Merge branch 'for-next' into for-linus

Conflicts:

	kernel/irq/chip.c

21 files changed, 29 insertions, 34 deletions

diff --git a/sound/Kconfig b/sound/Kconfig
index 439e15c..fcad760f 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -1,6 +1,3 @@
-# sound/Config.in
-#
-
 menuconfig SOUND
 	tristate "Sound card support"
 	depends on HAS_IOMEM
@@ -58,7 +55,7 @@ config SOUND_OSS_CORE_PRECLAIM
 	  Please read Documentation/feature-removal-schedule.txt for
 	  details.
 
-	  If unusre, say Y.
+	  If unsure, say Y.
 
 source "sound/oss/dmasound/Kconfig"
 
@@ -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 a076a6c..a828baa 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 02e30d7..ddad60e 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -137,7 +137,7 @@ struct snd_miro {
 static void snd_miro_proc_init(struct snd_miro * miro);
 
 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 5cd5553..8480075 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -185,7 +185,7 @@ 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 bcf2a06..ea0b1ae 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 06e9e88..bb14e4c 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 c62b7d1..8d13092 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 09d24c7..a65e119 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 36e08bd..360e380 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 8ba3068..7b0446f 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -947,7 +947,7 @@ 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 780e1a7..8917071 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 7058371..f7d5657 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6619,7 +6619,7 @@ 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 fd948bf..9c0f78e 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 0dce331..a1b10d1 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 c33b92e..8ce1c9b 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 fe1307b..d72347d 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 d998799..bc03368 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 1966e0d..3c7ccb7 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 83b8028..81d6f98 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 49c9981..dbca7c9 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 63c8f45..67c9123 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,