[PATCH] ASoC: codecs: remove unneeded variable: "ret"
From: Gu Shengxian
Date: Wed Jul 07 2021 - 07:51:48 EST
From: Gu Shengxian <gushengxian@xxxxxxxxxx>
The variable: "ret" is only defined and returned.
So it could be removed.
Fix some spelling mistakes.
Signed-off-by: Gu Shengxian <gushengxian@xxxxxxxxxx>
---
sound/soc/codecs/ad1836.c | 2 +-
sound/soc/codecs/adau1372.c | 2 +-
sound/soc/codecs/adau1701.c | 2 +-
sound/soc/codecs/adau17x1.c | 2 +-
sound/soc/codecs/adau1977.c | 2 +-
sound/soc/codecs/ak4554.c | 2 +-
sound/soc/codecs/ak4613.c | 2 +-
sound/soc/codecs/alc5632.c | 2 +-
sound/soc/codecs/arizona.c | 2 +-
sound/soc/codecs/cpcap.c | 2 +-
sound/soc/codecs/cs35l33.c | 2 +-
sound/soc/codecs/cs35l34.c | 2 +-
sound/soc/codecs/cs35l36.c | 2 +-
sound/soc/codecs/cs4270.c | 2 +-
sound/soc/codecs/cs42l42.c | 2 +-
sound/soc/codecs/cs42l73.c | 2 +-
sound/soc/codecs/cs42xx8.c | 2 +-
sound/soc/codecs/cx20442.c | 4 ++--
sound/soc/codecs/cx2072x.c | 6 +++---
sound/soc/codecs/cx2072x.h | 2 +-
sound/soc/codecs/da7210.c | 2 +-
sound/soc/codecs/da7213.c | 2 +-
sound/soc/codecs/hdac_hda.c | 2 +-
sound/soc/codecs/hdac_hdmi.c | 6 +++---
sound/soc/codecs/max98088.c | 2 +-
sound/soc/codecs/max98373.c | 2 +-
sound/soc/codecs/max98390.c | 2 +-
sound/soc/codecs/max98927.c | 4 ++--
sound/soc/codecs/mt6359-accdet.c | 2 +-
sound/soc/codecs/mt6359.c | 10 +++++-----
sound/soc/codecs/wcd938x.c | 6 ++----
31 files changed, 42 insertions(+), 44 deletions(-)
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 08a5651bed9f..2db3e42fc6c1 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -265,7 +265,7 @@ static int ad1836_probe(struct snd_soc_component *component)
regmap_write(ad1836->regmap, AD1836_DAC_CTRL2, 0x0);
/* high-pass filter enable, power-on adc */
regmap_write(ad1836->regmap, AD1836_ADC_CTRL1, 0x100);
- /* unmute adc channles, adc aux mode */
+ /* unmute adc channels, adc aux mode */
regmap_write(ad1836->regmap, AD1836_ADC_CTRL2, 0x180);
/* volume */
for (i = 1; i <= num_dacs; ++i) {
diff --git a/sound/soc/codecs/adau1372.c b/sound/soc/codecs/adau1372.c
index 6811a8b3866d..6e9061c60f9f 100644
--- a/sound/soc/codecs/adau1372.c
+++ b/sound/soc/codecs/adau1372.c
@@ -684,7 +684,7 @@ static int adau1372_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
/* I2S mode */
if (slots == 0) {
- /* The other settings dont matter in I2S mode */
+ /* The other settings don't matter in I2S mode */
regmap_update_bits(adau1372->regmap, ADAU1372_REG_SAI0,
ADAU1372_SAI0_SAI_MASK, ADAU1372_SAI0_SAI_I2S);
adau1372->rate_constraints.mask = ADAU1372_RATE_MASK_TDM2;
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index 5ce74697564a..ab6fcfca7506 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -689,7 +689,7 @@ static int adau1701_probe(struct snd_soc_component *component)
*/
adau1701->pll_clkdiv = ADAU1707_CLKDIV_UNSET;
- /* initalize with pre-configured pll mode settings */
+ /* initialize with pre-configured pll mode settings */
ret = adau1701_reset(component, adau1701->pll_clkdiv, 0);
if (ret < 0)
goto exit_regulators_disable;
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 8aae7ab74091..c6df4272363c 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -876,7 +876,7 @@ static int adau17x1_setup_firmware(struct snd_soc_component *component,
* point in performing the below steps as the call to
* sigmadsp_setup(...) will return directly when it finds the sample
* rate to be the same as before. By checking this we can prevent an
- * audiable popping noise which occours when toggling DSP_RUN.
+ * audible popping noise which occurs when toggling DSP_RUN.
*/
if (adau->sigmadsp->current_samplerate == rate)
return 0;
diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c
index e347a48131d1..9e40a223a7fa 100644
--- a/sound/soc/codecs/adau1977.c
+++ b/sound/soc/codecs/adau1977.c
@@ -241,7 +241,7 @@ static int adau1977_reset(struct adau1977 *adau1977)
}
/*
- * Returns the appropriate setting for ths FS field in the CTRL0 register
+ * Returns the appropriate setting for the FS field in the CTRL0 register
* depending on the rate.
*/
static int adau1977_lookup_fs(unsigned int rate)
diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c
index 8e60e2b56ad6..1e79ac831f69 100644
--- a/sound/soc/codecs/ak4554.c
+++ b/sound/soc/codecs/ak4554.c
@@ -19,7 +19,7 @@
*
* CPU/Codec DAI image
*
- * CPU-DAI1 (plaback only fmt = RIGHT_J) --+-- ak4554
+ * CPU-DAI1 (playback only fmt = RIGHT_J) --+-- ak4554
* |
* CPU-DAI2 (capture only fmt = LEFT_J) ---+
*/
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
index 4d2e78101f28..ed8a069129a5 100644
--- a/sound/soc/codecs/ak4613.c
+++ b/sound/soc/codecs/ak4613.c
@@ -521,7 +521,7 @@ static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd,
*
* Calling ak4613_dummy_write() function might be delayed.
* In such case, ak4613 volume might be temporarily 0dB when
- * beggining of playback.
+ * beginning of playback.
* see also
* ak4613_dummy_write()
*/
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index 79813882a955..df6a6da681cf 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -149,7 +149,7 @@ static const DECLARE_TLV_DB_RANGE(boost_tlv,
);
/* 0db min scale, 6 db steps, no mute */
static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
-/* 0db min scalem 0.75db steps, no mute */
+/* 0db min scale 0.75db steps, no mute */
static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 75, 0);
static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = {
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index e32871b3f68a..f7f6c5925a41 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -2261,7 +2261,7 @@ static int arizona_calc_fll(struct arizona_fll *fll,
arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, fll->fout);
- /* Fvco should be over the targt; don't check the upper bound */
+ /* Fvco should be over the target; don't check the upper bound */
div = ARIZONA_FLL_MIN_OUTDIV;
while (fll->fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) {
div++;
diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c
index 05bbacd0d174..fa4e024804a5 100644
--- a/sound/soc/codecs/cpcap.c
+++ b/sound/soc/codecs/cpcap.c
@@ -800,7 +800,7 @@ static const struct snd_soc_dapm_widget cpcap_dapm_widgets[] = {
SND_SOC_DAPM_PGA("EMU Left PGA",
CPCAP_REG_RXOA, CPCAP_BIT_EMU_SPKR_L_EN, 0, NULL, 0),
- /* Headet Charge Pump */
+ /* Headset Charge Pump */
SND_SOC_DAPM_SUPPLY("Headset Charge Pump",
CPCAP_REG_RXOA, CPCAP_BIT_ST_HS_CP_EN, 0, NULL, 0),
diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c
index 2a6f5e46d031..7dd80cb8cae6 100644
--- a/sound/soc/codecs/cs35l33.c
+++ b/sound/soc/codecs/cs35l33.c
@@ -581,7 +581,7 @@ static int cs35l33_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
| CS35L33_X_LOC);
}
- /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/
+ /* disconnect {vp,vbst}_mon routes: enable later if set in tx_mask*/
snd_soc_dapm_del_routes(dapm, cs35l33_vp_vbst_mon_route,
ARRAY_SIZE(cs35l33_vp_vbst_mon_route));
diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c
index ed678241c22b..b8f19a5d1c10 100644
--- a/sound/soc/codecs/cs35l34.c
+++ b/sound/soc/codecs/cs35l34.c
@@ -298,7 +298,7 @@ static int cs35l34_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
CS35L34_X_STATE | CS35L34_X_LOC,
CS35L34_X_STATE | CS35L34_X_LOC);
- /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/
+ /* disconnect {vp,vbst}_mon routes: enable later if set in tx_mask*/
while (slot >= 0) {
/* configure VMON_TX_LOC */
if (slot_num == 0)
diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c
index d83c1b318c1c..8bfc680a1177 100644
--- a/sound/soc/codecs/cs35l36.c
+++ b/sound/soc/codecs/cs35l36.c
@@ -1246,7 +1246,7 @@ static int cs35l36_component_probe(struct snd_soc_component *component)
* L37 is 12V
* If L36 we need to clamp some values for safety
* after probe has setup dt values. We want to make
- * sure we dont miss any values set in probe
+ * sure we don't miss any values set in probe
*/
if (cs35l36->chip_version == CS35L36_10V_L36) {
regmap_update_bits(cs35l36->regmap,
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 2d239e983a83..20c33e7edb22 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -176,7 +176,7 @@ static const struct snd_soc_dapm_route cs4270_dapm_routes[] = {
* @speed_mode is the corresponding bit pattern to be written to the
* MODE bits of the Mode Control Register
*
- * @mclk is the corresponding bit pattern to be wirten to the MCLK bits of
+ * @mclk is the corresponding bit pattern to be written to the MCLK bits of
* the Mode Control Register.
*
* In situations where a single ratio is represented by multiple speed
diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index eff013f295be..111fc0c04015 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -1410,7 +1410,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data)
int report = 0;
- /* Read sticky registers to clear interurpt */
+ /* Read sticky registers to clear interrupt */
for (i = 0; i < ARRAY_SIZE(stickies); i++) {
regmap_read(cs42l42->regmap, irq_params_table[i].status_addr,
&(stickies[i]));
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 018463f34e12..95d50fa22274 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1118,7 +1118,7 @@ static int cs42l73_set_bias_level(struct snd_soc_component *component,
mdelay(cs42l73->shutdwn_delay);
cs42l73->shutdwn_delay = 0;
} else {
- mdelay(15); /* Min amount of time requred to power
+ mdelay(15); /* Min amount of time required to power
* down.
*/
}
diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
index 5d6ef660f851..bbfe7651b469 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/sound/soc/codecs/cs42xx8.c
@@ -184,7 +184,7 @@ struct cs42xx8_ratios {
};
/*
- * According to reference mannual, define the cs42xx8_ratio struct
+ * According to reference manual, define the cs42xx8_ratio struct
* MFreq2 | MFreq1 | MFreq0 | Description | SSM | DSM | QSM |
* 0 | 0 | 0 |1.029MHz to 12.8MHz | 256 | 128 | 64 |
* 0 | 0 | 1 |1.536MHz to 19.2MHz | 384 | 192 | 96 |
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index ec8d6e74b467..824c09f3fd1a 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -197,10 +197,10 @@ static int cx20442_write(struct snd_soc_component *component, unsigned int reg,
}
/*
- * Line discpline related code
+ * Line discipline related code
*
* Any of the callback functions below can be used in two ways:
- * 1) registerd by a machine driver as one of line discipline operations,
+ * 1) registered by a machine driver as one of line discipline operations,
* 2) called from a machine's provided line discipline callback function
* in case when extra machine specific code must be run as well.
*/
diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c
index 1f5c57fab1d8..2691d747692f 100644
--- a/sound/soc/codecs/cx2072x.c
+++ b/sound/soc/codecs/cx2072x.c
@@ -565,7 +565,7 @@ static int cx2072x_reg_read(void *context, unsigned int reg,
return 0;
}
-/* get suggested pre_div valuce from mclk frequency */
+/* get suggested pre_div value from mclk frequency */
static unsigned int get_div_from_mclk(unsigned int mclk)
{
unsigned int div = 8;
@@ -1571,7 +1571,7 @@ static struct snd_soc_dai_driver soc_codec_cx2072x_dai[] = {
.ops = &cx2072x_dai_ops,
.symmetric_rate = 1,
},
- { /* plabayck only, return echo reference to Conexant DSP chip */
+ { /* playback only, return echo reference to Conexant DSP chip */
.name = "cx2072x-dsp",
.id = CX2072X_DAI_DSP,
.probe = cx2072x_dsp_dai_probe,
@@ -1584,7 +1584,7 @@ static struct snd_soc_dai_driver soc_codec_cx2072x_dai[] = {
},
.ops = &cx2072x_dai_ops,
},
- { /* plabayck only, return echo reference through I2S TX */
+ { /* playback only, return echo reference through I2S TX */
.name = "cx2072x-aec",
.id = 3,
.capture = {
diff --git a/sound/soc/codecs/cx2072x.h b/sound/soc/codecs/cx2072x.h
index ebdd567fa225..09e3a92b184f 100644
--- a/sound/soc/codecs/cx2072x.h
+++ b/sound/soc/codecs/cx2072x.h
@@ -177,7 +177,7 @@
#define CX2072X_PLBK_DRC_PARM_LEN 9
#define CX2072X_CLASSD_AMP_LEN 6
-/* DAI interfae type */
+/* DAI interface type */
#define CX2072X_DAI_HIFI 1
#define CX2072X_DAI_DSP 2
#define CX2072X_DAI_DSP_PWM 3 /* 4 ch, including mic and AEC */
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 8af344b2fdbf..2b6ed0a5a697 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -1151,7 +1151,7 @@ static int da7210_probe(struct snd_soc_component *component)
snd_soc_component_write(component, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ |
DA7210_PLL_BYP);
- /* Diable PLL and bypass it */
+ /* Disable PLL and bypass it */
snd_soc_component_write(component, DA7210_PLL, DA7210_PLL_FS_48000);
/* Activate all enabled subsystem */
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 3ab89387b4e6..5c3af89ff21e 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -778,7 +778,7 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w,
return 0;
case SND_SOC_DAPM_POST_PMD:
- /* Revert 32KHz PLL lock udpates if applied previously */
+ /* Revert 32KHz PLL lock updates if applied previously */
pll_ctrl = snd_soc_component_read(component, DA7213_PLL_CTRL);
if (pll_ctrl & DA7213_PLL_32K_MODE) {
snd_soc_component_write(component, 0xF0, 0x8B);
diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c
index 390dd6c7f6a5..7298244ba92d 100644
--- a/sound/soc/codecs/hdac_hda.c
+++ b/sound/soc/codecs/hdac_hda.c
@@ -487,7 +487,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component)
/*
* hdac_device core already sets the state to active and calls
* get_noresume. So enable runtime and set the device to suspend.
- * pm_runtime_enable is also called during codec registeration
+ * pm_runtime_enable is also called during codec registration
*/
pm_runtime_put(&hdev->dev);
pm_runtime_suspend(&hdev->dev);
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 66408a98298b..36b194a51fed 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1051,7 +1051,7 @@ static void hdac_hdmi_add_pinmux_cvt_route(struct hdac_device *hdev,
* Widgets are added in the below sequence
* Converter widgets for num converters enumerated
* Pin-port widgets for num ports for Pins enumerated
- * Pin-port mux widgets to represent connenction list of pin widget
+ * Pin-port mux widgets to represent connection list of pin widget
*
* For each port, one Mux and One output widget is added
* Total widgets elements = num_cvt + (num_ports * 2);
@@ -1256,7 +1256,7 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin,
return;
/*
- * In case of non MST pin, get_eld info API expectes port
+ * In case of non MST pin, get_eld info API expects port
* to be -1.
*/
mutex_lock(&hdmi->pin_mutex);
@@ -2039,7 +2039,7 @@ static int hdmi_codec_resume(struct device *dev)
/*
* As the ELD notify callback request is not entertained while the
* device is in suspend state. Need to manually check detection of
- * all pins here. pin capablity change is not support, so use the
+ * all pins here. pin capability change is not support, so use the
* already set pin caps.
*
* NOTE: this is safe to call even if the codec doesn't actually resume.
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index f8e49e45ce33..a4923601dd72 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -95,7 +95,7 @@ static const struct reg_default max98088_reg[] = {
{ 0x30, 0x00 }, /* 30 DAI1 playback level */
{ 0x31, 0x00 }, /* 31 DAI2 playback level */
- { 0x32, 0x00 }, /* 32 DAI2 playbakc level */
+ { 0x32, 0x00 }, /* 32 DAI2 playback level */
{ 0x33, 0x00 }, /* 33 left ADC level */
{ 0x34, 0x00 }, /* 34 right ADC level */
{ 0x35, 0x00 }, /* 35 MIC1 level */
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index e14fe98349a5..8eaba126f534 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -307,7 +307,7 @@ SOC_ENUM("Limiter Release Rate", max98373_limiter_release_rate_enum),
};
static const struct snd_soc_dapm_route max98373_audio_map[] = {
- /* Plabyack */
+ /* Playback */
{"DAI Sel Mux", "Left", "Amp Enable"},
{"DAI Sel Mux", "Right", "Amp Enable"},
{"DAI Sel Mux", "LeftRight", "Amp Enable"},
diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c
index 94773ccee9d5..1c8e81499378 100644
--- a/sound/soc/codecs/max98390.c
+++ b/sound/soc/codecs/max98390.c
@@ -686,7 +686,7 @@ static const struct snd_soc_dapm_widget max98390_dapm_widgets[] = {
};
static const struct snd_soc_dapm_route max98390_audio_map[] = {
- /* Plabyack */
+ /* Playback */
{"DAI Sel Mux", "Left", "Amp Enable"},
{"DAI Sel Mux", "Right", "Amp Enable"},
{"DAI Sel Mux", "LeftRight", "Amp Enable"},
diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c
index 8b206ee77709..8846b99218f6 100644
--- a/sound/soc/codecs/max98927.c
+++ b/sound/soc/codecs/max98927.c
@@ -696,7 +696,7 @@ static int max98927_probe(struct snd_soc_component *component)
regmap_write(max98927->regmap,
MAX98927_R0026_PCM_TO_SPK_MONOMIX_B,
0x1);
- /* Set inital volume (+13dB) */
+ /* Set initial volume (+13dB) */
regmap_write(max98927->regmap,
MAX98927_R0036_AMP_VOL_CTRL,
0x38);
@@ -911,7 +911,7 @@ static int max98927_i2c_probe(struct i2c_client *i2c,
/* voltage/current slot configuration */
max98927_slot_config(i2c, max98927);
- /* codec registeration */
+ /* codec registration */
ret = devm_snd_soc_register_component(&i2c->dev,
&soc_component_dev_max98927,
max98927_dai, ARRAY_SIZE(max98927_dai));
diff --git a/sound/soc/codecs/mt6359-accdet.c b/sound/soc/codecs/mt6359-accdet.c
index 78314187d37e..ad3cf4b35488 100644
--- a/sound/soc/codecs/mt6359-accdet.c
+++ b/sound/soc/codecs/mt6359-accdet.c
@@ -752,7 +752,7 @@ static void config_eint_init_by_mode(struct mt6359_accdet *priv)
/* ESD switches on */
regmap_update_bits(priv->regmap, RG_ACCDETSPARE_ADDR,
1 << 8, 1 << 8);
- /* before playback, set NCP pull low before nagative voltage */
+ /* before playback, set NCP pull low before negative voltage */
regmap_update_bits(priv->regmap, RG_NCP_PDDIS_EN_ADDR,
RG_NCP_PDDIS_EN_MASK_SFT, BIT(RG_NCP_PDDIS_EN_SFT));
diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c
index 2d6a4a29b850..89ff46374f1f 100644
--- a/sound/soc/codecs/mt6359.c
+++ b/sound/soc/codecs/mt6359.c
@@ -68,7 +68,7 @@ static void mt6359_reset_capture_gpio(struct mt6359_priv *priv)
0x3 << 0, 0x0);
}
-/* use only when doing mtkaif calibraiton at the boot time */
+/* use only when doing mtkaif calibration at the boot time */
static void mt6359_set_dcxo(struct mt6359_priv *priv, bool enable)
{
regmap_update_bits(priv->regmap, MT6359_DCXO_CW12,
@@ -76,7 +76,7 @@ static void mt6359_set_dcxo(struct mt6359_priv *priv, bool enable)
(enable ? 1 : 0) << RG_XO_AUDIO_EN_M_SFT);
}
-/* use only when doing mtkaif calibraiton at the boot time */
+/* use only when doing mtkaif calibration at the boot time */
static void mt6359_set_clksq(struct mt6359_priv *priv, bool enable)
{
/* Enable/disable CLKSQ 26MHz */
@@ -85,7 +85,7 @@ static void mt6359_set_clksq(struct mt6359_priv *priv, bool enable)
(enable ? 1 : 0) << RG_CLKSQ_EN_SFT);
}
-/* use only when doing mtkaif calibraiton at the boot time */
+/* use only when doing mtkaif calibration at the boot time */
static void mt6359_set_aud_global_bias(struct mt6359_priv *priv, bool enable)
{
regmap_update_bits(priv->regmap, MT6359_AUDDEC_ANA_CON13,
@@ -93,7 +93,7 @@ static void mt6359_set_aud_global_bias(struct mt6359_priv *priv, bool enable)
(enable ? 0 : 1) << RG_AUDGLB_PWRDN_VA32_SFT);
}
-/* use only when doing mtkaif calibraiton at the boot time */
+/* use only when doing mtkaif calibration at the boot time */
static void mt6359_set_topck(struct mt6359_priv *priv, bool enable)
{
regmap_update_bits(priv->regmap, MT6359_AUD_TOP_CKPDN_CON0,
@@ -1731,7 +1731,7 @@ static int mt_pga_3_event(struct snd_soc_dapm_widget *w,
return 0;
}
-/* It is based on hw's control sequenece to add some delay when PMU/PMD */
+/* It is based on hw's control sequence to add some delay when PMU/PMD */
static int mt_delay_250_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol,
int event)
diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c
index 78b76eceff8f..5fd708e013f9 100644
--- a/sound/soc/codecs/wcd938x.c
+++ b/sound/soc/codecs/wcd938x.c
@@ -1623,7 +1623,6 @@ static int wcd938x_codec_aux_dac_event(struct snd_soc_dapm_widget *w,
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component);
- int ret = 0;
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
@@ -1651,7 +1650,7 @@ static int wcd938x_codec_aux_dac_event(struct snd_soc_dapm_widget *w,
WCD938X_ANA_RX_DIV4_CLK_EN_MASK, 0);
break;
}
- return ret;
+ return 0;
}
@@ -1866,7 +1865,6 @@ static int wcd938x_codec_enable_aux_pa(struct snd_soc_dapm_widget *w,
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component);
int hph_mode = wcd938x->hph_mode;
- int ret = 0;
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
@@ -1902,7 +1900,7 @@ static int wcd938x_codec_enable_aux_pa(struct snd_soc_dapm_widget *w,
WCD938X_EN_CUR_DET_MASK, 1);
break;
}
- return ret;
+ return 0;
}
static int wcd938x_codec_enable_ear_pa(struct snd_soc_dapm_widget *w,
--
2.25.1