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

80 files changed, 126 insertions, 126 deletions

diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index fd2188c3df2..58804c7acfc 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -170,7 +170,7 @@ static void tas_set_volume(struct tas *tas)
 	/* analysing the volume and mixer tables shows
 	 * that they are similar enough when we shift
 	 * the mixer table down by 4 bits. The error
-	 * is miniscule, in just one item the error
+	 * is minuscule, in just one item the error
 	 * is 1, at a value of 0x07f17b (mixer table
 	 * value is 0x07f17a) */
 	tmp = tas_gaintable[left];
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index 917e4055ee3..150cb7edffe 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -253,7 +253,7 @@ static int snd_pcm_lib_preallocate_pages1(struct snd_pcm_substream *substream,
  * snd_pcm_lib_preallocate_pages - pre-allocation for the given DMA type
  * @substream: the pcm substream instance
  * @type: DMA type (SNDRV_DMA_TYPE_*)
- * @data: DMA type dependant data
+ * @data: DMA type dependent data
  * @size: the requested pre-allocation size in bytes
  * @max: the max. allowed pre-allocation size
  *
@@ -278,10 +278,10 @@ int snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream,
 EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages);
 
 /**
- * snd_pcm_lib_preallocate_pages_for_all - pre-allocation for continous memory type (all substreams)
+ * snd_pcm_lib_preallocate_pages_for_all - pre-allocation for continuous memory type (all substreams)
  * @pcm: the pcm instance
  * @type: DMA type (SNDRV_DMA_TYPE_*)
- * @data: DMA type dependant data
+ * @data: DMA type dependent data
  * @size: the requested pre-allocation size in bytes
  * @max: the max. allowed pre-allocation size
  *
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index fe5c8036beb..1a07750f383 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -460,7 +460,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
 				   PM_QOS_CPU_DMA_LATENCY, usecs);
 	return 0;
  _error:
-	/* hardware might be unuseable from this time,
+	/* hardware might be unusable from this time,
 	   so we force application to retry to set
 	   the correct hardware parameter settings */
 	runtime->status->state = SNDRV_PCM_STATE_OPEN;
diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c
index f3bdc54b429..1d7d90ca455 100644
--- a/sound/core/seq/seq_dummy.c
+++ b/sound/core/seq/seq_dummy.c
@@ -50,7 +50,7 @@
 
 	option snd-seq-dummy ports=4
 
-  The modle option "duplex=1" enables duplex operation to the port.
+  The model option "duplex=1" enables duplex operation to the port.
   In duplex mode, a pair of ports are created instead of single port,
   and events are tunneled between pair-ports.  For example, input to
   port A is sent to output port of another port B and vice versa.
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index a89948ae9e8..a39d3d8c2f9 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -233,7 +233,7 @@ static void slave_free(struct snd_kcontrol *kcontrol)
  * Add a slave control to the group with the given master control
  *
  * All slaves must be the same type (returning the same information
- * via info callback).  The fucntion doesn't check it, so it's your
+ * via info callback).  The function doesn't check it, so it's your
  * responsibility.
  *
  * Also, some additional limitations:
diff --git a/sound/drivers/pcm-indirect2.c b/sound/drivers/pcm-indirect2.c
index 3c93c23e488..e73fafd761b 100644
--- a/sound/drivers/pcm-indirect2.c
+++ b/sound/drivers/pcm-indirect2.c
@@ -264,7 +264,7 @@ snd_pcm_indirect2_playback_transfer(struct snd_pcm_substream *substream,
 		if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2))
 			diff += runtime->boundary;
 		/* number of bytes "added" by ALSA increases the number of
-		 * bytes which are ready to "be transfered to HW"/"played"
+		 * bytes which are ready to "be transferred to HW"/"played"
 		 * Then, set rec->appl_ptr to not count bytes twice next time.
 		 */
 		rec->sw_ready += (int)frames_to_bytes(runtime, diff);
@@ -330,7 +330,7 @@ snd_pcm_indirect2_playback_transfer(struct snd_pcm_substream *substream,
 		/* copy bytes from intermediate buffer position sw_data to the
 		 * HW and return number of bytes actually written
 		 * Furthermore, set hw_ready to 0, if the fifo isn't empty
-		 * now => more could be transfered to fifo
+		 * now => more could be transferred to fifo
 		 */
 		bytes = copy(substream, rec, bytes);
 		rec->bytes2hw += bytes;
diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c
index 35a2f71a6af..5e897b236ce 100644
--- a/sound/drivers/vx/vx_pcm.c
+++ b/sound/drivers/vx/vx_pcm.c
@@ -1189,7 +1189,7 @@ void vx_pcm_update_intr(struct vx_core *chip, unsigned int events)
 
 
 /*
- * vx_init_audio_io - check the availabe audio i/o and allocate pipe arrays
+ * vx_init_audio_io - check the available audio i/o and allocate pipe arrays
  */
 static int vx_init_audio_io(struct vx_core *chip)
 {
diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c
index 0c40951b652..5d61f5a2913 100644
--- a/sound/isa/sb/emu8000.c
+++ b/sound/isa/sb/emu8000.c
@@ -370,7 +370,7 @@ init_arrays(struct snd_emu8000 *emu)
 
 /*
  * Size the onboard memory.
- * This is written so as not to need arbitary delays after the write. It
+ * This is written so as not to need arbitrary delays after the write. It
  * seems that the only way to do this is to use the one channel and keep
  * reallocating between read and write.
  */
diff --git a/sound/isa/wavefront/wavefront_midi.c b/sound/isa/wavefront/wavefront_midi.c
index f14a7c0b699..65329f3abc3 100644
--- a/sound/isa/wavefront/wavefront_midi.c
+++ b/sound/isa/wavefront/wavefront_midi.c
@@ -537,7 +537,7 @@ snd_wavefront_midi_start (snd_wavefront_card_t *card)
 	}
 
 	/* Turn on Virtual MIDI, but first *always* turn it off,
-	   since otherwise consectutive reloads of the driver will
+	   since otherwise consecutive reloads of the driver will
 	   never cause the hardware to generate the initial "internal" or 
 	   "external" source bytes in the MIDI data stream. This
 	   is pretty important, since the internal hardware generally will
diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 9191b32d913..2a42cc37795 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -424,7 +424,7 @@ void snd_wss_mce_down(struct snd_wss *chip)
 
 	/*
 	 * Wait for (possible -- during init auto-calibration may not be set)
-	 * calibration process to start. Needs upto 5 sample periods on AD1848
+	 * calibration process to start. Needs up to 5 sample periods on AD1848
 	 * which at the slowest possible rate of 5.5125 kHz means 907 us.
 	 */
 	msleep(1);
diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c
index 854c303264d..0cd23d94888 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 7df48a25c4e..4b958b1c497 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 87e2c72651f..c918313c220 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 ceedb1eff20..8cdb2cfe65c 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 84ef4d06c1c..fb5d7250de3 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 9890cf2066f..5c773dff5ac 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 44357d877a2..09d46484bc1 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 f0e0caa5320..12ba28e7b93 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;
 		}
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index 4382d0fa6b9..d8f6fd65ebb 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 f53a31e939c..f8ccc9677c6 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 6fc025c448d..255429c32c1 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 3e3c2ef6efd..8c8aac4c567 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 620525bdac5..22e9f08dea6 100644
--- a/