm_context *dapm,
-	struct snd_soc_dapm_path *path, const char *control_name)
+	struct snd_soc_dapm_path *path, const char *control_name,
+	struct snd_soc_dapm_widget *w)
 {
-	const struct snd_kcontrol_new *kcontrol = &path->sink->kcontrol_news[0];
+	const struct snd_kcontrol_new *kcontrol = &w->kcontrol_news[0];
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int val, item;
 	int i;
@@ -781,6 +785,7 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
 				wname_in_long_name = false;
 				kcname_in_long_name = true;
 				break;
+			case snd_soc_dapm_demux:
 			case snd_soc_dapm_mux:
 				wname_in_long_name = true;
 				kcname_in_long_name = false;
@@ -886,17 +891,32 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w)
 {
 	struct snd_soc_dapm_context *dapm = w->dapm;
 	struct snd_soc_dapm_path *path;
+	struct list_head *paths;
+	const char *type;
 	int ret;
 
+	switch (w->id) {
+	case snd_soc_dapm_mux:
+		paths = &w->sources;
+		type = "mux";
+		break;
+	case snd_soc_dapm_demux:
+		paths = &w->sinks;
+		type = "demux";
+		break;
+	default:
+		return -EINVAL;
+	}
+
 	if (w->num_kcontrols != 1) {
 		dev_err(dapm->dev,
-			"ASoC: mux %s has incorrect number of controls\n",
+			"ASoC: %s %s has incorrect number of controls\n", type,
 			w->name);
 		return -EINVAL;
 	}
 
-	if (list_empty(&w->sources)) {
-		dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name);
+	if (list_empty(paths)) {
+		dev_err(dapm->dev, "ASoC: %s %s has no paths\n", type, w->name);
 		return -EINVAL;
 	}
 
@@ -904,9 +924,16 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w)
 	if (ret < 0)
 		return ret;
 
-	list_for_each_entry(path, &w->sources, list_sink) {
-		if (path->name)
-			dapm_kcontrol_add_path(w->kcontrols[0], path);
+	if (w->id == snd_soc_dapm_mux) {
+		list_for_each_entry(path, &w->sources, list_sink) {
+			if (path->name)
+				dapm_kcontrol_add_path(w->kcontrols[0], path);
+		}
+	} else {
+		list_for_each_entry(path, &w->sinks, list_source) {
+			if (path->name)
+				dapm_kcontrol_add_path(w->kcontrols[0], path);
+		}
 	}
 
 	return 0;
@@ -2414,6 +2441,50 @@ static void dapm_update_widget_flags(struct snd_soc_dapm_widget *w)
 	}
 }
 
+static int snd_soc_dapm_check_dynamic_path(struct snd_soc_dapm_context *dapm,
+	struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink,
+	const char *control)
+{
+	bool dynamic_source = false;
+	bool dynamic_sink = false;
+
+	if (!control)
+		return 0;
+
+	switch (source->id) {
+	case snd_soc_dapm_demux:
+		dynamic_source = true;
+		break;
+	default:
+		break;
+	}
+
+	switch (sink->id) {
+	case snd_soc_dapm_mux:
+	case snd_soc_dapm_switch:
+	case snd_soc_dapm_mixer:
+	case snd_soc_dapm_mixer_named_ctl:
+		dynamic_sink = true;
+		break;
+	default:
+		break;
+	}
+
+	if (dynamic_source && dynamic_sink) {
+		dev_err(dapm->dev,
+			"Direct connection between demux and mixer/mux not supported for path %s -> [%s] -> %s\n",
+			source->name, control, sink->name);
+		return -EINVAL;
+	} else if (!dynamic_source && !dynamic_sink) {
+		dev_err(dapm->dev,
+			"Control not supported for path %s -> [%s] -> %s\n",
+			source->name, control, sink->name);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
 static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 	struct snd_soc_dapm_widget *wsource, struct snd_soc_dapm_widget *wsink,
 	const char *control,
@@ -2444,6 +2515,10 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 		return -EINVAL;
 	}
 
+	ret = snd_soc_dapm_check_dynamic_path(dapm, wsource, wsink, control);
+	if (ret)
+		return ret;
+
 	path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL);
 	if (!path)
 		return -ENOMEM;
@@ -2463,10 +2538,19 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 	if (control == NULL) {
 		path->connect = 1;
 	} else {
-		/* connect dynamic paths */
+		switch (wsource->id) {
+		case snd_soc_dapm_demux:
+			ret = dapm_connect_mux(dapm, path, control, wsource);
+			if (ret)
+				goto err;
+			break;
+		default:
+			break;
+		}
+
 		switch (wsink->id) {
 		case snd_soc_dapm_mux:
-			ret = dapm_connect_mux(dapm, path, control);
+			ret = dapm_connect_mux(dapm, path, control, wsink);
 			if (ret != 0)
 				goto err;
 			break;
@@ -2478,11 +2562,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 				goto err;
 			break;
 		default:
-			dev_err(dapm->dev,
-				"Control not supported for path %s -> [%s] -> %s\n",
-				wsource->name, control, wsink->name);
-			ret = -EINVAL;
-			goto err;
+			break;
 		}
 	}
 
@@ -2815,6 +2895,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card)
 			dapm_new_mixer(w);
 			break;
 		case snd_soc_dapm_mux:
+		case snd_soc_dapm_demux:
 			dapm_new_mux(w);
 			break;
 		case snd_soc_dapm_pga:
@@ -3219,6 +3300,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 		w->power_check = dapm_always_on_check_power;
 		break;
 	case snd_soc_dapm_mux:
+	case snd_soc_dapm_demux:
 	case snd_soc_dapm_switch:
 	case snd_soc_dapm_mixer:
 	case snd_soc_dapm_mixer_named_ctl:
-- 
cgit v1.2.3


From 964a0b896a7c78622801afcee77ed3d240352747 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Fri, 8 May 2015 10:50:10 +0100
Subject: ASoC: dapm: Add missing mutex unlock

The is a missing mutex unlock on the error path in
snd_soc_dapm_get_enum_double. This was introduced in commit
561ed680b764 ("ASoC: dapm: Add support for autodisable mux controls").
This patch adds the missing unlock.

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a2e5f2278caa..765416174388 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3069,8 +3069,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 	if (e->reg != SND_SOC_NOPM && dapm_kcontrol_is_powered(kcontrol)) {
 		int ret = soc_dapm_read(dapm, e->reg, &reg_val);
-		if (ret)
+		if (ret) {
+			mutex_unlock(&card->dapm_mutex);
 			return ret;
+		}
 	} else {
 		reg_val = dapm_kcontrol_get_value(kcontrol);
 	}
-- 
cgit v1.2.3


From 70c751095d5481d246ae7ec622ed35a76ce6ff0c Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 7 May 2015 11:33:58 +0100
Subject: ASoC: dapm: Break out of widget search when source and sink are
 located

Currently snd_soc_dapm_add_route will continue to search the widget list
even after both the source and sink for the route have been located.
This patch breaks out of the search when both are located giving a
small improvement in probe time for drivers.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 10 ++++++++--
 1 file changed, 8 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 765416174388..63748526d630 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2617,14 +2617,20 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 	list_for_each_entry(w, &dapm->card->widgets, list) {
 		if (!wsink && !(strcmp(w->name, sink))) {
 			wtsink = w;
-			if (w->dapm == dapm)
+			if (w->dapm == dapm) {
 				wsink = w;
+				if (wsource)
+					break;
+			}
 			continue;
 		}
 		if (!wsource && !(strcmp(w->name, source))) {
 			wtsource = w;
-			if (w->dapm == dapm)
+			if (w->dapm == dapm) {
 				wsource = w;
+				if (wsink)
+					break;
+			}
 		}
 	}
 	/* use widget from another DAPM context if not found from this */
-- 
cgit v1.2.3


From 5f54ea214b2847e48f7d8077892d8f1126810d19 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 May 2015 09:42:26 +0200
Subject: ASoC: 88pm860x: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/88pm860x-codec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index c0b2686a6aac..ee31fa77af7b 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/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);
-- 
cgit v1.2.3


From c59878a4131c6db060d7c8b5074328c64330aa7d Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 May 2015 09:42:27 +0200
Subject: ASoC: ab8500: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ab8500-codec.c | 20 +++++++++++---------
 1 file changed, 11 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 88ca9cb0ce79..c7d243db010a 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/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);
 
-- 
cgit v1.2.3


From 6d701b6dedde988c517a625002dbb865080960e5 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 May 2015 09:42:28 +0200
Subject: ASoC: ak4641: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ak4641.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 3b22b587a820..2d0ff4595ea0 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/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);
-- 
cgit v1.2.3


From 3f36f3c72540a7fae7f0c534176cb123ff0f822f Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 May 2015 09:42:29 +0200
Subject: ASoC: cx20442: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cx20442.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 13041ccf1010..d6f4abbbf8a7 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/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);
-- 
cgit v1.2.3


From 2aff57e3334493c70f25abbecc31c9b36cd2700f Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 May 2015 09:42:30 +0200
Subject: ASoC: es8328: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es8328.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 996e3f4e7343..6a091016e0fc 100644
--- a/sound/soc/codecs/es8328.c
+++ b/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,
-- 
cgit v1.2.3


From 40d62f23b10ecdc997be85a783c0dd40156dea10 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 May 2015 09:42:31 +0200
Subject: ASoC: jz4740: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/jz4740.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index 8425d262e566..9363fdbca9cd 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/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 |
-- 
cgit v1.2.3


From 41b76881371bf7a5f2e63a6224d962a884dba9f4 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 May 2015 09:42:32 +0200
Subject: ASoC: ml26124: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ml26124.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index f1d5778e6599..62dda2488f14 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/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);
-- 
cgit v1.2.3


From 81024b11178e22c1d3ddfbbc2d142fb294e71466 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 May 2015 09:42:33 +0200
Subject: ASoC: uda134x: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/uda134x.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index dbecbc05cf7b..913edf283239 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -477,6 +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;
@@ -525,7 +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);
-- 
cgit v1.2.3


From 9f0617187ac2431e2efe85fccee749e6a31e9725 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 May 2015 09:42:34 +0200
Subject: ASoC: uda1380: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Also drop the unnecessary check at the beginning of the
uda1380_set_bias_level() which compares the current level to the target
level and aborts if they are the same. Since the core will not call the
set_bias_level() callback if we already are in the target state the result
of the check is always false.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/uda1380.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index cc5b1769958a..d708a9c43259 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/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);
-- 
cgit v1.2.3


From 8533eb24a9515c2a9e6779cfd377ab0c46ed8a77 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 May 2015 09:42:35 +0200
Subject: ASoC: sgtl5000: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sgtl5000.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index b01c985a2307..661ed4d22007 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/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);
-- 
cgit v1.2.3


From ca60bc41fb97b8bdda8bba3fdefac6d51ab9ffb4 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 May 2015 09:42:36 +0200
Subject: ASoC: sirf-audio-codec: Replace direct snd_soc_codec dapm field
 access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sirf-audio-codec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c
index 0a8e43c98a07..29cb44256044 100644
--- a/sound/soc/codecs/sirf-audio-codec.c
+++ b/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);
 
-- 
cgit v1.2.3


From 45a110a1377d9f7afbbf53e351b72cf813ac426e Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Mon, 11 May 2015 13:50:30 +0100
Subject: ASoC: dapm: Add cache to speed up adding of routes

Some CODECs have a significant number of DAPM routes and for each route,
when it is added to the card, the entire card widget list must be
searched. When adding routes it is very likely, however, that adjacent
routes will require adjacent widgets. For example all the routes for a
mux are likely added in a block and the sink widget will be the same
each time and it is also quite likely that the source widgets are
sequential located in the widget list.

This patch adds a cache to the DAPM context, this cache will hold the
source and sink widgets from the last call to snd_soc_dapm_add_route for
that context. A small search of the widget list will be made from those
points for both the sink and source. Currently this search only checks
both the last widget and the one adjacent to it.

On wm8280 which has approximately 500 widgets and 30000 routes (one of
the largest CODECs in mainline), the number of paths that hit the cache
is 24000, which significantly improves probe time.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dapm.h |  7 +++++++
 sound/soc/soc-dapm.c     | 39 +++++++++++++++++++++++++++++++++++++++
 2 files changed, 46 insertions(+)

(limited to 'sound')

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 96c5e0ec81d1..b9170e2bc5ab 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -593,6 +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;
@@ -614,6 +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
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 63748526d630..10fb7087c405 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -572,6 +572,35 @@ static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm)
 		snd_soc_component_async_complete(dapm->component);
 }
 
+static struct snd_soc_dapm_widget *
+dapm_wcache_lookup(struct snd_soc_dapm_wcache *wcache, const char *name)
+{
+	struct snd_soc_dapm_widget *w = wcache->widget;
+	struct list_head *wlist;
+	const int depth = 2;
+	int i = 0;
+
+	if (w) {
+		wlist = &w->dapm->card->widgets;
+
+		list_for_each_entry_from(w, wlist, list) {
+			if (!strcmp(name, w->name))
+				return w;
+
+			if (++i == depth)
+				break;
+		}
+	}
+
+	return NULL;
+}
+
+static inline void dapm_wcache_update(struct snd_soc_dapm_wcache *wcache,
+				      struct snd_soc_dapm_widget *w)
+{
+	wcache->widget = w;
+}
+
 /**
  * snd_soc_dapm_force_bias_level() - Sets the DAPM bias level
  * @dapm: The DAPM context for which to set the level
@@ -2610,6 +2639,12 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 		source = route->source;
 	}
 
+	wsource = dapm_wcache_lookup(&dapm->path_source_cache, source);
+	wsink = dapm_wcache_lookup(&dapm->path_sink_cache, sink);
+
+	if (wsink && wsource)
+		goto skip_search;
+
 	/*
 	 * find src and dest widgets over all widgets but favor a widget from
 	 * current DAPM context
@@ -2650,6 +2685,10 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 		return -ENODEV;
 	}
 
+skip_search:
+	dapm_wcache_update(&dapm->path_sink_cache, wsink);
+	dapm_wcache_update(&dapm->path_source_cache, wsource);
+
 	ret = snd_soc_dapm_add_path(dapm, wsource, wsink, route->control,
 		route->connected);
 	if (ret)
-- 
cgit v1.2.3


From 22310d320e352c5dd08b40bcabaefa62e71ed652 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 15 May 2015 12:32:53 +0200
Subject: ASoC: sn95031: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

While we are at it also replace the if(x == A) ... else if(x == B) ...
construct with a switch-case.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sn95031.c | 11 ++++++++---
 1 file changed, 8 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index e4743684cc1d..3a7de0159f24 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/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;
 
-- 
cgit v1.2.3


From edc20cadcccc19953fe0ab117d854af04f4d9c8c Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 15 May 2015 12:32:54 +0200
Subject: ASoC: lm49453: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/lm49453.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index 166fd4c88ddb..6600aa0a33dc 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/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,
-- 
cgit v1.2.3


From b6070c11cdaa31ecc5273bdfb8d2239026f10a90 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 15 May 2015 12:32:55 +0200
Subject: ASoC: pcm512x: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/pcm512x.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index c305b2871c59..de16429f0a43 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/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;
-- 
cgit v1.2.3


From 378d1e432d9b1504d7ced936837e66dd7d246d45 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 15 May 2015 12:32:56 +0200
Subject: ASoC: tlv320aix31xx: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic31xx.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index e629273019d0..c4c960f592a1 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/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;
@@ -1049,7 +1049,7 @@ 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;
 	}
-- 
cgit v1.2.3


From 650a18acacf431cf979a49c904028afe636de6b9 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 15 May 2015 12:32:57 +0200
Subject: ASoC: tlv320aic3x: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic3x.c | 9 +++++----
 1 file changed, 5 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 57d709075746..a7cf19b53fb2 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/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,
-- 
cgit v1.2.3


From 37e931c17926c4a5268afa134be9d4a09c230e06 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 15 May 2015 12:32:58 +0200
Subject: ASoC: tlv320dac33: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320dac33.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 33e93f62de30..d67a311f0e75 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/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)
@@ -644,7 +644,7 @@ 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)
-- 
cgit v1.2.3


From 1682c8e5708ecbd9409123877784e82ca9557588 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 15 May 2015 12:32:59 +0200
Subject: ASoC: twl4030: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/twl4030.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index e725e13a7f59..90f5f04eca2d 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1588,7 +1588,7 @@ 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:
-- 
cgit v1.2.3


From d9dd37305e9d230856e851ea720eaba68d92a252 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 15 May 2015 12:33:00 +0200
Subject: ASoC: twl6040: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/twl6040.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index b8ecce206af8..9db7408f6e05 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/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 */
-- 
cgit v1.2.3


From b8faaba4a655d58b67ba28598c22a48aa844b489 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 15 May 2015 12:41:30 +0200
Subject: ASoC: Drop unnecessary bias level check on resume

The suspended flag will only be set if the CODEC bias level was either
STANDBY or OFF. This means we don't need to check for that on resume since
the condition will always be true.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 19 +++----------------
 1 file changed, 3 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 23732523f87c..95b5f034d864 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -750,23 +750,10 @@ static void soc_resume_deferred(struct work_struct *work)
 	}
 
 	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
-		/* If the CODEC was idle over suspend then it will have been
-		 * left with bias OFF or STANDBY and suspended so we must now
-		 * resume.  Otherwise the suspend was suppressed.
-		 */
 		if (codec->suspended) {
-			switch (codec->dapm.bias_level) {
-			case SND_SOC_BIAS_STANDBY:
-			case SND_SOC_BIAS_OFF:
-				if (codec->driver->resume)
-					codec->driver->resume(codec);
-				codec->suspended = 0;
-				break;
-			default:
-				dev_dbg(codec->dev,
-					"ASoC: CODEC was on over suspend\n");
-				break;
-			}
+			if (codec->driver->resume)
+				codec->driver->resume(codec);
+			codec->suspended = 0;
 		}
 	}
 
-- 
cgit v1.2.3


From cc740ec84dc888144fe31b44e3af7ad467ccfc70 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 14 May 2015 11:19:53 +0200
Subject: ASoC: sta32x: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sta32x.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 033b7d9f45f7..ffe6187dce85 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/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) {
-- 
cgit v1.2.3


From ea85b45b11b38ecc81edd1e42d22e2f7155db57a Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 14 May 2015 11:19:54 +0200
Subject: ASoC: sta350: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sta350.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c
index 50d8bbf90ce2..025f6639330e 100644
--- a/sound/soc/codecs/sta350.c
+++ b/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);
-- 
cgit v1.2.3


From 95fcb384e6738bcc37b4f7bf6d1272aba4e7d2b9 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 14 May 2015 11:19:55 +0200
Subject: ASoC: sta529: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sta529.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index c3217af1ca29..4f70378b2cfb 100644
--- a/sound/soc/codecs/sta529.c
+++ b/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);
-- 
cgit v1.2.3


From 9c414c62461d09e6dd64887a3db793b5163d82c0 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 14 May 2015 11:19:56 +0200
Subject: ASoC: da7213: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/da7213.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 925dd3c16d6c..238e48a3a4fe 100644
--- a/sound/soc/codecs/da7213.c
+++ b/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,
-- 
cgit v1.2.3


From a3ea8a66f64ace02f91006f06fe904be5780b7d1 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 14 May 2015 11:19:57 +0200
Subject: ASoC: da732x: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/da732x.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index 06519057bdff..207523686bd5 100644
--- a/sound/soc/codecs/da732x.c
+++ b/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);
-- 
cgit v1.2.3


From ed3347e83cce7edf6cd9b5e530b9da11908d2f83 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 14 May 2015 11:19:58 +0200
Subject: ASoC: da9055: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/da9055.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index 3bdc95a70112..66bb446473b8 100644
--- a/sound/soc/codecs/da9055.c
+++ b/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,
-- 
cgit v1.2.3


From 0fbcbef98d2209fde25463f12c8b9ca07f750974 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 14 May 2015 11:19:59 +0200
Subject: ASoC: max98088: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98088.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 3200aa80f1f2..d0f45348bfbb 100644
--- a/sound/soc/codecs/max98088.c
+++ b/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,
-- 
cgit v1.2.3


From b0b80c8075add488ca2632393670da31b174195d Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 14 May 2015 11:20:00 +0200
Subject: ASoC: max98090: Remove unnecessary snd_soc_dapm_sync()

max98090_jack_work() doesn't modify the DAPM graph other than what's done
in snd_soc_jack_report(). snd_soc_jack_report() already calls
snd_soc_dapm_sync() internally, so there is no need to call it manually
and can be removed.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98090.c | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index c5736b2f7c76..5a0bd8a0c9e9 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2186,7 +2186,6 @@ 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;
 
@@ -2265,8 +2264,6 @@ 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)
-- 
cgit v1.2.3


From 29ca43bc548e1b0060c8426b98a2ce9601cd5a17 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 14 May 2015 11:20:01 +0200
Subject: ASoC: max98090: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and all other
manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98090.c | 17 +++++++++--------
 1 file changed, 9 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 5a0bd8a0c9e9..c2306268cab8 100644
--- a/sound/soc/codecs/max98090.c
+++ b/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,
-- 
cgit v1.2.3


From 1179a3685022e954b0de1df12b5711229918c4ae Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 14 May 2015 11:20:02 +0200
Subject: ASoC: max98095: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98095.c | 15 ++++++++-------
 1 file changed, 8 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 66c7ca431a2e..2b8b8a5f385f 100644
--- a/sound/soc/codecs/max98095.c
+++ b/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) {
-- 
cgit v1.2.3


From 3054716d4ff720378cda96dbafcd87e99164782c Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 14 May 2015 11:20:03 +0200
Subject: ASoC: max9850: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max9850.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index f6b616b6ffca..481d58f1cb3f 100644
--- a/sound/soc/codecs/max9850.c
+++ b/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,
-- 
cgit v1.2.3


From ff344dcd80cf8a27eb6cb9b38d810fe5e1b6c34f Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 19 May 2015 21:49:05 +0200
Subject: ASoC: alc5623: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/alc5623.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index e92b5ae3cab2..0fc24e0d518c 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -893,7 +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);
 
-- 
cgit v1.2.3


From b1cd8457dadd52bdd3e38c6f34b5465f4430b34f Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 19 May 2015 21:49:06 +0200
Subject: ASoC: rt286: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt286.c | 32 +++++++++++++++++---------------
 1 file changed, 17 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index dbdbb9e8d4ba..c6cca0639e0d 100644
--- a/sound/soc/codecs/rt286.c
+++ b/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,
-- 
cgit v1.2.3


From 61aad0b91c537436932d399b14d1e9412e58c438 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 19 May 2015 21:49:07 +0200
Subject: ASoC: rt5631: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and
snd_soc_codec_init_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5631.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index e285d8ad260a..058167c80d71 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/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);
@@ -1614,7 +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;
 }
-- 
cgit v1.2.3


From 76aad74bdd050037bd28a02a56c30460532cdce6 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 19 May 2015 21:49:08 +0200
Subject: ASoC: rt5640: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5640.c | 11 ++++++-----
 1 file changed, 6 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 7d488d8b03d6..f40752a6c242 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/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,
@@ -1934,6 +1934,7 @@ 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;
@@ -1950,18 +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;
-- 
cgit v1.2.3


From eb13bd563a21c34696c942690586e64389b3e054 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 19 May 2015 21:49:09 +0200
Subject: ASoC: rt5651: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5651.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index f03c6fc1a7e9..a3506e193abc 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/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,
-- 
cgit v1.2.3


From 6d8135ff00385c6b5149e19615c031ab3021df04 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 19 May 2015 21:49:10 +0200
Subject: ASoC: rt5670: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5670.c | 25 +++++++++++++------------
 1 file changed, 13 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 9235711e86c2..840dd6d0003a 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/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,
@@ -2653,23 +2653,24 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec,
 
 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;
-- 
cgit v1.2.3


From 6b43c2eb9a7907c3e7ab9210ff6c62322d81e18c Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 19 May 2015 21:49:11 +0200
Subject: ASoC: rt5677: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5677.c | 11 ++++++-----
 1 file changed, 6 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 696ba587969e..c0211a1187a5 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -817,7 +817,7 @@ 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;
@@ -2476,7 +2476,7 @@ 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,
@@ -4350,7 +4350,7 @@ 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,
@@ -4602,17 +4602,18 @@ 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));
 	}
-- 
cgit v1.2.3


From 02aa78abec6ebe2ae4a2ec0687758a4e58ee9507 Mon Sep 17 00:00:00 2001
From: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Date: Mon, 25 May 2015 18:21:17 +0100
Subject: ASoC: DAPM: Add APIs to create individual DAPM controls.

The topology core needs to be able to create individual widget controls at
runtime and driver init. Add a regular locked and unlocked API calls to
facilitate this requirement.

The unlocked call is used by the topology core during component driver
probing where the card dapm_mutex is held by the ASoC core and the locked
version at non component driver probe time.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 34 ++++++++++++++++++++++++++--------
 1 file changed, 26 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 10fb7087c405..3c53db0034ef 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -52,8 +52,8 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 	const char *control,
 	int (*connected)(struct snd_soc_dapm_widget *source,
 			 struct snd_soc_dapm_widget *sink));
-static struct snd_soc_dapm_widget *
-snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
+struct snd_soc_dapm_widget *
+snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
 			 const struct snd_soc_dapm_widget *widget);
 
 /* dapm power sequences - make this per codec in the future */
@@ -350,7 +350,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 
 			data->value = template.on_val;
 
-			data->widget = snd_soc_dapm_new_control(widget->dapm,
+			data->widget =
+				snd_soc_dapm_new_control_unlocked(widget->dapm,
 				&template);
 			if (!data->widget) {
 				ret = -ENOMEM;
@@ -3264,8 +3265,25 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch);
 
-static struct snd_soc_dapm_widget *
+struct snd_soc_dapm_widget *
 snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
+	const struct snd_soc_dapm_widget *widget)
+{
+	struct snd_soc_dapm_widget *w;
+
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+	w = snd_soc_dapm_new_control_unlocked(dapm, widget);
+	if (!w)
+		dev_err(dapm->dev,
+			"ASoC: Failed to create DAPM control %s\n",
+			widget->name);
+
+	mutex_unlock(&dapm->card->dapm_mutex);
+	return w;
+}
+
+struct snd_soc_dapm_widget *
+snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
 			 const struct snd_soc_dapm_widget *widget)
 {
 	struct snd_soc_dapm_widget *w;
@@ -3411,7 +3429,7 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
 
 	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
 	for (i = 0; i < num; i++) {
-		w = snd_soc_dapm_new_control(dapm, widget);
+		w = snd_soc_dapm_new_control_unlocked(dapm, widget);
 		if (!w) {
 			dev_err(dapm->dev,
 				"ASoC: Failed to create DAPM control %s\n",
@@ -3649,7 +3667,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
 
 	dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name);
 
-	w = snd_soc_dapm_new_control(&card->dapm, &template);
+	w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template);
 	if (!w) {
 		dev_err(card->dev, "ASoC: Failed to create %s widget\n",
 			link_name);
@@ -3700,7 +3718,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 		dev_dbg(dai->dev, "ASoC: adding %s widget\n",
 			template.name);
 
-		w = snd_soc_dapm_new_control(dapm, &template);
+		w = snd_soc_dapm_new_control_unlocked(dapm, &template);
 		if (!w) {
 			dev_err(dapm->dev, "ASoC: Failed to create %s widget\n",
 				dai->driver->playback.stream_name);
@@ -3719,7 +3737,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 		dev_dbg(dai->dev, "ASoC: adding %s widget\n",
 			template.name);
 
-		w = snd_soc_dapm_new_control(dapm, &template);
+		w = snd_soc_dapm_new_control_unlocked(dapm, &template);
 		if (!w) {
 			dev_err(dapm->dev, "ASoC: Failed to create %s widget\n",
 				dai->driver->capture.stream_name);
-- 
cgit v1.2.3


From 43a0350f2122f24c3af21ff65574eba84fad13e4 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:20 +0200
Subject: ASoC: cs42l52: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l52.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 3c49a756b89b..4de52c9957ac 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/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);
 		}
@@ -955,7 +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;
-- 
cgit v1.2.3


From 46a35b0d4d26090aedc0e72ac701d0f5304e29a0 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:21 +0200
Subject: ASoC: cs42l56: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l56.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index a7638c52b509..1e11ba45a79f 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/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),
@@ -1025,7 +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;
-- 
cgit v1.2.3


From 353c10a91964a2dfde77224a284abf55d0856da1 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:22 +0200
Subject: ASoC: cs42l73: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42l73.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 156ec938f441..b7853b9d3a60 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/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);
 		}
-- 
cgit v1.2.3


From 02b8c59adedff17a3003a93f3cc395eb6e0d6e8c Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:23 +0200
Subject: ASoC: cs42xx8: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42xx8.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
index 670ebfe12903..e1d46862e81f 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/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:
-- 
cgit v1.2.3


From 1ac52145053bdddc0c831e11e8b220a958c10741 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:24 +0200
Subject: ASoC: arizona: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/arizona.c | 27 ++++++++++++++-------------
 1 file changed, 14 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index eff4b4d512b7..0cb2962ddb9e 100644
--- a/sound/soc/codecs/arizona.c
+++ b/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)
-- 
cgit v1.2.3


From e566b53251fb394501830397e82b5eb46841f36a Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:25 +0200
Subject: ASoC: wm0010: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm0010.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index 3358dd6811fa..6560a66b3f35 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/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);
-- 
cgit v1.2.3


From 002d1c4ed8ca319d638247250ebf3261d92f4e16 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:26 +0200
Subject: ASoC: wm5100: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm5100.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 96740379b711..98495dd61239 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/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) {
-- 
cgit v1.2.3


From 0740135a53f04cce7894e1751b34fe660d948cd1 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:27 +0200
Subject: ASoC: wm5102: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm5102.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 0c6d1bc0526e..f11523fc5bd0 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/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;
 }
-- 
cgit v1.2.3


From 72945b3d3c78ab2babeb8ed8f00c18441f417bb9 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:28 +0200
Subject: ASoC: wm5110: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().

While we are at it also remove the duplicated initialization of
priv->core.arizona->dapm.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm5110.c | 7 +++----
 1 file changed, 3 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index fbaeddb3e903..67960009f0c4 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/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;
 }
-- 
cgit v1.2.3


From 9b142894bec491e16d011733d4115855b5e47dd0 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:29 +0200
Subject: ASoC: wm8350: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8350.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index dd0d0248e641..41c62c1e62db 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/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)
-- 
cgit v1.2.3


From cf25c66c5b699abcd7a3a5862d42df85b346d148 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:30 +0200
Subject: ASoC: wm8400: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8400.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index adbfebe04c77..d7555085e7f4 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/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) {
-- 
cgit v1.2.3


From 38337a9df28464eac07e7df842ffafeb23a9c528 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:31 +0200
Subject: ASoC: wm8510: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8510.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index a380c10e867b..dac5beb4d023 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/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 */
-- 
cgit v1.2.3


From 7db634d918ca72307c4e7445420bc41b94c72847 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:32 +0200
Subject: ASoC: wm8523: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8523.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 34ebe95d93f1..8c5b9df3e542 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/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) {
-- 
cgit v1.2.3


From e8f48bc8cb7e714c565784b0039fdb88a1f3de76 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:33 +0200
Subject: ASoC: wm8580: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8580.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 5951d88e3dc9..759a7928ac3e 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/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 |
-- 
cgit v1.2.3


From f235d94fcadf185995d7ca57b7c2ae45879e2fe8 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:34 +0200
Subject: ASoC: wm8711: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8711.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index a4aab6e7f5cc..cc8251f09f8a 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/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);
-- 
cgit v1.2.3


From 5e80bb92f268078a946bccbb97983f040a128d3e Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:35 +0200
Subject: ASoC: wm8728: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8728.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index a737068d5576..f1a173e6ec33 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/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);
-- 
cgit v1.2.3


From fc31fda63b54c7b9574f983a95124abb8474ce0a Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:36 +0200
Subject: ASoC: wm8731: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8731.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index a13a20ac47af..915ea11ad4b6 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/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)
-- 
cgit v1.2.3


From 11fb3914bd7714971f4ec498ae325d327c6b8f47 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:37 +0200
Subject: ASoC: wm8737: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8737.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index 4a9407dadae3..ff4c8e979e01 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/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) {
-- 
cgit v1.2.3


From b31c9ef9f8627d7591bd2248e21cdf4347ad5e72 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:38 +0200
Subject: ASoC: wm8750: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8750.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index d6ff25a9d5af..56d89b0865fa 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/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 */
-- 
cgit v1.2.3


From 6093e926cc17c6b5da486a85e9f91bd1e70b45fa Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:39 +0200
Subject: ASoC: wm8753: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8753.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index b7d38f7ba636..feb2997a377a 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/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,
-- 
cgit v1.2.3


From ef075ca657e1fea66efbce32ff6947f82b3dc9e2 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:40 +0200
Subject: ASoC: wm8770: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8770.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index c24db8037201..66c1f151071d 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/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) {
-- 
cgit v1.2.3


From 265b8ac8b0ac8dd52b81f97615903530de1c750a Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:41 +0200
Subject: ASoC: wm8776: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8776.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index b0e3c3bbd440..ece9b4456767 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/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 */
-- 
cgit v1.2.3


From e7556037687be97396f1c610dd8cfb78d94fbc92 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:42 +0200
Subject: ASoC: wm8804: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8804.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 1e403f67cf16..c195c2e8af07 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/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;
-- 
cgit v1.2.3


From eee53c35bb0c30340489272412ac7e81ede7da59 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:43 +0200
Subject: ASoC: wm8900: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8900.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index e7d2ecd150cf..ecc7b4703617 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/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);
-- 
cgit v1.2.3


From 060ea2a0bda408c102421de3d2c645bacb772143 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:44 +0200
Subject: ASoC: wm8903: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8903.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 5e0bef62d974..b5322c1544fb 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/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 |
-- 
cgit v1.2.3


From f44a9842931a952829b114f846d603e93688a8d3 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:45 +0200
Subject: ASoC: wm8904: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8904.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index a7a8fa0567b1..265a4a58a2d1 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/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) {
-- 
cgit v1.2.3


From 2145554fea759a303d31f64e5befc50996f42dd0 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:46 +0200
Subject: ASoC: wm8940: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8940.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index f2d6a490713f..98ef0ba5c2a4 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/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);
-- 
cgit v1.2.3


From afcd11df6d5acb03339ed96c21e219c510e0de46 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:47 +0200
Subject: ASoC: wm8955: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8955.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index f400d5c7234c..3a5bf894ff6d 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/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) {
-- 
cgit v1.2.3


From 93f32f534e0fcbb5cad0734e599e960454caa303 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:48 +0200
Subject: ASoC: wm8960: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8960.c | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 6fa832b6365b..edd34db9bd25 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/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 */
@@ -705,7 +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,
@@ -776,7 +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,
-- 
cgit v1.2.3


From 049e17d7b9c82f2cc1171f7b2c32f0fe9e9fc6d1 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:49 +0200
Subject: ASoC: wm8961: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8961.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 6f95d7044aac..a057662632ff 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/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;
-- 
cgit v1.2.3


From 57ef7fa7b2c499ad1aece50b368679fe90fe348f Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:50 +0200
Subject: ASoC: wm8962: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Also drop the unnecessary comparison in the set_bias_level() callback that
checks if the device is already at the target level. The core already takes
care of this and will not call the callback if the device is already at the
target level.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8962.c | 20 +++++++++-----------
 1 file changed, 9 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 00793b7b0a83..c5748fd4f296 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/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;
 
@@ -2613,7 +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;
@@ -3117,7 +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;
@@ -3163,7 +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;
@@ -3414,6 +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;
@@ -3461,7 +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");
-- 
cgit v1.2.3


From 19773614205be8a60efa50b180758307ad6f16bf Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:51 +0200
Subject: ASoC: wm8971: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8971.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 94eb27ec572f..b51184c4e816 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/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);
-- 
cgit v1.2.3


From 5c6415d630a6c6b6b1e70aaaf4dba1062bcf8b7c Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:52 +0200
Subject: ASoC: wm8974: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8974.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index d2180c83a5cc..33b16a7ba82e 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/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 */
-- 
cgit v1.2.3


From 547f3f47f541faffa6b2dcec363730999e97445d Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:53 +0200
Subject: ASoC: wm8978: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8978.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index e2363b9a38a0..cfc8cdf49970 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/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);
-- 
cgit v1.2.3


From 71ffce008c0e78e66c357894725c7934fa81d0eb Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:54 +0200
Subject: ASoC: wm8983: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8983.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
index f9245715cebd..2fdd2c6cc09d 100644
--- a/sound/soc/codecs/wm8983.c
+++ b/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);
-- 
cgit v1.2.3


From 4b67780291bef6b7efc4046630f0ab4b8cf06584 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:55 +0200
Subject: ASoC: wm8985: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8985.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index 4e6901b5c819..8a85f5004d41 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/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) {
-- 
cgit v1.2.3


From 491c04eb86c768f61678fe7169ff13ce57dadef9 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:56 +0200
Subject: ASoC: wm8988: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8988.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 92680c6d247e..f13a995af277 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/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 */
-- 
cgit v1.2.3


From 015ff301935425e1f00194fd3af8fc356cc78c14 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:57 +0200
Subject: ASoC: wm8990: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8990.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index ff377cab5775..1993fd2a6f15 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/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);
-- 
cgit v1.2.3


From bfdd20a4979a5815e4175c896dd7f0ad63bc78db Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:58 +0200
Subject: ASoC: wm8991: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8991.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index abd439fb0820..44a677720828 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/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 |
-- 
cgit v1.2.3


From f8ae3cf81fb866a0d91e3319f53d6ed0a599616e Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:10:59 +0200
Subject: ASoC: wm8993: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8993.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 52ec4fe03b23..8a8db8605dc2 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/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)
@@ -1483,7 +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;
@@ -1537,7 +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;
 
-- 
cgit v1.2.3


From 8e09bac78a48f738f3a180fe213198ab225c807e Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:11:00 +0200
Subject: ASoC: wm8994: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8994.c | 58 +++++++++++++++++++++++------------------------
 1 file changed, 29 insertions(+), 29 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 2d32b542f103..99a758a54986 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -212,6 +212,7 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif)
 
 static int configure_clock(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int change, new;
 
@@ -239,7 +240,7 @@ static int configure_clock(struct snd_soc_codec *codec)
 	change = snd_soc_update_bits(codec, WM8994_CLOCKING_1,
 				     WM8994_SYSCLK_SRC, new);
 	if (change)
-		snd_soc_dapm_sync(&codec->dapm);
+		snd_soc_dapm_sync(dapm);
 
 	wm8958_micd_set_rate(codec);
 
@@ -2492,12 +2493,12 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
 			break;
 		}
 
-		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY)
 			active_reference(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) {
 			switch (control->type) {
 			case WM8958:
 				if (control->revision == 0) {
@@ -2521,7 +2522,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
 					    WM8994_LINEOUT2_DISCH);
 		}
 
-		if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_PREPARE)
 			active_dereference(codec);
 
 		/* MICBIAS into bypass mode on newer devices */
@@ -2541,7 +2542,7 @@ static int wm8994_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)
 			wm8994->cur_fw = NULL;
 		break;
 	}
@@ -2552,7 +2553,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
 int wm8994_vmid_mode(struct snd_soc_codec *codec, enum wm8994_vmid_mode mode)
 {
 	struct wm8994_priv *wm8994 = 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 (mode) {
 	case WM8994_VMID_NORMAL:
@@ -3354,6 +3355,7 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994)
 int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 		      int micbias)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	struct wm8994_micdet *micdet;
 	struct wm8994 *control = wm8994->wm8994;
@@ -3368,20 +3370,16 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 	case 1:
 		micdet = &wm8994->micdet[0];
 		if (jack)
-			ret = snd_soc_dapm_force_enable_pin(&codec->dapm,
-							    "MICBIAS1");
+			ret = snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1");
 		else
-			ret = snd_soc_dapm_disable_pin(&codec->dapm,
-						       "MICBIAS1");
+			ret = snd_soc_dapm_disable_pin(dapm, "MICBIAS1");
 		break;
 	case 2:
 		micdet = &wm8994->micdet[1];
 		if (jack)
-			ret = snd_soc_dapm_force_enable_pin(&codec->dapm,
-							    "MICBIAS1");
+			ret = snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1");
 		else
-			ret = snd_soc_dapm_disable_pin(&codec->dapm,
-						       "MICBIAS1");
+			ret = snd_soc_dapm_disable_pin(dapm, "MICBIAS1");
 		break;
 	default:
 		dev_warn(codec->dev, "Invalid MICBIAS %d\n", micbias);
@@ -3413,7 +3411,7 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 			    WM8994_MIC2_DET_DB_MASK | WM8994_MIC2_SHRT_DB_MASK,
 			    WM8994_MIC1_DET_DB | WM8994_MIC1_SHRT_DB);
 
-	snd_soc_dapm_sync(&codec->dapm);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
@@ -3503,6 +3501,7 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data)
 /* Should be called with accdet_lock held */
 static void wm1811_micd_stop(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 
 	if (!wm8994->jackdet)
@@ -3513,8 +3512,7 @@ static void wm1811_micd_stop(struct snd_soc_codec *codec)
 	wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_JACK);
 
 	if (wm8994->wm8994->pdata.jd_ext_cap)
-		snd_soc_dapm_disable_pin(&codec->dapm,
-					 "MICBIAS2");
+		snd_soc_dapm_disable_pin(dapm, "MICBIAS2");
 }
 
 static void wm8958_button_det(struct snd_soc_codec *codec, u16 status)
@@ -3623,14 +3621,14 @@ static void wm1811_mic_work(struct work_struct *work)
 						  mic_work.work);
 	struct wm8994 *control = wm8994->wm8994;
 	struct snd_soc_codec *codec = wm8994->hubs.codec;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	pm_runtime_get_sync(codec->dev);
 
 	/* If required for an external cap force MICBIAS on */
 	if (control->pdata.jd_ext_cap) {
-		snd_soc_dapm_force_enable_pin(&codec->dapm,
-					      "MICBIAS2");
-		snd_soc_dapm_sync(&codec->dapm);
+		snd_soc_dapm_force_enable_pin(dapm, "MICBIAS2");
+		snd_soc_dapm_sync(dapm);
 	}
 
 	mutex_lock(&wm8994->accdet_lock);
@@ -3662,6 +3660,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
 	struct wm8994_priv *wm8994 = data;
 	struct wm8994 *control = wm8994->wm8994;
 	struct snd_soc_codec *codec = wm8994->hubs.codec;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	int reg, delay;
 	bool present;
 
@@ -3722,7 +3721,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
 
 	/* Turn off MICBIAS if it was on for an external cap */
 	if (control->pdata.jd_ext_cap && !present)
-		snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2");
+		snd_soc_dapm_disable_pin(dapm, "MICBIAS2");
 
 	if (present)
 		snd_soc_jack_report(wm8994->micdet[0].jack,
@@ -3768,6 +3767,7 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 		      wm1811_micdet_cb det_cb, void *det_cb_data,
 		      wm1811_mic_id_cb id_cb, void *id_cb_data)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	struct wm8994 *control = wm8994->wm8994;
 	u16 micd_lvl_sel;
@@ -3781,8 +3781,8 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 	}
 
 	if (jack) {
-		snd_soc_dapm_force_enable_pin(&codec->dapm, "CLK_SYS");
-		snd_soc_dapm_sync(&codec->dapm);
+		snd_soc_dapm_force_enable_pin(dapm, "CLK_SYS");
+		snd_soc_dapm_sync(dapm);
 
 		wm8994->micdet[0].jack = jack;
 
@@ -3817,7 +3817,7 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 		snd_soc_update_bits(codec, WM8958_MIC_DETECT_2,
 				    WM8958_MICD_LVL_SEL_MASK, micd_lvl_sel);
 
-		WARN_ON(codec->dapm.bias_level > SND_SOC_BIAS_STANDBY);
+		WARN_ON(snd_soc_codec_get_bias_level(codec) > SND_SOC_BIAS_STANDBY);
 
 		/*
 		 * If we can use jack detection start off with that,
@@ -3844,8 +3844,8 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 		snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
 				    WM8958_MICD_ENA, 0);
 		wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_NONE);
-		snd_soc_dapm_disable_pin(&codec->dapm, "CLK_SYS");
-		snd_soc_dapm_sync(&codec->dapm);
+		snd_soc_dapm_disable_pin(dapm, "CLK_SYS");
+		snd_soc_dapm_sync(dapm);
 	}
 
 	return 0;
@@ -3983,9 +3983,9 @@ static irqreturn_t wm8994_temp_shut(int irq, void *data)
 
 static int wm8994_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct wm8994 *control = dev_get_drvdata(codec->dev->parent);
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	unsigned int reg;
 	int ret, i;
 
@@ -4016,7 +4016,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	wm8994->micdet_irq = control->pdata.micdet_irq;
 
 	/* By default use idle_bias_off, will override for WM8994 */
-	codec->dapm.idle_bias_off = 1;
+	dapm->idle_bias_off = 1;
 
 	/* Set revision-specific configuration */
 	switch (control->type) {
@@ -4024,7 +4024,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		/* Single ended line outputs should have VMID on. */
 		if (!control->pdata.lineout1_diff ||
 		    !control->pdata.lineout2_diff)
-			codec->dapm.idle_bias_off = 0;
+			dapm->idle_bias_off = 0;
 
 		switch (control->revision) {
 		case 2:
-- 
cgit v1.2.3


From a01ddd388d4789af6124889d11cc27f6263a9af1 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:11:01 +0200
Subject: ASoC: wm8995: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8995.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 47af27fb339a..687c4dd7ec99 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -721,6 +721,7 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif)
 
 static int configure_clock(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct wm8995_priv *wm8995;
 	int change, new;
 
@@ -751,7 +752,7 @@ static int configure_clock(struct snd_soc_codec *codec)
 	if (!change)
 		return 0;
 
-	snd_soc_dapm_sync(&codec->dapm);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
@@ -1965,7 +1966,7 @@ static int wm8995_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(wm8995->supplies),
 						    wm8995->supplies);
 			if (ret)
-- 
cgit v1.2.3


From 6a141e462ef878fd395e838d8bfc2624104dc66c Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:11:02 +0200
Subject: ASoC: wm8996: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8996.c | 15 +++++++++------
 1 file changed, 9 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 3dce50751469..370459fcf21c 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1590,7 +1590,7 @@ static int wm8996_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(wm8996->supplies),
 						    wm8996->supplies);
 			if (ret != 0) {
@@ -2245,7 +2245,7 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 		  wm8996_polarity_fn polarity_cb)
 {
 	struct wm8996_priv *wm8996 = 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);
 
 	wm8996->jack = jack;
 	wm8996->detecting = true;
@@ -2290,6 +2290,7 @@ EXPORT_SYMBOL_GPL(wm8996_detect);
 
 static void wm8996_hpdet_irq(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
 	int val, reg, report;
 
@@ -2343,12 +2344,14 @@ out:
 	snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA,
 			    WM8996_MICD_ENA);
 
-	snd_soc_dapm_disable_pin(&codec->dapm, "Bandgap");
-	snd_soc_dapm_sync(&codec->dapm);
+	snd_soc_dapm_disable_pin(dapm, "Bandgap");
+	snd_soc_dapm_sync(dapm);
 }
 
 static void wm8996_hpdet_start(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+
 	/* Unclamp the output, we can't measure while we're shorting it */
 	snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1,
 			    WM8996_HPOUT1L_RMV_SHORT |
@@ -2357,8 +2360,8 @@ static void wm8996_hpdet_start(struct snd_soc_codec *codec)
 			    WM8996_HPOUT1R_RMV_SHORT);
 
 	/* We need bandgap for HPDET */
-	snd_soc_dapm_force_enable_pin(&codec->dapm, "Bandgap");
-	snd_soc_dapm_sync(&codec->dapm);
+	snd_soc_dapm_force_enable_pin(dapm, "Bandgap");
+	snd_soc_dapm_sync(dapm);
 
 	/* Go into headphone detect left mode */
 	snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, 0);
-- 
cgit v1.2.3


From 8383dfd8893a8d4413549d03c55f3c337d6b8f1d Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:11:03 +0200
Subject: ASoC: wm8997: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8997.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c
index a4d11770630c..e9c4a9f35392 100644
--- a/sound/soc/codecs/wm8997.c
+++ b/sound/soc/codecs/wm8997.c
@@ -1055,13 +1055,14 @@ static struct snd_soc_dai_driver wm8997_dai[] = {
 
 static int wm8997_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec);
 
 	arizona_init_spk(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;
 }
-- 
cgit v1.2.3


From 1571f6ecfdb4890a2ba13a6c920694d589b015bb Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:11:04 +0200
Subject: ASoC: wm9081: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm9081.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 02d9a5012c1b..8a8b1c0f9142 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -838,7 +838,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_STANDBY:
 		/* Initial cold start */
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_cache_only(wm9081->regmap, false);
 			regcache_sync(wm9081->regmap);
 
-- 
cgit v1.2.3


From 718e23fde529cf7f4f945606217e49c5f2e31537 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:11:05 +0200
Subject: ASoC: wm9090: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm9090.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 03bca8581bc7..13d23fc797db 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -425,7 +425,7 @@ static const struct snd_soc_dapm_route audio_map_in2_diff[] = {
 static int wm9090_add_controls(struct snd_soc_codec *codec)
 {
 	struct wm9090_priv *wm9090 = 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 i;
 
 	snd_soc_dapm_new_controls(dapm, wm9090_dapm_widgets,
@@ -496,7 +496,7 @@ static int wm9090_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) {
 			/* Restore the register cache */
 			regcache_sync(wm9090->regmap);
 		}
-- 
cgit v1.2.3


From 4a6c2aa19d5b6dcd6078d1e0db2a88407b926ded Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 1 Jun 2015 10:11:06 +0200
Subject: ASoC: wm_hubs: Replace direct snd_soc_codec dapm field access

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm_hubs.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 8366e19657a7..fd86bd105460 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -1116,7 +1116,7 @@ static const struct snd_soc_dapm_route lineout2_se_routes[] = {
 
 int wm_hubs_add_analogue_controls(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);
 
 	/* Latch volume update bits & default ZC on */
 	snd_soc_update_bits(codec, WM8993_LEFT_LINE_INPUT_1_2_VOLUME,
@@ -1160,7 +1160,7 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec,
 				int lineout1_diff, int lineout2_diff)
 {
 	struct wm_hubs_data *hubs = 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);
 
 	hubs->codec = codec;
 
-- 
cgit v1.2.3


From 5353f65b859255a07e8bf5c096be4d5d268b46e8 Mon Sep 17 00:00:00 2001
From: Vladimir Zapolskiy <vz@mleia.com>
Date: Tue, 2 Jun 2015 00:57:53 +0300
Subject: ASoC: dapm: fix snd_soc_dapm_new_control() implicit declaration

The change fixes the following compilation problem:

  sound/soc/soc-dapm.c: In function 'dapm_kcontrol_data_alloc':
  sound/soc/soc-dapm.c:388:4: error: implicit declaration of function
    'snd_soc_dapm_new_control' [-Werror=implicit-function-declaration]
    data->widget = snd_soc_dapm_new_control(widget->dapm,
    ^

  sound/soc/soc-dapm.c:387:17: warning: assignment makes pointer
    from integer without a cast [enabled by default]
    data->widget = snd_soc_dapm_new_control(widget->dapm,
                 ^
  sound/soc/soc-dapm.c: At top level:
  sound/soc/soc-dapm.c:3269:1: error: conflicting types for
    'snd_soc_dapm_new_control'
  snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
  ^

In addition to the fix add static qualifier to
snd_soc_dapm_new_control() function to silence checkpatch.

Fixes: 02aa78abec ("ASoC: DAPM: Add APIs to create individual DAPM controls.")
Signed-off-by: Vladimir Zapolskiy <vz@mleia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 7 ++++++-
 1 file changed, 6 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 3c53db0034ef..92d57a952bd9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -52,6 +52,11 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 	const char *control,
 	int (*connected)(struct snd_soc_dapm_widget *source,
 			 struct snd_soc_dapm_widget *sink));
+
+static struct snd_soc_dapm_widget *
+snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
+			 const struct snd_soc_dapm_widget *widget);
+
 struct snd_soc_dapm_widget *
 snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
 			 const struct snd_soc_dapm_widget *widget);
@@ -3265,7 +3270,7 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch);
 
-struct snd_soc_dapm_widget *
+static struct snd_soc_dapm_widget *
 snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 	const struct snd_soc_dapm_widget *widget)
 {
-- 
cgit v1.2.3


From cc76e7def0fa27b5f42aea54e34c96b4bddaf30a Mon Sep 17 00:00:00 2001
From: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Date: Thu, 4 Jun 2015 15:13:09 +0100
Subject: ASoC: dapm: fix build errors for missing snd_soc_dapm_new_control
 symbol

Fix the following error:-

All error/warnings (new ones prefixed by >>):
>
>    sound/built-in.o: In function `soc_tplg_dapm_widget_create':
> >> :(.text+0x25a90): undefined reference to `snd_soc_dapm_new_control'

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 92d57a952bd9..1b4a6eb43174 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -53,7 +53,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 	int (*connected)(struct snd_soc_dapm_widget *source,
 			 struct snd_soc_dapm_widget *sink));
 
-static struct snd_soc_dapm_widget *
+struct snd_soc_dapm_widget *
 snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 			 const struct snd_soc_dapm_widget *widget);
 
@@ -3270,7 +3270,7 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch);
 
-static struct snd_soc_dapm_widget *
+struct snd_soc_dapm_widget *
 snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 	const struct snd_soc_dapm_widget *widget)
 {
-- 
cgit v1.2.3