* [PATCH v3] ASoC: fix typos in comments
@ 2026-09-17 5:23 Hemanth Selam
0 siblings, 0 replies; only message in thread
From: Hemanth Selam @ 2026-09-17 5:23 UTC (permalink / raw)
To: Mark Brown, Liam Girdwood
Cc: Claudiu Beznea, Andrei Simion, Jaroslav Kysela, Takashi Iwai,
Nicolas Ferre, Alexandre Belloni, Ray Jui, Scott Branden,
Broadcom internal kernel review list, Lars-Peter Clausen,
Nuno Sá,
David Rhodes, Richard Fitzgerald, Support Opensource,
Matthias Brugger, AngeloGioacchino Del Regno, Oder Chiou,
Shenghao Ding, Kevin Lu, Baojun Xu, Sen Wang, Cezary Rojewski,
Peter Ujfalusi, Bard Liao, Kai Vehmanen, Pierre-Louis Bossart,
Srinivas Kandagatla, Jarkko Nikula, Kunihiko Hayashi,
Masami Hiramatsu, linux-sound, linux-arm-kernel, linux-kernel,
patches, linux-mediatek, linux-arm-msm, linux-omap
Fix typos in comments, reported by scripts/checkpatch.pl using the
misspelling list in scripts/spelling.txt. Only touches comments, no code
changes.
v3 is v2 regenerated on top of broonie/sound.git for-next, with a
base-commit trailer so the base is unambiguous. v2 was reported as not
applying; I could not reproduce that against for-next, for-7.4 or
for-linus, so please let me know which branch you are applying to if this
one still fails.
Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
sound/soc/atmel/atmel-pcm-dma.c | 2 +-
sound/soc/bcm/cygnus-ssp.h | 2 +-
sound/soc/codecs/adau1701.c | 4 ++--
sound/soc/codecs/cs35l33.c | 2 +-
sound/soc/codecs/cs35l34.c | 2 +-
sound/soc/codecs/cs42l42.c | 2 +-
sound/soc/codecs/cs42l73.c | 4 ++--
sound/soc/codecs/cx20442.c | 2 +-
sound/soc/codecs/da7210.c | 2 +-
sound/soc/codecs/hdac_hda.c | 2 +-
sound/soc/codecs/hdac_hdmi.c | 2 +-
sound/soc/codecs/max98927.c | 4 ++--
sound/soc/codecs/mt6359.c | 8 ++++----
sound/soc/codecs/nau8825.c | 10 +++++-----
sound/soc/codecs/rt1320-sdw.c | 2 +-
sound/soc/codecs/rt5631.h | 8 ++++----
sound/soc/codecs/rt5640.c | 2 +-
sound/soc/codecs/rt5645.c | 2 +-
sound/soc/codecs/rt5651.c | 2 +-
sound/soc/codecs/rt5670.c | 2 +-
sound/soc/codecs/tlv320dac33.c | 2 +-
sound/soc/intel/atom/sst-mfld-dsp.h | 2 +-
sound/soc/intel/atom/sst/sst.h | 2 +-
sound/soc/intel/boards/sof_cirrus_common.c | 2 +-
sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +-
sound/soc/ti/davinci-i2s.c | 2 +-
sound/soc/ti/davinci-mcasp.c | 2 +-
sound/soc/ti/omap-mcbsp.c | 2 +-
sound/soc/uniphier/aio.h | 2 +-
sound/soc/ux500/ux500_msp_i2s.c | 2 +-
30 files changed, 43 insertions(+), 43 deletions(-)
diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c
index 7306e04da513..e4b73ae3df4f 100644
--- a/sound/soc/atmel/atmel-pcm-dma.c
+++ b/sound/soc/atmel/atmel-pcm-dma.c
@@ -47,7 +47,7 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = {
* atmel_pcm_dma_irq: SSC interrupt handler for DMAENGINE enabled SSC
*
* We use DMAENGINE to send/receive data to/from SSC so this ISR is only to
- * check if any overrun occured.
+ * check if any overrun occurred.
*/
static void atmel_pcm_dma_irq(u32 ssc_sr,
struct snd_pcm_substream *substream)
diff --git a/sound/soc/bcm/cygnus-ssp.h b/sound/soc/bcm/cygnus-ssp.h
index 4925e03c3c30..7848e15efbe1 100644
--- a/sound/soc/bcm/cygnus-ssp.h
+++ b/sound/soc/bcm/cygnus-ssp.h
@@ -27,7 +27,7 @@ struct ringbuf_regs {
unsigned wraddr;
unsigned baseaddr;
unsigned endaddr;
- unsigned fmark; /* freemark for play, fullmark for caputure */
+ unsigned fmark; /* freemark for play, fullmark for capture */
unsigned period_bytes;
unsigned buf_size;
};
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index a1311c7b56b1..d55cdc5d4413 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -4,7 +4,7 @@
*
* Copyright 2011 Analog Devices Inc.
* Author: Lars-Peter Clausen <lars@metafoo.de>
- * based on an inital version by Cliff Cai <cliff.cai@analog.com>
+ * based on an initial version by Cliff Cai <cliff.cai@analog.com>
*/
#include <linux/module.h>
@@ -701,7 +701,7 @@ 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/cs35l33.c b/sound/soc/codecs/cs35l33.c
index d0f9231f5e44..b8c19f175701 100644
--- a/sound/soc/codecs/cs35l33.c
+++ b/sound/soc/codecs/cs35l33.c
@@ -578,7 +578,7 @@ 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 3f265bd760f7..a72be629a31e 100644
--- a/sound/soc/codecs/cs35l34.c
+++ b/sound/soc/codecs/cs35l34.c
@@ -297,7 +297,7 @@ 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/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 866c21944729..84065d153ebf 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -1682,7 +1682,7 @@ irqreturn_t cs42l42_irq_thread(int irq, void *data)
if (cs42l42->suspended || !cs42l42->init_done)
return IRQ_NONE;
- /* 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 72076ccf4570..72c4213ed74b 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -761,7 +761,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"ASPOUTL", "ASP-IP Volume", "ASPL Output Mixer"},
{"ASPOUTR", "ASP-IP Volume", "ASPR Output Mixer"},
- /* Auxillary Capture */
+ /* Auxiliary Capture */
{"XSPL Output Mixer", NULL, "Input Left Capture"},
{"XSPR Output Mixer", NULL, "Input Right Capture"},
@@ -1126,7 +1126,7 @@ 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/cx20442.c b/sound/soc/codecs/cx20442.c
index 0dea84cd8ef2..d3c7a9370220 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -200,7 +200,7 @@ static int cx20442_write(struct snd_soc_component *component, unsigned int reg,
* Line discpline 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/da7210.c b/sound/soc/codecs/da7210.c
index 597510de561a..40d759938edc 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -1158,7 +1158,7 @@ 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/hdac_hda.c b/sound/soc/codecs/hdac_hda.c
index 1c06fdbf0e71..7414be08d3d5 100644
--- a/sound/soc/codecs/hdac_hda.c
+++ b/sound/soc/codecs/hdac_hda.c
@@ -520,7 +520,7 @@ 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 d9e6fa6d3e99..b3f677e7ddc9 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1753,7 +1753,7 @@ 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/max98927.c b/sound/soc/codecs/max98927.c
index d8dfaefdca7e..285c9d1ea721 100644
--- a/sound/soc/codecs/max98927.c
+++ b/sound/soc/codecs/max98927.c
@@ -672,7 +672,7 @@ static int max98927_probe(struct snd_soc_component *component)
0x80);
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);
regmap_write(max98927->regmap, MAX98927_R003C_SPK_GAIN, 0x05);
/* Enable DC blocker */
@@ -870,7 +870,7 @@ 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.c b/sound/soc/codecs/mt6359.c
index b15bdb15dbb0..c51eae602ca0 100644
--- a/sound/soc/codecs/mt6359.c
+++ b/sound/soc/codecs/mt6359.c
@@ -82,7 +82,7 @@ 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,
@@ -90,7 +90,7 @@ 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 */
@@ -99,7 +99,7 @@ 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,
@@ -107,7 +107,7 @@ 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,
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index 5165aeba4b67..2eb0efc983a0 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -345,10 +345,10 @@ static u32 nau8825_intlog10_dec3(u32 value)
/**
* nau8825_xtalk_sidetone - computes cross talk suppression sidetone gain.
*
- * @sig_org: orignal signal level
+ * @sig_org: original signal level
* @sig_cros: cross talk signal level
*
- * The orignal and cross talk signal vlues need to be characterized.
+ * The original and cross talk signal vlues need to be characterized.
* Once these values have been characterized, this sidetone value
* can be converted to decibel with the equation below.
* sidetone = 20 * log (original signal level / crosstalk signal level)
@@ -671,11 +671,11 @@ static void nau8825_xtalk_imm_stop(struct nau8825 *nau8825)
* Thus, the measurement function has four states to complete whole sequence.
* 1. Prepare state : Prepare the resource for detection and transfer to HPR
* IMM stat to make JKR1(HPR) impedance measure.
- * 2. HPR IMM state : Read out orignal signal level of JKR1(HPR) and transfer
+ * 2. HPR IMM state : Read out original signal level of JKR1(HPR) and transfer
* to HPL IMM state to make JKTIP(HPL) impedance measure.
* 3. HPL IMM state : Read out cross talk signal level of JKTIP(HPL) and
* transfer to IMM state to determine suppression sidetone gain.
- * 4. IMM state : Computes cross talk suppression sidetone gain with orignal
+ * 4. IMM state : Computes cross talk suppression sidetone gain with original
* and cross talk signal level. Apply this gain and then restore codec
* configuration. Then transfer to Done state for ending.
*/
@@ -724,7 +724,7 @@ static void nau8825_xtalk_measure(struct nau8825 *nau8825)
nau8825->xtalk_state = NAU8825_XTALK_IMM;
break;
case NAU8825_XTALK_IMM:
- /* In impedance measure state, the orignal and cross talk
+ /* In impedance measure state, the original and cross talk
* signal level vlues are ready. The side tone gain is deter-
* mined with these signal level. After all, restore codec
* configuration.
diff --git a/sound/soc/codecs/rt1320-sdw.c b/sound/soc/codecs/rt1320-sdw.c
index 444a91755be9..b13622e54ae0 100644
--- a/sound/soc/codecs/rt1320-sdw.c
+++ b/sound/soc/codecs/rt1320-sdw.c
@@ -2084,7 +2084,7 @@ static int rt1320_rae_load(struct rt1320_sdw_priv *rt1320)
regmap_update_bits(rt1320->regmap, 0x20005818, 0x80, 0x80);
/* RAE run */
regmap_update_bits(rt1320->regmap, 0x2000301c, 0x01, 0x01);
- /* Phase sync eanble */
+ /* Phase sync enable */
regmap_update_bits(rt1320->regmap, 0xc047, 0x80, 0x80);
break;
}
diff --git a/sound/soc/codecs/rt5631.h b/sound/soc/codecs/rt5631.h
index 8a6b99a48c7c..00734207ac9f 100644
--- a/sound/soc/codecs/rt5631.h
+++ b/sound/soc/codecs/rt5631.h
@@ -415,7 +415,7 @@
#define RT5631_ADDA_FILTER_CLK_SEL_256FS (0 << 7) /* 256FS */
#define RT5631_ADDA_FILTER_CLK_SEL_384FS (1 << 7) /* 384FS */
-/* Power managment addition 1 (0x3A) */
+/* Power management addition 1 (0x3A) */
#define RT5631_PWR_MAIN_I2S_EN (0x1 << 15)
#define RT5631_PWR_MAIN_I2S_BIT 15
#define RT5631_PWR_CLASS_D (0x1 << 12)
@@ -435,7 +435,7 @@
#define RT5631_PWR_DAC_R_TO_MIXER (0x1 << 5)
#define RT5631_PWR_DAC_R_TO_MIXER_BIT 5
-/* Power managment addition 2 (0x3B) */
+/* Power management addition 2 (0x3B) */
#define RT5631_PWR_OUTMIXER_L (0x1 << 15)
#define RT5631_PWR_OUTMIXER_L_BIT 15
#define RT5631_PWR_OUTMIXER_R (0x1 << 14)
@@ -461,7 +461,7 @@
#define RT5631_PWR_PLL2 (0x1 << 0)
#define RT5631_PWR_PLL2_BIT 0
-/* Power managment addition 3(0x3C) */
+/* Power management addition 3(0x3C) */
#define RT5631_PWR_VREF (0x1 << 15)
#define RT5631_PWR_VREF_BIT 15
#define RT5631_PWR_FAST_VREF_CTRL (0x1 << 14)
@@ -489,7 +489,7 @@
#define RT5631_PWR_HP_AMP_DRIVING (0x1 << 0)
#define RT5631_PWR_HP_AMP_DRIVING_BIT 0
-/* Power managment addition 4(0x3E) */
+/* Power management addition 4(0x3E) */
#define RT5631_PWR_SPK_L_VOL (0x1 << 15)
#define RT5631_PWR_SPK_L_VOL_BIT 15
#define RT5631_PWR_SPK_R_VOL (0x1 << 14)
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 4fa1eee8e87f..03d0ac3359f5 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1216,7 +1216,7 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
0, rt5640_spk_l_mix, ARRAY_SIZE(rt5640_spk_l_mix)),
SND_SOC_DAPM_MIXER("SPK MIXR", RT5640_PWR_MIXER, RT5640_PWR_SM_R_BIT,
0, rt5640_spk_r_mix, ARRAY_SIZE(rt5640_spk_r_mix)),
- /* Ouput Volume */
+ /* Output Volume */
SND_SOC_DAPM_PGA("SPKVOL L", RT5640_PWR_VOL,
RT5640_PWR_SV_L_BIT, 0, NULL, 0),
SND_SOC_DAPM_PGA("SPKVOL R", RT5640_PWR_VOL,
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index bf9f278c3de9..bb448254275f 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -2209,7 +2209,7 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = {
0, rt5645_out_l_mix, ARRAY_SIZE(rt5645_out_l_mix)),
SND_SOC_DAPM_MIXER("OUT MIXR", RT5645_PWR_MIXER, RT5645_PWR_OM_R_BIT,
0, rt5645_out_r_mix, ARRAY_SIZE(rt5645_out_r_mix)),
- /* Ouput Volume */
+ /* Output Volume */
SND_SOC_DAPM_SWITCH("SPKVOL L", RT5645_PWR_VOL, RT5645_PWR_SV_L_BIT, 0,
&spk_l_vol_control),
SND_SOC_DAPM_SWITCH("SPKVOL R", RT5645_PWR_VOL, RT5645_PWR_SV_R_BIT, 0,
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 9723b4c7540c..b911757376f5 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -1024,7 +1024,7 @@ static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = {
0, rt5651_out_l_mix, ARRAY_SIZE(rt5651_out_l_mix)),
SND_SOC_DAPM_MIXER("OUT MIXR", RT5651_PWR_MIXER, RT5651_PWR_OM_R_BIT,
0, rt5651_out_r_mix, ARRAY_SIZE(rt5651_out_r_mix)),
- /* Ouput Volume */
+ /* Output Volume */
SND_SOC_DAPM_SWITCH("OUTVOL L", RT5651_PWR_VOL,
RT5651_PWR_OV_L_BIT, 0, &outvol_l_control),
SND_SOC_DAPM_SWITCH("OUTVOL R", RT5651_PWR_VOL,
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index f772f093479a..62d4e7cea5af 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -1901,7 +1901,7 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = {
0, rt5670_out_l_mix, ARRAY_SIZE(rt5670_out_l_mix)),
SND_SOC_DAPM_MIXER("OUT MIXR", RT5670_PWR_MIXER, RT5670_PWR_OM_R_BIT,
0, rt5670_out_r_mix, ARRAY_SIZE(rt5670_out_r_mix)),
- /* Ouput Volume */
+ /* Output Volume */
SND_SOC_DAPM_MIXER("HPOVOL MIXL", RT5670_PWR_VOL,
RT5670_PWR_HV_L_BIT, 0,
rt5670_hpvoll_mix, ARRAY_SIZE(rt5670_hpvoll_mix)),
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 55a95b62a650..b91ab42882d1 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -914,7 +914,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream,
/* OSC calibration time */
dac33_write(component, DAC33_CALIB_TIME, 96);
- /* adjustment treshold & step */
+ /* adjustment threshold & step */
dac33_write(component, DAC33_INT_OSC_CTRL_B, DAC33_ADJTHRSHLD(2) |
DAC33_ADJSTEP(1));
diff --git a/sound/soc/intel/atom/sst-mfld-dsp.h b/sound/soc/intel/atom/sst-mfld-dsp.h
index c8f0816edb53..59606259a06e 100644
--- a/sound/soc/intel/atom/sst-mfld-dsp.h
+++ b/sound/soc/intel/atom/sst-mfld-dsp.h
@@ -189,7 +189,7 @@ struct ipc_dsp_hdr {
union ipc_header_high {
struct {
u32 msg_id:8; /* Message ID - Max 256 Message Types */
- u32 task_id:4; /* Task ID associated with this comand */
+ u32 task_id:4; /* Task ID associated with this command */
u32 drv_id:4; /* Identifier for the driver to track*/
u32 rsvd1:8; /* Reserved */
u32 result:4; /* Reserved */
diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h
index c43946c5ecee..e24d9dad7ef6 100644
--- a/sound/soc/intel/atom/sst/sst.h
+++ b/sound/soc/intel/atom/sst/sst.h
@@ -182,7 +182,7 @@ struct sst_block {
* @pcm_substream : PCM substream
* @period_elapsed : PCM period elapsed callback
* @sfreq : stream sampling freq
- * @cumm_bytes : cummulative bytes decoded
+ * @cumm_bytes : cumulative bytes decoded
*/
struct stream_info {
unsigned int status;
diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c
index 88fc6cb2bfd4..ec4bda40ae00 100644
--- a/sound/soc/intel/boards/sof_cirrus_common.c
+++ b/sound/soc/intel/boards/sof_cirrus_common.c
@@ -151,7 +151,7 @@ static const char * const cs35l41_name_prefixes[] = { "WL", "WR", "TL", "TR" };
* UID 0x2 -> TL
* UID 0x3 -> TR
* Note: If there are less than 4 Amps, UIDs still map to WL/WR/TL/TR. Dynamic code will only create
- * dai links for UIDs which exist, and ignore non-existant ones. Only 2 or 4 amps are expected.
+ * dai links for UIDs which exist, and ignore non-existent ones. Only 2 or 4 amps are expected.
* Return number of codecs found.
*/
static int cs35l41_compute_codec_conf(void)
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 4f09fdd40905..cd4f82bb8cc7 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -551,7 +551,7 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
prtd->stream_id,
CMD_CLOSE);
/*
- * vaild stream ids start from 1, So we are
+ * valid stream ids start from 1, So we are
* toggling this between 1 and 2.
*/
prtd->stream_id = (prtd->stream_id == 1 ? 2 : 1);
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index ec7eb2f19b6f..3614f614117d 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -49,7 +49,7 @@
* incompatible with ASP and with either McBSP.
*
* In short: this uses ASP to implement I2S, not McBSP.
- * And it won't be the only DaVinci implemention of I2S.
+ * And it won't be the only DaVinci implementation of I2S.
*/
#define DAVINCI_MCBSP_DRR_REG 0x00
#define DAVINCI_MCBSP_DXR_REG 0x04
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index ad9e194a3d0d..d461ab419655 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -1818,7 +1818,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
!mcasp->async_mode)
max_channels = mcasp->channels;
/*
- * But we can always allow channels upto the amount of
+ * But we can always allow channels up to the amount of
* the available tdm_slots.
*/
if (max_channels < tdm_slots)
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 5734388a4e4b..889e357b64d0 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -1237,7 +1237,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
break;
case OMAP_MCBSP_SYSCLK_CLKR_EXT:
regs->pcr0 |= SCLKME;
- /* Disable ouput on CLKR pin in master mode */
+ /* Disable output on CLKR pin in master mode */
regs->pcr0 &= ~CLKRM;
break;
default:
diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h
index 1900ea01ce3d..87b27256cf80 100644
--- a/sound/soc/uniphier/aio.h
+++ b/sound/soc/uniphier/aio.h
@@ -183,7 +183,7 @@ struct uniphier_aio_selector {
*
* These are examples for sound data paths:
*
- * For caputure device:
+ * For capture device:
* (outer of AIO) -> iport -> iif -> ch -> rb -> (CPU)
* For playback device:
* (CPU) -> rb -> ch -> oif -> oport -> (outer of AIO)
diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
index 683b485fb570..bd2cfc618290 100644
--- a/sound/soc/ux500/ux500_msp_i2s.c
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -19,7 +19,7 @@
#include "ux500_msp_i2s.h"
- /* Protocol desciptors */
+ /* Protocol descriptors */
static const struct msp_protdesc prot_descs[] = {
{ /* I2S */
MSP_SINGLE_PHASE,
base-commit: 260a144af70c8babbf2f15d1a56bacfa83f8aef6
--
2.48.1
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