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-rw-r--r--sound/oss/ac97_codec.c6
-rw-r--r--sound/oss/audio.c4
-rw-r--r--sound/oss/dmasound/dmasound_core.c2
-rw-r--r--sound/oss/midibuf.c2
-rw-r--r--sound/oss/sb_card.c2
-rw-r--r--sound/oss/sb_ess.c2
-rw-r--r--sound/oss/swarm_cs4297a.c2
-rw-r--r--sound/oss/vidc.c2
8 files changed, 11 insertions, 11 deletions
diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c
index 854c303..0cd23d9 100644
--- a/sound/oss/ac97_codec.c
+++ b/sound/oss/ac97_codec.c
@@ -28,7 +28,7 @@
*
* History
* May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
- * Removed non existant WM9700
+ * Removed non existent WM9700
* Added support for WM9705, WM9708, WM9709, WM9710, WM9711
* WM9712 and WM9717
* Mar 28, 2002 Randolph Bentson <bentson@holmsjoen.com>
@@ -441,7 +441,7 @@ static void ac97_set_mixer(struct ac97_codec *codec, unsigned int oss_mixer, uns
}
/* read or write the recmask, the ac97 can really have left and right recording
- inputs independantly set, but OSS doesn't seem to want us to express that to
+ inputs independently set, but OSS doesn't seem to want us to express that to
the user. the caller guarantees that we have a supported bit set, and they
must be holding the card's spinlock */
static int ac97_recmask_io(struct ac97_codec *codec, int rw, int mask)
@@ -754,7 +754,7 @@ int ac97_probe_codec(struct ac97_codec *codec)
if((codec->codec_ops == &null_ops) && (f & 4))
codec->codec_ops = &default_digital_ops;
- /* A device which thinks its a modem but isnt */
+ /* A device which thinks its a modem but isn't */
if(codec->flags & AC97_DELUDED_MODEM)
codec->modem = 0;
diff --git a/sound/oss/audio.c b/sound/oss/audio.c
index 7df48a2..4b958b1 100644
--- a/sound/oss/audio.c
+++ b/sound/oss/audio.c
@@ -514,7 +514,7 @@ int audio_ioctl(int dev, struct file *file, unsigned int cmd, void __user *arg)
count += dmap->bytes_in_use; /* Pointer wrap not handled yet */
count += dmap->byte_counter;
- /* Substract current count from the number of bytes written by app */
+ /* Subtract current count from the number of bytes written by app */
count = dmap->user_counter - count;
if (count < 0)
count = 0;
@@ -931,7 +931,7 @@ static int dma_ioctl(int dev, unsigned int cmd, void __user *arg)
if (count < dmap_out->fragment_size && dmap_out->qhead != 0)
count += dmap_out->bytes_in_use; /* Pointer wrap not handled yet */
count += dmap_out->byte_counter;
- /* Substract current count from the number of bytes written by app */
+ /* Subtract current count from the number of bytes written by app */
count = dmap_out->user_counter - count;
if (count < 0)
count = 0;
diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c
index 87e2c72..c918313 100644
--- a/sound/oss/dmasound/dmasound_core.c
+++ b/sound/oss/dmasound/dmasound_core.c
@@ -1021,7 +1021,7 @@ static int sq_ioctl(struct file *file, u_int cmd, u_long arg)
case SNDCTL_DSP_SYNC:
/* This call, effectively, has the same behaviour as SNDCTL_DSP_RESET
except that it waits for output to finish before resetting
- everything - read, however, is killed imediately.
+ everything - read, however, is killed immediately.
*/
result = 0 ;
if (file->f_mode & FMODE_WRITE) {
diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c
index ceedb1e..8cdb2cf 100644
--- a/sound/oss/midibuf.c
+++ b/sound/oss/midibuf.c
@@ -295,7 +295,7 @@ int MIDIbuf_write(int dev, struct file *file, const char __user *buf, int count)
for (i = 0; i < n; i++)
{
- /* BROKE BROKE BROKE - CANT DO THIS WITH CLI !! */
+ /* BROKE BROKE BROKE - CAN'T DO THIS WITH CLI !! */
/* yes, think the same, so I removed the cli() brackets
QUEUE_BYTE is protected against interrupts */
if (copy_from_user((char *) &tmp_data, &(buf)[c], 1)) {
diff --git a/sound/oss/sb_card.c b/sound/oss/sb_card.c
index 84ef4d0..fb5d725 100644
--- a/sound/oss/sb_card.c
+++ b/sound/oss/sb_card.c
@@ -1,7 +1,7 @@
/*
* sound/oss/sb_card.c
*
- * Detection routine for the ISA Sound Blaster and compatable sound
+ * Detection routine for the ISA Sound Blaster and compatible sound
* cards.
*
* This file is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c
index 9890cf2..5c773df 100644
--- a/sound/oss/sb_ess.c
+++ b/sound/oss/sb_ess.c
@@ -168,7 +168,7 @@
* corresponding playback levels, unless recmask says they aren't recorded. In
* the latter case the recording volumes are 0.
* Now recording levels of inputs can be controlled, by changing the playback
- * levels. Futhermore several devices can be recorded together (which is not
+ * levels. Furthermore several devices can be recorded together (which is not
* possible with the ES1688).
* Besides the separate recording level control for each input, the common
* recording level can also be controlled by RECLEV as described above.
diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c
index 44357d8..09d4648 100644
--- a/sound/oss/swarm_cs4297a.c
+++ b/sound/oss/swarm_cs4297a.c
@@ -875,7 +875,7 @@ static void start_adc(struct cs4297a_state *s)
if (s->prop_adc.fmt & AFMT_S8 || s->prop_adc.fmt & AFMT_U8) {
//
// now only use 16 bit capture, due to truncation issue
- // in the chip, noticable distortion occurs.
+ // in the chip, noticeable distortion occurs.
// allocate buffer and then convert from 16 bit to
// 8 bit for the user buffer.
//
diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c
index f0e0caa..12ba28e 100644
--- a/sound/oss/vidc.c
+++ b/sound/oss/vidc.c
@@ -227,7 +227,7 @@ static int vidc_audio_set_speed(int dev, int rate)
} else {
/*printk("VIDC: internal %d %d %d\n", rate, rate_int, hwrate);*/
hwctrl=0x00000003;
- /* Allow rougly 0.4% tolerance */
+ /* Allow roughly 0.4% tolerance */
if (diff_int > (rate/256))
rate=rate_int;
}
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