From mboxrd@z Thu Jan 1 00:00:00 1970 Return-Path: X-Spam-Checker-Version: SpamAssassin 3.4.0 (2014-02-07) on aws-us-west-2-korg-lkml-1.web.codeaurora.org X-Spam-Level: X-Spam-Status: No, score=-15.7 required=3.0 tests=BAYES_00,DKIM_SIGNED, DKIM_VALID,DKIM_VALID_AU,FREEMAIL_FORGED_FROMDOMAIN,FREEMAIL_FROM, HEADER_FROM_DIFFERENT_DOMAINS,INCLUDES_CR_TRAILER,INCLUDES_PATCH, MAILING_LIST_MULTI,SPF_HELO_NONE,SPF_PASS,USER_AGENT_GIT autolearn=ham autolearn_force=no version=3.4.0 Received: from mail.kernel.org (mail.kernel.org [198.145.29.99]) by smtp.lore.kernel.org (Postfix) with ESMTP id 0DF33C07E95 for ; Wed, 7 Jul 2021 11:51:49 +0000 (UTC) Received: from vger.kernel.org (vger.kernel.org [23.128.96.18]) by mail.kernel.org (Postfix) with ESMTP id D936661CC5 for ; Wed, 7 Jul 2021 11:51:48 +0000 (UTC) Received: (majordomo@vger.kernel.org) by vger.kernel.org via listexpand id S231449AbhGGLy1 (ORCPT ); Wed, 7 Jul 2021 07:54:27 -0400 Received: from lindbergh.monkeyblade.net ([23.128.96.19]:33372 "EHLO lindbergh.monkeyblade.net" rhost-flags-OK-OK-OK-OK) by vger.kernel.org with ESMTP id S231358AbhGGLy0 (ORCPT ); Wed, 7 Jul 2021 07:54:26 -0400 Received: from mail-pl1-x62f.google.com (mail-pl1-x62f.google.com [IPv6:2607:f8b0:4864:20::62f]) by lindbergh.monkeyblade.net (Postfix) with ESMTPS id 019AAC061574 for ; Wed, 7 Jul 2021 04:51:46 -0700 (PDT) Received: by mail-pl1-x62f.google.com with SMTP id x3so922270pll.5 for ; Wed, 07 Jul 2021 04:51:45 -0700 (PDT) DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=gmail.com; s=20161025; h=from:to:cc:subject:date:message-id:mime-version :content-transfer-encoding; bh=oqyImscKIGHZdWry/NO6O15x9HwF7lH+KNgTMvp4IP0=; b=nbjaO/HS0vRkQUSzhOQo9OGmH9xkCc9dGZpPoV3gE/kGsalVhEFP8DqSM3ZS72N06C qHTWo46p/T+D1LLDr2KOYgCDAo6ybCAIFjTUcbofaIEyw4V60P6obGnOQdme2OjLBw1v ifxVbix1no0tLjRBnPpMxH1REEt4UI68aCE5to70vjO0sdEETxocHhkhCeTyr52mL2IX r8T49p4tjDZFOiA9Q0vvmbrsx79H0cdgHBD5glF6MGZyq2cog+i67iwfkIIsJ5rtYs5/ AKr5tvIgFDPqDG2O0Qo5RupZsG8li/fSoDlcu/Sb0GPJLeI8tpHBMBPNQiBfQMH+HgpE C51A== X-Google-DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=1e100.net; s=20161025; h=x-gm-message-state:from:to:cc:subject:date:message-id:mime-version :content-transfer-encoding; bh=oqyImscKIGHZdWry/NO6O15x9HwF7lH+KNgTMvp4IP0=; b=SngaOi9zKZvR0vAehTw5zgxdIpCXnJibSdPCY8Ll5SXyvxuh7ygF+rRQSW/Yc3VHNp exSl4BfGa9IcG9j3wKOQwKfL6tru+nyU7zIJhmFTQaIm9q4L2gWJmarCzmnpMBpbxwN9 Bj/Rx2UOkt3iwA+iEpTdaZAm7JGk+STW2XFVqhqjbVA/KQQsttYIWWdA5NVwRRtKk0z7 TFkZeuj+peTwHfpttHXL3i1y2gAPwYEwW5Mqv3rHQMeJDlrTsVINntTMWTUsMEk0fQQa he46dOp/wCZ6HeV42HGTdGoZlcrz2kYydAZTse/pdZGcc718qignlvm4fyeQF/wqlcg8 6YlQ== X-Gm-Message-State: AOAM531QL6zJ89gzCo5WwzGjz4FnQ8kgLEPNubhMOge2YkyKeMcrxu20 03gAyKBDz+hoBX56FBBgxXU= X-Google-Smtp-Source: ABdhPJwCqWNYOYzIehnlB6QDB9F908y6VC7TdG0vhGLlJW0hfnDzf08sbaJiDGmxvy3A7PITrAtuuQ== X-Received: by 2002:a17:902:b7c2:b029:128:c1cd:241e with SMTP id v2-20020a170902b7c2b0290128c1cd241emr21172571plz.14.1625658705391; Wed, 07 Jul 2021 04:51:45 -0700 (PDT) Received: from ubuntu.localdomain ([103.220.76.197]) by smtp.gmail.com with ESMTPSA id i24sm17795473pfr.56.2021.07.07.04.51.42 (version=TLS1_3 cipher=TLS_AES_256_GCM_SHA384 bits=256/256); Wed, 07 Jul 2021 04:51:44 -0700 (PDT) From: Gu Shengxian To: perex@perex.cz, tiwai@suse.com, james.schulman@cirrus.com, david.rhodes@cirrus.com, matthias.bgg@gmail.com Cc: linux-kernel@vger.kernel.org, patches@opensource.cirrus.com, Gu Shengxian Subject: [PATCH] ASoC: codecs: remove unneeded variable: "ret" Date: Wed, 7 Jul 2021 19:51:30 +0800 Message-Id: <20210707115131.9060-1-gushengxian507419@gmail.com> X-Mailer: git-send-email 2.25.1 MIME-Version: 1.0 Content-Transfer-Encoding: 8bit Precedence: bulk List-ID: X-Mailing-List: linux-kernel@vger.kernel.org From: Gu Shengxian The variable: "ret" is only defined and returned. So it could be removed. Fix some spelling mistakes. Signed-off-by: Gu Shengxian --- 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