From mboxrd@z Thu Jan 1 00:00:00 1970 Received: from mail-ot1-f47.google.com (mail-ot1-f47.google.com [209.85.210.47]) (using TLSv1.2 with cipher ECDHE-RSA-AES128-GCM-SHA256 (128/128 bits)) (No client certificate requested) by smtp.subspace.kernel.org (Postfix) with ESMTPS id BBC373E9580 for ; Mon, 7 Sep 2026 06:32:08 +0000 (UTC) Authentication-Results: smtp.subspace.kernel.org; arc=none smtp.client-ip=209.85.210.47 ARC-Seal:i=1; a=rsa-sha256; d=subspace.kernel.org; s=arc-20240116; t=1788762732; cv=none; b=DPGHXoKoAbzaWn75wAygpSArAMoo9dAw9ObAS40qzKuES66g/EiWcyfwkV/tN+SvbfkmWWAU8dM8FadeaZ6o+WZxJ/3ptwWqB4hVHxUt/hxHGp6ZkB7qz8bHTF7X1AjynPimXAV1/OebbAr7E/lY0fC32DF5w1olmmLT+2reb60= ARC-Message-Signature:i=1; a=rsa-sha256; d=subspace.kernel.org; s=arc-20240116; t=1788762732; c=relaxed/simple; bh=Ka9zf/mgpATO1G7iBmQhGjL0xwu8Ftbt1RDUJQeq1xw=; h=From:To:Cc:Subject:Date:Message-ID:MIME-Version; b=Nie/fYJAXX71gaQ1rqB5UE3t1Q9hvm85pjx7H24pnF6EiM7JzCbApru0h5bl0wKOgQ0ssjA8UDXom/LrQ/o5HtuZ2eXb06/kKRyTMxomiavo19zSoSW/8RDxRR5uON9EtQKMrV8Vw1o4zmIik4opABXvFRkVtdvw1awHd4pdd7g= ARC-Authentication-Results:i=1; smtp.subspace.kernel.org; dmarc=pass (p=none dis=none) header.from=gmail.com; spf=pass smtp.mailfrom=gmail.com; dkim=pass (2048-bit key) header.d=gmail.com header.i=@gmail.com header.b=J77+r4Dj; arc=none smtp.client-ip=209.85.210.47 Authentication-Results: smtp.subspace.kernel.org; dmarc=pass (p=none dis=none) header.from=gmail.com Authentication-Results: smtp.subspace.kernel.org; spf=pass smtp.mailfrom=gmail.com Authentication-Results: smtp.subspace.kernel.org; dkim=pass (2048-bit key) header.d=gmail.com header.i=@gmail.com header.b="J77+r4Dj" Received: by mail-ot1-f47.google.com with SMTP id 46e09a7af769-7f84a55cc06so2309980a34.2 for ; Sun, 06 Sep 2026 23:32:08 -0700 (PDT) DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=gmail.com; s=20251104; t=1788762727; x=1789367527; darn=vger.kernel.org; h=content-transfer-encoding:mime-version:message-id:date:subject:cc :to:from:from:to:cc:subject:date:message-id:reply-to:content-type; bh=N2d8OuPAlxNZ8hjWv+WV2WB8KqXzC5UIFEWCoCdAk0U=; b=J77+r4DjuUNrk8GU+Vudmy7AjuZpCUbCRI4skHBBklVb/G050qD1I4weFLw+eaBT0X GvCccLXshr2dAEhRxXMf8oi8j6ibLEMzKNoVkXT6w4pj4rTUtXzZrxEasU4H6WUqQjCi fZqB0iAWgXd7RleG9U/VQrOwcVq0TK0jN5qpftujnRO4i1oL8o77QdmMGBvvjPp4NxZy s4AETR6mJ1q4kP4vGLYQfUYk7LshotlWvBoRvxoiGp5M9Hm4je0f+xR3+VJoDBNkA79s 0dcQOTJKbbYGOyXZq8MiIY+ugvUpDHVmM6S/jNcocMmUT7MPSkiokaqpLIUdru9y2PsT l4/w== X-Google-DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=1e100.net; s=20251104; t=1788762727; x=1789367527; h=content-transfer-encoding:mime-version:message-id:date:subject:cc :to:from:x-gm-gg:x-gm-message-state:from:to:cc:subject:date :message-id:reply-to:content-type; bh=N2d8OuPAlxNZ8hjWv+WV2WB8KqXzC5UIFEWCoCdAk0U=; b=aMPlcV1ga9xymy7GcAExFPEMdvqPzeXq6HlgxVXAK2pnuTSVPllxp1dCWihWhhSeeV JW3TfCVzI7Ka6w/Z66zuWMmjkv0bRL5gEf+cCcCk9DU6pbeumrcVVJeFURTholpL1ftu N8beHaPLzGZzU78aLHg06Nwk0xg0dcpFB6LQuZtacoIXaFeWX7YnYSTSXDdAL8psoFn2 JOHcWpZvL3be9opJRTcdhxSLZ00ygmRlozznZs6OBrVKwje9EWZeQH7rmCN+BJjxOhW+ +ja8V26u2Q8USE1hkrVhsbQWJU92APef+qX7gAUfHotpOLl4TrcXM+4CzOJi+0uHGzYG w6yQ== X-Forwarded-Encrypted: i=1; AKwUvBxxgBY1IAZhpc2d0ELTh75UdTo3UpVtATUZxbCP/RL0wXvPMwA1ezaSHZMkkybuzJ+/43Cfv2sHymgtpvc=@vger.kernel.org X-Gm-Message-State: AFuF++mcj+ON/mA3qDjV/aFrgaRCHZg+Y8RL6Z2Gda06329bCgRQOsHX JiZZxdymiA5U+cqtGTCYNw3z1nmPGhswrr07cfPEB+Ql7EArV5G2X+Ib X-Gm-Gg: AYBFou0XebSTvQ+udvLhfanNnC/Pf7zsvKu6juc5cIAUrA3bG0++/PeN1yHSzb90jfO QRBWnFT2eHvZpfe4MVDUHTc9AtZ/0rTQmI1vq8ObzpD+LelJYom/fi81Mm4DNlFGdeN7C5a9j88 43wnOUG+yukCJbp0Jw7rd+rCZxf5tfJ010b5FwtALS2mlSTwzly+ZNh15SrV/8TLvTuQlYgglIW 586qsf38EahqmaTVtLKXgijS7T1STxyveGM3t2lV2DA1pFM4XwIZ0R50VASEqDAadCQwVnFBbkP ZPVyCiQHOECZ46+h2eBKpd8fHCQMJjk7cdnqZ5arktDi0SDeCX+HtN+CWZ6xcdQY8hdDweNHQ/8 NjYAT/P0HNdal7bgGPMZyYG8T4FBfVPuLkPzgAKXxO5ZZUpzt6KiCWFmzbw87NwO/h/PWcT4LkB 54bahGwkOriMoB9r/mF8PuyWwTrEUeoRHWVDd5oiS+6x0RcS76jbTo4HtRwgwdlD/aaz2Up8vTr KURWfvkXMPusbjOjiKIhHwN X-Received: by 2002:a05:6820:4d04:b0:6a0:f44b:1265 with SMTP id 006d021491bc7-6b6fa8e0837mr12749038eaf.4.1788762727148; Sun, 06 Sep 2026 23:32:07 -0700 (PDT) Received: from volcano9f6e-hostos.amd.com ([165.204.217.251]) by smtp.gmail.com with ESMTPSA id a92af1059eb24-143243767e1sm21042867c88.6.2026.09.06.23.32.04 (version=TLS1_3 cipher=TLS_AES_256_GCM_SHA384 bits=256/256); Sun, 06 Sep 2026 23:32:06 -0700 (PDT) From: Hemanth Selam To: Jaroslav Kysela , Takashi Iwai , Clemens Ladisch , Takashi Sakamoto Cc: linux-sound@vger.kernel.org, linux-kernel@vger.kernel.org Subject: [PATCH] ALSA: fix typos in comments Date: Mon, 7 Sep 2026 12:02:02 +0530 Message-ID: <20260907063202.20577-1-hemanth.selam@gmail.com> X-Mailer: git-send-email 2.48.1 Precedence: bulk X-Mailing-List: linux-kernel@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/sound/emu10k1.h | 4 ++-- include/sound/emux_legacy.h | 2 +- sound/core/seq/oss/seq_oss_init.c | 2 +- sound/firewire/bebob/bebob_hwdep.c | 2 +- sound/hda/codecs/hdmi/hdmi.c | 2 +- sound/hda/codecs/realtek/alc882.c | 2 +- sound/hda/codecs/sigmatel.c | 6 +++--- sound/hda/common/codec.c | 2 +- sound/i2c/other/ak4113.c | 2 +- sound/i2c/other/ak4114.c | 2 +- sound/i2c/other/ak4xxx-adda.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 2 +- sound/mips/hal2.c | 4 ++-- sound/pci/ac97/ac97_codec.c | 2 +- sound/pci/ac97/ac97_patch.c | 2 +- sound/pci/au88x0/au88x0.h | 2 +- sound/pci/au88x0/au88x0_mpu401.c | 2 +- sound/pci/aw2/aw2-saa7146.c | 4 ++-- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/cs46xx_lib.c | 6 +++--- sound/pci/cs46xx/dsp_spos_scb_lib.c | 8 ++++---- sound/pci/echoaudio/echoaudio.c | 4 ++-- sound/pci/echoaudio/echoaudio.h | 4 ++-- sound/pci/lx6464es/lx_core.c | 2 +- sound/pci/lx6464es/lx_defs.h | 2 +- sound/pci/pcxhr/pcxhr_core.c | 4 ++-- sound/pci/riptide/riptide.c | 4 ++-- sound/pci/rme32.c | 2 +- sound/pci/rme9652/hdsp.c | 2 +- sound/pci/trident/trident_memory.c | 2 +- sound/pci/vx222/vx222_ops.c | 2 +- sound/pcmcia/vx/vxp_ops.c | 2 +- sound/ppc/tumbler.c | 2 +- sound/usb/mixer.c | 4 ++-- sound/usb/mixer_us16x08.h | 2 +- 35 files changed, 50 insertions(+), 50 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 4f94565c9d15..6474239a11c1 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -633,8 +633,8 @@ SUB_REG(PEFE, FILTERAMOUNT, 0x000000ff) /* Filter envlope amount */ #define A_ADCCR_RCHANENABLE 0x00000020 #define A_ADCCR_LCHANENABLE 0x00000010 -#define A_ADCCR_SAMPLERATE_MASK 0x0000000F /* Audigy sample rate convertor output rate */ -#define ADCCR_SAMPLERATE_MASK 0x00000007 /* Sample rate convertor output rate */ +#define A_ADCCR_SAMPLERATE_MASK 0x0000000F /* Audigy sample rate converter output rate */ +#define ADCCR_SAMPLERATE_MASK 0x00000007 /* Sample rate converter output rate */ #define ADCCR_SAMPLERATE_48 0x00000000 /* 48kHz sample rate */ #define ADCCR_SAMPLERATE_44 0x00000001 /* 44.1kHz sample rate */ #define ADCCR_SAMPLERATE_32 0x00000002 /* 32kHz sample rate */ diff --git a/include/sound/emux_legacy.h b/include/sound/emux_legacy.h index 1127e30d33bb..55a7fbfb9bfa 100644 --- a/include/sound/emux_legacy.h +++ b/include/sound/emux_legacy.h @@ -5,7 +5,7 @@ /* * Copyright (c) 1999-2000 Takashi Iwai * - * Definitions of OSS compatible headers for Emu8000 device informations + * Definitions of OSS compatible headers for Emu8000 device information */ #include diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index 1aece46c8b06..6586e07431c3 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -376,7 +376,7 @@ delete_seq_queue(int queue) /* - * free device informations - private_free callback of port + * free device information - private_free callback of port */ static void free_devinfo(void *private) diff --git a/sound/firewire/bebob/bebob_hwdep.c b/sound/firewire/bebob/bebob_hwdep.c index 216d1fceb6e7..1c523525fad2 100644 --- a/sound/firewire/bebob/bebob_hwdep.c +++ b/sound/firewire/bebob/bebob_hwdep.c @@ -8,7 +8,7 @@ /* * This codes give three functionality. * - * 1.get firewire node infomation + * 1.get firewire node information * 2.get notification about starting/stopping stream * 3.lock/unlock stream */ diff --git a/sound/hda/codecs/hdmi/hdmi.c b/sound/hda/codecs/hdmi/hdmi.c index 1e7a05c87733..a8aac0b36c73 100644 --- a/sound/hda/codecs/hdmi/hdmi.c +++ b/sound/hda/codecs/hdmi/hdmi.c @@ -1671,7 +1671,7 @@ int snd_hda_hdmi_generic_pcm_prepare(struct hda_pcm_stream *hinfo, * After S3, the audio driver restores pin:cvt selections * but this can happen before gfx is ready and such selection * is overlooked by HW. Thus multiple pins can share a same - * default convertor and mute control will affect each other, + * default converter and mute control will affect each other, * which can cause a resumed audio playback become silent * after S3. */ diff --git a/sound/hda/codecs/realtek/alc882.c b/sound/hda/codecs/realtek/alc882.c index fd466b6985f0..800c3d6f5134 100644 --- a/sound/hda/codecs/realtek/alc882.c +++ b/sound/hda/codecs/realtek/alc882.c @@ -372,7 +372,7 @@ static const struct hda_fixup alc882_fixups[] = { [ALC883_FIXUP_ACER_EAPD] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { - /* eanable EAPD on Acer laptops */ + /* enable EAPD on Acer laptops */ { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, { } diff --git a/sound/hda/codecs/sigmatel.c b/sound/hda/codecs/sigmatel.c index ee3bd21adc36..9a8ba43667f7 100644 --- a/sound/hda/codecs/sigmatel.c +++ b/sound/hda/codecs/sigmatel.c @@ -361,7 +361,7 @@ static unsigned int stac_vref_led_power_filter(struct hda_codec *codec, return snd_hda_gen_path_power_filter(codec, nid, power_state); } -/* update mute-LED accoring to the master switch */ +/* update mute-LED according to the master switch */ static void stac_update_led_status(struct hda_codec *codec, bool muted) { struct sigmatel_spec *spec = codec->spec; @@ -1710,7 +1710,7 @@ static const struct hda_pintbl ref92hd73xx_pin_configs[] = { { 0x11, 0x01014030 }, // CD in { 0x12, 0x02319040 }, - // Digial Mic ins + // Digital Mic ins { 0x13, 0x90a000f0 }, { 0x14, 0x90a000f0 }, // Digital outs @@ -3817,7 +3817,7 @@ static const struct hda_pintbl dell_3st_pin_configs[] = { static void stac927x_fixup_ref_no_jd(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - /* no jack detecion for ref-no-jd model */ + /* no jack detection for ref-no-jd model */ if (action == HDA_FIXUP_ACT_PRE_PROBE) codec->no_jack_detect = 1; } diff --git a/sound/hda/common/codec.c b/sound/hda/common/codec.c index 7d17d773cfbf..b1965fb66814 100644 --- a/sound/hda/common/codec.c +++ b/sound/hda/common/codec.c @@ -3772,7 +3772,7 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_open); /** - * snd_hda_multi_out_analog_prepare - Preapre the analog outputs. + * snd_hda_multi_out_analog_prepare - Prepare the analog outputs. * @codec: the HDA codec * @mout: hda_multi_out object * @stream_tag: stream tag to assign diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index 6ef5fffda8d7..9405d985a87d 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -130,7 +130,7 @@ void snd_ak4113_reinit(struct ak4113 *chip) scoped_guard(mutex, &chip->reinit_mutex) { ak4113_init_regs(chip); } - /* bring up statistics / event queing */ + /* bring up statistics / event queuing */ if (atomic_dec_and_test(&chip->wq_processing)) schedule_delayed_work(&chip->work, HZ / 10); } diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index 71efb29b6c7c..4c3dbbf8c316 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -135,7 +135,7 @@ void snd_ak4114_reinit(struct ak4114 *chip) scoped_guard(mutex, &chip->reinit_mutex) { ak4114_init_regs(chip); } - /* bring up statistics / event queing */ + /* bring up statistics / event queuing */ if (atomic_dec_and_test(&chip->wq_processing)) schedule_delayed_work(&chip->work, HZ / 10); } diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 9dd36b82a6ac..e7410f0f8660 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -147,7 +147,7 @@ EXPORT_SYMBOL(snd_akm4xxx_reset); * Volume conversion table for non-linear volumes * from -63.5dB (mute) to 0dB step 0.5dB * - * Used for AK4524/AK4620 input/ouput attenuation, AK4528, and + * Used for AK4524/AK4620 input/output attenuation, AK4528, and * AK5365 input attenuation */ static const unsigned char vol_cvt_datt[128] = { diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index abaa3ed3ab5c..8a6db40ff893 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -378,7 +378,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(21), 0x82, 0xff); /* - * This bit sets OPTI931 to automaticaly select FM + * This bit sets OPTI931 to automatically select FM * or digital input signal. */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index 2beb6a0dc3b7..84f265335cf9 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -375,7 +375,7 @@ static void hal2_setup_dac(struct snd_hal2 *hal2) hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX); /* Setup the HAL2 for playback */ hal2_set_dac_rate(hal2); - /* Set endianess */ + /* Set endianness */ hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECTX); /* Set DMA bus */ hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr)); @@ -400,7 +400,7 @@ static void hal2_setup_adc(struct snd_hal2 *hal2) hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR); /* Setup the HAL2 for record */ hal2_set_adc_rate(hal2); - /* Set endianess */ + /* Set endianness */ hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECR); /* Set DMA bus */ hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr)); diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 0bb65be021d9..207ce083593f 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2960,7 +2960,7 @@ int snd_ac97_tune_hardware(struct snd_ac97 *ac97, { int result; - /* quirk overriden? */ + /* quirk overridden? */ if (override && strcmp(override, "-1") && strcmp(override, "default")) { result = apply_quirk_str(ac97, override); if (result < 0) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 64cc39dd2008..8577a2e0510c 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -430,7 +430,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97) /* Patch for Yamaha YMF753, Copyright (c) by David Shust, dshust@shustring.com. This chip has nonstandard and extended behaviour with regard to its S/PDIF output. The AC'97 spec states that the S/PDIF signal is to be output at pin 48. - The YMF753 will ouput the S/PDIF signal to pin 43, 47 (EAPD), or 48. + The YMF753 will output the S/PDIF signal to pin 43, 47 (EAPD), or 48. By default, no output pin is selected, and the S/PDIF signal is not output. There is also a bit to mute S/PDIF output in a vendor-specific register. */ diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index 6cbb2bc4a048..f97a0303da3e 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -28,7 +28,7 @@ /* Vortex MPU401 defines. */ #define MIDI_CLOCK_DIV 0x61 -/* Standart MPU401 defines. */ +/* Standard MPU401 defines. */ #define MPU401_RESET 0xff #define MPU401_ENTER_UART 0x3f #define MPU401_ACK 0xfe diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 164f6b7039ab..dff4bf80ab96 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -22,7 +22,7 @@ /* Vortex MPU401 defines. */ #define MIDI_CLOCK_DIV 0x61 -/* Standart MPU401 defines. */ +/* Standard MPU401 defines. */ #define MPU401_RESET 0xff #define MPU401_ENTER_UART 0x3f #define MPU401_ACK 0xfe diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c index c84f1a45194f..6857a0ec2924 100644 --- a/sound/pci/aw2/aw2-saa7146.c +++ b/sound/pci/aw2/aw2-saa7146.c @@ -147,7 +147,7 @@ void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146 *chip, unsigned long dw_page, dw_limit; /* Configure DMA for substream - Configuration informations: ALSA has allocated continuous memory + Configuration information: ALSA has allocated continuous memory pages. So we don't need to use MMU of saa7146. */ @@ -204,7 +204,7 @@ void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146 *chip, unsigned long dw_page, dw_limit; /* Configure DMA for substream - Configuration informations: ALSA has allocated continuous memory + Configuration information: ALSA has allocated continuous memory pages. So we don't need to use MMU of saa7146. */ diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index f51f4bb63766..27ceda6278c5 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -324,7 +324,7 @@ MODULE_PARM_DESC(dual_codec, "Secondary Codec ID (0 = disabled)."); #define BA0_SSCR_XLPSRC (1<<8) /* External SRC Loopback Mode */ #define BA0_SSCR_LPSRC (1<<7) /* SRC Loopback Mode */ #define BA0_SSCR_CDTX (1<<5) /* CD Transfer Data */ -#define BA0_SSCR_HVC (1<<3) /* Harware Volume Control Enable */ +#define BA0_SSCR_HVC (1<<3) /* Hardware Volume Control Enable */ #define BA0_FMLVC 0x0754 /* FM Synthesis Left Volume Control */ #define BA0_FMRVC 0x0758 /* FM Synthesis Right Volume Control */ diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 19a6927c079d..a16586f7d913 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -315,7 +315,7 @@ int snd_cs46xx_download(struct snd_cs46xx *chip, dst = chip->region.idx[bank+1].remap_addr + offset; len /= sizeof(u32); - /* writel already converts 32-bit value to right endianess */ + /* writel already converts 32-bit value to right endianness */ while (len-- > 0) { writel(*src++, dst); dst += sizeof(u32); @@ -474,7 +474,7 @@ int snd_cs46xx_clear_BA1(struct snd_cs46xx *chip, dst = chip->region.idx[bank+1].remap_addr + offset; len /= sizeof(u32); - /* writel already converts 32-bit value to right endianess */ + /* writel already converts 32-bit value to right endianness */ while (len-- > 0) { writel(0, dst); dst += sizeof(u32); @@ -3039,7 +3039,7 @@ static int snd_cs46xx_chip_init(struct snd_cs46xx *chip) #endif /* - * Assert the vaid frame signal so that we can start sending commands + * Assert the valid frame signal so that we can start sending commands * to the AC97 codec. */ snd_cs46xx_pokeBA0(chip, BA0_ACCTL, ACCTL_VFRM | ACCTL_ESYN | ACCTL_RSTN); diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index fd19365026b4..d483f9a609be 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -160,15 +160,15 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * { struct dsp_spos_instance * ins = chip->dsp_spos_instance; - /* check integrety */ + /* check integrity */ if (snd_BUG_ON(scb->index < 0 || scb->index >= ins->nscb || (ins->scbs + scb->index) != scb)) return; #if 0 - /* can't remove a SCB with childs before - removing childs first */ + /* can't remove a SCB with children before + removing children first */ if (snd_BUG_ON(scb->sub_list_ptr != ins->the_null_scb || scb->next_scb_ptr != ins->the_null_scb)) goto _end; @@ -1656,7 +1656,7 @@ int cs46xx_dsp_disable_spdif_out (struct snd_cs46xx *chip) return -EBUSY; } - /* check integrety */ + /* check integrity */ if (snd_BUG_ON(!ins->asynch_tx_scb)) return -EINVAL; if (snd_BUG_ON(!ins->spdif_pcm_input_scb)) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 214d68657707..8b4637231deb 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -270,7 +270,7 @@ static int pcm_open(struct snd_pcm_substream *substream, return -ENOMEM; pipe->index = -1; /* Not configured yet */ - /* Set up hw capabilities and contraints */ + /* Set up hw capabilities and constraints */ memcpy(&pipe->hw, &pcm_hardware_skel, sizeof(struct snd_pcm_hardware)); dev_dbg(chip->card->dev, "max_channels=%d\n", max_channels); pipe->constr.list = channels_list; @@ -1720,7 +1720,7 @@ static const struct snd_kcontrol_new snd_echo_vumeters = { -/*** Channels info - it exports informations about the number of channels ***/ +/*** Channels info - it exports information about the number of channels ***/ static int snd_echo_channels_info_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 7a5744bf3155..ac4507b7cc78 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -394,8 +394,8 @@ struct echoaudio { u16 digital_modes; /* Bitmask of supported modes * (see ECHOCAPS_HAS_DIGITAL_MODE_*) */ - u16 input_clock_types; /* Suppoted input clock types */ - u16 output_clock_types; /* Suppoted output clock types - + u16 input_clock_types; /* Supported input clock types */ + u16 output_clock_types; /* Supported output clock types - * Layla20 only */ u16 device_id, subdevice_id; u16 *dsp_code; /* Current DSP code loaded, diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 9909f1067871..ae265dd19333 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -364,7 +364,7 @@ int lx_dsp_get_mac(struct lx6464es *chip) macmsb = lx_dsp_reg_read(chip, eReg_ADMACESMSB) & 0x00FFFFFF; maclsb = lx_dsp_reg_read(chip, eReg_ADMACESLSB) & 0x00FFFFFF; - /* todo: endianess handling */ + /* todo: endianness handling */ chip->mac_address[5] = ((u8 *)(&maclsb))[0]; chip->mac_address[4] = ((u8 *)(&maclsb))[1]; chip->mac_address[3] = ((u8 *)(&maclsb))[2]; diff --git a/sound/pci/lx6464es/lx_defs.h b/sound/pci/lx6464es/lx_defs.h index eca5367ba561..89e433d3f7a2 100644 --- a/sound/pci/lx6464es/lx_defs.h +++ b/sound/pci/lx6464es/lx_defs.h @@ -315,7 +315,7 @@ enum stream_flags { #define ED_XILINX_ERROR (ED_RT | 0x07) #define ED_COBRANET_ITF_NOT_RESPONDING (ED_RT | 0x08) -/* Complete BOARD error code for the invaid standard object class */ +/* Complete BOARD error code for the invalid standard object class */ #define EB_ISO (ERROR_VALUE | E_SOURCE_BOARD | \ E_CLASS_INVALID_STD_OBJECT) #define EB_INVALID_EFFECT (EB_ISO | 0x00) diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index b3b9ab4f303e..f855d3bfad4b 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -802,7 +802,7 @@ static int pcxhr_prepair_pipe_start(struct pcxhr_mgr *mgr, err); return err; } - /* if the pipe couldn't be prepaired for start, + /* if the pipe couldn't be prepared for start, * retry it later */ if (rmh.stat[0] == 0) @@ -913,7 +913,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, if (err) return err; if (state == 0) - break; /* success, all pipes prepaired */ + break; /* success, all pipes prepared */ mdelay(1); /* wait 1 millisecond and retry */ } } else { diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 99c00e46ce23..697fece744e7 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -264,7 +264,7 @@ MODULE_PARM_DESC(opl3_port, "OPL3 port # for Riptide driver."); #define SEND_SI2S(p,b) sendcmd(p,PARM,SI2S,WORD2(b),RET(0)) #define EOB_STATUS 0x80000000 /* status flags : block boundary */ -#define EOS_STATUS 0x40000000 /* : stoppped */ +#define EOS_STATUS 0x40000000 /* : stopped */ #define EOC_STATUS 0x20000000 /* : stream end */ #define ERR_STATUS 0x10000000 #define EMPTY_STATUS 0x08000000 @@ -453,7 +453,7 @@ struct snd_riptide { int in_suspend; }; -struct sgd { /* scatter gather desriptor */ +struct sgd { /* scatter gather descriptor */ __le32 dwNextLink; __le32 dwSegPtrPhys; __le32 dwSegLen; diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 454a30a2c07e..1d5390621908 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -8,7 +8,7 @@ * Thanks to : Anders Torger , * Henk Hesselink * for writing the digi96-driver - * and RME for all informations. + * and RME for all information. * * **************************************************************************** * diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 31cc2d91c8d2..0ade73057731 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3664,7 +3664,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "\n"); - /* Informations about H9632 specific controls */ + /* Information about H9632 specific controls */ if (hdsp->io_type == H9632) { char *tmp; diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index 81f6348191dc..4ed0e7d68322 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -27,7 +27,7 @@ #if PAGE_SIZE == 4096 /* page size == SNDRV_TRIDENT_PAGE_SIZE */ #define ALIGN_PAGE_SIZE PAGE_SIZE /* minimum page size for allocation */ -#define MAX_ALIGN_PAGES SNDRV_TRIDENT_MAX_PAGES /* maxmium aligned pages */ +#define MAX_ALIGN_PAGES SNDRV_TRIDENT_MAX_PAGES /* maximum aligned pages */ /* fill TLB entrie(s) corresponding to page with ptr */ #define set_tlb_bus(trident,page,addr) __set_tlb_bus(trident,page,addr) /* fill TLB entrie(s) corresponding to page with silence pointer */ diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index b6459dbdb1b0..40f3716c478f 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -431,7 +431,7 @@ static int vx2_load_dsp(struct vx_core *vx, int index, const struct firmware *ds /* * vx_test_and_ack - test and acknowledge interrupt * - * called from irq hander, too + * called from irq handler, too * * spinlock held! */ diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c index 4211e7239138..e52b61125629 100644 --- a/sound/pcmcia/vx/vxp_ops.c +++ b/sound/pcmcia/vx/vxp_ops.c @@ -265,7 +265,7 @@ static int vxp_load_dsp(struct vx_core *vx, int index, const struct firmware *fw /* * vx_test_and_ack - test and acknowledge interrupt * - * called from irq hander, too + * called from irq handler, too * * spinlock held! */ diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 139c7b25927a..7ba52e7a24d7 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1134,7 +1134,7 @@ static long tumbler_find_device(const char *device, const char *platform, gp->active_val = 0x4; gp->inactive_val = 0x5; /* Here are some crude hacks to extract the GPIO polarity and - * open collector informations out of the do-platform script + * open collector information out of the do-platform script * as we don't yet have an interpreter for these things */ if (platform) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ecaa8bc08d7c..33a6a1281410 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -786,7 +786,7 @@ static int parse_term_uac2_iterm_unit(struct mixer_build *state, return err; /* save input term properties after recursion, - * to ensure they are not overriden by the recursion calls + * to ensure they are not overridden by the recursion calls */ term->id = id; term->type = le16_to_cpu(d->wTerminalType); @@ -809,7 +809,7 @@ static int parse_term_uac3_iterm_unit(struct mixer_build *state, return err; /* save input term properties after recursion, - * to ensure they are not overriden by the recursion calls + * to ensure they are not overridden by the recursion calls */ term->id = id; term->type = le16_to_cpu(d->wTerminalType); diff --git a/sound/usb/mixer_us16x08.h b/sound/usb/mixer_us16x08.h index 7b8583dd3b07..e98fe55265d1 100644 --- a/sound/usb/mixer_us16x08.h +++ b/sound/usb/mixer_us16x08.h @@ -60,7 +60,7 @@ #define SND_US16X08_ID_EQHIGHMIDLEVEL 0x03 #define SND_US16X08_ID_EQHIGHLEVEL 0x04 -/* EQ frequence IDs */ +/* EQ frequency IDs */ #define SND_US16X08_ID_EQLOWFREQ 0x11 #define SND_US16X08_ID_EQLOWMIDFREQ 0x12 #define SND_US16X08_ID_EQHIGHMIDFREQ 0x13 -- 2.48.1