Fixed many typos in output and source code comments

libdisplay/display_wave.c

@@ -120,7 +120,7 @@ void display_wave(dispwav_t *disp, sample_t *samples, int length, double range)
 		if (pos == width + 2) {
 			memset(&screen, ' ', sizeof(screen));
 			for (j = 0; j < width; j++) {
-				/* Input value is scaled to range -1 .. 1 and then substracted from 1,
+				/* Input value is scaled to range -1 .. 1 and then subtracted from 1,
 				 * so the result ranges from 0 .. 2.
 				 * HEIGHT-1 is multiplied with the range, so a HEIGHT of 3 would allow
 				 * 0..4 (5 steps) and a HEIGHT of 11 would allow 0..20 (21 steps).

libfsk/fsk.c

@@ -157,7 +157,7 @@ void fsk_receive(fsk_t *fsk, sample_t *sample, int length)
 	int bit;
 	double level, quality;
 
-	/* demod samples to offset arround center frequency */
+	/* demod samples to offset around center frequency */
 	fm_demodulate_real(&fsk->demod, frequency, length, sample, I, Q);
 
 	for (i = 0; i < length; i++) {
@@ -187,7 +187,7 @@ void fsk_receive(fsk_t *fsk, sample_t *sample, int length)
 					fsk->rx_bitpos = 0.5;
 			}
 		}
-		/* if bit counter reaches 1, we substract 1 and sample the bit */
+		/* if bit counter reaches 1, we subtract 1 and sample the bit */
 		if (fsk->rx_bitpos >= 1.0) {
 			/* peak level is the length of I/Q vector
 			 * since we filter out the unwanted modulation product, the vector is only half of length */

libgoertzel/goertzel.c

@@ -68,7 +68,7 @@ void audio_goertzel_init(goertzel_t *goertzel, double freq, int samplerate)
  *
  * samples: pointer to sample buffer
  * length: length of buffer
- * offset: for ring buffer, start here and wrap arround to 0 when length has been hit
+ * offset: for ring buffer, start here and wrap around to 0 when length has been hit
  * coeff: array of coefficients (coeff << 15)
  * result: array of result levels (average value of the sine, that is 1 / (PI/2) of the sine's peak)
  * k: number of frequencies to check

libsample/sample.c

@@ -30,14 +30,14 @@ static double int_16_speech_level = SPEECH_LEVEL * 0.7079; /* 16 dBm below dBm0,
  * support high numbers. 'double' or 'float' types are sufficient.
  *
  * When using sample_t inside signal processing of each base station, the
- * level of +- 1 is relative to the normal speach evenlope.
+ * level of +- 1 is relative to the normal speech evenlope.
  *
  * When converting sample_t to int16_t, the level of +- 1 is reduced by factor.
  * This way the speech may be louder before clipping happens.
  *
  * When using sample_t to modulate (SDR or sound card), the level is changed,
  * so it represents the frequency deviation in Hz. The deviation of speech
- * envelope is network dependant.
+ * envelope is network dependent.
  */
 
 void samples_to_int16(int16_t *spl, sample_t *samples, int length)

libsdr/sdr.c

@@ -967,7 +967,7 @@ int sdr_get_tosend(void *inst, int latspl)
 	count /= sdr->oversample;
 
 	if (sdr->threads) {
-		/* substract what we have in write buffer, because this is not jent sent to the SDR */
+		/* subtract what we have in write buffer, because this is not jent sent to the SDR */
 		int fill;
 
 		fill = (sdr->thread_write.in - sdr->thread_write.out + sdr->thread_write.buffer_size) % sdr->thread_write.buffer_size;

libsdr/sdr_config.c

@@ -62,7 +62,7 @@ void sdr_config_print_help(void)
 	printf("    --sdr-device-args <args>\n");
 	printf("    --sdr-stream-args <args>\n");
 	printf("    --sdr-tune-args <args>\n");
-	printf("        Optional SDR device arguments, seperated by comma\n");
+	printf("        Optional SDR device arguments, separated by comma\n");
 	printf("        e.g. --sdr-device-args <key>=<value>[,<key>=<value>[,...]]\n");
 	printf("    --sdr-samplerate <samplerate>\n");
 	printf("        Sample rate to use with SDR. By default it equals the regular sample\n");

libsdr/soapy.c

@@ -501,7 +501,7 @@ int soapy_get_tosend(int latspl)
 	tosend = latspl - (tx_count - rx_count);
 	/* in case of underrun: */
 	if (tosend > latspl) {
-// It is normal that we have underruns, prior inital filling of buffer.
+// It is normal that we have underruns, prior initial filling of buffer.
 // FIXME: better solution to detect underrun
 //		PDEBUG(DSOAPY, DEBUG_ERROR, "SDR TX underrun!\n");
 		tosend = 0;

libsquelch/squelch.c

@@ -104,7 +104,7 @@ enum squelch_result squelch(squelch_t *squelch, double rf_level_db, double durat
 		}
 	}
 
-	/* enough RF level, so we unmute when mute_count reched 0 */
+	/* enough RF level, so we unmute when mute_count reached 0 */
 	if (rf_level_db >= squelch->threshold_db) {
 		squelch->mute_count -= duration;
 		if (squelch->mute_count <= 0.0) {