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author | Linus Torvalds <torvalds@linux-foundation.org> | 2011-04-07 11:14:49 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2011-04-07 11:14:49 -0700 |
commit | 42933bac11e811f02200c944d8562a15f8ec4ff0 (patch) | |
tree | fcdd9afe56eb0e746565ddd1f92f22d36678b843 /sound/oss | |
parent | 2b9accbee563f535046ff2cd382d0acaa92e130c (diff) | |
parent | 25985edcedea6396277003854657b5f3cb31a628 (diff) | |
download | kernel_samsung_espresso10-42933bac11e811f02200c944d8562a15f8ec4ff0.zip kernel_samsung_espresso10-42933bac11e811f02200c944d8562a15f8ec4ff0.tar.gz kernel_samsung_espresso10-42933bac11e811f02200c944d8562a15f8ec4ff0.tar.bz2 |
Merge branch 'for-linus2' of git://git.profusion.mobi/users/lucas/linux-2.6
* 'for-linus2' of git://git.profusion.mobi/users/lucas/linux-2.6:
Fix common misspellings
Diffstat (limited to 'sound/oss')
-rw-r--r-- | sound/oss/ac97_codec.c | 6 | ||||
-rw-r--r-- | sound/oss/audio.c | 4 | ||||
-rw-r--r-- | sound/oss/dmasound/dmasound_core.c | 2 | ||||
-rw-r--r-- | sound/oss/midibuf.c | 2 | ||||
-rw-r--r-- | sound/oss/sb_card.c | 2 | ||||
-rw-r--r-- | sound/oss/sb_ess.c | 2 | ||||
-rw-r--r-- | sound/oss/swarm_cs4297a.c | 2 | ||||
-rw-r--r-- | sound/oss/vidc.c | 2 |
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; } |