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Fri, 04 Sep 2026 04:03:30 -0700 (PDT) Received: from volcano9f6e-hostos.amd.com ([165.204.217.251]) by smtp.gmail.com with ESMTPSA id 5a478bee46e88-3339a62be87sm5320081eec.6.2026.09.04.04.03.17 (version=TLS1_3 cipher=TLS_AES_256_GCM_SHA384 bits=256/256); Fri, 04 Sep 2026 04:03:29 -0700 (PDT) From: Hemanth Selam To: Liam Girdwood , Mark Brown , Rob Herring , Krzysztof Kozlowski , Conor Dooley , Claudiu Beznea , Andrei Simion , Jaroslav Kysela , Takashi Iwai , Nicolas Ferre , Alexandre Belloni , Ray Jui , Scott Branden , Broadcom internal kernel review list , Lars-Peter Clausen , =?UTF-8?q?Nuno=20S=C3=A1?= , 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 Cc: linux-kernel@vger.kernel.org, linux-sound@vger.kernel.org, devicetree@vger.kernel.org, linux-arm-kernel@lists.infradead.org, patches@opensource.cirrus.com, linux-mediatek@lists.infradead.org, linux-arm-msm@vger.kernel.org, linux-omap@vger.kernel.org Subject: [PATCH] ASoC: fix typos in comments Date: Fri, 4 Sep 2026 16:33:01 +0530 Message-ID: <20260904110315.11650-1-hemanth.selam@gmail.com> X-Mailer: git-send-email 2.48.1 Precedence: bulk X-Mailing-List: devicetree@vger.kernel.org List-Id: List-Subscribe: List-Unsubscribe: MIME-Version: 1.0 Content-Transfer-Encoding: 8bit Fix typos in comments, reported by scripts/checkpatch.pl using the misspelling list in scripts/spelling.txt. Only touches comments, no code changes. Assisted-by: Cursor:claude-opus-5 Signed-off-by: Hemanth Selam --- include/dt-bindings/sound/audio-graph.h | 2 +- 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 +- 31 files changed, 44 insertions(+), 44 deletions(-) diff --git a/include/dt-bindings/sound/audio-graph.h b/include/dt-bindings/sound/audio-graph.h index bdb70c6b7332..44c803d61506 100644 --- a/include/dt-bindings/sound/audio-graph.h +++ b/include/dt-bindings/sound/audio-graph.h @@ -21,6 +21,6 @@ #define SND_SOC_TRIGGER_LINK 0 #define SND_SOC_TRIGGER_COMPONENT 1 #define SND_SOC_TRIGGER_DAI 2 -#define SND_SOC_TRIGGER_SIZE 3 /* shoud be last */ +#define SND_SOC_TRIGGER_SIZE 3 /* should be last */ #endif /* __AUDIO_GRAPH_H */ 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 329ab01b62c0..3f02b8a0edc9 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 - * based on an inital version by Cliff Cai + * based on an initial version by Cliff Cai */ #include @@ -688,7 +688,7 @@ 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 e217da124549..ce6116cde70f 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 fadf68860601..6152b54d9fd5 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1673,7 +1673,7 @@ 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 6ddc6549492a..84481809915d 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 4df1a25e318b..df9b3b01ef96 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/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 524300d8fb0b..217921e8fe05 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -662,7 +662,7 @@ 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 */ @@ -860,7 +860,7 @@ 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 ae83b778bf93..91ef9ec39f2f 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 90333779971d..eebd18a98fd5 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 74fd05176eff..8643f616a886 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 7313ea71c958..115af391691d 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 5c729135f71b..8de06a9309ed 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 1d120b5dc2be..780f9f9d0561 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 9bd2ddd8dacd..9a2c3828d4df 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 059967f0e632..fda0a7dfe281 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 79239131b659..ebe8bc77a9a6 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 26af616c33f5..e3ca93bd9dd6 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 fbfeefa418ca..b85e4b9c3ea1 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, -- 2.48.1