Merge commit 'v2.6.39-rc3' into for-2.6.39

89 files changed, 244 insertions, 177 deletions

diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index fd2188c3df2b..58804c7acfcf 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_lib.c b/sound/core/pcm_lib.c
index a82e3756a72d..64449cb8f873 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -375,6 +375,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 	}
 
 	if (runtime->no_period_wakeup) {
+		snd_pcm_sframes_t xrun_threshold;
 		/*
 		 * Without regular period interrupts, we have to check
 		 * the elapsed time to detect xruns.
@@ -383,7 +384,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 		if (jdelta < runtime->hw_ptr_buffer_jiffies / 2)
 			goto no_delta_check;
 		hdelta = jdelta - delta * HZ / runtime->rate;
-		while (hdelta > runtime->hw_ptr_buffer_jiffies / 2 + 1) {
+		xrun_threshold = runtime->hw_ptr_buffer_jiffies / 2 + 1;
+		while (hdelta > xrun_threshold) {
 			delta += runtime->buffer_size;
 			hw_base += runtime->buffer_size;
 			if (hw_base >= runtime->boundary)
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index 917e4055ee30..150cb7edffee 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 fe5c8036beba..1a07750f3836 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 f3bdc54b429a..1d7d90ca455e 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 a89948ae9e8d..a39d3d8c2f9c 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 3c93c23e4883..e73fafd761b3 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 35a2f71a6af5..5e897b236cec 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/firewire/speakers.c b/sound/firewire/speakers.c
index 0fce9218abb1..5466de8527bd 100644
--- a/sound/firewire/speakers.c
+++ b/sound/firewire/speakers.c
@@ -778,10 +778,9 @@ static int __devexit fwspk_remove(struct device *dev)
 {
 	struct fwspk *fwspk = dev_get_drvdata(dev);
 
-	snd_card_disconnect(fwspk->card);
-
 	mutex_lock(&fwspk->mutex);
 	amdtp_out_stream_pcm_abort(&fwspk->stream);
+	snd_card_disconnect(fwspk->card);
 	fwspk_stop_stream(fwspk);
 	mutex_unlock(&fwspk->mutex);
 
diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c
index 0c40951b6523..5d61f5a29130 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 f14a7c0b6998..65329f3abc30 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 9191b32d9130..2a42cc377957 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 854c303264dc..0cd23d94888f 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 7df48a25c4ee..4b958b1c497c 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 87e2c72651f5..c918313c2206 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 ceedb1eff203..8cdb2cfe65c8 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 84ef4d06c1c2..fb5d7250de38 100644
--- a/