Fix common misspellings

Fixes generated by 'codespell' and manually reviewed.

Signed-off-by: Lucas De Marchi <lucas.demarchi@profusion.mobi>

15 files changed, 21 insertions, 21 deletions

diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 4f377c9..eecffb5 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -481,7 +481,7 @@ struct _pll_div {
 };
 
 /* Note : pll code from original alc5623 driver. Not sure of how good it is */
-/* usefull only for master mode */
+/* useful only for master mode */
 static const struct _pll_div codec_master_pll_div[] = {
 
 	{  2048000,  8192000,	0x0ea0},
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index 72de47e..2c2a681 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -161,7 +161,7 @@ static const struct snd_kcontrol_new lm4857_controls[] = {
 		lm4857_get_mode, lm4857_set_mode),
 };
 
-/* There is a demux inbetween the the input signal and the output signals.
+/* There is a demux between the input signal and the output signals.
  * Currently there is no easy way to model it in ASoC and since it does not make
  * much of a difference in practice simply connect the input direclty to the
  * outputs. */
diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h
index 62b1f22..67f19c3 100644
--- a/sound/soc/codecs/tlv320aic26.h
+++ b/sound/soc/codecs/tlv320aic26.h
@@ -14,14 +14,14 @@
 #define AIC26_PAGE_ADDR(page, offset)	((page << 6) | offset)
 #define AIC26_NUM_REGS			AIC26_PAGE_ADDR(3, 0)
 
-/* Page 0: Auxillary data registers */
+/* Page 0: Auxiliary data registers */
 #define AIC26_REG_BAT1			AIC26_PAGE_ADDR(0, 0x05)
 #define AIC26_REG_BAT2			AIC26_PAGE_ADDR(0, 0x06)
 #define AIC26_REG_AUX			AIC26_PAGE_ADDR(0, 0x07)
 #define AIC26_REG_TEMP1			AIC26_PAGE_ADDR(0, 0x09)
 #define AIC26_REG_TEMP2			AIC26_PAGE_ADDR(0, 0x0A)
 
-/* Page 1: Auxillary control registers */
+/* Page 1: Auxiliary control registers */
 #define AIC26_REG_AUX_ADC		AIC26_PAGE_ADDR(1, 0x00)
 #define AIC26_REG_STATUS		AIC26_PAGE_ADDR(1, 0x01)
 #define AIC26_REG_REFERENCE		AIC26_PAGE_ADDR(1, 0x03)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 3bedab2..6c43c13 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -884,7 +884,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	if (bypass_pll)
 		return 0;
 
-	/* Use PLL, compute apropriate setup for j, d, r and p, the closest
+	/* Use PLL, compute appropriate setup for j, d, r and p, the closest
 	 * one wins the game. Try with d==0 first, next with d!=0.
 	 * Constraints for j are according to the datasheet.
 	 * The sysclk is divided by 1000 to prevent integer overflows.
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 00b6d87..f01f141 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1020,7 +1020,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 		/*
 		 * For FIFO bypass mode:
 		 * Enable the FIFO bypass (Disable the FIFO use)
-		 * Set the BCLK as continous
+		 * Set the BCLK as continuous
 		 */
 		fifoctrl_a |= DAC33_FBYPAS;
 		aictrl_b |= DAC33_BCLKON;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 8512800..575238d 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -281,7 +281,7 @@ static inline void twl4030_check_defaults(struct snd_soc_codec *codec)
 				 i, val, twl4030_reg[i]);
 		}
 	}
-	dev_dbg(codec->dev, "Found %d non maching registers. %s\n",
+	dev_dbg(codec->dev, "Found %d non-matching registers. %s\n",
 		 difference, difference ? "Not OK" : "OK");
 }
 
@@ -2018,7 +2018,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
 	u8 mode;
 
 	/* If the system master clock is not 26MHz, the voice PCM interface is
-	 * not avilable.
+	 * not available.
 	 */
 	if (twl4030->sysclk != 26000) {
 		dev_err(codec->dev, "The board is configured for %u Hz, while"
@@ -2028,7 +2028,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
 	}
 
 	/* If the codec mode is not option2, the voice PCM interface is not
-	 * avilable.
+	 * available.
 	 */
 	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
 		& TWL4030_OPT_MODE;
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 8f6b5ee..4bbc0a7 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -772,7 +772,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 			reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD);
 			snd_soc_write(codec, WM8580_PWRDN1, reg);
 
-			/* Make VMID high impedence */
+			/* Make VMID high impedance */
 			reg = snd_soc_read(codec,  WM8580_ADC_CONTROL1);
 			reg &= ~0x100;
 			snd_soc_write(codec, WM8580_ADC_CONTROL1, reg);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 3f09dee..ffa2ffe 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1312,7 +1312,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
 	SNDRV_PCM_FMTBIT_S24_LE)
 
 /*
- * The WM8753 supports upto 4 different and mutually exclusive DAI
+ * The WM8753 supports up to 4 different and mutually exclusive DAI
  * configurations. This gives 2 PCM's available for use, hifi and voice.
  * NOTE: The Voice PCM cannot play or capture audio to the CPU as it's DAI
  * is connected between the wm8753 and a BT codec or GSM modem.
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 443ae58..9b3bba4 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1895,7 +1895,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	pr_debug("Fvco=%dHz\n", target);
 
-	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	/* Find an appropriate FLL_FRATIO and factor it out of the target */
 	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
 		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
 			fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 5e0214d..3c71987 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -176,7 +176,7 @@ static int wm8995_pll_factors(struct device *dev,
 	return 0;
 }
 
-/* Lookup table specifiying SRATE (table 25 in datasheet); some of the
+/* Lookup table specifying SRATE (table 25 in datasheet); some of the
  * output frequencies have been rounded to the standard frequencies
  * they are intended to match where the error is slight. */
 static struct {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 3b71dd6..500011e 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3137,7 +3137,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	pr_debug("FLL Fvco=%dHz\n", target);
 
-	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	/* Find an appropriate FLL_FRATIO and factor it out of the target */
 	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
 		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
 			fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 28fdfd6..3c2ee1b 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -981,7 +981,7 @@ static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai,
 		reg = snd_soc_read(codec, WM8991_CLOCKING_2);
 		snd_soc_write(codec, WM8991_CLOCKING_2, reg | WM8991_SYSCLK_SRC);
 
-		/* set up N , fractional mode and pre-divisor if neccessary */
+		/* set up N , fractional mode and pre-divisor if necessary */
 		snd_soc_write(codec, WM8991_PLL1, pll_div.n | WM8991_SDM |
 			      (pll_div.div2 ? WM8991_PRESCALE : 0));
 		snd_soc_write(codec, WM8991_PLL2, (u8)(pll_div.k>>8));
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 379fa22..056aef9 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -324,7 +324,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	pr_debug("Fvco=%dHz\n", target);
 
-	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	/* Find an appropriate FLL_FRATIO and factor it out of the target */
 	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
 		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
 			fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3dc64c8..3290333 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -82,18 +82,18 @@ struct wm8994_priv {
 
 	int mbc_ena[3];
 
-	/* Platform dependant DRC configuration */
+	/* Platform dependent DRC configuration */
 	const char **drc_texts;
 	int drc_cfg[WM8994_NUM_DRC];
 	struct soc_enum drc_enum;
 
-	/* Platform dependant ReTune mobile configuration */
+	/* Platform dependent ReTune mobile configuration */
 	int num_retune_mobile_texts;
 	const char **retune_mobile_texts;
 	int retune_mobile_cfg[WM8994_NUM_EQ];
 	struct soc_enum retune_mobile_enum;
 
-	/* Platform dependant MBC configuration */
+	/* Platform dependent MBC configuration */
 	int mbc_cfg;
 	const char **mbc_texts;
 	struct soc_enum mbc_enum;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 55cdf29..91c6b39 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -305,7 +305,7 @@ static int speaker_mode_get(struct snd_kcontrol *kcontrol,
 /*
  * Stop any attempts to change speaker mode while the speaker is enabled.
  *
- * We also have some special anti-pop controls dependant on speaker
+ * We also have some special anti-pop controls dependent on speaker
  * mode which must be changed along with the mode.
  */
 static int speaker_mode_put(struct snd_kcontrol *kcontrol,
@@ -456,7 +456,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	pr_debug("Fvco=%dHz\n", target);
 
-	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	/* Find an appropriate FLL_FRATIO and factor it out of the target */
 	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
 		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
 			fll_div->fll_fratio = fll_fratios[i].fll_fratio;