* [PATCH] ALSA: fix typos in comments
@ 2026-09-07 6:32 Hemanth Selam
0 siblings, 0 replies; only message in thread
From: Hemanth Selam @ 2026-09-07 6:32 UTC (permalink / raw)
To: Jaroslav Kysela, Takashi Iwai, Clemens Ladisch, Takashi Sakamoto
Cc: linux-sound, linux-kernel
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 <hemanth.selam@gmail.com>
---
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 <tiwai@suse.de>
*
- * Definitions of OSS compatible headers for Emu8000 device informations
+ * Definitions of OSS compatible headers for Emu8000 device information
*/
#include <sound/seq_oss_legacy.h>
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 <torger@ludd.luth.se>,
* Henk Hesselink <henk@anda.nl>
* 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
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