ASoC: Merge branch 'for-2.6.39' into for-2.6.40

Fix trivial conflict caused by silly spelling fix patch.

Conflicts:
	sound/soc/codecs/wm8994.c

88 files changed, 201 insertions, 158 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_lib.c b/sound/core/pcm_lib.c
index a82e3756a72..64449cb8f87 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 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/firewire/speakers.c b/sound/firewire/speakers.c
index 0fce9218abb..5466de8527b 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 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/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -466,7 +466,7 @@ 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/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index af678be0aa1..3b9fd115da3 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -607,7 +607,7 @@ struct hpi_data_compat32 {
 #endif
 
 struct hpi_buffer {
-  /** placehoder for backward compatability (see dwBufferSize) */
+  /** placehoder for backward compatibility (see dwBufferSize) */
 	struct hpi_msg_format reserved;
 	u32 command; /**< HPI_BUFFER_CMD_xxx*/
 	u32 pci_address; /**< PCI physical address of buffer for DSP DMA */
diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c
index bcbdf30a6aa..360028b9abf 100644
--- a/sound/pci/asihpi/hpimsgx.c
+++ b/sound/pci/asihpi/hpimsgx.c
@@ -722,7 +722,7 @@ static u16 HPIMSGX__init(struct hpi_message *phm,
 		return phr->error;
 	}
 	if (hr.error == 0) {
-		/* the adapter was created succesfully
+		/* the adapter was created successfully
 		   save the mapping for future use */
 		hpi_entry_points[hr.u.s.adapter_index] = entry_point_func;
 		/* prepare adapter (pre-open streams etc.) */
diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h
index ecb8f4daf40..02f6e08f759 100644
--- a/sound/pci/au88x0/au88x0.h
+++ b/sound/pci/au88x0/au88x0.h
@@ -104,7 +104,7 @@
 #define MIX_PLAYB(x) (vortex->mixplayb[x])
 #define MIX_SPDIF(x) (vortex->mixspdif[x])
 
-#define NR_WTPB 0x20		/* WT channels per eahc bank. */
+#define NR_WTPB 0x20		/* WT channels per each bank. */
 
 /* Structs */
 typedef struct {
diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c
index f4aa8ff6f5f..9ae8b3b1765 100644
--- a/sound/pci/au88x0/au88x0_a3d.c
+++ b/sound/pci/au88x0/au88x0_a3d.c
@@ -53,7 +53,7 @@ a3dsrc_GetTimeConsts(a3dsrc_t * a, short *HrtfTrack, short *ItdTrack,
 }
 
 #endif
-/* Atmospheric absorbtion. */
+/* Atmospheric absorption. */
 
 static void
 a3dsrc_SetAtmosTarget(a3dsrc_t * a, short aa, short b, short c, short d,
@@ -835,7 +835,7 @@ snd_vortex_a3d_filter_put(struct snd_kcontrol *kcontrol,
 		params[i] = ucontrol->value.integer.value[i];
 	/* Translate generic filter params to a3d filter params. */
 	vortex_a3d_translate_filter(a->filter, params);
-	/* Atmospheric absorbtion and filtering. */
+	/* Atmospheric absorption and filtering. */
 	a3dsrc_SetAtmosTarget(a, a->filter[0],
 			      a->filter[1], a->filter[2],
 			      a->filter[3], a->filter[4]);
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index 5439d662d10..33f0ba5559a 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -515,7 +515,7 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr)
 		return -ENODEV;
 
 	/* idx indicates which kind of PCM device. ADB, SPDIF, I2S and A3D share the 
-	 * same dma engine. WT uses it own separate dma engine whcih cant capture. */
+	 * same dma engine. WT uses it own separate dma engine which can't capture. */
 	if (idx == VORTEX_PCM_ADB)
 		nr_capt = nr;
 	else
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 5715c4d0557..9b7a6346037 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -140,7 +140,7 @@
  *  Possible remedies:
  *  - use speaker (amplifier) output instead of headphone output
  *    (in case crackling is due to overloaded output clipping)
- *  - plug card into a different PCI slot, preferrably one that isn't shared
+ *  - plug card into a different PCI slot, preferably one that isn't shared
  *    too much (this helps a lot, but not completely!)
  *  - get rid of PCI VGA card, use AGP instead
  *  - upgrade or downgrade BIOS
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h
index fc53b9bca26..e8e8ccc9640 100644
--- a/sound/pci/ca0106/ca0106.h
+++ b/sound/pci/ca0106/ca0106.h
@@ -51,7 +51,7 @@
  *    Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
  *
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -175,7 +175,7 @@
 /* CA0106 pointer-offset register set, accessed through the PTR and DATA registers                     */
 /********************************************************************************************************/
                                                                                                                            
-/* Initally all registers from 0x00 to 0x3f have zero contents. */
+/* Initially all registers from 0x00 to 0x3f have zero contents. */
 #define PLAYBACK_LIST_ADDR	0x00		/* Base DMA address of a list of pointers to each period/size */
 						/* One list entry: 4 bytes for DMA address, 
 						 * 4 bytes for period_size << 16.
@@ -223,7 +223,7 @@
  * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3
  * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground
  * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground.
- * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Sheild on all three, 4 -> Red.
+ * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Shield on all three, 4 -> Red.
  * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card.
  */
 /* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 01b49388faf..43775923969 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -117,7 +117,7 @@
  *    DAC: Unknown
  *    Trying to handle it like the SB0410.
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 630aa499818..84f3f92436b 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -42,7 +42,7 @@
  *  0.0.18
  *    Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index ba96428c9f4..c694464b116 100644
--- a/sound/pci/ca0106/ca0106_proc.c
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -42,7 +42,7 @@
  *  0.0.18
  *    Implement support for Line-in capture on SB Live 24bit.
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index b5bb036ef73..f4e573555da 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -73,7 +73,7 @@ MODULE_PARM_DESC(mpu_port, "MPU-401 port.");
 module_param_array(fm_port, long, NULL, 0444);
 MODULE_PARM_DESC(fm_port, "FM port.");
 module_param_array(soft_ac3, bool, NULL, 0444);
-MODULE_PARM_DESC(soft_ac3, "Sofware-conversion of raw SPDIF packets (model 033 only).");
+MODULE_PARM_DESC(soft_ac3, "Software-conversion of raw SPDIF packets (model 033 only).");
 #ifdef SUPPORT_JOYSTICK
 module_param_array(joystick_port, int, NULL, 0444);
 MODULE_PARM_DESC(joystick_port, "Joystick port address.");
@@ -656,8 +656,8 @@ out:
 }
 
 /*
- * Program pll register bits, I assume that the 8 registers 0xf8 upto 0xff
- * are mapped onto the 8 ADC/DAC sampling frequency which can be choosen
+ * Program pll register bits, I assume that the 8 registers 0xf8 up to 0xff
+ * are mapped onto the 8 ADC/DAC sampling frequency which can be chosen
  * at the register CM_REG_FUNCTRL1 (0x04).
  * Problem: other ways are also possible (any information about that?)
  */
@@ -666,7 +666,7 @@ static void snd_cmipci_set_pll(struct cmipci *cm, unsigned int rate, unsigned in
 	unsigned int reg = CM_REG_PLL + slot;
 	/*
 	 * Guess that this programs at reg. 0x04 the pos 15:13/12:10
-	 * for DSFC/ASFC (000 upto 111).
+	 * for DSFC/ASFC (000 up to 111).
 	 */
 
 	/* FIXME: Init (Do we've to set an other register first before programming?) */
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index b9321544c31..13f33c0719d 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1627,7 +1627,7 @@ static struct ct_atc atc_preset __devinitdata = {
  *  Creates and initializes a hardware manager.
  *
  *  Creates kmallocated ct_atc structure. Initializes hardware.
- *  Returns 0 if suceeds, or negative error code if fails.
+ *  Returns 0 if succeeds, or negative error code if fails.
  */
 
 int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c
index 0cf400f879f..a5c957db5ce 100644
--- a/sound/pci/ctxfi/cthw20k1.c
+++ b/sound/pci/ctxfi/cthw20k1.c
@@ -1285,7 +1285,7 @@ static int hw_trn_init(struct hw *hw, const struct trn_conf *info)
 	hw_write_20kx(hw, PTPALX, ptp_phys_low);
 	hw_write_20kx(hw, PTPAHX, ptp_phys_high);
 	hw_write_20kx(hw, TRNCTL, trnctl);
-	hw_write_20kx(hw, TRNIS, 0x200c01); /* realy needed? */
+	hw_write_20kx(hw, TRNIS, 0x200c01); /* really needed? */
 
 	return 0;
 }
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index 957a311514c..c250614dadd 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -248,7 +248,7 @@ static int is_valid_page(struct snd_emu10k1 *emu, dma_addr_t addr)
 /*
  * map the given memory block on PTB.
  * if the block is already mapped, update the link order.
- * if no empty pages are found, tries to release unsed memory blocks
+ * if no empty pages are found, tries to release unused memory blocks
  * and retry the mapping.
  */
 int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk)
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index 61b8ab39800..a81dc44228e 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -69,7 +69,7 @@
  *    ADC: Philips 1361T (Stereo 24bit)
  *    DAC: CS4382-K (8-channel, 24bit, 192Khz)
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h
index 00f4817533b..4e0ee1a9747 100644
--- a/sound/pci/emu10k1/p16v.h
+++ b/sound/pci/emu10k1/p16v.h
@@ -59,7 +59,7 @@
  *    ADC: Philips 1361T (Stereo 24bit)
  *    DAC: CS4382-K (8-channel, 24bit, 192Khz)
  *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
  *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
  *
  *   This program is free software; you can redistribute it and/or modify
@@ -86,7 +86,7 @@
  * The sample rate is also controlled by the same registers that control the rate of the EMU10K2 sample rate converters.
  */
 
-/* Initally all registers from 0x00 to 0x3f have zero contents. */
+/* Initially all registers from 0x00 to 0x3f have zero contents. */
 #define PLAYBACK_LIST_ADDR	0x00		/* Base DMA address of a list of pointers to each period/size */
 						/* One list entry: 4 bytes for DMA address, 
 						 * 4 bytes for period_size << 16.
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 537cfba829a..863eafea691 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -229,6 +229,7 @@ MODULE_PARM_DESC(lineio, "Line In to Rear Out (0 = auto, 1 = force).");
 #define ES_REG_1371_CODEC 0x14	/* W/R: Codec Read/Write register address */
 #define   ES_1371_CODEC_RDY	   (1<<31)	/* codec ready */
 #define   ES_1371_CODEC_WIP	   (1<<30)	/* codec register access in progress */
+#define   EV_1938_CODEC_MAGIC	   (1<<26)
 #define   ES_1371_CODEC_PIRD	   (1<<23)	/* codec read/write select register */
 #define   ES_1371_CODEC_WRITE(a,d) ((((a)&0x7f)<<16)|(((d)&0xffff)<<0))
 #define   ES_1371_CODEC_READS(a)   ((((a)&0x7f)<<16)|ES_1371_CODEC_PIRD)
@@ -603,12 +604,18 @@ static void snd_es1370_codec_write(struct snd_ak4531 *ak4531,
 
 #ifdef CHIP1371
 
+static inline bool is_ev1938(struct ensoniq *ensoniq)
+{
+	return ensoniq->pci->device == 0x8938;
+}
+
 static void snd_es1371_codec_write(struct snd_ac97 *ac97,
 				   unsigned short reg, unsigned short val)
 {
 	struct ensoniq *ensoniq = ac97->private_data;
-	unsigned int t, x;
+	unsigned int t, x, flag;
 
+	flag = is_ev1938(ensoniq) ? EV_1938_CODEC_MAGIC : 0;
 	mutex_lock(&ensoniq->src_mutex);
 	for (t = 0; t < POLL_COUNT; t++) {
 		if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP)) {
@@ -630,7 +637,8 @@ static void snd_es1371_codec_write(struct snd_ac97 *ac97,
 				    0x00010000)
 					break;
 			}
-			outl(ES_1371_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1371_CODEC));
+			outl(ES_1371_CODEC_WRITE(reg, val) | flag,
+			     ES_REG(ensoniq, 1371_CODEC));
 			/* restore SRC reg */
 			snd_es1371_wait_src_ready(ensoniq);
 			outl(x, ES_REG(ensoniq, 1371_SMPRATE));
@@ -647,8 +655,9 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97,
 					    unsigned short reg)
 {
 	struct ensoniq *ensoniq = ac97->private_data;
-	unsigned int t, x, fail = 0;
+	unsigned int t, x, flag, fail = 0;
 
+	flag = is_ev1938(ensoniq) ? EV_1938_CODEC_MAGIC : 0;
       __again:
 	mutex_lock(&ensoniq->src_mutex);
 	for (t = 0; t < POLL_COUNT; t++) {
@@ -671,7 +680,8 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97,
 				    0x00010000)
 					break;
 			}
-			outl(ES_1371_CODEC_READS(reg), ES_REG(ensoniq, 1371_CODEC));
+			outl(ES_1371_CODEC_READS(reg) | flag,
+			     ES_REG(ensoniq, 1371_CODEC));
 			/* restore SRC reg */
 			snd_es1371_wait_src_ready(ensoniq);
 			outl(x, ES_REG(ensoniq, 1371_SMPRATE));
@@ -683,6 +693,11 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97,
 			/* now wait for the stinkin' data (RDY) */
 			for (t = 0; t < POLL_COUNT; t++) {
 				if ((x = inl(ES_REG(ensoniq, 1371_CODEC))) & ES_1371_CODEC_RDY) {
+					if (is_ev1938(ensoniq)) {
+						for (t = 0; t < 100; t++)
+							inl(ES_REG(ensoniq, CONTROL));
+						x = inl(ES_REG(ensoniq, 1371_CODEC));
+					}
 					mutex_unlock(&ensoniq->src_mutex);
 					return ES_1371_CODEC_READ(x);
 				}
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 2c79e96d032..430f41db604 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3661,7 +3661,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec)
  * with the proper parameters for set up.
  * ops.cleanup should be called in hw_free for clean up of streams.
  *
- * This function returns 0 if successfull, or a negative error code.
+ * This function returns 0 if successful, or a negative error code.
  */
 int __devinit snd_hda_build_pcms(struct hda_bus *bus)
 {
@@ -4851,7 +4851,7 @@ EXPORT_SYMBOL_HDA(snd_hda_suspend);
  *
  * Returns 0 if successful.
  *
- * This fucntion is defined only when POWER_SAVE isn't set.
+ * This function is defined only when POWER_SAVE isn't set.
  * In the power-save mode, the codec is resumed dynamically.
  */
 int snd_hda_resume(struct hda_bus *bus)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index d08cf31596f..ad97d937d3a 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3034,6 +3034,8 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
  	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+	SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
+	SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
 	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
 	{}
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 251773e45f6..715615a88a8 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1280,6 +1280,39 @@ static int simple_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					     stream_tag, format, substream);
 }
 
+static void nvhdmi_8ch_7x_set_info_frame_parameters(struct hda_codec *codec,
+						    int channels)
+{
+	unsigned int chanmask;
+	int chan = channels ? (channels - 1) : 1;
+
+	switch (channels) {
+	default:
+	case 0:
+	case 2:
+		chanmask = 0x00;
+		break;
+	case 4:
+		chanmask = 0x08;
+		break;
+	case 6:
+		chanmask = 0x0b;
+		break;
+	case 8:
+		chanmask = 0x13;
+		break;
+	}
+
+	/* Set the audio infoframe channel allocation and checksum fields.  The
+	 * channel count is computed implicitly by the hardware. */
+	snd_hda_codec_write(codec, 0x1, 0,
+			Nv_VERB_SET_Channel_Allocation, chanmask);
+
+	snd_hda_codec_write(codec, 0x1, 0,
+			Nv_VERB_SET_Info_Frame_Checksum,
+			(0x71 - chan - chanmask));
+}
+
 static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo,
 				   struct hda_codec *codec,
 				   struct snd_pcm_substream *substream)
@@ -1298,6 +1331,10 @@ static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo,
 				AC_VERB_SET_STREAM_FORMAT, 0);
 	}
 
+	/* The audio hardware sends a channel count of 0x7 (8ch) when all the
+	 * streams are disabled. */
+	nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8);
+
 	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
 }
 
@@ -1308,37 +1345,16 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
 				     struct snd_pcm_substream *substream)
 {
 	int chs;
-	unsigned int dataDCC1, dataDCC2, chan, chanmask, channel_id;
+	unsigned int dataDCC1, dataDCC2, channel_id;
 	int i;
 
 	mutex_lock(&codec->spdif_mutex);
 
 	chs = substream->runtime->channels;
-	chan = chs ? (chs - 1) : 1;
 
-	switch (chs) {
-	default:
-	case 0:
-	case 2:
-		chanmask = 0x00;
-		break;
-	case 4:
-		chanmask = 0x08;
-		break;
-	case 6:
-		chanmask = 0x0b;
-		break;
-	case 8:
-		chanmask = 0x13;
-		break;
-	}
 	dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT;
 	dataDCC2 = 0x2;
 
-	/* set the Audio InforFrame Channel Allocation */
-	snd_hda_codec_write(codec, 0x1, 0,
-			Nv_VERB_SET_Channel_Allocation, chanmask);
-
 	/* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
 	if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
 		snd_hda_codec_write(codec,
@@ -1413,10 +1429,7 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo,
 		}
 	}
 
-	/* set the Audio Info Frame Checksum */
-	snd_hda_codec_write(codec, 0x1, 0,
-			Nv_VERB_SET_Info_Frame_Checksum,
-			(0x71 - chan - chanmask));
+	nvhdmi_8ch_7x_set_info_frame_parameters(codec, chs);
 
 	mutex_unlock(&codec->spdif_mutex);
 	return 0;
@@ -1512,6 +1525,11 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec)
 	spec->multiout.max_channels = 8;
 	spec->pcm_playback = &nvhdmi_pcm_playback_8ch_7x;
 	codec->patch_ops = nvhdmi_patch_ops_8ch_7x;
+
+	/* Initialize the audio infoframe channel mask and checksum to something
+	 * valid */
+	nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8);
+
 	return 0;
 }
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 12c6f4508c5..52928d9a72d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -549,7 +549,7 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
 
 /*
  * Control the mode of pin widget settings via the mixer.  "pc" is used
- * instead of "%" to avoid consequences of accidently treating the % as
+ * instead of "%" to avoid consequences of accidentally treating the % as
  * being part of a format specifier.  Maximum allowed length of a value is
  * 63 characters plus NULL terminator.
  *
@@ -9836,7 +9836,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 
 	SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
 
-	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavilion", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
@@ -14124,7 +14124,7 @@ static hda_nid_t alc269vb_capsrc_nids[1] = {
 };
 
 static hda_nid_t alc269_adc_candidates[] = {
-	0x08, 0x09, 0x07,
+	0x08, 0x09, 0x07, 0x11,
 };
 
 #define alc269_modes		alc260_modes
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 05fcd60cc46..94d19c03a7f 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2475,7 +2475,7 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
  
 	spec->hp_switch = ucontrol->value.integer.value[0] ? nid : 0;
 
-	/* check to be sure that the ports are upto date with
+	/* check to be sure that the ports are up to date with
 	 * switch changes
 	 */
 	stac_issue_unsol_event(codec, nid);
@@ -3408,6 +3408,9 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
 	hda_nid_t conn[HDA_MAX_NUM_INPUTS];
 	int i, nums;
 
+	if (!(get_wcaps(codec, mux) & AC_WCAP_CONN_LIST))
+		return -1;
+
 	nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
 	for (i = 0; i < nums; i++)
 		if (conn[i] == nid)
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 2f6252266a0..3e4f8c12ffc 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -148,7 +148,7 @@ static void aureon_pca9554_write(struct snd_ice1712 *ice, unsigned char reg,
 	udelay(100);
 	/*
 	 * send device address, command and value,
-	 * skipping ack cycles inbetween
+	 * skipping ack cycles in between
 	 */
 	for (j = 0; j < 3; j++) {
 		switch (j) {
@@ -2143,7 +2143,7 @@ static int __devinit aureon_init(struct snd_ice1712 *ice)
 		ice->num_total_adcs = 2;
 	}
 
-	/* to remeber the register values of CS8415 */
+	/* to remember the register values of CS8415 */
 	ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
 	if (!ice->akm)
 		return -ENOMEM;
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 4fc6d8bc637..f4594d76b6e 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2755,7 +2755,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
 			return err;
 		}
 		if (c->mpu401_1_name)
-			/*  Prefered name available in card_info */
+			/*  Preferred name available in card_info */
 			snprintf(ice->rmidi[0]->name,
 				 sizeof(ice->rmidi[0]->name),
 				 "%s %d", c->mpu401_1_name, card->number);
@@ -2772,7 +2772,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
 				return err;
 			}
 			if (c->mpu401_2_name)
-				/*  Prefered name available in card_info */
+				/*  Preferred name available in card_info */
 				snprintf(ice->rmidi[1]->name,
 					 sizeof(ice->rmidi[1]->name),
 					 "%s %d", c->mpu401_2_name,
diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c
index cdb873f5da5..92c1160d7ab 100644
--- a/sound/pci/ice1712/pontis.c
+++ b/sound/pci/ice1712/pontis.c
@@ -768,7 +768,7 @@ static int __devinit pontis_init(struct snd_ice1712 *ice)
 	ice->num_total_dacs = 2;
 	ice->num_total_adcs = 2;
 
-	/* to remeber the register values */
+	/* to remember the register values */
 	ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
 	if (! ice->akm)
 		return -ENOMEM;
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 6a9fee3ee78..764cc93dbca 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -1046,7 +1046,7 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice)
 	* don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten
 	*/
 	ice->gpio.saved[0] = 0;
-	/* to remeber the register values */
+	/* to remember the register values */
 
 	ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
 	if (! ice->akm)
@@ -1128,7 +1128,7 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice)
 	* don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten
 	*/
 	ice->gpio.saved[0] = 0;
-	/* to remeber the register values */
+	/* to remember the register values */
 
 	ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
 	if (! ice->akm)
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 629a5494347..6c896dbfd79 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -534,7 +534,7 @@ static int snd_intel8x0_codec_semaphore(struct intel8x0 *chip, unsigned int code
 		udelay(10);
 	} while (time--);
 
-	/* access to some forbidden (non existant) ac97 registers will not
+	/* access to some forbidden (non existent) ac97 registers will not
 	 * reset the semaphore. So even if you don't get the semaphore, still
 	 * continue the access. We don't need the semaphore anyway. */
 	snd_printk(KERN_ERR "codec_semaphore: semaphore is not ready [0x%x][0x%x]\n",
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 2ae8d29500a..27709f0cd2a 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -331,7 +331,7 @@ static int snd_intel8x0m_codec_semaphore(struct intel8x0m *chip, unsigned int co
 		udelay(10);
 	} while (time--);
 
-	/* access to some forbidden (non existant) ac97 registers will not
+	/* access to some forbidden (non existent) ac97 registers will not
 	 * reset the semaphore. So even if you don't get the semaphore, still
 	 * continue the access. We don't need the semaphore anyway. */
 	snd_printk(KERN_ERR "codec_semaphore: semaphore is not ready [0x%x][0x%x]\n",
diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c
index d3350f38396..3df0f530f67 100644
--- a/sound/pci/mixart/mixart_core.c
+++ b/sound/pci/mixart/mixart_core.c
@@ -265,7 +265,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int
 	if (! timeout) {
 		/* error - no ack */
 		mutex_unlock(&mgr->msg_mutex);
-		snd_printk(KERN_ERR "error: no reponse on msg %x\n", msg_frame);
+		snd_printk(KERN_ERR "error: no response on msg %x\n", msg_frame);
 		return -EIO;
 	}
 
@@ -278,7 +278,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int
 	err = get_msg(mgr, &resp, msg_frame);
 
 	if( request->message_id != resp.message_id )
-		snd_printk(KERN_ERR "REPONSE ERROR!\n");
+		snd_printk(KERN_ERR "RESPONSE ERROR!\n");
 
 	mutex_unlock(&mgr->msg_mutex);
 	return err;
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index 833e7180ad2..304411c1fe4 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -1042,11 +1042,11 @@ void pcxhr_msg_tasklet(unsigned long arg)
 	int i, j;
 
 	if (mgr->src_it_dsp & PCXHR_IRQ_FREQ_CHANGE)
-		snd_printdd("TASKLET : PCXHR_IRQ_FREQ_CHANGE event occured\n");
+		snd_printdd("TASKLET : PCXHR_IRQ_FREQ_CHANGE event occurred\n");
 	if (mgr->src_it_dsp & PCXHR_IRQ_TIME_CODE)
-		snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occured\n");
+		snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occurred\n");
 	if (mgr->src_it_dsp & PCXHR_IRQ_NOTIFY)
-		snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occured\n");
+		snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occurred\n");
 	if (mgr->src_it_dsp & (PCXHR_IRQ_FREQ_CHANGE | PCXHR_IRQ_TIME_CODE)) {
 		/* clear events FREQ_CHANGE and TIME_CODE */
 		pcxhr_init_rmh(prmh, CMD_TEST_IT);
@@ -1055,7 +1055,7 @@ void pcxhr_msg_tasklet(unsigned long arg)
 			    err, prmh->stat[0]);
 	}
 	if (mgr->src_it_dsp & PCXHR_IRQ_ASYNC) {
-		snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occured\n");
+		snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occurred\n");
 
 		pcxhr_init_rmh(prmh, CMD_ASYNC);
 		prmh->cmd[0] |= 1;	/* add SEL_ASYNC_EVENTS */
@@ -1233,7 +1233,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 	reg = PCXHR_INPL(mgr, PCXHR_PLX_L2PCIDB);
 	PCXHR_OUTPL(mgr, PCXHR_PLX_L2PCIDB, reg);
 
-	/* timer irq occured */
+	/* timer irq occurred */
 	if (reg & PCXHR_IRQ_TIMER) {
 		int timer_toggle = reg & PCXHR_IRQ_TIMER;
 		/* is a 24 bit counter */
@@ -1288,7 +1288,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 	if (reg & PCXHR_IRQ_MASK) {
 		if (reg & PCXHR_IRQ_ASYNC) {
 			/* as we didn't request any async notifications,
-			 * some kind of xrun error will probably occured
+			 * some kind of xrun error will probably occurred
 			 */
 			/* better resynchronize all streams next interrupt : */
 			mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID;
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index d5f5b440fc4..9ff247fc887 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -150,7 +150,7 @@ MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard.");
 #define RME96_RCR_BITPOS_F1 28
 #define RME96_RCR_BITPOS_F2 29
 
-/* Additonal register bits */
+/* Additional register bits */
 #define RME96_AR_WSEL       (1 << 0)
 #define RME96_AR_ANALOG     (1 << 1)
 #define RME96_AR_FREQPAD_0  (1 << 2)
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index a323eafb9e0..949691a876d 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -391,7 +391,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 
 /* Status2 Register bits */ /* MADI ONLY */
 
-#define HDSPM_version0 (1<<0)	/* not realy defined but I guess */
+#define HDSPM_version0 (1<<0)	/* not really defined but I guess */
 #define HDSPM_version1 (1<<1)	/* in former cards it was ??? */
 #define HDSPM_version2 (1<<2)
 
@@ -936,7 +936,7 @@ struct hdspm {
 	struct snd_kcontrol *playback_mixer_ctls[HDSPM_MAX_CHANNELS];
 	/* but input to much, so not used */
 	struct snd_kcontrol *input_mixer_ctls[HDSPM_MAX_CHANNELS];
-	/* full mixer accessable over mixer ioctl or hwdep-device */
+	/* full mixer accessible over mixer ioctl or hwdep-device */
 	struct hdspm_mixer *mixer;
 
 	struct hdspm_tco *tco;  /* NULL if no TCO detected */
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 1b8f6742b5f..2b5c7a95ae1 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -308,7 +308,7 @@ static irqreturn_t sis_interrupt(int irq, void *dev)
 	u32 intr, status;
 
 	/* We only use the DMA interrupts, and we don't enable any other
-	 * source of interrupts. But, it is possible to see an interupt
+	 * source of interrupts. But, it is possible to see an interrupt
 	 * status that didn't actually interrupt us, so eliminate anything
 	 * we're not expecting to avoid falsely claiming an IRQ, and an
 	 * ensuing endless loop.
@@ -773,7 +773,7 @@ static void sis_prepare_timing_voice(struct voice *voice,
 		vperiod = 0;
 	}
 
-	/* The interrupt handler implements the timing syncronization, so
+	/* The interrupt handler implements the timing synchronization, so
 	 * setup its state.
 	 */
 	timing->flags |= VOICE_SYNC_TIMING;
@@ -1139,7 +1139,7 @@ static int sis_chip_init(struct sis7019 *sis)
 	 */
 	outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR);
 
-	/* Reset the syncronization groups for all of the channels
+	/* Reset the synchronization groups for all of the channels
 	 * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc.
 	 * we'll need to change how we handle these. Until then, we just
 	 * assign sub-mixer 0 to all playback channels, and avoid any
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index edce8a27e3e..bc823a54755 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -358,7 +358,7 @@ static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id)
 		 * filling dummy data, serial automatically start to
 		 * consume them and then will generate normal buffer
 		 * empty interrupts.
-		 * If both buffer underflow and buffer empty are occured,
+		 * If both buffer underflow and buffer empty are occurred,
 		 * it is better to do nomal data transfer than empty one
 		 */
 		snd_ps3_program_dma(card,
diff --git a/sound/ppc/snd_ps3_reg.h b/sound/ppc/snd_ps3_reg.h
index 03fdee4aaaf..2e630207956 100644
--- a/sound/ppc/snd_ps3_reg.h
+++ b/sound/ppc/snd_ps3_reg.h
@@ -125,7 +125,7 @@
    transfers.  Any interrupts associated with the canceled transfers
    will occur as if the transfer had finished.
    Since this bit is designed to recover from DMA related issues
-   which are caused by unpredictable situations, it is prefered to wait
+   which are caused by unpredictable situations, it is preferred to wait
    for normal DMA transfer end without using this bit.
 */
 #define PS3_AUDIO_CONFIG_CLEAR          (1 << 8)  /* RWIVF */
@@ -316,13 +316,13 @@ DISABLED=Interrupt generation disabled.
 
 /*
 Audio Port Interrupt Status Register
-Indicates Interrupt status, which interrupt has occured, and can clear
+Indicates Interrupt status, which interrupt has occurred, and can clear
 each interrupt in this register.
 Writing 1b to a field containing 1b clears field and de-asserts interrupt.
 Writing 0b to a field has no effect.
 Field vaules are the following:
-0 - Interrupt hasn't occured.
-1 - Interrupt has occured.
+0 - Interrupt hasn't occurred.
+1 - Interrupt has occurred.
 
 
  31            24 23           16 15            8 7             0
@@ -473,7 +473,7 @@ Channel N is out of action by setting 0 to asoen.
 /*
 Sampling Rate
 Specifies the divide ratio of the bit clock (clock output
-from bclko) used by the 3-wire Audio Output Clock, whcih
+from bclko) used by the 3-wire Audio Output Clock, which
 is applied to the master clock selected by mcksel.
 Data output is synchronized with this clock.
 */
@@ -756,7 +756,7 @@ The STATUS field can be used to monitor the progress of a DMA request.
 DONE indicates the previous request has completed.
 EVENT indicates that the DMA engine is waiting for the EVENT to occur.
 PENDING indicates that the DMA engine has not started processing this
-request, but the EVENT has occured.
+request, but the EVENT has occurred.
 DMA indicates that the data transfer is in progress.
 NOTIFY indicates that the notifier signalling end of transfer is being written.
 CLEAR indicated that the previous transfer was cleared.
@@ -824,7 +824,7 @@ AUDIOFIFO = Audio WriteData FIFO,
 
 /*
 PS3_AUDIO_DMASIZE specifies the number of 128-byte blocks + 1 to transfer.
-So a value of 0 means 128-bytes will get transfered.
+So a value of 0 means 128-bytes will get transferred.
 
 
  31            24 23           16 15            8 7             0
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 5d230cee3fa..7fbfa051f6e 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -672,7 +672,7 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai)
 	/* re-enable interrupts */
 	ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr);
 
-	/* Re-enable recieve and transmit as appropriate */
+	/* Re-enable receive and transmit as appropriate */
 	cr = 0;
 	cr |=
 	    (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0;
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 4f377c9e868..eecffb54894 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -481,7 +481,7 @@ struct _pll_div {
 };
 
 /* Note : pll code from original alc5623 driver. Not sure of how good it is */
-/* usefull only for master mode */
+/* useful only for master mode */
 static const struct _pll_div codec_master_pll_div[] = {
 
 	{  2048000,  8192000,	0x0ea0},
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index 72de47e5d04..2c2a681da0d 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -161,7 +161,7 @@ static const struct snd_kcontrol_new lm4857_controls[] = {
 		lm4857_get_mode, lm4857_set_mode),
 };
 
-/* There is a demux inbetween the the input signal and the output signals.
+/* There is a demux between the input signal and the output signals.
  * Currently there is no easy way to model it in ASoC and since it does not make
  * much of a difference in practice simply connect the input direclty to the
  * outputs. */
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index f70977d7dbe..84ffdebb8a8 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -26,7 +26,9 @@
 #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
 
 #include <linux/platform_device.h>
+#include <linux/delay.h>
 #include <linux/slab.h>
+
 #include <asm/intel_scu_ipc.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h
index 62b1f226142..67f19c3bebe 100644
--- a/sound/soc/codecs/tlv320aic26.h
+++ b/sound/soc/codecs/tlv320aic26.h
@@ -14,14 +14,14 @@
 #define AIC26_PAGE_ADDR(page, offset)	((page << 6) | offset)
 #define AIC26_NUM_REGS			AIC26_PAGE_ADDR(3, 0)
 
-/* Page 0: Auxillary data registers */
+/* Page 0: Auxiliary data registers */
 #define AIC26_REG_BAT1			AIC26_PAGE_ADDR(0, 0x05)
 #define AIC26_REG_BAT2			AIC26_PAGE_ADDR(0, 0x06)
 #define AIC26_REG_AUX			AIC26_PAGE_ADDR(0, 0x07)
 #define AIC26_REG_TEMP1			AIC26_PAGE_ADDR(0, 0x09)
 #define AIC26_REG_TEMP2			AIC26_PAGE_ADDR(0, 0x0A)
 
-/* Page 1: Auxillary control registers */
+/* Page 1: Auxiliary control registers */
 #define AIC26_REG_AUX_ADC		AIC26_PAGE_ADDR(1, 0x00)
 #define AIC26_REG_STATUS		AIC26_PAGE_ADDR(1, 0x01)
 #define AIC26_REG_REFERENCE		AIC26_PAGE_ADDR(1, 0x03)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 3bedab26892..6c43c13f043 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -884,7 +884,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	if (bypass_pll)
 		return 0;
 
-	/* Use PLL, compute apropriate setup for j, d, r and p, the closest
+	/* Use PLL, compute appropriate setup for j, d, r and p, the closest
 	 * one wins the game. Try with d==0 first, next with d!=0.
 	 * Constraints for j are according to the datasheet.
 	 * The sysclk is divided by 1000 to prevent integer overflows.
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index eb1a0b4e09b..082e9d51963 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1027,7 +1027,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 		/*
 		 * For FIFO bypass mode:
 		 * Enable the FIFO bypass (Disable the FIFO use)
-		 * Set the BCLK as continous
+		 * Set the BCLK as continuous
 		 */
 		fifoctrl_a |= DAC33_FBYPAS;
 		aictrl_b |= DAC33_BCLKON;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 8512800f632..575238d68e5 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -281,7 +281,7 @@ static inline void twl4030_check_defaults(struct snd_soc_codec *codec)
 				 i, val, twl4030_reg[i]);
 		}
 	}
-	dev_dbg(codec->dev, "Found %d non maching registers. %s\n",
+	dev_dbg(codec->dev, "Found %d non-matching registers. %s\n",
 		 difference, difference ? "Not OK" : "OK");
 }
 
@@ -2018,7 +2018,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
 	u8 mode;
 
 	/* If the system master clock is not 26MHz, the voice PCM interface is
-	 * not avilable.
+	 * not available.
 	 */
 	if (twl4030->sysclk != 26000) {
 		dev_err(codec->dev, "The board is configured for %u Hz, while"
@@ -2028,7 +2028,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
 	}
 
 	/* If the codec mode is not option2, the voice PCM interface is not
-	 * avilable.
+	 * available.
 	 */
 	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
 		& TWL4030_OPT_MODE;
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 8f6b5ee6645..4bbc0a79f01 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -772,7 +772,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 			reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD);
 			snd_soc_write(codec, WM8580_PWRDN1, reg);
 
-			/* Make VMID high impedence */
+			/* Make VMID high impedance */
 			reg = snd_soc_read(codec,  WM8580_ADC_CONTROL1);
 			reg &= ~0x100;
 			snd_soc_write(codec, WM8580_ADC_CONTROL1, reg);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 3f09deea8d9..ffa2ffe5ec1 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1312,7 +1312,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
 	SNDRV_PCM_FMTBIT_S24_LE)
 
 /*
- * The WM8753 supports upto 4 different and mutually exclusive DAI
+ * The WM8753 supports up to 4 different and mutually exclusive DAI
  * configurations. This gives 2 PCM's available for use, hifi and voice.
  * NOTE: The Voice PCM cannot play or capture audio to the CPU as it's DAI
  * is connected between the wm8753 and a BT codec or GSM modem.
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 443ae580445..9b3bba4df5b 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1895,7 +1895,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	pr_debug("Fvco=%dHz\n", target);
 
-	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	/* Find an appropriate FLL_FRATIO and factor it out of the target */
 	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
 		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
 			fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 5e0214d6293..3c7198779c3 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -176,7 +176,7 @@ static int wm8995_pll_factors(struct device *dev,
 	return 0;
 }
 
-/* Lookup table specifiying SRATE (table 25 in datasheet); some of the
+/* Lookup table specifying SRATE (table 25 in datasheet); some of the
  * output frequencies have been rounded to the standard frequencies
  * they are intended to match where the error is slight. */
 static struct {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 3b71dd65c96..500011eb8b2 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3137,7 +3137,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	pr_debug("FLL Fvco=%dHz\n", target);
 
-	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	/* Find an appropriate FLL_FRATIO and factor it out of the target */
 	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
 		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
 			fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 28fdfd66661..3c2ee1bb73c 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -981,7 +981,7 @@ static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai,
 		reg = snd_soc_read(codec, WM8991_CLOCKING_2);
 		snd_soc_write(codec, WM8991_CLOCKING_2, reg | WM8991_SYSCLK_SRC);
 
-		/* set up N , fractional mode and pre-divisor if neccessary */
+		/* set up N , fractional mode and pre-divisor if necessary */
 		snd_soc_write(codec, WM8991_PLL1, pll_div.n | WM8991_SDM |
 			      (pll_div.div2 ? WM8991_PRESCALE : 0));
 		snd_soc_write(codec, WM8991_PLL2, (u8)(pll_div.k>>8));
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 379fa22c5b6..056aef90434 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -324,7 +324,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	pr_debug("Fvco=%dHz\n", target);
 
-	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	/* Find an appropriate FLL_FRATIO and factor it out of the target */
 	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
 		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
 			fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 55cdf298202..91c6b39de50 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -305,7 +305,7 @@ static int speaker_mode_get(struct snd_kcontrol *kcontrol,
 /*
  * Stop any attempts to change speaker mode while the speaker is enabled.
  *
- * We also have some special anti-pop controls dependant on speaker
+ * We also have some special anti-pop controls dependent on speaker
  * mode which must be changed along with the mode.
  */
 static int speaker_mode_put(struct snd_kcontrol *kcontrol,
@@ -456,7 +456,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	pr_debug("Fvco=%dHz\n", target);
 
-	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	/* Find an appropriate FLL_FRATIO and factor it out of the target */
 	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
 		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
 			fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index c331d65587d..5b13feca753 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -16,7 +16,7 @@
  * sane processor vendors have a FIFO per AC97 slot, the i.MX has only
  * one FIFO which combines all valid receive slots. We cannot even select
  * which slots we want to receive. The WM9712 with which this driver
- * was developped with always sends GPIO status data in slot 12 which
+ * was developed with always sends GPIO status data in slot 12 which
  * we receive in our (PCM-) data stream. The only chance we have is to
  * manually skip this data in the FIQ handler. With sampling rates different
  * from 48000Hz not every frame has valid receive data, so the ratio
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index 0fd6a630db0..e13c6ce4632 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -132,7 +132,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream)
 	priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
 	snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw);
 
-	/* Ensure that all constraints linked to dma burst are fullfilled */
+	/* Ensure that all constraints linked to dma burst are fulfilled */
 	err = snd_pcm_hw_constraint_minmax(runtime,
 			SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
 			priv->burst * 2,
@@ -170,7 +170,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream)
 
 		/*
 		 * Enable Error interrupts. We're only ack'ing them but
-		 * it's usefull for diagnostics
+		 * it's useful for diagnostics
 		 */
 		writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK);
 	}
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index d827edb3d54..9765fb81a5e 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -446,7 +446,7 @@ static int sst_platform_remove(struct platform_device *pdev)
 
 	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sst_platform_dai));
 	snd_soc_unregister_platform(&pdev->dev);
-	pr_debug("sst_platform_remove sucess\n");
+	pr_debug("sst_platform_remove success\n");
 	return 0;
 }
 
@@ -469,7 +469,7 @@ module_init(sst_soc_platform_init);
 static void __exit sst_soc_platform_exit(void)
 {
 	platform_driver_unregister(&sst_platform_driver);
-	pr_debug("sst_soc_platform_exit sucess\n");
+	pr_debug("sst_soc_platform_exit success\n");
 }
 module_exit(sst_soc_platform_exit);
 
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 3167be68962..462cbcbea74 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -248,7 +248,7 @@ static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
  */
 
 /* To actually apply any modem controlled configuration changes to the codec,
- * we must connect codec DAI pins to the modem for a moment.  Be carefull not
+ * we must connect codec DAI pins to the modem for a moment.  Be careful not
  * to interfere with our digital mute function that shares the same hardware. */
 static struct timer_list cx81801_timer;
 static bool cx81801_cmd_pending;
@@ -402,9 +402,9 @@ static struct tty_ldisc_ops cx81801_ops = {
 
 
 /*
- * Even if not very usefull, the sound card can still work without any of the
+ * Even if not very useful, the sound card can still work without any of the
  * above functonality activated.  You can still control its audio input/output
- * constellation and speakerphone gain from userspace by issueing AT commands
+ * constellation and speakerphone gain from userspace by issuing AT commands
  * over the modem port.
  */
 
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 78bfdb3f5d7..45223097563 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -228,7 +228,7 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Handset Mic"),
 };
 
-/* GTA02 specific routes and controlls */
+/* GTA02 specific routes and controls */
 
 #ifdef CONFIG_MACH_NEO1973_GTA02
 
@@ -372,7 +372,7 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-/* GTA01 specific controlls */
+/* GTA01 specific controls */
 
 #ifdef CONFIG_MACH_NEO1973_GTA01
 
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 58431589539..23c0e83d4c1 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1330,3 +1330,4 @@ module_exit(fsi_mobile_exit);
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip FSI audio driver");
 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
+MODULE_ALIAS("platform:fsi-pcm-audio");
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 6203a72d57a..7c17b98d584 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -331,7 +331,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
 			goto err;
 
 		if (gpios[i].wake) {
-			ret = set_irq_wake(gpio_to_irq(gpios[i].gpio), 1);
+			ret = irq_set_irq_wake(gpio_to_irq(gpios[i].gpio), 1);
 			if (ret != 0)
 				printk(KERN_ERR
 				  "Failed to mark GPIO %d as wake source: %d\n",
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index 9081a54a9c6..86c1a310376 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -76,7 +76,7 @@ struct ihex_record {
 	u16 address;
 	u8 len;
 	u8 data[256];
-	char error; /* true if an error occured parsing this record */
+	char error; /* true if an error occurred parsing this record */
 
 	u8 max_len; /* maximum record length in whole ihex */
 
@@ -107,7 +107,7 @@ static u8 usb6fire_fw_ihex_hex(const u8 *data, u8 *crc)
 
 /*
  * returns true if record is available, false otherwise.
- * iff an error occured, false will be returned and record->error will be true.
+ * iff an error occurred, false will be returned and record->error will be true.
  */
 static bool usb6fire_fw_ihex_next_record(struct ihex_record *record)
 {
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index b4b39c0b6c9..f9289102886 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1301,6 +1301,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi,
 	case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */
 	case USB_ID(0x15ca, 0x1806): /* Textech USB Midi Cable */
 	case USB_ID(0x1a86, 0x752d): /* QinHeng CH345 "USB2.0-MIDI" */
+	case USB_ID(0xfc08, 0x0101): /* Unknown vendor Cable */
 		ep->max_transfer = 4;
 		break;
 		/*
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 5e477571660..6ec33b62e6c 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1182,7 +1182,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 /*
  * parse a feature unit
  *
- * most of controlls are defined here.
+ * most of controls are defined here.
  */
 static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void *_ftr)
 {
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 355759bad58..ec07e62e53f 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -266,7 +266,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
  * audio-interface quirks
  *
  * returns zero if no standard audio/MIDI parsing is needed.
- * returns a postive value if standard audio/midi interfaces are parsed
+ * returns a positive value if standard audio/midi interfaces are parsed
  * after this.
  * returns a negative value at error.
  */
diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c
index 287ef73b123..a51340f6f2d 100644
--- a/sound/usb/usx2y/usx2yhwdeppcm.c
+++ b/sound/usb/usx2y/usx2yhwdeppcm.c
@@ -20,7 +20,7 @@
  at standard samplerates,
  what led to this part of the usx2y module: 
  It provides the alsa kernel half of the usx2y-alsa-jack driver pair.
- The pair uses a hardware dependant alsa-device for mmaped pcm transport.
+ The pair uses a hardware dependent alsa-device for mmaped pcm transport.
  Advantage achieved:
          The usb_hc moves pcm data from/into memory via DMA.
          That memory is mmaped by jack's usx2y driver.
@@ -38,7 +38,7 @@
          2periods works but is useless cause of crackling).
 
  This is a first "proof of concept" implementation.
- Later, functionalities should migrate to more apropriate places:
+ Later, functionalities should migrate to more appropriate places:
  Userland:
  - The jackd could mmap its float-pcm buffers directly from alsa-lib.
  - alsa-lib could provide power of 2 period sized shaping combined with int/float