Fix common misspellings
Fixes generated by 'codespell' and manually reviewed.
Signed-off-by: Lucas De Marchi <lucas.demarchi@profusion.mobi>
diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index fd2188c..58804c7 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -170,7 +170,7 @@
/* 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 917e405..150cb7e 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -253,7 +253,7 @@
* 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 @@
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 fe5c8036..1a07750 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -460,7 +460,7 @@
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 f3bdc54..1d7d90c 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 a89948a..a39d3d8 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -233,7 +233,7 @@
* 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 3c93c23..e73fafd 100644
--- a/sound/drivers/pcm-indirect2.c
+++ b/sound/drivers/pcm-indirect2.c
@@ -264,7 +264,7 @@
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 @@
/* 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 35a2f71..5e897b2 100644
--- a/sound/drivers/vx/vx_pcm.c
+++ b/sound/drivers/vx/vx_pcm.c
@@ -1189,7 +1189,7 @@
/*
- * 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 0c40951..5d61f5a2 100644
--- a/sound/isa/sb/emu8000.c
+++ b/sound/isa/sb/emu8000.c
@@ -370,7 +370,7 @@
/*
* 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 f14a7c0..65329f3 100644
--- a/sound/isa/wavefront/wavefront_midi.c
+++ b/sound/isa/wavefront/wavefront_midi.c
@@ -537,7 +537,7 @@
}
/* 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 9191b32..2a42cc3 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -424,7 +424,7 @@
/*
* 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 854c303..0cd23d9 100644
--- a/sound/oss/ac97_codec.c
+++ b/sound/oss/ac97_codec.c
@@ -28,7 +28,7 @@
*
* History
* May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
- * Removed non existant WM9700
+ * Removed non existent WM9700
* Added support for WM9705, WM9708, WM9709, WM9710, WM9711
* WM9712 and WM9717
* Mar 28, 2002 Randolph Bentson <bentson@holmsjoen.com>
@@ -441,7 +441,7 @@
}
/* 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 @@
if((codec->codec_ops == &null_ops) && (f & 4))
codec->codec_ops = &default_digital_ops;
- /* A device which thinks its a modem but isnt */
+ /* A device which thinks its a modem but isn't */
if(codec->flags & AC97_DELUDED_MODEM)
codec->modem = 0;
diff --git a/sound/oss/audio.c b/sound/oss/audio.c
index 7df48a2..4b958b1c 100644
--- a/sound/oss/audio.c
+++ b/sound/oss/audio.c
@@ -514,7 +514,7 @@
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 @@
if (count < dmap_out->fragment_size && dmap_out->qhead != 0)
count += dmap_out->bytes_in_use; /* Pointer wrap not handled yet */
count += dmap_out->byte_counter;
- /* Substract current count from the number of bytes written by app */
+ /* Subtract current count from the number of bytes written by app */
count = dmap_out->user_counter - count;
if (count < 0)
count = 0;
diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c
index 87e2c72..c918313 100644
--- a/sound/oss/dmasound/dmasound_core.c
+++ b/sound/oss/dmasound/dmasound_core.c
@@ -1021,7 +1021,7 @@
case SNDCTL_DSP_SYNC:
/* This call, effectively, has the same behaviour as SNDCTL_DSP_RESET
except that it waits for output to finish before resetting
- everything - read, however, is killed imediately.
+ everything - read, however, is killed immediately.
*/
result = 0 ;
if (file->f_mode & FMODE_WRITE) {
diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c
index ceedb1e..8cdb2cf 100644
--- a/sound/oss/midibuf.c
+++ b/sound/oss/midibuf.c
@@ -295,7 +295,7 @@
for (i = 0; i < n; i++)
{
- /* BROKE BROKE BROKE - CANT DO THIS WITH CLI !! */
+ /* BROKE BROKE BROKE - CAN'T DO THIS WITH CLI !! */
/* yes, think the same, so I removed the cli() brackets
QUEUE_BYTE is protected against interrupts */
if (copy_from_user((char *) &tmp_data, &(buf)[c], 1)) {
diff --git a/sound/oss/sb_card.c b/sound/oss/sb_card.c
index 84ef4d0..fb5d725 100644
--- a/sound/oss/sb_card.c
+++ b/sound/oss/sb_card.c
@@ -1,7 +1,7 @@
/*
* sound/oss/sb_card.c
*
- * Detection routine for the ISA Sound Blaster and compatable sound
+ * Detection routine for the ISA Sound Blaster and compatible sound
* cards.
*
* This file is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c
index 9890cf2..5c773df 100644
--- a/sound/oss/sb_ess.c
+++ b/sound/oss/sb_ess.c
@@ -168,7 +168,7 @@
* corresponding playback levels, unless recmask says they aren't recorded. In
* the latter case the recording volumes are 0.
* Now recording levels of inputs can be controlled, by changing the playback
- * levels. Futhermore several devices can be recorded together (which is not
+ * levels. Furthermore several devices can be recorded together (which is not
* possible with the ES1688).
* Besides the separate recording level control for each input, the common
* recording level can also be controlled by RECLEV as described above.
diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c
index 44357d8..09d4648 100644
--- a/sound/oss/swarm_cs4297a.c
+++ b/sound/oss/swarm_cs4297a.c
@@ -875,7 +875,7 @@
if (s->prop_adc.fmt & AFMT_S8 || s->prop_adc.fmt & AFMT_U8) {
//
// now only use 16 bit capture, due to truncation issue
- // in the chip, noticable distortion occurs.
+ // in the chip, noticeable distortion occurs.
// allocate buffer and then convert from 16 bit to
// 8 bit for the user buffer.
//
diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c
index f0e0caa..12ba28e 100644
--- a/sound/oss/vidc.c
+++ b/sound/oss/vidc.c
@@ -227,7 +227,7 @@
} 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 4382d0f..d8f6fd6 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 f53a31e..f8ccc967 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -963,7 +963,7 @@
/*? 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 6fc025c..255429c 100644
--- a/sound/pci/asihpi/hpi.h
+++ b/sound/pci/asihpi/hpi.h
@@ -725,7 +725,7 @@
#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 3e3c2ef..8c8aac4 100644
--- a/sound/pci/asihpi/hpi6000.c
+++ b/sound/pci/asihpi/hpi6000.c
@@ -423,7 +423,7 @@
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 620525b..22e9f08 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -466,7 +466,7 @@
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 af678be..3b9fd11 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -607,7 +607,7 @@
#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 bcbdf30..360028b 100644
--- a/sound/pci/asihpi/hpimsgx.c
+++ b/sound/pci/asihpi/hpimsgx.c
@@ -722,7 +722,7 @@
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 ecb8f4d..02f6e08 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 f4aa8ff6..9ae8b3b 100644
--- a/sound/pci/au88x0/au88x0_a3d.c
+++ b/sound/pci/au88x0/au88x0_a3d.c
@@ -53,7 +53,7 @@
}
#endif
-/* Atmospheric absorbtion. */
+/* Atmospheric absorption. */
static void
a3dsrc_SetAtmosTarget(a3dsrc_t * a, short aa, short b, short c, short d,
@@ -835,7 +835,7 @@
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 5439d66..33f0ba5 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -515,7 +515,7 @@
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 5715c4d0..9b7a634 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 fc53b9b..e8e8ccc 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 01b4938..4377592 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 630aa49..84f3f92 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 ba96428..c694464b 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 b5bb036..f4e5735 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -73,7 +73,7 @@
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 @@
}
/*
- * 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 @@
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 b932154..13f33c0 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1627,7 +1627,7 @@
* 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 0cf400f..a5c957d 100644
--- a/sound/pci/ctxfi/cthw20k1.c
+++ b/sound/pci/ctxfi/cthw20k1.c
@@ -1285,7 +1285,7 @@
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 957a311..c250614 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -248,7 +248,7 @@
/*
* 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 61b8ab3..a81dc44 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 00f4817..4e0ee1a 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/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 2c79e96..430f41d 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3661,7 +3661,7 @@
* 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 @@
*
* 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_realtek.c b/sound/pci/hda/patch_realtek.c
index 0ef0035..1e5a786 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -549,7 +549,7 @@
/*
* 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 @@
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),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 05fcd60..1395991 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2475,7 +2475,7 @@
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);
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 2f62522..3e4f8c1 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -148,7 +148,7 @@
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 @@
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 4fc6d8b..f4594d7 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2755,7 +2755,7 @@
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 @@
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 cdb873f..92c1160 100644
--- a/sound/pci/ice1712/pontis.c
+++ b/sound/pci/ice1712/pontis.c
@@ -768,7 +768,7 @@
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 6a9fee3..764cc93d 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -1046,7 +1046,7 @@
* 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 @@
* 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 629a549..6c896db 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -534,7 +534,7 @@
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 2ae8d29..27709f0 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -331,7 +331,7 @@
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 d3350f3..3df0f53 100644
--- a/sound/pci/mixart/mixart_core.c
+++ b/sound/pci/mixart/mixart_core.c
@@ -265,7 +265,7 @@
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 @@
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 833e718..304411c 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -1042,11 +1042,11 @@
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 @@
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 @@
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 @@
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 d5f5b44..9ff247f 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -150,7 +150,7 @@
#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 a323eaf..949691a 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -391,7 +391,7 @@
/* 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 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 1b8f674..2b5c7a95 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -308,7 +308,7 @@
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 @@
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 @@
*/
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 edce8a2..bc823a5 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -358,7 +358,7 @@
* 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 03fdee4..2e63020 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 @@
/*
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 @@
/*
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 @@
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 @@
/*
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 5d230ce..7fbfa05 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -672,7 +672,7 @@
/* 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 4f377c9..eecffb5 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -481,7 +481,7 @@
};
/* 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 72de47e..2c2a681 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -161,7 +161,7 @@
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/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h
index 62b1f22..67f19c3 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 3bedab2..6c43c13 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -884,7 +884,7 @@
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 00b6d87..f01f141 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1020,7 +1020,7 @@
/*
* 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 8512800..575238d 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -281,7 +281,7 @@
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 @@
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 @@
}
/* 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 8f6b5ee..4bbc0a7 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -772,7 +772,7 @@
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 3f09dee..ffa2ffe 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1312,7 +1312,7 @@
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 443ae58..9b3bba4 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1895,7 +1895,7 @@
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 5e0214d..3c71987 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -176,7 +176,7 @@
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 3b71dd6..500011e 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3137,7 +3137,7 @@
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 28fdfd6..3c2ee1b 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -981,7 +981,7 @@
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 379fa22..056aef9 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -324,7 +324,7 @@
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/wm8994.c b/sound/soc/codecs/wm8994.c
index 3dc64c8..3290333 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -82,18 +82,18 @@
int mbc_ena[3];
- /* Platform dependant DRC configuration */
+ /* Platform dependent DRC configuration */
const char **drc_texts;
int drc_cfg[WM8994_NUM_DRC];
struct soc_enum drc_enum;
- /* Platform dependant ReTune mobile configuration */
+ /* Platform dependent ReTune mobile configuration */
int num_retune_mobile_texts;
const char **retune_mobile_texts;
int retune_mobile_cfg[WM8994_NUM_EQ];
struct soc_enum retune_mobile_enum;
- /* Platform dependant MBC configuration */
+ /* Platform dependent MBC configuration */
int mbc_cfg;
const char **mbc_texts;
struct soc_enum mbc_enum;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 55cdf29..91c6b39 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -305,7 +305,7 @@
/*
* 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 @@
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 bc92ec6..ac2ded9 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 0fd6a63..e13c6ce 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -132,7 +132,7 @@
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 @@
/*
* 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 ee2c22475..b2e9198 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -440,7 +440,7 @@
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;
}
@@ -463,7 +463,7 @@
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 3167be6..462cbcb 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -248,7 +248,7 @@
*/
/* 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 @@
/*
- * 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 78bfdb3..4522309 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -228,7 +228,7 @@
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 @@
return 0;
}
-/* GTA01 specific controlls */
+/* GTA01 specific controls */
#ifdef CONFIG_MACH_NEO1973_GTA01
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index 9081a54..86c1a31 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -76,7 +76,7 @@
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 @@
/*
* 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/mixer.c b/sound/usb/mixer.c
index 5e47757..6ec33b6 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1182,7 +1182,7 @@
/*
* 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 355759b..ec07e62 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -266,7 +266,7 @@
* 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 287ef73..a51340f 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
