ALSA: usb-audio: Fix some typos

Fix the following typos in comments and in the code:
 - KHz -> kHz
 - procssed -> processed
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200629032607.255419-1-alexander@tsoy.meSigned-off-by: Takashi Iwai <tiwai@suse.de>

sound/usb/mixer_s1810c.c

@@ -554,11 +554,11 @@ int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer)
 	dev_info(&dev->dev,
 		 "Presonus Studio 1810c, device_setup: %u\n", chip->setup);
 	if (chip->setup == 1)
-		dev_info(&dev->dev, "(8out/18in @ 48KHz)\n");
+		dev_info(&dev->dev, "(8out/18in @ 48kHz)\n");
 	else if (chip->setup == 2)
-		dev_info(&dev->dev, "(6out/8in @ 192KHz)\n");
+		dev_info(&dev->dev, "(6out/8in @ 192kHz)\n");
 	else
-		dev_info(&dev->dev, "(8out/14in @ 96KHz)\n");
+		dev_info(&dev->dev, "(8out/14in @ 96kHz)\n");
 
 	ret = snd_s1810c_init_mixer_maps(chip);
 	if (ret < 0)

sound/usb/mixer_scarlett_gen2.c

@@ -332,7 +332,7 @@ static const struct scarlett2_device_info s18i8_gen2_info = {
 		},
 		[SCARLETT2_PORT_TYPE_SPDIF] = {
 			.id = 0x180,
-			/* S/PDIF outputs aren't available at 192KHz
+			/* S/PDIF outputs aren't available at 192kHz
 			 * but are included in the USB mux I/O
 			 * assignment message anyway
 			 */
@@ -401,7 +401,7 @@ static const struct scarlett2_device_info s18i20_gen2_info = {
 			.dst_descr = "Analogue Output %02d Playback"
 		},
 		[SCARLETT2_PORT_TYPE_SPDIF] = {
-			/* S/PDIF outputs aren't available at 192KHz
+			/* S/PDIF outputs aren't available at 192kHz
 			 * but are included in the USB mux I/O
 			 * assignment message anyway
 			 */

sound/usb/pcm.c

@@ -1694,8 +1694,8 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
 	int processed = urb->transfer_buffer_length / ep->stride;
 	int est_delay;
 
-	/* ignore the delay accounting when procssed=0 is given, i.e.
-	 * silent payloads are procssed before handling the actual data
+	/* ignore the delay accounting when processed=0 is given, i.e.
+	 * silent payloads are processed before handling the actual data
 	 */
 	if (!processed)
 		return;

sound/usb/quirks.c

@@ -1144,14 +1144,14 @@ static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev)
 #define MAUDIO_SET		0x01 /* parse device_setup */
 #define MAUDIO_SET_COMPATIBLE	0x80 /* use only "win-compatible" interfaces */
 #define MAUDIO_SET_DTS		0x02 /* enable DTS Digital Output */
-#define MAUDIO_SET_96K		0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */
+#define MAUDIO_SET_96K		0x04 /* 48-96kHz rate if set, 8-48kHz otherwise */
 #define MAUDIO_SET_24B		0x08 /* 24bits sample if set, 16bits otherwise */
 #define MAUDIO_SET_DI		0x10 /* enable Digital Input */
 #define MAUDIO_SET_MASK		0x1f /* bit mask for setup value */
-#define MAUDIO_SET_24B_48K_DI	 0x19 /* 24bits+48KHz+Digital Input */
-#define MAUDIO_SET_24B_48K_NOTDI 0x09 /* 24bits+48KHz+No Digital Input */
-#define MAUDIO_SET_16B_48K_DI	 0x11 /* 16bits+48KHz+Digital Input */
-#define MAUDIO_SET_16B_48K_NOTDI 0x01 /* 16bits+48KHz+No Digital Input */
+#define MAUDIO_SET_24B_48K_DI	 0x19 /* 24bits+48kHz+Digital Input */
+#define MAUDIO_SET_24B_48K_NOTDI 0x09 /* 24bits+48kHz+No Digital Input */
+#define MAUDIO_SET_16B_48K_DI	 0x11 /* 16bits+48kHz+Digital Input */
+#define MAUDIO_SET_16B_48K_NOTDI 0x01 /* 16bits+48kHz+No Digital Input */
 
 static int quattro_skip_setting_quirk(struct snd_usb_audio *chip,
 				      int iface, int altno)
@@ -1597,7 +1597,7 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep)
 
 	/*
 	 * M-Audio Fast Track C400/C600 - when packets are not skipped, real
-	 * world latency varies by approx. +/- 50 frames (at 96KHz) each time
+	 * world latency varies by approx. +/- 50 frames (at 96kHz) each time
 	 * the stream is (re)started. When skipping packets 16 at endpoint
 	 * start up, the real world latency is stable within +/- 1 frame (also
 	 * across power cycles).
@@ -1831,7 +1831,7 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip,
 		/*
 		 * MaxPacketsOnly attribute is erroneously set in endpoint
 		 * descriptors. As a result this card produces noise with
-		 * all sample rates other than 96 KHz.
+		 * all sample rates other than 96 kHz.
 		 */
 		fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX;
 		break;