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-rw-r--r--sound/soc/blackfin/Kconfig11
-rw-r--r--sound/soc/codecs/88pm860x-codec.c3
-rw-r--r--sound/soc/codecs/ad1980.c4
-rw-r--r--sound/soc/codecs/da732x.c12
-rw-r--r--sound/soc/codecs/da9055.c11
-rw-r--r--sound/soc/codecs/isabelle.c52
-rw-r--r--sound/soc/codecs/max98090.c21
-rw-r--r--sound/soc/codecs/rt5640.c1
-rw-r--r--sound/soc/codecs/si476x.c2
-rw-r--r--sound/soc/codecs/sta32x.c76
-rw-r--r--sound/soc/codecs/wm8400.c34
-rw-r--r--sound/soc/codecs/wm8770.c4
-rw-r--r--sound/soc/codecs/wm8900.c44
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c2
-rw-r--r--sound/soc/codecs/wm8993.c1
-rw-r--r--sound/soc/codecs/wm8994.c135
-rw-r--r--sound/soc/davinci/davinci-evm.c1
-rw-r--r--sound/soc/davinci/davinci-mcasp.c83
-rw-r--r--sound/soc/fsl/fsl_esai.c4
-rw-r--r--sound/soc/fsl/fsl_esai.h2
-rw-r--r--sound/soc/fsl/imx-mc13783.c1
-rw-r--r--sound/soc/fsl/imx-sgtl5000.c10
-rw-r--r--sound/soc/fsl/imx-wm8962.c11
-rw-r--r--sound/soc/omap/n810.c4
-rw-r--r--sound/soc/samsung/Kconfig6
-rw-r--r--sound/soc/soc-dapm.c139
-rw-r--r--sound/soc/soc-pcm.c3
-rw-r--r--sound/soc/txx9/txx9aclc-ac97.c8
28 files changed, 422 insertions, 263 deletions
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 54f74f8cbb7..4544d8eb145 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -11,7 +11,7 @@ config SND_BF5XX_I2S
config SND_BF5XX_SOC_SSM2602
tristate "SoC SSM2602 Audio Codec Add-On Card support"
- depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
+ depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
select SND_BF5XX_SOC_I2S if !BF60x
select SND_BF6XX_SOC_I2S if BF60x
select SND_SOC_SSM2602
@@ -21,10 +21,9 @@ config SND_BF5XX_SOC_SSM2602
config SND_SOC_BFIN_EVAL_ADAU1701
tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards"
- depends on SND_BF5XX_I2S
+ depends on SND_BF5XX_I2S && I2C
select SND_BF5XX_SOC_I2S
select SND_SOC_ADAU1701
- select I2C
help
Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ
board connected to one of the Blackfin evaluation boards like the
@@ -45,7 +44,7 @@ config SND_SOC_BFIN_EVAL_ADAU1373
config SND_SOC_BFIN_EVAL_ADAV80X
tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
- depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
+ depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
select SND_BF5XX_SOC_I2S
select SND_SOC_ADAV80X
help
@@ -58,7 +57,7 @@ config SND_SOC_BFIN_EVAL_ADAV80X
config SND_BF5XX_SOC_AD1836
tristate "SoC AD1836 Audio support for BF5xx"
- depends on SND_BF5XX_I2S
+ depends on SND_BF5XX_I2S && SPI_MASTER
select SND_BF5XX_SOC_I2S
select SND_SOC_AD1836
help
@@ -66,7 +65,7 @@ config SND_BF5XX_SOC_AD1836
config SND_BF5XX_SOC_AD193X
tristate "SoC AD193X Audio support for Blackfin"
- depends on SND_BF5XX_I2S
+ depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
select SND_BF5XX_SOC_I2S
select SND_SOC_AD193X
help
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 75d0ad5d2dc..647a72cda00 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -1328,6 +1328,9 @@ static int pm860x_probe(struct snd_soc_codec *codec)
pm860x->codec = codec;
codec->control_data = pm860x->regmap;
+ ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
+ if (ret)
+ return ret;
for (i = 0; i < 4; i++) {
ret = request_threaded_irq(pm860x->irq[i], NULL,
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 7257a8885f4..34d965a4a04 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -57,8 +57,8 @@ static const u16 ad1980_reg[] = {
static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line",
"Stereo Mix", "Mono Mix", "Phone"};
-static const struct soc_enum ad1980_cap_src =
- SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel);
+static SOC_ENUM_DOUBLE_DECL(ad1980_cap_src,
+ AC97_REC_SEL, 8, 0, ad1980_rec_sel);
static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = {
SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index f295b656991..f4d965ebc29 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -1268,11 +1268,23 @@ static struct snd_soc_dai_driver da732x_dai[] = {
},
};
+static bool da732x_volatile(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case DA732X_REG_HPL_DAC_OFF_CNTL:
+ case DA732X_REG_HPR_DAC_OFF_CNTL:
+ return true;
+ default:
+ return false;
+ }
+}
+
static const struct regmap_config da732x_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = DA732X_MAX_REG,
+ .volatile_reg = da732x_volatile,
.reg_defaults = da732x_reg_cache,
.num_reg_defaults = ARRAY_SIZE(da732x_reg_cache),
.cache_type = REGCACHE_RBTREE,
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index 52b79a487ac..422812613a2 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1523,8 +1523,15 @@ static int da9055_remove(struct i2c_client *client)
return 0;
}
+/*
+ * DO NOT change the device Ids. The naming is intentionally specific as both
+ * the CODEC and PMIC parts of this chip are instantiated separately as I2C
+ * devices (both have configurable I2C addresses, and are to all intents and
+ * purposes separate). As a result there are specific DA9055 Ids for CODEC
+ * and PMIC, which must be different to operate together.
+ */
static const struct i2c_device_id da9055_i2c_id[] = {
- { "da9055", 0 },
+ { "da9055-codec", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, da9055_i2c_id);
@@ -1532,7 +1539,7 @@ MODULE_DEVICE_TABLE(i2c, da9055_i2c_id);
/* I2C codec control layer */
static struct i2c_driver da9055_i2c_driver = {
.driver = {
- .name = "da9055",
+ .name = "da9055-codec",
.owner = THIS_MODULE,
},
.probe = da9055_i2c_probe,
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index 5839048ec46..cb736ddc446 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -140,13 +140,17 @@ static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"};
static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"};
static const struct soc_enum isabelle_rx1_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts),
- SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts),
+ SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3,
+ ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts),
+ SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5,
+ ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts),
};
static const struct soc_enum isabelle_rx2_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts),
- SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts),
+ SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2,
+ ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts),
+ SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4,
+ ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts),
};
/* Headset DAC playback switches */
@@ -161,13 +165,17 @@ static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"};
static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"};
static const struct soc_enum isabelle_atx_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts),
- SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts),
+ SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7,
+ ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts),
+ SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0,
+ ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts),
};
static const struct soc_enum isabelle_vtx_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts),
- SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts),
+ SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6,
+ ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts),
+ SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0,
+ ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts),
};
static const struct snd_kcontrol_new atx_mux_controls =
@@ -183,17 +191,13 @@ static const char *isabelle_amic1_texts[] = {
/* Left analog microphone selection */
static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"};
-static const struct soc_enum isabelle_amic1_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5,
- ARRAY_SIZE(isabelle_amic1_texts),
- isabelle_amic1_texts),
-};
+static SOC_ENUM_SINGLE_DECL(isabelle_amic1_enum,
+ ISABELLE_AMIC_CFG_REG, 5,
+ isabelle_amic1_texts);
-static const struct soc_enum isabelle_amic2_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4,
- ARRAY_SIZE(isabelle_amic2_texts),
- isabelle_amic2_texts),
-};
+static SOC_ENUM_SINGLE_DECL(isabelle_amic2_enum,
+ ISABELLE_AMIC_CFG_REG, 4,
+ isabelle_amic2_texts);
static const struct snd_kcontrol_new amic1_control =
SOC_DAPM_ENUM("Route", isabelle_amic1_enum);
@@ -206,16 +210,20 @@ static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"};
static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"};
static const struct soc_enum isabelle_st_audio_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1,
+ SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7,
+ ARRAY_SIZE(isabelle_st_audio_texts),
isabelle_st_audio_texts),
- SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1,
+ SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7,
+ ARRAY_SIZE(isabelle_st_audio_texts),
isabelle_st_audio_texts),
};
static const struct soc_enum isabelle_st_voice_enum[] = {
- SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1,
+ SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7,
+ ARRAY_SIZE(isabelle_st_voice_texts),
isabelle_st_voice_texts),
- SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1,
+ SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7,
+ ARRAY_SIZE(isabelle_st_voice_texts),
isabelle_st_voice_texts),
};
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 51f9b3d16b4..9f714ea8661 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -336,6 +336,7 @@ static bool max98090_readable_register(struct device *dev, unsigned int reg)
case M98090_REG_RECORD_TDM_SLOT:
case M98090_REG_SAMPLE_RATE:
case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E:
+ case M98090_REG_REVISION_ID:
return true;
default:
return false;
@@ -1769,16 +1770,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
- if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
- ret = regcache_sync(max98090->regmap);
-
- if (ret != 0) {
- dev_err(codec->dev,
- "Failed to sync cache: %d\n", ret);
- return ret;
- }
- }
-
if (max98090->jack_state == M98090_JACK_STATE_HEADSET) {
/*
* Set to normal bias level.
@@ -1792,6 +1783,16 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ ret = regcache_sync(max98090->regmap);
+ if (ret != 0) {
+ dev_err(codec->dev,
+ "Failed to sync cache: %d\n", ret);
+ return ret;
+ }
+ }
+ break;
+
case SND_SOC_BIAS_OFF:
/* Set internal pull-up to lowest power mode */
snd_soc_update_bits(codec, M98090_REG_JACK_DETECT,
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index a3fb4117963..886924934aa 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2093,6 +2093,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id);
#ifdef CONFIG_ACPI
static struct acpi_device_id rt5640_acpi_match[] = {
{ "INT33CA", 0 },
+ { "10EC5640", 0 },
{ },
};
MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match);
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index 52e7cb08434..fa2b8e07f42 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -210,7 +210,7 @@ out:
static int si476x_codec_probe(struct snd_soc_codec *codec)
{
codec->control_data = dev_get_regmap(codec->dev->parent, NULL);
- return 0;
+ return snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
}
static struct snd_soc_dai_ops si476x_dai_ops = {
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 06edb396e73..2735361a4c3 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -187,42 +187,42 @@ static const unsigned int sta32x_limiter_drc_release_tlv[] = {
13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0),
};
-static const struct soc_enum sta32x_drc_ac_enum =
- SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT,
- 2, sta32x_drc_ac);
-static const struct soc_enum sta32x_auto_eq_enum =
- SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT,
- 3, sta32x_auto_eq_mode);
-static const struct soc_enum sta32x_auto_gc_enum =
- SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT,
- 4, sta32x_auto_gc_mode);
-static const struct soc_enum sta32x_auto_xo_enum =
- SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT,
- 16, sta32x_auto_xo_mode);
-static const struct soc_enum sta32x_preset_eq_enum =
- SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT,
- 32, sta32x_preset_eq_mode);
-static const struct soc_enum sta32x_limiter_ch1_enum =
- SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT,
- 3, sta32x_limiter_select);
-static const struct soc_enum sta32x_limiter_ch2_enum =
- SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT,
- 3, sta32x_limiter_select);
-static const struct soc_enum sta32x_limiter_ch3_enum =
- SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT,
- 3, sta32x_limiter_select);
-static const struct soc_enum sta32x_limiter1_attack_rate_enum =
- SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT,
- 16, sta32x_limiter_attack_rate);
-static const struct soc_enum sta32x_limiter2_attack_rate_enum =
- SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT,
- 16, sta32x_limiter_attack_rate);
-static const struct soc_enum sta32x_limiter1_release_rate_enum =
- SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT,
- 16, sta32x_limiter_release_rate);
-static const struct soc_enum sta32x_limiter2_release_rate_enum =
- SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT,
- 16, sta32x_limiter_release_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_drc_ac_enum,
+ STA32X_CONFD, STA32X_CONFD_DRC_SHIFT,
+ sta32x_drc_ac);
+static SOC_ENUM_SINGLE_DECL(sta32x_auto_eq_enum,
+ STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT,
+ sta32x_auto_eq_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_auto_gc_enum,
+ STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT,
+ sta32x_auto_gc_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_auto_xo_enum,
+ STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT,
+ sta32x_auto_xo_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_preset_eq_enum,
+ STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT,
+ sta32x_preset_eq_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch1_enum,
+ STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT,
+ sta32x_limiter_select);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch2_enum,
+ STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT,
+ sta32x_limiter_select);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch3_enum,
+ STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT,
+ sta32x_limiter_select);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_attack_rate_enum,
+ STA32X_L1AR, STA32X_LxA_SHIFT,
+ sta32x_limiter_attack_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_attack_rate_enum,
+ STA32X_L2AR, STA32X_LxA_SHIFT,
+ sta32x_limiter_attack_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_release_rate_enum,
+ STA32X_L1AR, STA32X_LxR_SHIFT,
+ sta32x_limiter_release_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum,
+ STA32X_L2AR, STA32X_LxR_SHIFT,
+ sta32x_limiter_release_rate);
/* byte array controls for setting biquad, mixer, scaling coefficients;
* for biquads all five coefficients need to be set in one go,
@@ -331,7 +331,7 @@ static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec)
static int sta32x_cache_sync(struct snd_soc_codec *codec)
{
- struct sta32x_priv *sta32x = codec->control_data;
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
unsigned int mute;
int rc;
@@ -434,7 +434,7 @@ SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0,
SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum),
SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum),
SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum),
-SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum),
+SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter2_release_rate_enum),
/* depending on mode, the attack/release thresholds have
* two different enum definitions; provide both
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 48dc7d2fee3..6d684d934f4 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -117,19 +117,23 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
static const char *wm8400_digital_sidetone[] =
{"None", "Left ADC", "Right ADC", "Reserved"};
-static const struct soc_enum wm8400_left_digital_sidetone_enum =
-SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE,
- WM8400_ADC_TO_DACL_SHIFT, 2, wm8400_digital_sidetone);
+static SOC_ENUM_SINGLE_DECL(wm8400_left_digital_sidetone_enum,
+ WM8400_DIGITAL_SIDE_TONE,
+ WM8400_ADC_TO_DACL_SHIFT,
+ wm8400_digital_sidetone);
-static const struct soc_enum wm8400_right_digital_sidetone_enum =
-SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE,
- WM8400_ADC_TO_DACR_SHIFT, 2, wm8400_digital_sidetone);
+static SOC_ENUM_SINGLE_DECL(wm8400_right_digital_sidetone_enum,
+ WM8400_DIGITAL_SIDE_TONE,
+ WM8400_ADC_TO_DACR_SHIFT,
+ wm8400_digital_sidetone);
static const char *wm8400_adcmode[] =
{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
-static const struct soc_enum wm8400_right_adcmode_enum =
-SOC_ENUM_SINGLE(WM8400_ADC_CTRL, WM8400_ADC_HPF_CUT_SHIFT, 3, wm8400_adcmode);
+static SOC_ENUM_SINGLE_DECL(wm8400_right_adcmode_enum,
+ WM8400_ADC_CTRL,
+ WM8400_ADC_HPF_CUT_SHIFT,
+ wm8400_adcmode);
static const struct snd_kcontrol_new wm8400_snd_controls[] = {
/* INMIXL */
@@ -422,9 +426,10 @@ SOC_DAPM_SINGLE("RINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT,
static const char *wm8400_ainlmux[] =
{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
-static const struct soc_enum wm8400_ainlmux_enum =
-SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINLMODE_SHIFT,
- ARRAY_SIZE(wm8400_ainlmux), wm8400_ainlmux);
+static SOC_ENUM_SINGLE_DECL(wm8400_ainlmux_enum,
+ WM8400_INPUT_MIXER1,
+ WM8400_AINLMODE_SHIFT,
+ wm8400_ainlmux);
static const struct snd_kcontrol_new wm8400_dapm_ainlmux_controls =
SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum);
@@ -435,9 +440,10 @@ SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum);
static const char *wm8400_ainrmux[] =
{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
-static const struct soc_enum wm8400_ainrmux_enum =
-SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINRMODE_SHIFT,
- ARRAY_SIZE(wm8400_ainrmux), wm8400_ainrmux);
+static SOC_ENUM_SINGLE_DECL(wm8400_ainrmux_enum,
+ WM8400_INPUT_MIXER1,
+ WM8400_AINRMODE_SHIFT,
+ wm8400_ainrmux);
static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls =
SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum);
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index 89a18d82f30..5bce2101348 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -196,8 +196,8 @@ static const char *ain_text[] = {
"AIN5", "AIN6", "AIN7", "AIN8"
};
-static const struct soc_enum ain_enum =
- SOC_ENUM_DOUBLE(WM8770_ADCMUX, 0, 4, 8, ain_text);
+static SOC_ENUM_DOUBLE_DECL(ain_enum,
+ WM8770_ADCMUX, 0, 4, ain_text);
static const struct snd_kcontrol_new ain_mux =
SOC_DAPM_ENUM("Capture Mux", ain_enum);
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index e98bc7038a0..43c2201cb90 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -304,53 +304,53 @@ static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1);
static const char *mic_bias_level_txt[] = { "0.9*AVDD", "0.65*AVDD" };
-static const struct soc_enum mic_bias_level =
-SOC_ENUM_SINGLE(WM8900_REG_INCTL, 8, 2, mic_bias_level_txt);
+static SOC_ENUM_SINGLE_DECL(mic_bias_level,
+ WM8900_REG_INCTL, 8, mic_bias_level_txt);
static const char *dac_mute_rate_txt[] = { "Fast", "Slow" };
-static const struct soc_enum dac_mute_rate =
-SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 7, 2, dac_mute_rate_txt);
+static SOC_ENUM_SINGLE_DECL(dac_mute_rate,
+ WM8900_REG_DACCTRL, 7, dac_mute_rate_txt);
static const char *dac_deemphasis_txt[] = {
"Disabled", "32kHz", "44.1kHz", "48kHz"
};
-static const struct soc_enum dac_deemphasis =
-SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 4, 4, dac_deemphasis_txt);
+static SOC_ENUM_SINGLE_DECL(dac_deemphasis,
+ WM8900_REG_DACCTRL, 4, dac_deemphasis_txt);
static const char *adc_hpf_cut_txt[] = {
"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"
};
-static const struct soc_enum adc_hpf_cut =
-SOC_ENUM_SINGLE(WM8900_REG_ADCCTRL, 5, 4, adc_hpf_cut_txt);
+static SOC_ENUM_SINGLE_DECL(adc_hpf_cut,
+ WM8900_REG_ADCCTRL, 5, adc_hpf_cut_txt);
static const char *lr_txt[] = {
"Left", "Right"
};
-static const struct soc_enum aifl_src =
-SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 15, 2, lr_txt);
+static SOC_ENUM_SINGLE_DECL(aifl_src,
+ WM8900_REG_AUDIO1, 15, lr_txt);
-static const struct soc_enum aifr_src =
-SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 14, 2, lr_txt);
+static SOC_ENUM_SINGLE_DECL(aifr_src,
+ WM8900_REG_AUDIO1, 14, lr_txt);
-static const struct soc_enum dacl_src =
-SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 15, 2, lr_txt);
+static SOC_ENUM_SINGLE_DECL(dacl_src,
+ WM8900_REG_AUDIO2, 15, lr_txt);
-static const struct soc_enum dacr_src =
-SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 14, 2, lr_txt);
+static SOC_ENUM_SINGLE_DECL(dacr_src,
+ WM8900_REG_AUDIO2, 14, lr_txt);
static const char *sidetone_txt[] = {
"Disabled", "Left ADC", "Right ADC"
};
-static const struct soc_enum dacl_sidetone =
-SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 2, 3, sidetone_txt);
+static SOC_ENUM_SINGLE_DECL(dacl_sidetone,
+ WM8900_REG_SIDETONE, 2, sidetone_txt);
-static const struct soc_enum dacr_sidetone =
-SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 0, 3, sidetone_txt);
+static SOC_ENUM_SINGLE_DECL(dacr_sidetone,
+ WM8900_REG_SIDETONE, 0, sidetone_txt);
static const struct snd_kcontrol_new wm8900_snd_controls[] = {
SOC_ENUM("Mic Bias Level", mic_bias_level),
@@ -496,8 +496,8 @@ SOC_DAPM_SINGLE("RINPUT3 Switch", WM8900_REG_INCTL, 0, 1, 0),
static const char *wm8900_lp_mux[] = { "Disabled", "Enabled" };
-static const struct soc_enum wm8900_lineout2_lp_mux =
-SOC_ENUM_SINGLE(WM8900_REG_LOUTMIXCTL1, 1, 2, wm8900_lp_mux);
+static SOC_ENUM_SINGLE_DECL(wm8900_lineout2_lp_mux,
+ WM8900_REG_LOUTMIXCTL1, 1, wm8900_lp_mux);
static const struct snd_kcontrol_new wm8900_lineout2_lp =
SOC_DAPM_ENUM("Route", wm8900_lineout2_lp_mux);
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index b7488f190d2..d4248e00160 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -153,7 +153,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
data32 &= 0xffffff;
- wm8994_bulk_write(codec->control_data,
+ wm8994_bulk_write(wm8994->wm8994,
data32 & 0xffffff,
block_len / 2,
(void *)(data + 8));
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 433d59a0f3e..2ee23a39622 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1562,7 +1562,6 @@ static int wm8993_remove(struct snd_soc_codec *codec)
struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF);
- regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies);
return 0;
}
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index b9be9cbc460..adb72063d44 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -265,21 +265,21 @@ static const char *sidetone_hpf_text[] = {
"2.7kHz", "1.35kHz", "675Hz", "370Hz", "180Hz", "90Hz", "45Hz"
};
-static const struct soc_enum sidetone_hpf =
- SOC_ENUM_SINGLE(WM8994_SIDETONE, 7, 7, sidetone_hpf_text);
+static SOC_ENUM_SINGLE_DECL(sidetone_hpf,
+ WM8994_SIDETONE, 7, sidetone_hpf_text);
static const char *adc_hpf_text[] = {
"HiFi", "Voice 1", "Voice 2", "Voice 3"
};
-static const struct soc_enum aif1adc1_hpf =
- SOC_ENUM_SINGLE(WM8994_AIF1_ADC1_FILTERS, 13, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(aif1adc1_hpf,
+ WM8994_AIF1_ADC1_FILTERS, 13, adc_hpf_text);
-static const struct soc_enum aif1adc2_hpf =
- SOC_ENUM_SINGLE(WM8994_AIF1_ADC2_FILTERS, 13, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(aif1adc2_hpf,
+ WM8994_AIF1_ADC2_FILTERS, 13, adc_hpf_text);
-static const struct soc_enum aif2adc_hpf =
- SOC_ENUM_SINGLE(WM8994_AIF2_ADC_FILTERS, 13, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(aif2adc_hpf,
+ WM8994_AIF2_ADC_FILTERS, 13, adc_hpf_text);
static const DECLARE_TLV_DB_SCALE(aif_tlv, 0, 600, 0);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
@@ -501,39 +501,39 @@ static const char *aif_chan_src_text[] = {
"Left", "Right"
};
-static const struct soc_enum aif1adcl_src =
- SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 15, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif1adcl_src,
+ WM8994_AIF1_CONTROL_1, 15, aif_chan_src_text);
-static const struct soc_enum aif1adcr_src =
- SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 14, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif1adcr_src,
+ WM8994_AIF1_CONTROL_1, 14, aif_chan_src_text);
-static const struct soc_enum aif2adcl_src =
- SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 15, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2adcl_src,
+ WM8994_AIF2_CONTROL_1, 15, aif_chan_src_text);
-static const struct soc_enum aif2adcr_src =
- SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 14, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2adcr_src,
+ WM8994_AIF2_CONTROL_1, 14, aif_chan_src_text);
-static const struct soc_enum aif1dacl_src =
- SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 15, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif1dacl_src,
+ WM8994_AIF1_CONTROL_2, 15, aif_chan_src_text);
-static const struct soc_enum aif1dacr_src =
- SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 14, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif1dacr_src,
+ WM8994_AIF1_CONTROL_2, 14, aif_chan_src_text);
-static const struct soc_enum aif2dacl_src =
- SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 15, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2dacl_src,
+ WM8994_AIF2_CONTROL_2, 15, aif_chan_src_text);
-static const struct soc_enum aif2dacr_src =
- SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 14, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2dacr_src,
+ WM8994_AIF2_CONTROL_2, 14, aif_chan_src_text);
static const char *osr_text[] = {
"Low Power", "High Performance",
};
-static const struct soc_enum dac_osr =
- SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 0, 2, osr_text);
+static SOC_ENUM_SINGLE_DECL(dac_osr,
+ WM8994_OVERSAMPLING, 0, osr_text);
-static const struct soc_enum adc_osr =
- SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 1, 2, osr_text);
+static SOC_ENUM_SINGLE_DECL(adc_osr,
+ WM8994_OVERSAMPLING, 1, osr_text);
static const struct snd_kcontrol_new wm8994_snd_controls[] = {
SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME,
@@ -690,17 +690,20 @@ static const char *wm8958_ng_text[] = {
"30ms", "125ms", "250ms", "500ms",
};
-static const struct soc_enum wm8958_aif1dac1_ng_hold =
- SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE,
- WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text);
+static SOC_ENUM_SINGLE_DECL(wm8958_aif1dac1_ng_hold,
+ WM8958_AIF1_DAC1_NOISE_GATE,
+ WM8958_AIF1DAC1_NG_THR_SHIFT,
+ wm8958_ng_text);
-static const struct soc_enum wm8958_aif1dac2_ng_hold =
- SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE,
- WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text);
+static SOC_ENUM_SINGLE_DECL(wm8958_aif1dac2_ng_hold,
+ WM8958_AIF1_DAC2_NOISE_GATE,
+ WM8958_AIF1DAC2_NG_THR_SHIFT,
+ wm8958_ng_text);
-static const struct soc_enum wm8958_aif2dac_ng_hold =
- SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE,
- WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text);
+static SOC_ENUM_SINGLE_DECL(wm8958_aif2dac_ng_hold,
+ WM8958_AIF2_DAC_NOISE_GATE,
+ WM8958_AIF2DAC_NG_THR_SHIFT,
+ wm8958_ng_text);
static const struct snd_kcontrol_new wm8958_snd_controls[] = {
SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv),
@@ -1341,8 +1344,8 @@ static const char *adc_mux_text[] = {
"DMIC",
};
-static const struct soc_enum adc_enum =
- SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text);
+static SOC_ENUM_SINGLE_DECL(adc_enum,
+ 0, 0, adc_mux_text);
static const struct snd_kcontrol_new adcl_mux =
SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum);
@@ -1478,14 +1481,14 @@ static const char *sidetone_text[] = {
"ADC/DMIC1", "DMIC2",
};
-static const struct soc_enum sidetone1_enum =
- SOC_ENUM_SINGLE(WM8994_SIDETONE, 0, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetone1_enum,
+ WM8994_SIDETONE, 0, sidetone_text);
static const struct snd_kcontrol_new sidetone1_mux =
SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum);
-static const struct soc_enum sidetone2_enum =
- SOC_ENUM_SINGLE(WM8994_SIDETONE, 1, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetone2_enum,
+ WM8994_SIDETONE, 1, sidetone_text);
static const struct snd_kcontrol_new sidetone2_mux =
SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum);
@@ -1498,22 +1501,24 @@ static const char *loopback_text[] = {
"None", "ADCDAT",
};
-static const struct soc_enum aif1_loopback_enum =
- SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2,
- loopback_text);
+static SOC_ENUM_SINGLE_DECL(aif1_loopback_enum,
+ WM8994_AIF1_CONTROL_2,
+ WM8994_AIF1_LOOPBACK_SHIFT,
+ loopback_text);
static const struct snd_kcontrol_new aif1_loopback =
SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum);
-static const struct soc_enum aif2_loopback_enum =
- SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2,
- loopback_text);
+static SOC_ENUM_SINGLE_DECL(aif2_loopback_enum,
+ WM8994_AIF2_CONTROL_2,
+ WM8994_AIF2_LOOPBACK_SHIFT,
+ loopback_text);
static const struct snd_kcontrol_new aif2_loopback =
SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum);
-static const struct soc_enum aif1dac_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text);
+static SOC_ENUM_SINGLE_DECL(aif1dac_enum,
+ WM8994_POWER_MANAGEMENT_6, 0, aif1dac_text);
static const struct snd_kcontrol_new aif1dac_mux =
SOC_DAPM_ENUM("AIF1DAC Mux", aif1dac_enum);
@@ -1522,8 +1527,8 @@ static const char *aif2dac_text[] = {
"AIF2DACDAT", "AIF3DACDAT",
};
-static const struct soc_enum aif2dac_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 1, 2, aif2dac_text);
+static SOC_ENUM_SINGLE_DECL(aif2dac_enum,
+ WM8994_POWER_MANAGEMENT_6, 1, aif2dac_text);
static const struct snd_kcontrol_new aif2dac_mux =
SOC_DAPM_ENUM("AIF2DAC Mux", aif2dac_enum);
@@ -1532,8 +1537,8 @@ static const char *aif2adc_text[] = {
"AIF2ADCDAT", "AIF3DACDAT",
};
-static const struct soc_enum aif2adc_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 2, 2, aif2adc_text);
+static SOC_ENUM_SINGLE_DECL(aif2adc_enum,
+ WM8994_POWER_MANAGEMENT_6, 2, aif2adc_text);
static const struct snd_kcontrol_new aif2adc_mux =
SOC_DAPM_ENUM("AIF2ADC Mux", aif2adc_enum);
@@ -1542,14 +1547,14 @@ static const char *aif3adc_text[] = {
"AIF1ADCDAT", "AIF2ADCDAT", "AIF2DACDAT", "Mono PCM",
};
-static const struct soc_enum wm8994_aif3adc_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 3, aif3adc_text);
+static SOC_ENUM_SINGLE_DECL(wm8994_aif3adc_enum,
+ WM8994_POWER_MANAGEMENT_6, 3, aif3adc_text);
static const struct snd_kcontrol_new wm8994_aif3adc_mux =
SOC_DAPM_ENUM("AIF3ADC Mux", wm8994_aif3adc_enum);
-static const struct soc_enum wm8958_aif3adc_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 4, aif3adc_text);
+static SOC_ENUM_SINGLE_DECL(wm8958_aif3adc_enum,
+ WM8994_POWER_MANAGEMENT_6, 3, aif3adc_text);
static const struct snd_kcontrol_new wm8958_aif3adc_mux =
SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum);
@@ -1558,8 +1563,8 @@ static const char *mono_pcm_out_text[] = {
"None", "AIF2ADCL", "AIF2ADCR",
};
-static const struct soc_enum mono_pcm_out_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 9, 3, mono_pcm_out_text);
+static SOC_ENUM_SINGLE_DECL(mono_pcm_out_enum,
+ WM8994_POWER_MANAGEMENT_6, 9, mono_pcm_out_text);
static const struct snd_kcontrol_new mono_pcm_out_mux =
SOC_DAPM_ENUM("Mono PCM Out Mux", mono_pcm_out_enum);
@@ -1569,14 +1574,14 @@ static const char *aif2dac_src_text[] = {
};
/* Note that these two control shouldn't be simultaneously switched to AIF3 */
-static const struct soc_enum aif2dacl_src_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 7, 2, aif2dac_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2dacl_src_enum,
+ WM8994_POWER_MANAGEMENT_6, 7, aif2dac_src_text);
static const struct snd_kcontrol_new aif2dacl_src_mux =
SOC_DAPM_ENUM("AIF2DACL Mux", aif2dacl_src_enum);
-static const struct soc_enum aif2dacr_src_enum =
- SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 8, 2, aif2dac_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2dacr_src_enum,
+ WM8994_POWER_MANAGEMENT_6, 8, aif2dac_src_text);
static const struct snd_kcontrol_new aif2dacr_src_mux =
SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum);
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 70ff3772079..5e3bc3c6801 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -399,6 +399,7 @@ static struct platform_driver davinci_evm_driver = {
.driver = {
.name = "davinci_evm",
.owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
.of_match_table = of_match_ptr(davinci_evm_dt_ids),
},
};
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index b7858bfa029..670afa29e30 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -263,7 +263,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
unsigned int fmt)
{
struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+ int ret = 0;
+ pm_runtime_get_sync(mcasp->dev);
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_B:
case SND_SOC_DAIFMT_AC97:
@@ -317,7 +319,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
break;
default:
- return -EINVAL;
+ ret = -EINVAL;
+ goto out;
}
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -354,10 +357,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
break;
default:
- return -EINVAL;
+ ret = -EINVAL;
+ break;
}
-
- return 0;
+out:
+ pm_runtime_put_sync(mcasp->dev);
+ return ret;
}
static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
@@ -448,7 +453,7 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
return 0;
}
-static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
+static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
int channels)
{
int i;
@@ -524,12 +529,18 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
return 0;
}
-static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream)
+static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream)
{
int i, active_slots;
u32 mask = 0;
u32 busel = 0;
+ if ((mcasp->tdm_slots < 2) || (mcasp->tdm_slots > 32)) {
+ dev_err(mcasp->dev, "tdm slot %d not supported\n",
+ mcasp->tdm_slots);
+ return -EINVAL;
+ }
+
active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots;
for (i = 0; i < active_slots; i++)
mask |= (1 << i);
@@ -539,35 +550,21 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream)
if (!mcasp->dat_port)
busel = TXSEL;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- /* bit stream is MSB first with no delay */
- /* DSP_B mode */
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
- mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
-
- if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32))
- mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
- FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF));
- else
- printk(KERN_ERR "playback tdm slot %d not supported\n",
- mcasp->tdm_slots);
- } else {
- /* bit stream is MSB first with no delay */
- /* DSP_B mode */
- mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
-
- if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32))
- mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
- FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF));
- else
- printk(KERN_ERR "capture tdm slot %d not supported\n",
- mcasp->tdm_slots);
- }
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+ FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF));
+
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
+ mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
+ mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
+ FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF));
+
+ return 0;
}
/* S/PDIF */
-static void davinci_hw_dit_param(struct davinci_mcasp *mcasp)
+static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp)
{
/* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0
and LSB first */
@@ -589,6 +586,8 @@ static void davinci_hw_dit_param(struct davinci_mcasp *mcasp)
/* Enable the DIT */
mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN);
+
+ return 0;
}
static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
@@ -605,13 +604,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
u8 slots = mcasp->tdm_slots;
u8 active_serializers;
int channels;
+ int ret;
struct snd_interval *pcm_channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
channels = pcm_channels->min;
active_serializers = (channels + slots - 1) / slots;
- if (davinci_hw_common_param(mcasp, substream->stream, channels) == -EINVAL)
+ if (mcasp_common_hw_param(mcasp, substream->stream, channels) == -EINVAL)
return -EINVAL;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
fifo_level = mcasp->txnumevt * active_serializers;
@@ -619,9 +619,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
fifo_level = mcasp->rxnumevt * active_serializers;
if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
- davinci_hw_dit_param(mcasp);
+ ret = mcasp_dit_hw_param(mcasp);
else
- davinci_hw_param(mcasp, substream->stream);
+ ret = mcasp_i2s_hw_param(mcasp, substream->stream);
+
+ if (ret)
+ return ret;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_U8:
@@ -678,19 +681,9 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- ret = pm_runtime_get_sync(mcasp->dev);
- if (IS_ERR_VALUE(ret))
- dev_err(mcasp->dev, "pm_runtime_get_sync() failed\n");
davinci_mcasp_start(mcasp, substream->stream);
break;
-
case SNDRV_PCM_TRIGGER_SUSPEND:
- davinci_mcasp_stop(mcasp, substream->stream);
- ret = pm_runtime_put_sync(mcasp->dev);
- if (IS_ERR_VALUE(ret))
- dev_err(mcasp->dev, "pm_runtime_put_sync() failed\n");
- break;
-
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
davinci_mcasp_stop(mcasp, substream->stream);
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index d0c72ed261e..c84026c9913 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -326,7 +326,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA,
ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask));
regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB,
- ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask));
+ ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask));
regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots));
@@ -334,7 +334,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA,
ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask));
regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB,
- ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask));
+ ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask));
esai_priv->slot_width = slot_width;
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
index 9c9f957fcae..75e14033e8d 100644
--- a/sound/soc/fsl/fsl_esai.h
+++ b/sound/soc/fsl/fsl_esai.h
@@ -322,7 +322,7 @@
#define ESAI_xSMB_xS_SHIFT 0
#define ESAI_xSMB_xS_WIDTH 16
#define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT)
-#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK)
+#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMB_xS_MASK)
/* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */
#define ESAI_PRRC_PDC_SHIFT 0
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 79cee782dbb..a2fd7321b5a 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -160,7 +160,6 @@ static struct platform_driver imx_mc13783_audio_driver = {
.driver = {
.name = "imx_mc13783",
.owner = THIS_MODULE,
- .pm = &snd_soc_pm_ops,
},
.probe = imx_mc13783_probe,
.remove = imx_mc13783_remove
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index f2beae78969..1cb22dd034e 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -33,8 +33,7 @@ struct imx_sgtl5000_data {
static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd)
{
- struct imx_sgtl5000_data *data = container_of(rtd->card,
- struct imx_sgtl5000_data, card);
+ struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(rtd->card);
struct device *dev = rtd->card->dev;
int ret;
@@ -159,13 +158,15 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
data->card.dapm_widgets = imx_sgtl5000_dapm_widgets;
data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets);
+ platform_set_drvdata(pdev, &data->card);
+ snd_soc_card_set_drvdata(&data->card, data);
+
ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
goto fail;
}
- platform_set_drvdata(pdev, data);
of_node_put(ssi_np);
of_node_put(codec_np);
@@ -184,7 +185,8 @@ fail:
static int imx_sgtl5000_remove(struct platform_device *pdev)
{
- struct imx_sgtl5000_data *data = platform_get_drvdata(pdev);
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(card);
clk_put(data->codec_clk);
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 3fd76bc391d..3a3d17ce6ba 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -71,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
{
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
struct imx_priv *priv = &card_priv;
- struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev);
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
struct device *dev = &priv->pdev->dev;
unsigned int pll_out;
int ret;
@@ -137,7 +137,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card)
{
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
struct imx_priv *priv = &card_priv;
- struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev);
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
struct device *dev = &priv->pdev->dev;
int ret;
@@ -264,13 +264,15 @@ static int imx_wm8962_probe(struct platform_device *pdev)
data->card.late_probe = imx_wm8962_late_probe;
data->card.set_bias_level = imx_wm8962_set_bias_level;
+ platform_set_drvdata(pdev, &data->card);
+ snd_soc_card_set_drvdata(&data->card, data);
+
ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
goto clk_fail;
}
- platform_set_drvdata(pdev, data);
of_node_put(ssi_np);
of_node_put(codec_np);
@@ -289,7 +291,8 @@ fail:
static int imx_wm8962_remove(struct platform_device *pdev)
{
- struct imx_wm8962_data *data = platform_get_drvdata(pdev);
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
if (!IS_ERR(data->codec_clk))
clk_disable_unprepare(data->codec_clk);
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 3fde9e40271..d163e18d85d 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -305,7 +305,9 @@ static int __init n810_soc_init(void)
int err;
struct device *dev;
- if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
+ if (!of_have_populated_dt() ||
+ (!of_machine_is_compatible("nokia,n810") &&
+ !of_machine_is_compatible("nokia,n810-wimax")))
return -ENODEV;
n810_snd_device = platform_device_alloc("soc-audio", -1);
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 454f41cfc82..35075740039 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -59,7 +59,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750
select SND_SOC_WM8750
select SND_S3C2412_SOC_I2S
help
- Sat Y if you want to add support for SoC audio on the Jive.
+ Say Y if you want to add support for SoC audio on the Jive.
config SND_SOC_SAMSUNG_SMDK_WM8580
tristate "SoC I2S Audio support for WM8580 on SMDK"
@@ -145,11 +145,11 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380
config SND_SOC_SAMSUNG_SMDK_WM9713
tristate "SoC AC97 Audio support for SMDK with WM9713"
- depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210)
+ depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110)
select SND_SOC_WM9713
select SND_SAMSUNG_AC97
help
- Sat Y if you want to add support for SoC audio on the SMDK.
+ Say Y if you want to add support for SoC audio on the SMDK.
config SND_SOC_SMARTQ
tristate "SoC I2S Audio support for SmartQ board"
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index dc8ff13187f..b9dc6acbba8 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1218,7 +1218,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
ret = regulator_allow_bypass(w->regulator, false);
if (ret != 0)
dev_warn(w->dapm->dev,
- "ASoC: Failed to bypass %s: %d\n",
+ "ASoC: Failed to unbypass %s: %d\n",
w->name, ret);
}
@@ -1228,7 +1228,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
ret = regulator_allow_bypass(w->regulator, true);
if (ret != 0)
dev_warn(w->dapm->dev,
- "ASoC: Failed to unbypass %s: %d\n",
+ "ASoC: Failed to bypass %s: %d\n",
w->name, ret);
}
@@ -3210,15 +3210,11 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
const char *pin = (const char *)kcontrol->private_value;
- mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-
if (ucontrol->value.integer.value[0])
snd_soc_dapm_enable_pin(&card->dapm, pin);
else
snd_soc_dapm_disable_pin(&card->dapm, pin);
- mutex_unlock(&card->dapm_mutex);
-
snd_soc_dapm_sync(&card->dapm);
return 0;
}
@@ -3248,7 +3244,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
ret = regulator_allow_bypass(w->regulator, true);
if (ret != 0)
dev_warn(w->dapm->dev,
- "ASoC: Failed to unbypass %s: %d\n",
+ "ASoC: Failed to bypass %s: %d\n",
w->name, ret);
}
break;
@@ -3767,23 +3763,52 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
}
/**
+ * snd_soc_dapm_enable_pin_unlocked - enable pin.
+ * @dapm: DAPM context
+ * @pin: pin name
+ *
+ * Enables input/output pin and its parents or children widgets iff there is
+ * a valid audio route and active audio stream.
+ *
+ * Requires external locking.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
+ const char *pin)
+{
+ return snd_soc_dapm_set_pin(dapm, pin, 1);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin_unlocked);
+
+/**
* snd_soc_dapm_enable_pin - enable pin.
* @dapm: DAPM context
* @pin: pin name
*
* Enables input/output pin and its parents or children widgets iff there is
* a valid audio route and active audio stream.
+ *
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
*/
int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm, const char *pin)
{
- return snd_soc_dapm_set_pin(dapm, pin, 1);
+ int ret;
+
+ mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_dapm_set_pin(dapm, pin, 1);
+
+ mutex_unlock(&dapm->card->dapm_mutex);
+
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
/**
- * snd_soc_dapm_force_enable_pin - force a pin to be enabled
+ * snd_soc_dapm_force_enable_pin_unlocked - force a pin to be enabled
* @dapm: DAPM context
* @pin: pin name
*
@@ -3791,11 +3816,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
* intended for use with microphone bias supplies used in microphone
* jack detection.
*
+ * Requires external locking.
+ *
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
*/
-int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
- const char *pin)
+int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
+ const char *pin)
{
struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true);
@@ -3811,25 +3838,103 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
return 0;
}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin_unlocked);
+
+/**
+ * snd_soc_dapm_force_enable_pin - force a pin to be enabled
+ * @dapm: DAPM context
+ * @pin: pin name
+ *
+ * Enables input/output pin regardless of any other state. This is
+ * intended for use with microphone bias supplies used in microphone
+ * jack detection.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
+ const char *pin)
+{
+ int ret;
+
+ mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_dapm_force_enable_pin_unlocked(dapm, pin);
+
+ mutex_unlock(&dapm->card->dapm_mutex);
+
+ return ret;
+}
EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin);
/**
+ * snd_soc_dapm_disable_pin_unlocked - disable pin.
+ * @dapm: DAPM context
+ * @pin: pin name
+ *
+ * Disables input/output pin and its parents or children widgets.
+ *
+ * Requires external locking.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_disable_pin_unlocked(struct snd_soc_dapm_context *dapm,
+ const char *pin)
+{
+ return snd_soc_dapm_set_pin(dapm, pin, 0);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin_unlocked);
+
+/**
* snd_soc_dapm_disable_pin - disable pin.
* @dapm: DAPM context
* @pin: pin name
*
* Disables input/output pin and its parents or children widgets.
+ *
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
*/
int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm,
const char *pin)
{
- return snd_soc_dapm_set_pin(dapm, pin, 0);
+ int ret;
+
+ mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_dapm_set_pin(dapm, pin, 0);
+
+ mutex_unlock(&dapm->card->dapm_mutex);
+
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
/**
+ * snd_soc_dapm_nc_pin_unlocked - permanently disable pin.
+ * @dapm: DAPM context
+ * @pin: pin name
+ *
+ * Marks the specified pin as being not connected, disabling it along
+ * any parent or child widgets. At present this is identical to
+ * snd_soc_dapm_disable_pin() but in future it will be extended to do
+ * additional things such as disabling controls which only affect
+ * paths through the pin.
+ *
+ * Requires external locking.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_nc_pin_unlocked(struct snd_soc_dapm_context *dapm,
+ const char *pin)
+{
+ return snd_soc_dapm_set_pin(dapm, pin, 0);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin_unlocked);
+
+/**
* snd_soc_dapm_nc_pin - permanently disable pin.
* @dapm: DAPM context
* @pin: pin name
@@ -3845,7 +3950,15 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
*/
int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin)
{
- return snd_soc_dapm_set_pin(dapm, pin, 0);
+ int ret;
+
+ mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+ ret = snd_soc_dapm_set_pin(dapm, pin, 0);
+
+ mutex_unlock(&dapm->card->dapm_mutex);
+
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 47e1ce771e6..28522bd03b8 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1989,6 +1989,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
if (paths < 0) {
+ dpcm_path_put(&list);
dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
fe->dai_link->name, "playback");
mutex_unlock(&card->mutex);
@@ -2018,6 +2019,7 @@ capture:
paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
if (paths < 0) {
+ dpcm_path_put(&list);
dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
fe->dai_link->name, "capture");
mutex_unlock(&card->mutex);
@@ -2082,6 +2084,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
fe->dpcm[stream].runtime = fe_substream->runtime;
if (dpcm_path_get(fe, stream, &list) <= 0) {
+ dpcm_path_put(&list);
dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
fe->dai_link->name, stream ? "capture" : "playback");
}
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index e0305a14856..9edd68db9f4 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -183,14 +183,16 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev)
irq = platform_get_irq(pdev, 0);
if (irq < 0)
return irq;
+
+ drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
+ if (!drvdata)
+ return -ENOMEM;
+
r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
drvdata->base = devm_ioremap_resource(&pdev->dev, r);
if (IS_ERR(drvdata->base))
return PTR_ERR(drvdata->base);
- drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
- if (!drvdata)
- return -ENOMEM;
platform_set_drvdata(pdev, drvdata);
drvdata->physbase = r->start;
if (sizeof(drvdata->physbase) > sizeof(r->start) &&