Fix common misspellings

Fixes generated by 'codespell' and manually reviewed.

Signed-off-by: Lucas De Marchi <lucas.demarchi@profusion.mobi>

35 files changed, 56 insertions, 56 deletions

diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index 4382d0f..d8f6fd6 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -29,7 +29,7 @@
  *	PM support
  *	MIDI support
  *	Game Port support
- *	SG DMA support (this will need *alot* of work)
+ *	SG DMA support (this will need *a lot* of work)
  */
 
 #include <linux/init.h>
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index f53a31e..f8ccc96 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -963,7 +963,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream)
 
 	/*? also check ASI5000 samplerate source
 	    If external, only support external rate.
-	    If internal and other stream playing, cant switch
+	    If internal and other stream playing, can't switch
 	*/
 
 	init_timer(&dpcm->timer);
diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h
index 6fc025c..255429c 100644
--- a/sound/pci/asihpi/hpi.h
+++ b/sound/pci/asihpi/hpi.h
@@ -725,7 +725,7 @@ enum HPI_AESEBU_ERRORS {
 #define HPI_PAD_TITLE_LEN               64
 /** The text string containing the comment. */
 #define HPI_PAD_COMMENT_LEN             256
-/** The PTY when the tuner has not recieved any PTY. */
+/** The PTY when the tuner has not received any PTY. */
 #define HPI_PAD_PROGRAM_TYPE_INVALID    0xffff
 /** \} */
 
diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c
index 3e3c2ef..8c8aac4 100644
--- a/sound/pci/asihpi/hpi6000.c
+++ b/sound/pci/asihpi/hpi6000.c
@@ -423,7 +423,7 @@ static void subsys_create_adapter(struct hpi_message *phm,
 
 	ao.priv = kzalloc(sizeof(struct hpi_hw_obj), GFP_KERNEL);
 	if (!ao.priv) {
-		HPI_DEBUG_LOG(ERROR, "cant get mem for adapter object\n");
+		HPI_DEBUG_LOG(ERROR, "can't get mem for adapter object\n");
 		phr->error = HPI_ERROR_MEMORY_ALLOC;
 		return;
 	}
diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index 620525b..22e9f08 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -466,7 +466,7 @@ static void subsys_create_adapter(struct hpi_message *phm,
 
 	ao.priv = kzalloc(sizeof(struct hpi_hw_obj), GFP_KERNEL);
 	if (!ao.priv) {
-		HPI_DEBUG_LOG(ERROR, "cant get mem for adapter object\n");
+		HPI_DEBUG_LOG(ERROR, "can't get mem for adapter object\n");
 		phr->error = HPI_ERROR_MEMORY_ALLOC;
 		return;
 	}
diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index af678be..3b9fd11 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -607,7 +607,7 @@ struct hpi_data_compat32 {
 #endif
 
 struct hpi_buffer {
-  /** placehoder for backward compatability (see dwBufferSize) */
+  /** placehoder for backward compatibility (see dwBufferSize) */
 	struct hpi_msg_format reserved;
 	u32 command; /**< HPI_BUFFER_CMD_xxx*/
 	u32 pci_address; /**< PCI physical address of buffer for DSP DMA */
diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c
index bcbdf30..360028b 100644
--- a/sound/pci/asihpi/hpimsgx.c
+++ b/sound/pci/asihpi/hpimsgx.c
@@ -722,7 +722,7 @@ static u16 HPIMSGX__init(struct hpi_message *phm,
 		return phr->error;
 	}
 	if (hr.error == 0) {
-		/* the adapter was created succesfully
+		/* the adapter was created successfully
 		   save the mapping for future use */
 		hpi_entry_points[hr.u.s.adapter_index] = entry_point_func;
 		/* prepare adapter (pre-open streams etc.) */
diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h
index ecb8f4d..02f6e08 100644
--- a/sound/pci/au88x0/au88x0.h
+++ b/sound/pci/au88x0/au88x0.h
@@ -104,7 +104,7 @@
 #define MIX_PLAYB(x) (vortex->mixplayb[x])
 #define MIX_SPDIF(x) (vortex->mixspdif[x])
 
-#define NR_WTPB 0x20		/* WT channels per eahc bank. */
+#define NR_WTPB 0x20		/* WT channels per each bank. */
 
 /* Structs */
 typedef struct {
diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c
index f4aa8ff..9ae8b3b 100644
--- a/sound/pci/au88x0/au88x0_a3d.c
+++ b/sound/pci/au88x0/au88x0_a3d.c
@@ -53,7 +53,7 @@ a3dsrc_GetTimeConsts(a3dsrc_t * a, short *HrtfTrack, short *ItdTrack,
 }
 
 #endif
-/* Atmospheric absorbtion. */
+/* Atmospheric absorption. */
 
 static void
 a3dsrc_SetAtmosTarget(a3dsrc_t * a, short aa, short b, short c, short d,
@@ -835,7 +835,7 @@ snd_vortex_a3d_filter_put(struct snd_kcontrol *kcontrol,
 		params[i] = ucontrol->value.integer.value[i];
 	/* Translate generic filter params to a3d filter params. */
 	vortex_a3d_translate_filter(a->filter, params);
-	/* Atmospheric absorbtion and filtering. */
+	/* Atmospheric absorption and filtering. */
 	a3dsrc_SetAtmosTarget(a, a->filter[0],
 			      a->filter[1], a->filter[2],
 			      a->filter[3], a->filter[4]);
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index 5439d66..33f0ba5 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -515,7 +515,7 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr)
 		return -ENODEV;
 
 	/* idx indicates which kind of PCM device. ADB, SPDIF, I2S and A3D share the 
-	 * same dma engine. WT uses it own separate dma engine whcih cant capture. */
+	 * same dma engine. WT uses it own separate dma engine which can't capture. */
 	if (idx == VORTEX_PCM_ADB)
 		nr_capt = nr;
 	else
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 5715c4d0..9b7a634 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -140,7 +140,7 @@
  *  Possible remedies:
  *  - use speaker (amplifier) output instead of headphone output
  *    (in case crackling is due to overloaded output clipping)
- *  - plug card into a different PCI slot, preferrably one that isn't shared
+ *  - plug card into a different PCI slot, preferably one that isn't shared
  *    too much (this helps a lot, but not completely!)
  *  - get rid of PCI VGA card, use AGP instead
  *  - upgrade or downgrade BIOS
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h
index fc53b9b..e8e8ccc 100644
--- a/sound/pci/ca0106/ca0106.h
+++ b/sound/pci/ca0106/ca0106.h
@@ -51,7 +51,7 @@
  *    Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
  *
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -175,7 +175,7 @@
 /* CA0106 pointer-offset register set, accessed through the PTR and DATA registers                     */
 /********************************************************************************************************/
                                                                                                                            
-/* Initally all registers from 0x00 to 0x3f have zero contents. */
+/* Initially all registers from 0x00 to 0x3f have zero contents. */
 #define PLAYBACK_LIST_ADDR	0x00		/* Base DMA address of a list of pointers to each period/size */
 						/* One list entry: 4 bytes for DMA address, 
 						 * 4 bytes for period_size << 16.
@@ -223,7 +223,7 @@
  * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3
  * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground
  * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground.
- * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Sheild on all three, 4 -> Red.
+ * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Shield on all three, 4 -> Red.
  * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card.
  */
 /* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 01b4938..4377592 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -117,7 +117,7 @@
  *    DAC: Unknown
  *    Trying to handle it like the SB0410.
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 630aa49..84f3f92 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -42,7 +42,7 @@
  *  0.0.18
  *    Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index ba96428..c694464 100644
--- a/sound/pci/ca0106/ca0106_proc.c
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -42,7 +42,7 @@
  *  0.0.18
  *    Implement support for Line-in capture on SB Live 24bit.
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index b5bb036..f4e5735 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -73,7 +73,7 @@ MODULE_PARM_DESC(mpu_port, "MPU-401 port.");
 module_param_array(fm_port, long, NULL, 0444);
 MODULE_PARM_DESC(fm_port, "FM port.");
 module_param_array(soft_ac3, bool, NULL, 0444);
-MODULE_PARM_DESC(soft_ac3, "Sofware-conversion of raw SPDIF packets (model 033 only).");
+MODULE_PARM_DESC(soft_ac3, "Software-conversion of raw SPDIF packets (model 033 only).");
 #ifdef SUPPORT_JOYSTICK
 module_param_array(joystick_port, int, NULL, 0444);
 MODULE_PARM_DESC(joystick_port, "Joystick port address.");
@@ -656,8 +656,8 @@ out:
 }
 
 /*
- * Program pll register bits, I assume that the 8 registers 0xf8 upto 0xff
- * are mapped onto the 8 ADC/DAC sampling frequency which can be choosen
+ * Program pll register bits, I assume that the 8 registers 0xf8 up to 0xff
+ * are mapped onto the 8 ADC/DAC sampling frequency which can be chosen
  * at the register CM_REG_FUNCTRL1 (0x04).
  * Problem: other ways are also possible (any information about that?)
  */
@@ -666,7 +666,7 @@ static void snd_cmipci_set_pll(struct cmipci *cm, unsigned int rate, unsigned in
 	unsigned int reg = CM_REG_PLL + slot;
 	/*
 	 * Guess that this programs at reg. 0x04 the pos 15:13/12:10
-	 * for DSFC/ASFC (000 upto 111).
+	 * for DSFC/ASFC (000 up to 111).
 	 */
 
 	/* FIXME: Init (Do we've to set an other register first before programming?) */
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index b932154..13f33c0 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1627,7 +1627,7 @@ static struct ct_atc atc_preset __devinitdata = {
  *  Creates and initializes a hardware manager.
  *
  *  Creates kmallocated ct_atc structure. Initializes hardware.
- *  Returns 0 if suceeds, or negative error code if fails.
+ *  Returns 0 if succeeds, or negative error code if fails.
  */
 
 int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c
index 0cf400f..a5c957d 100644
--- a/sound/pci/ctxfi/cthw20k1.c
+++ b/sound/pci/ctxfi/cthw20k1.c
@@ -1285,7 +1285,7 @@ static int hw_trn_init(struct hw *hw, const struct trn_conf *info)
 	hw_write_20kx(hw, PTPALX, ptp_phys_low);
 	hw_write_20kx(hw, PTPAHX, ptp_phys_high);
 	hw_write_20kx(hw, TRNCTL, trnctl);
-	hw_write_20kx(hw, TRNIS, 0x200c01); /* realy needed? */
+	hw_write_20kx(hw, TRNIS, 0x200c01); /* really needed? */
 
 	return 0;
 }
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index 957a311..c250614 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -248,7 +248,7 @@ static int is_valid_page(struct snd_emu10k1 *emu, dma_addr_t addr)
 /*
  * map the given memory block on PTB.
  * if the block is already mapped, update the link order.
- * if no empty pages are found, tries to release unsed memory blocks
+ * if no empty pages are found, tries to release unused memory blocks
  * and retry the mapping.
  */
 int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk)
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index 61b8ab3..a81dc44 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -69,7 +69,7 @@
  *    ADC: Philips 1361T (Stereo 24bit)
  *    DAC: CS4382-K (8-channel, 24bit, 192Khz)
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h
index 00f4817..4e0ee1a 100644
--- a/sound/pci/emu10k1/p16v.h
+++ b/sound/pci/emu10k1/p16v.h
@@ -59,7 +59,7 @@
  *    ADC: Philips 1361T (Stereo 24bit)
  *    DAC: CS4382-K (8-channel, 24bit, 192Khz)
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -86,7 +86,7 @@
  * The sample rate is also controlled by the same registers that control the rate of the EMU10K2 sample rate converters.
  */
 
-/* Initally all registers from 0x00 to 0x3f have zero contents. */
+/* Initially all registers from 0x00 to 0x3f have zero contents. */
 #define PLAYBACK_LIST_ADDR	0x00		/* Base DMA address of a list of pointers to each period/size */
 						/* One list entry: 4 bytes for DMA address, 
 						 * 4 bytes for period_size << 16.
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 2c79e96..430f41d 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3661,7 +3661,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec)
  * with the proper parameters for set up.
  * ops.cleanup should be called in hw_free for clean up of streams.
  *
- * This function returns 0 if successfull, or a negative error code.
+ * This function returns 0 if successful, or a negative error code.
  */
 int __devinit snd_hda_build_pcms(struct hda_bus *bus)
 {
@@ -4851,7 +4851,7 @@ EXPORT_SYMBOL_HDA(snd_hda_suspend);
  *
  * Returns 0 if successful.
  *
- * This fucntion is defined only when POWER_SAVE isn't set.
+ * This function is defined only when POWER_SAVE isn't set.
  * In the power-save mode, the codec is resumed dynamically.
  */
 int snd_hda_resume(struct hda_bus *bus)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0ef0035..1e5a786 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -549,7 +549,7 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
 
 /*
  * Control the mode of pin widget settings via the mixer.  "pc" is used
- * instead of "%" to avoid consequences of accidently treating the % as
+ * instead of "%" to avoid consequences of accidentally treating the % as
  * being part of a format specifier.  Maximum allowed length of a value is
  * 63 characters plus NULL terminator.
  *
@@ -9836,7 +9836,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 
 	SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
 
-	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavilion", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 05fcd60..1395991 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2475,7 +2475,7 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
  
 	spec->hp_switch = ucontrol->value.integer.value[0] ? nid : 0;
 
-	/* check to be sure that the ports are upto date with
+	/* check to be sure that the ports are up to date with
 	 * switch changes
 	 */
 	stac_issue_unsol_event(codec, nid);
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 2f62522..3e4f8c1 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -148,7 +148,7 @@ static void aureon_pca9554_write(struct snd_ice1712 *ice, unsigned char reg,
 	udelay(100);
 	/*
 	 * send device address, command and value,
-	 * skipping ack cycles inbetween
+	 * skipping ack cycles in between
 	 */
 	for (j = 0; j < 3; j++) {
 		switch (j) {
@@ -2143,7 +2143,7 @@ static int __devinit aureon_init(struct snd_ice1712 *ice)
 		ice->num_total_adcs = 2;
 	}
 
-	/* to remeber the register values of CS8415 */
+	/* to remember the register values of CS8415 */
 	ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
 	if (!ice->akm)
 		return -ENOMEM;
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 4fc6d8b..f4594d7 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2755,7 +2755,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
 			return err;
 		}
 		if (c->mpu401_1_name)
-			/*  Prefered name available in card_info */
+			/*  Preferred name available in card_info */
 			snprintf(ice->rmidi[0]->name,
 				 sizeof(ice->rmidi[0]->name),
 				 "%s %d", c->mpu401_1_name, card->number);
@@ -2772,7 +2772,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
 				return err;
 			}
 			if (c->mpu401_2_name)
-				/*  Prefered name available in card_info */
+				/*  Preferred name available in card_info */
 				snprintf(ice->rmidi[1]->name,
 					 sizeof(ice->rmidi[1]->name),
 					 "%s %d", c->mpu401_2_name,
diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c
index cdb873f..92c1160 100644
--- a/sound/pci/ice1712/pontis.c
+++ b/sound/pci/ice1712/pontis.c
@@ -768,7 +768,7 @@ static int __devinit pontis_init(struct snd_ice1712 *ice)
 	ice->num_total_dacs = 2;
 	ice->num_total_adcs = 2;
 
-	/* to remeber the register values */
+	/* to remember the register values */
 	ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
 	if (! ice->akm)
 		return -ENOMEM;
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 6a9fee3..764cc93 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -1046,7 +1046,7 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice)
 	* don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten
 	*/
 	ice->gpio.saved[0] = 0;
-	/* to remeber the register values */
+	/* to remember the register values */
 
 	ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
 	if (! ice->akm)
@@ -1128,7 +1128,7 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice)
 	* don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten
 	*/
 	ice->gpio.saved[0] = 0;
-	/* to remeber the register values */
+	/* to remember the register values */
 
 	ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
 	if (! ice->akm)
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 629a549..6c896db 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -534,7 +534,7 @@ static int snd_intel8x0_codec_semaphore(struct intel8x0 *chip, unsigned int code
 		udelay(10);
 	} while (time--);
 
-	/* access to some forbidden (non existant) ac97 registers will not
+	/* access to some forbidden (non existent) ac97 registers will not
 	 * reset the semaphore. So even if you don't get the semaphore, still
 	 * continue the access. We don't need the semaphore anyway. */
 	snd_printk(KERN_ERR "codec_semaphore: semaphore is not ready [0x%x][0x%x]\n",
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 2ae8d29..27709f0 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -331,7 +331,7 @@ static int snd_intel8x0m_codec_semaphore(struct intel8x0m *chip, unsigned int co
 		udelay(10);
 	} while (time--);
 
-	/* access to some forbidden (non existant) ac97 registers will not
+	/* access to some forbidden (non existent) ac97 registers will not
 	 * reset the semaphore. So even if you don't get the semaphore, still
 	 * continue the access. We don't need the semaphore anyway. */
 	snd_printk(KERN_ERR "codec_semaphore: semaphore is not ready [0x%x][0x%x]\n",
diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c
index d3350f3..3df0f53 100644
--- a/sound/pci/mixart/mixart_core.c
+++ b/sound/pci/mixart/mixart_core.c
@@ -265,7 +265,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int
 	if (! timeout) {
 		/* error - no ack */
 		mutex_unlock(&mgr->msg_mutex);
-		snd_printk(KERN_ERR "error: no reponse on msg %x\n", msg_frame);
+		snd_printk(KERN_ERR "error: no response on msg %x\n", msg_frame);
 		return -EIO;
 	}
 
@@ -278,7 +278,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int
 	err = get_msg(mgr, &resp, msg_frame);
 
 	if( request->message_id != resp.message_id )
-		snd_printk(KERN_ERR "REPONSE ERROR!\n");
+		snd_printk(KERN_ERR "RESPONSE ERROR!\n");
 
 	mutex_unlock(&mgr->msg_mutex);
 	return err;
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index 833e718..304411c 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -1042,11 +1042,11 @@ void pcxhr_msg_tasklet(unsigned long arg)
 	int i, j;
 
 	if (mgr->src_it_dsp & PCXHR_IRQ_FREQ_CHANGE)
-		snd_printdd("TASKLET : PCXHR_IRQ_FREQ_CHANGE event occured\n");
+		snd_printdd("TASKLET : PCXHR_IRQ_FREQ_CHANGE event occurred\n");
 	if (mgr->src_it_dsp & PCXHR_IRQ_TIME_CODE)
-		snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occured\n");
+		snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occurred\n");
 	if (mgr->src_it_dsp & PCXHR_IRQ_NOTIFY)
-		snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occured\n");
+		snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occurred\n");
 	if (mgr->src_it_dsp & (PCXHR_IRQ_FREQ_CHANGE | PCXHR_IRQ_TIME_CODE)) {
 		/* clear events FREQ_CHANGE and TIME_CODE */
 		pcxhr_init_rmh(prmh, CMD_TEST_IT);
@@ -1055,7 +1055,7 @@ void pcxhr_msg_tasklet(unsigned long arg)
 			    err, prmh->stat[0]);
 	}
 	if (mgr->src_it_dsp & PCXHR_IRQ_ASYNC) {
-		snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occured\n");
+		snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occurred\n");
 
 		pcxhr_init_rmh(prmh, CMD_ASYNC);
 		prmh->cmd[0] |= 1;	/* add SEL_ASYNC_EVENTS */
@@ -1233,7 +1233,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 	reg = PCXHR_INPL(mgr, PCXHR_PLX_L2PCIDB);
 	PCXHR_OUTPL(mgr, PCXHR_PLX_L2PCIDB, reg);
 
-	/* timer irq occured */
+	/* timer irq occurred */
 	if (reg & PCXHR_IRQ_TIMER) {
 		int timer_toggle = reg & PCXHR_IRQ_TIMER;
 		/* is a 24 bit counter */
@@ -1288,7 +1288,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 	if (reg & PCXHR_IRQ_MASK) {
 		if (reg & PCXHR_IRQ_ASYNC) {
 			/* as we didn't request any async notifications,
-			 * some kind of xrun error will probably occured
+			 * some kind of xrun error will probably occurred
 			 */
 			/* better resynchronize all streams next interrupt : */
 			mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID;
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index d5f5b44..9ff247f 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -150,7 +150,7 @@ MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard.");
 #define RME96_RCR_BITPOS_F1 28
 #define RME96_RCR_BITPOS_F2 29
 
-/* Additonal register bits */
+/* Additional register bits */
 #define RME96_AR_WSEL       (1 << 0)
 #define RME96_AR_ANALOG     (1 << 1)
 #define RME96_AR_FREQPAD_0  (1 << 2)
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index a323eaf..949691a 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -391,7 +391,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 
 /* Status2 Register bits */ /* MADI ONLY */
 
-#define HDSPM_version0 (1<<0)	/* not realy defined but I guess */
+#define HDSPM_version0 (1<<0)	/* not really defined but I guess */
 #define HDSPM_version1 (1<<1)	/* in former cards it was ??? */
 #define HDSPM_version2 (1<<2)
 
@@ -936,7 +936,7 @@ struct hdspm {
 	struct snd_kcontrol *playback_mixer_ctls[HDSPM_MAX_CHANNELS];
 	/* but input to much, so not used */
 	struct snd_kcontrol *input_mixer_ctls[HDSPM_MAX_CHANNELS];
-	/* full mixer accessable over mixer ioctl or hwdep-device */
+	/* full mixer accessible over mixer ioctl or hwdep-device */
 	struct hdspm_mixer *mixer;
 
 	struct hdspm_tco *tco;  /* NULL if no TCO detected */
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 1b8f674..2b5c7a95 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -308,7 +308,7 @@ static irqreturn_t sis_interrupt(int irq, void *dev)
 	u32 intr, status;
 
 	/* We only use the DMA interrupts, and we don't enable any other
-	 * source of interrupts. But, it is possible to see an interupt
+	 * source of interrupts. But, it is possible to see an interrupt
 	 * status that didn't actually interrupt us, so eliminate anything
 	 * we're not expecting to avoid falsely claiming an IRQ, and an
 	 * ensuing endless loop.
@@ -773,7 +773,7 @@ static void sis_prepare_timing_voice(struct voice *voice,
 		vperiod = 0;
 	}
 
-	/* The interrupt handler implements the timing syncronization, so
+	/* The interrupt handler implements the timing synchronization, so
 	 * setup its state.
 	 */
 	timing->flags |= VOICE_SYNC_TIMING;
@@ -1139,7 +1139,7 @@ static int sis_chip_init(struct sis7019 *sis)
 	 */
 	outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR);
 
-	/* Reset the syncronization groups for all of the channels
+	/* Reset the synchronization groups for all of the channels
 	 * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc.
 	 * we'll need to change how we handle these. Until then, we just
 	 * assign sub-mixer 0 to all playback channels, and avoid any