tree-wide: fix assorted typos all over the place
That is "success", "unknown", "through", "performance", "[re|un]mapping"
, "access", "default", "reasonable", "[con]currently", "temperature"
, "channel", "[un]used", "application", "example","hierarchy", "therefore"
, "[over|under]flow", "contiguous", "threshold", "enough" and others.
Signed-off-by: André Goddard Rosa <andre.goddard@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
diff --git a/sound/Kconfig b/sound/Kconfig
index 4b5365a..fcad760 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -55,7 +55,7 @@
Please read Documentation/feature-removal-schedule.txt for
details.
- If unusre, say Y.
+ If unsure, say Y.
source "sound/oss/dmasound/Kconfig"
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index a076a6c..a828baa 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -177,7 +177,7 @@
{ .id = "CSC0437", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
/* Digital PC 5000 Onboard - CS4236B */
{ .id = "CSC0735", .devs = { { "CSC0000" }, { "CSC0010" } } },
- /* some uknown CS4236B */
+ /* some unknown CS4236B */
{ .id = "CSC0b35", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
/* Intel PR440FX Onboard sound */
{ .id = "CSC0b36", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
index 02e30d7..ddad60e 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -137,7 +137,7 @@
static void snd_miro_proc_init(struct snd_miro * miro);
static char * snd_opti9xx_names[] = {
- "unkown",
+ "unknown",
"82C928", "82C929",
"82C924", "82C925",
"82C930", "82C931", "82C933"
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index 5cd5553..8480075 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -185,7 +185,7 @@
#endif
static char * snd_opti9xx_names[] = {
- "unkown",
+ "unknown",
"82C928", "82C929",
"82C924", "82C925",
"82C930", "82C931", "82C933"
diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c
index 06e9e88..bb14e4c 100644
--- a/sound/oss/dmasound/dmasound_paula.c
+++ b/sound/oss/dmasound/dmasound_paula.c
@@ -657,7 +657,7 @@
len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n",
dmasound.volume_right);
if (len >= space) {
- printk(KERN_ERR "dmasound_paula: overlowed state buffer alloc.\n") ;
+ printk(KERN_ERR "dmasound_paula: overflowed state buffer alloc.\n") ;
len = space ;
}
return len;
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index c62b7d1..8d13092 100644
--- a/sound/pci/ca0106/ca0106_proc.c
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -233,7 +233,7 @@
snd_iprintf(buffer, "user-defined\n");
break;
default:
- snd_iprintf(buffer, "unkown\n");
+ snd_iprintf(buffer, "unknown\n");
break;
}
snd_iprintf(buffer, "Sample Bits: ");
diff --git a/sound/pci/cs46xx/imgs/cwcdma.asp b/sound/pci/cs46xx/imgs/cwcdma.asp
index 09d24c7..a65e119 100644
--- a/sound/pci/cs46xx/imgs/cwcdma.asp
+++ b/sound/pci/cs46xx/imgs/cwcdma.asp
@@ -26,10 +26,11 @@
//
//
// The purpose of this code is very simple: make it possible to tranfser
-// the samples 'as they are' with no alteration from a PCMreader SCB (DMA from host)
-// to any other SCB. This is useful for AC3 throug SPDIF. SRC (source rate converters)
-// task always alters the samples in some how, however it's from 48khz -> 48khz. The
-// alterations are not audible, but AC3 wont work.
+// the samples 'as they are' with no alteration from a PCMreader
+// SCB (DMA from host) to any other SCB. This is useful for AC3 through SPDIF.
+// SRC (source rate converters) task always alters the samples in somehow,
+// however it's from 48khz -> 48khz.
+// The alterations are not audible, but AC3 wont work.
//
// ...
// |
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 36e08bd..360e380 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -184,7 +184,7 @@
* The hardware has 3 channels for playback and 1 for capture.
* - channel 0 is the front channel
* - channel 1 is the rear channel
- * - channel 2 is the center/lfe chanel
+ * - channel 2 is the center/lfe channel
* Volume is controlled by the AC97 for the front and rear channels by
* the PCM Playback Volume, Sigmatel Surround Playback Volume and
* Surround Playback Volume. The Sigmatel 4-Speaker Stereo switch affects
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 780e1a7..8917071 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -66,7 +66,7 @@
struct hda_pcm pcm_rec[2]; /* PCM information */
- /* pin deafault configuration */
+ /* pin default configuration */
hda_nid_t pin_nid[NUM_PINS];
unsigned int def_conf[NUM_PINS];
unsigned int pin_def_confs;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ff20048..872731e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6619,7 +6619,7 @@
/* Front Mic (0x01) unused */
{ "Line", 0x2 },
/* Line 2 (0x03) unused */
- /* CD (0x04) unsused? */
+ /* CD (0x04) unused? */
},
};
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 0dce331..a1b10d1 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -3017,7 +3017,7 @@
insel = "Coaxial";
break;
default:
- insel = "Unkown";
+ insel = "Unknown";
}
switch (hdspm->control_register & HDSPM_SyncRefMask) {
@@ -3028,7 +3028,7 @@
syncref = "MADI";
break;
default:
- syncref = "Unkown";
+ syncref = "Unknown";
}
snd_iprintf(buffer, "Inputsel = %s, SyncRef = %s\n", insel,
syncref);
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index c33b92e..8ce1c9b 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -101,7 +101,7 @@
pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
if (reg >= UDA134X_REGS_NUM) {
- printk(KERN_ERR "%s unkown register: reg: %u",
+ printk(KERN_ERR "%s unknown register: reg: %u",
__func__, reg);
return -EINVAL;
}
@@ -552,7 +552,7 @@
ARRAY_SIZE(uda1341_snd_controls));
break;
default:
- printk(KERN_ERR "%s unkown codec type: %d",
+ printk(KERN_ERR "%s unknown codec type: %d",
__func__, pd->model);
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index fe1307b..d72347d 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -607,7 +607,7 @@
SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0),
SOC_ENUM("DRC Compressor Slope R0", drc_slope_r0),
SOC_ENUM("DRC Compressor Slope R1", drc_slope_r1),
-SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8903_DRC_3, 5, 124, 1,
+SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8903_DRC_3, 5, 124, 1,
drc_tlv_thresh),
SOC_SINGLE_TLV("DRC Volume", WM8903_DRC_3, 0, 30, 1, drc_tlv_amp),
SOC_SINGLE_TLV("DRC Minimum Gain Volume", WM8903_DRC_1, 2, 3, 1, drc_tlv_min),
@@ -617,11 +617,11 @@
SOC_ENUM("DRC FF Delay", drc_ff_delay),
SOC_SINGLE("DRC Anticlip Switch", WM8903_DRC_0, 1, 1, 0),
SOC_SINGLE("DRC QR Switch", WM8903_DRC_0, 2, 1, 0),
-SOC_SINGLE_TLV("DRC QR Threashold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max),
+SOC_SINGLE_TLV("DRC QR Threshold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max),
SOC_ENUM("DRC QR Decay Rate", drc_qr_decay),
SOC_SINGLE("DRC Smoothing Switch", WM8903_DRC_0, 3, 1, 0),
SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8903_DRC_0, 0, 1, 0),
-SOC_ENUM("DRC Smoothing Threashold", drc_smoothing),
+SOC_ENUM("DRC Smoothing Threshold", drc_smoothing),
SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup),
SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT,
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index d998799..bc03368 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -689,7 +689,7 @@
SOC_SINGLE("DRC Switch", WM8993_DRC_CONTROL_1, 15, 1, 0),
SOC_ENUM("DRC Path", drc_path),
-SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8993_DRC_CONTROL_2,
+SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8993_DRC_CONTROL_2,
2, 60, 1, drc_comp_threash),
SOC_SINGLE_TLV("DRC Compressor Amplitude Volume", WM8993_DRC_CONTROL_3,
11, 30, 1, drc_comp_amp),
@@ -709,7 +709,7 @@
SOC_ENUM("DRC Quick Release Rate", drc_qr_rate),
SOC_SINGLE("DRC Smoothing Switch", WM8993_DRC_CONTROL_1, 11, 1, 0),
SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8993_DRC_CONTROL_1, 8, 1, 0),
-SOC_ENUM("DRC Smoothing Hysteresis Threashold", drc_smooth),
+SOC_ENUM("DRC Smoothing Hysteresis Threshold", drc_smooth),
SOC_SINGLE_TLV("DRC Startup Volume", WM8993_DRC_CONTROL_4, 8, 18, 0,
drc_startup_tlv),
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c
index 1966e0d..3c7ccb7 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec.c
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.c
@@ -270,7 +270,7 @@
gpio_direction_output(pd->amp_gain[1], 0);
}
- /* note, curently we assume GPA0 isn't valid amp */
+ /* note, currently we assume GPA0 isn't valid amp */
if (pdata->amp_gpio > 0) {
ret = gpio_request(pd->amp_gpio, "gpio-amp");
if (ret) {
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 83b8028..81d6f98 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -196,7 +196,7 @@
0 /* destination skip after chunk (impossible) */,
4 /* 16 byte burst size */,
-1 /* don't conserve bandwidth */,
- 0 /* low watermark irq descriptor theshold */,
+ 0 /* low watermark irq descriptor threshold */,
0 /* disable hardware timestamps */,
1 /* enable channel */);
diff --git a/sound/sound_core.c b/sound/sound_core.c
index 49c9981..dbca7c9 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -353,7 +353,7 @@
* @dev: device pointer
*
* Allocate a special sound device by minor number from the sound
- * subsystem. The allocated number is returned on succes. On failure
+ * subsystem. The allocated number is returned on success. On failure
* a negative error code is returned.
*/