Merge branch 'fixes' of git://git.alsa-project.org/alsa-kernel into for-linus

9 files changed, 14 insertions, 13 deletions

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 20cb60afb20..c1192062300 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -2122,7 +2122,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
 		}
 		/* nothing should be in powerdown mode */
 		snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0);
-		end_time = jiffies + msecs_to_jiffies(120);
+		end_time = jiffies + msecs_to_jiffies(5000);
 		do {
 			if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f)
 				goto __ready_ok;
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index 15523e60351..0470461cc03 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 09d24c76f03..a65e1193c89 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 6b8ae7b5cd5..1d369ff7380 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 2439e84dcb2..4b200da1bd1 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 85c81feb10c..a45c1169762 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 deecdd2d5d3..888b6313eec 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6621,7 +6621,7 @@ 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 0c9413d5341..98bc3b7681b 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 0dce331a2a3..a1b10d1a384 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);