Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (42 commits)
  tree-wide: fix misspelling of "definition" in comments
  reiserfs: fix misspelling of "journaled"
  doc: Fix a typo in slub.txt.
  inotify: remove superfluous return code check
  hdlc: spelling fix in find_pvc() comment
  doc: fix regulator docs cut-and-pasteism
  mtd: Fix comment in Kconfig
  doc: Fix IRQ chip docs
  tree-wide: fix assorted typos all over the place
  drivers/ata/libata-sff.c: comment spelling fixes
  fix typos/grammos in Documentation/edac.txt
  sysctl: add missing comments
  fs/debugfs/inode.c: fix comment typos
  sgivwfb: Make use of ARRAY_SIZE.
  sky2: fix sky2_link_down copy/paste comment error
  tree-wide: fix typos "couter" -> "counter"
  tree-wide: fix typos "offest" -> "offset"
  fix kerneldoc for set_irq_msi()
  spidev: fix double "of of" in comment
  comment typo fix: sybsystem -> subsystem
  ...

21 files changed, 28 insertions, 33 deletions

diff --git a/sound/Kconfig b/sound/Kconfig
index b3e53e616ec9..fcad760f5691 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -1,6 +1,3 @@
-# sound/Config.in
-#
-
 menuconfig SOUND
         tristate "Sound card support"
         depends on HAS_IOMEM
@@ -136,4 +133,3 @@ config AC97_BUS
           sound subsystem and other function drivers completely unrelated to
           sound although they're sharing the AC97 bus. Concerned drivers
           should "select" this.
-
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index 93fa6720d197..cc15d1d65a22 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -177,7 +177,7 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = {
         { .id = "CSC0437", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
         /* Digital PC 5000 Onboard - CS4236B */
         { .id = "CSC0735", .devs = { { "CSC0000" }, { "CSC0010" } } },
-        /* some uknown CS4236B */
+        /* some unknown CS4236B */
         { .id = "CSC0b35", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
         /* Intel PR440FX Onboard sound */
         { .id = "CSC0b36", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
index 6123c7531110..b865e45a8f9b 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -133,7 +133,7 @@ struct snd_miro {
 static struct snd_miro_aci aci_device;
 
 static char * snd_opti9xx_names[] = {
-        "unkown",
+        "unknown",
         "82C928", "82C929",
         "82C924", "82C925",
         "82C930", "82C931", "82C933"
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index d08c38906449..d622d2906862 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -177,7 +177,7 @@ MODULE_DEVICE_TABLE(pnp_card, snd_opti9xx_pnpids);
 #endif
 
 static char * snd_opti9xx_names[] = {
-        "unkown",
+        "unknown",
         "82C928",       "82C929",
         "82C924",       "82C925",
         "82C930",       "82C931",       "82C933"
diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig
index 135a2b77cc4a..a513651fa149 100644
--- a/sound/oss/Kconfig
+++ b/sound/oss/Kconfig
@@ -1,5 +1,3 @@
-# drivers/sound/Config.in
-#
 # 18 Apr 1998, Michael Elizabeth Chastain, <mailto:mec@shout.net>
 # More hacking for modularisation.
 #
diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c
index 06e9e88e4c05..bb14e4c67e89 100644
--- a/sound/oss/dmasound/dmasound_paula.c
+++ b/sound/oss/dmasound/dmasound_paula.c
@@ -657,7 +657,7 @@ static int AmiStateInfo(char *buffer, size_t space)
         len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n",
                        dmasound.volume_right);
         if (len >= space) {
-                printk(KERN_ERR "dmasound_paula: overlowed state buffer alloc.\n") ;
+                printk(KERN_ERR "dmasound_paula: overflowed state buffer alloc.\n") ;
                 len = space ;
         }
         return len;
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index 15523e60351c..0470461cc03e 100644
--- a/sound/pci/ca0106/ca0106_proc.c
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -233,7 +233,7 @@ static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 val
                         snd_iprintf(buffer, "user-defined\n");
                         break;
                 default:
-                        snd_iprintf(buffer, "unkown\n");
+                        snd_iprintf(buffer, "unknown\n");
                         break;
                 }
                 snd_iprintf(buffer, "Sample Bits: ");
diff --git a/sound/pci/cs46xx/imgs/cwcdma.asp b/sound/pci/cs46xx/imgs/cwcdma.asp
index 09d24c76f034..a65e1193c89a 100644
--- a/sound/pci/cs46xx/imgs/cwcdma.asp
+++ b/sound/pci/cs46xx/imgs/cwcdma.asp
@@ -26,10 +26,11 @@
 //
 //
 // The purpose of this code is very simple: make it possible to tranfser
-// the samples 'as they are' with no alteration from a PCMreader SCB (DMA from host)
-// to any other SCB. This is useful for AC3 throug SPDIF. SRC (source rate converters) 
-// task always alters the samples in some how, however it's from 48khz -> 48khz. The
-// alterations are not audible, but AC3 wont work. 
+// the samples 'as they are' with no alteration from a PCMreader
+// SCB (DMA from host) to any other SCB. This is useful for AC3 through SPDIF.
+// SRC (source rate converters) task always alters the samples in somehow,
+// however it's from 48khz -> 48khz.
+// The alterations are not audible, but AC3 wont work. 
 //
 //        ...
 //         |
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 6b8ae7b5cd54..1d369ff73805 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -184,7 +184,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard.");
  * The hardware has 3 channels for playback and 1 for capture.
  *  - channel 0 is the front channel
  *  - channel 1 is the rear channel
- *  - channel 2 is the center/lfe chanel
+ *  - channel 2 is the center/lfe channel
  * Volume is controlled by the AC97 for the front and rear channels by
  * the PCM Playback Volume, Sigmatel Surround Playback Volume and 
  * Surround Playback Volume. The Sigmatel 4-Speaker Stereo switch affects
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 2439e84dcb21..4b200da1bd18 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -938,7 +938,7 @@ static void init_input(struct hda_codec *codec)
                 coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */
         if (is_active_pin(codec, CS_DMIC1_PIN_NID))
                 coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 
-                                 * No effect if SPDIF_OUT2 is slected in 
+                                 * No effect if SPDIF_OUT2 is selected in 
                                  * IDX_SPDIF_CTL.
                                   */
         cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 85c81feb10cf..a45c1169762b 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -66,7 +66,7 @@ struct cmi_spec {
 
         struct hda_pcm pcm_rec[2];      /* PCM information */
 
-        /* pin deafault configuration */
+        /* pin default configuration */
         hda_nid_t pin_nid[NUM_PINS];
         unsigned int def_conf[NUM_PINS];
         unsigned int pin_def_confs;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d967836f36bb..db4a7367b625 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6618,7 +6618,7 @@ static struct hda_input_mux alc889A_mb31_capture_source = {
                 /* Front Mic (0x01) unused */
                 { "Line", 0x2 },
                 /* Line 2 (0x03) unused */
-                /* CD (0x04) unsused? */
+                /* CD (0x04) unused? */
         },
 };
 
diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c
index 0c9413d5341b..98bc3b7681b5 100644
--- a/sound/pci/ice1712/juli.c
+++ b/sound/pci/ice1712/juli.c
@@ -380,7 +380,7 @@ static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = {
          * inputs) are fed from Xilinx.
          *
          * I even checked traces on board and coded a support in driver for
-         * an alternative possiblity - the unused I2S ICE output channels
+         * an alternative possibility - the unused I2S ICE output channels
          * switched to HW-IN/SPDIF-IN and providing the monitoring signal to
          * the DAC - to no avail. The I2S outputs seem to be unconnected.
          *
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 0dce331a2a3b..a1b10d1a384d 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -3017,7 +3017,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
                 insel = "Coaxial";
                 break;
         default:
-                insel = "Unkown";
+                insel = "Unknown";
         }
 
         switch (hdspm->control_register & HDSPM_SyncRefMask) {
@@ -3028,7 +3028,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
                 syncref = "MADI";
                 break;
         default:
-                syncref = "Unkown";
+                syncref = "Unknown";
         }
         snd_iprintf(buffer, "Inputsel = %s, SyncRef = %s\n", insel,
                     syncref);
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index aa40d985138f..3e99fe5131dd 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
         pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
 
         if (reg >= UDA134X_REGS_NUM) {
-                printk(KERN_ERR "%s unkown register: reg: %u",
+                printk(KERN_ERR "%s unknown register: reg: %u",
                        __func__, reg);
                 return -EINVAL;
         }
@@ -552,7 +552,7 @@ static int uda134x_soc_probe(struct platform_device *pdev)
                                         ARRAY_SIZE(uda1341_snd_controls));
         break;
         default:
-                printk(KERN_ERR "%s unkown codec type: %d",
+                printk(KERN_ERR "%s unknown codec type: %d",
                         __func__, pd->model);
         return -EINVAL;
         }
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index b8cae1758642..ce5515e3f2b0 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -607,7 +607,7 @@ SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1,
 SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0),
 SOC_ENUM("DRC Compressor Slope R0", drc_slope_r0),
 SOC_ENUM("DRC Compressor Slope R1", drc_slope_r1),
-SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8903_DRC_3, 5, 124, 1,
+SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8903_DRC_3, 5, 124, 1,
                drc_tlv_thresh),
 SOC_SINGLE_TLV("DRC Volume", WM8903_DRC_3, 0, 30, 1, drc_tlv_amp),
 SOC_SINGLE_TLV("DRC Minimum Gain Volume", WM8903_DRC_1, 2, 3, 1, drc_tlv_min),
@@ -617,11 +617,11 @@ SOC_ENUM("DRC Decay Rate", drc_decay),
 SOC_ENUM("DRC FF Delay", drc_ff_delay),
 SOC_SINGLE("DRC Anticlip Switch", WM8903_DRC_0, 1, 1, 0),
 SOC_SINGLE("DRC QR Switch", WM8903_DRC_0, 2, 1, 0),
-SOC_SINGLE_TLV("DRC QR Threashold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max),
+SOC_SINGLE_TLV("DRC QR Threshold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max),
 SOC_ENUM("DRC QR Decay Rate", drc_qr_decay),
 SOC_SINGLE("DRC Smoothing Switch", WM8903_DRC_0, 3, 1, 0),
 SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8903_DRC_0, 0, 1, 0),
-SOC_ENUM("DRC Smoothing Threashold", drc_smoothing),
+SOC_ENUM("DRC Smoothing Threshold", drc_smoothing),
 SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup),
 
 SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT,
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 5e32f2ed5fc2..2981afae842c 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -689,7 +689,7 @@ SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8993_DIGITAL_SIDE_TONE,
 
 SOC_SINGLE("DRC Switch", WM8993_DRC_CONTROL_1, 15, 1, 0),
 SOC_ENUM("DRC Path", drc_path),
-SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8993_DRC_CONTROL_2,
+SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8993_DRC_CONTROL_2,
                2, 60, 1, drc_comp_threash),
 SOC_SINGLE_TLV("DRC Compressor Amplitude Volume", WM8993_DRC_CONTROL_3,
                11, 30, 1, drc_comp_amp),
@@ -709,7 +709,7 @@ SOC_SINGLE_TLV("DRC Quick Release Volume", WM8993_DRC_CONTROL_3, 2, 3, 0,
 SOC_ENUM("DRC Quick Release Rate", drc_qr_rate),
 SOC_SINGLE("DRC Smoothing Switch", WM8993_DRC_CONTROL_1, 11, 1, 0),
 SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8993_DRC_CONTROL_1, 8, 1, 0),
-SOC_ENUM("DRC Smoothing Hysteresis Threashold", drc_smooth),
+SOC_ENUM("DRC Smoothing Hysteresis Threshold", drc_smooth),
 SOC_SINGLE_TLV("DRC Startup Volume", WM8993_DRC_CONTROL_4, 8, 18, 0,
                drc_startup_tlv),
 
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c
index 507b2ed5d58b..d441c3b64631 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec.c
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.c
@@ -270,7 +270,7 @@ static int attach_gpio_amp(struct device *dev,
                 gpio_direction_output(pd->amp_gain[1], 0);
         }
 
-        /* note, curently we assume GPA0 isn't valid amp */
+        /* note, currently we assume GPA0 isn't valid amp */
         if (pdata->amp_gpio > 0) {
                 ret = gpio_request(pd->amp_gpio, "gpio-amp");
                 if (ret) {
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 0eb1722f6581..1d61109e09fa 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -196,7 +196,7 @@ static int s6000_pcm_start(struct snd_pcm_substream *substream)
                            0 /* destination skip after chunk (impossible) */,
                            4 /* 16 byte burst size */,
                            -1 /* don't conserve bandwidth */,
-                           0 /* low watermark irq descriptor theshold */,
+                           0 /* low watermark irq descriptor threshold */,
                            0 /* disable hardware timestamps */,
                            1 /* enable channel */);
 
diff --git a/sound/sound_core.c b/sound/sound_core.c
index 49c998186592..dbca7c909a31 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -353,7 +353,7 @@ static struct sound_unit *chains[SOUND_STEP];
  *      @dev: device pointer
  *
  *      Allocate a special sound device by minor number from the sound
- *      subsystem. The allocated number is returned on succes. On failure
+ *      subsystem. The allocated number is returned on success. On failure
  *      a negative error code is returned.
  */
  
diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c
index 63c8f45c0c22..67c91230c197 100644
--- a/sound/synth/emux/soundfont.c
+++ b/sound/synth/emux/soundfont.c
@@ -374,7 +374,7 @@ sf_zone_new(struct snd_sf_list *sflist, struct snd_soundfont *sf)
 
 
 /*
- * increment sample couter
+ * increment sample counter
  */
 static void
 set_sample_counter(struct snd_sf_list *sflist, struct snd_soundfont *sf,