[PATCH v2 00/23] ALSA: fix typos in comments

[PATCH v2 00/23] ALSA: fix typos in comments

[Hemanth Selam]

v1 was a single patch touching 35 files, which was too coarse and would
get in the way of backporting real fixes later on.

This v2 splits it per driver, so each patch stays confined to one driver
and can be dropped or backported on its own.  Drivers made up of several
files (hda, usb-audio, i2c, ac97, ...) are kept together in a single
patch rather than going all the way down to one patch per file.

Comments only, no functional change.  The combined diff of this series
is identical to v1 except for a trailing whitespace dropped from one
cs46xx comment line that checkpatch complained about.

Built with allmodconfig on x86, sound/ compiles clean with no new
warnings.  sound/mips/hal2.c and sound/ppc/tumbler.c are arch-gated so
they were not compile-tested here; both are comment-only as well.

v1: https://lore.kernel.org/all/20260907063202.20577-1-hemanth.selam@gmail.com/

Hemanth Selam (23):
  ALSA: core: seq: oss: fix typos in comments
  ALSA: hda: fix typos in comments
  ALSA: usb-audio: fix typos in comments
  ALSA: i2c: fix typos in comments
  ALSA: emu10k1: fix typos in comments
  ALSA: emux: fix typos in comments
  ALSA: firewire: bebob: fix typos in comments
  ALSA: opti9xx: fix typos in comments
  ALSA: hal2: fix typos in comments
  ALSA: ppc: tumbler: fix typos in comments
  ALSA: ac97: fix typos in comments
  ALSA: au88x0: fix typos in comments
  ALSA: aw2: fix typos in comments
  ALSA: cs4281: fix typos in comments
  ALSA: cs46xx: fix typos in comments
  ALSA: echoaudio: fix typos in comments
  ALSA: lx6464es: fix typos in comments
  ALSA: pcxhr: fix typos in comments
  ALSA: riptide: fix typos in comments
  ALSA: rme32: fix typos in comments
  ALSA: hdsp: fix typos in comments
  ALSA: trident: fix typos in comments
  ALSA: vx: fix typos in comments

 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(-)


base-commit: 2e4c9ab2574eeaa7b893c7d35b4a9f60fd681ddd
-- 
2.48.1

[PATCH v2 01/23] ALSA: core: seq: oss: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/core/seq/oss/seq_oss_init.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

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)
-- 
2.48.1

[PATCH v2 02/23] ALSA: hda: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 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 +-
 4 files changed, 6 insertions(+), 6 deletions(-)

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
-- 
2.48.1

[PATCH v2 03/23] ALSA: usb-audio: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/usb/mixer.c         | 4 ++--
 sound/usb/mixer_us16x08.h | 2 +-
 2 files changed, 3 insertions(+), 3 deletions(-)

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

[PATCH v2 04/23] ALSA: i2c: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/i2c/other/ak4113.c      | 2 +-
 sound/i2c/other/ak4114.c      | 2 +-
 sound/i2c/other/ak4xxx-adda.c | 2 +-
 3 files changed, 3 insertions(+), 3 deletions(-)

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] = {
-- 
2.48.1

[PATCH v2 05/23] ALSA: emu10k1: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 include/sound/emu10k1.h | 4 ++--
 1 file changed, 2 insertions(+), 2 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					*/
-- 
2.48.1

[PATCH v2 06/23] ALSA: emux: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 include/sound/emux_legacy.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

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>
-- 
2.48.1

[PATCH v2 07/23] ALSA: firewire: bebob: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/firewire/bebob/bebob_hwdep.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

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
  */
-- 
2.48.1

[PATCH v2 08/23] ALSA: opti9xx: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/isa/opti9xx/opti92x-ad1848.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

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);
-- 
2.48.1

[PATCH v2 09/23] ALSA: hal2: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/mips/hal2.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

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));
-- 
2.48.1

[PATCH v2 10/23] ALSA: ppc: tumbler: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/ppc/tumbler.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

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)
-- 
2.48.1

[PATCH v2 11/23] ALSA: ac97: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/ac97/ac97_codec.c | 2 +-
 sound/pci/ac97/ac97_patch.c | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

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.
 	*/
-- 
2.48.1

[PATCH v2 12/23] ALSA: au88x0: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/au88x0/au88x0.h        | 2 +-
 sound/pci/au88x0/au88x0_mpu401.c | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

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
-- 
2.48.1

[PATCH v2 13/23] ALSA: aw2: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/aw2/aw2-saa7146.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

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.
 	 */
 
-- 
2.48.1

[PATCH v2 14/23] ALSA: cs4281: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/cs4281.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

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 */
-- 
2.48.1

[PATCH v2 15/23] ALSA: cs46xx: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/cs46xx/cs46xx_lib.c       | 6 +++---
 sound/pci/cs46xx/dsp_spos_scb_lib.c | 8 ++++----
 2 files changed, 7 insertions(+), 7 deletions(-)

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..71eee5cb478f 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))
-- 
2.48.1

[PATCH v2 16/23] ALSA: echoaudio: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/echoaudio/echoaudio.c | 4 ++--
 sound/pci/echoaudio/echoaudio.h | 4 ++--
 2 files changed, 4 insertions(+), 4 deletions(-)

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,
-- 
2.48.1

[PATCH v2 17/23] ALSA: lx6464es: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/lx6464es/lx_core.c | 2 +-
 sound/pci/lx6464es/lx_defs.h | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

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)
-- 
2.48.1

[PATCH v2 18/23] ALSA: pcxhr: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/pcxhr/pcxhr_core.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

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 {
-- 
2.48.1

[PATCH v2 19/23] ALSA: riptide: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/riptide/riptide.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

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;
-- 
2.48.1

[PATCH v2 20/23] ALSA: rme32: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/rme32.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

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.
  * 
  * ****************************************************************************
  * 
-- 
2.48.1

[PATCH v2 21/23] ALSA: hdsp: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/rme9652/hdsp.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

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;
 
-- 
2.48.1

[PATCH v2 22/23] ALSA: trident: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/trident/trident_memory.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

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 */
-- 
2.48.1

[PATCH v2 23/23] ALSA: vx: fix typos in comments

[Hemanth Selam]

Correct spelling mistakes in comments. No functional change.

Signed-off-by: Hemanth Selam <hemanth.selam@gmail.com>
---
 sound/pci/vx222/vx222_ops.c | 2 +-
 sound/pcmcia/vx/vxp_ops.c   | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

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!
  */
-- 
2.48.1

Re: [PATCH v2 00/23] ALSA: fix typos in comments

[Takashi Iwai]

On Tue, 15 Sep 2026 10:44:01 +0200,
Hemanth Selam wrote:
> 
> v1 was a single patch touching 35 files, which was too coarse and would
> get in the way of backporting real fixes later on.
> 
> This v2 splits it per driver, so each patch stays confined to one driver
> and can be dropped or backported on its own.  Drivers made up of several
> files (hda, usb-audio, i2c, ac97, ...) are kept together in a single
> patch rather than going all the way down to one patch per file.
> 
> Comments only, no functional change.  The combined diff of this series
> is identical to v1 except for a trailing whitespace dropped from one
> cs46xx comment line that checkpatch complained about.
> 
> Built with allmodconfig on x86, sound/ compiles clean with no new
> warnings.  sound/mips/hal2.c and sound/ppc/tumbler.c are arch-gated so
> they were not compile-tested here; both are comment-only as well.
> 
> v1: https://lore.kernel.org/all/20260907063202.20577-1-hemanth.selam@gmail.com/
> 
> Hemanth Selam (23):
>   ALSA: core: seq: oss: fix typos in comments
>   ALSA: hda: fix typos in comments
>   ALSA: usb-audio: fix typos in comments
>   ALSA: i2c: fix typos in comments
>   ALSA: emu10k1: fix typos in comments
>   ALSA: emux: fix typos in comments
>   ALSA: firewire: bebob: fix typos in comments
>   ALSA: opti9xx: fix typos in comments
>   ALSA: hal2: fix typos in comments
>   ALSA: ppc: tumbler: fix typos in comments
>   ALSA: ac97: fix typos in comments
>   ALSA: au88x0: fix typos in comments
>   ALSA: aw2: fix typos in comments
>   ALSA: cs4281: fix typos in comments
>   ALSA: cs46xx: fix typos in comments
>   ALSA: echoaudio: fix typos in comments
>   ALSA: lx6464es: fix typos in comments
>   ALSA: pcxhr: fix typos in comments
>   ALSA: riptide: fix typos in comments
>   ALSA: rme32: fix typos in comments
>   ALSA: hdsp: fix typos in comments
>   ALSA: trident: fix typos in comments
>   ALSA: vx: fix typos in comments

Applied to for-next branch now.  Thanks.


Takashi