Fixed many typos in output and source code comments

src/amps/amps.c

@@ -184,7 +184,7 @@ void amps_number2min(const char *number, uint32_t *min1, uint16_t *min2)
 	int i;
 
 	if (nlen != 10) {
-		fprintf(stderr, "illegal lenght %d. Must be 10, aborting!", nlen);
+		fprintf(stderr, "illegal length %d. Must be 10, aborting!", nlen);
 		abort();
 	}
 
@@ -580,7 +580,7 @@ int amps_create(int channel, enum amps_chan_type chan_type, const char *audiodev
 
 	amps->pre_emphasis = pre_emphasis;
 	amps->de_emphasis = de_emphasis;
-	/* the AMPS uses a frequency rage of 300..3000 Hz, but we still use the default low pass filter, wich is not too far above */
+	/* the AMPS uses a frequency rage of 300..3000 Hz, but we still use the default low pass filter, which is not too far above */
 	rc = init_emphasis(&amps->estate, samplerate, CUT_OFF_EMPHASIS_DEFAULT, CUT_OFF_HIGHPASS_DEFAULT, CUT_OFF_LOWPASS_DEFAULT);
 	if (rc < 0)
 		goto error;

src/amps/dsp.c

@@ -39,7 +39,7 @@
  * The average level change offsets of the dotting sequence is used to set the
  * window for the first bit.  When all samples for the window are received, a
  * raise in level is detected as 1, fall in level is detected as 0. This is done
- * by substracting the average sample value of the left side of the window by
+ * by subtracting the average sample value of the left side of the window by
  * the average sample value of the right side.  After the bit has been detected,
  * the samples for the next window will be received and detected.
  *
@@ -108,7 +108,7 @@
 #define AMPS_BITRATE		10000
 /* for some reason, 4000 Hz deviation works better */
 #define TACS_DBM0_DEVIATION	4000.0  /* 2300 Hz deviation at 1 kHz (according to panasonic manual) */
-#define TACS_MAX_DEVIATION	6400.0	/* (according to texas intruments and other sources) */
+#define TACS_MAX_DEVIATION	6400.0	/* (according to texas instruments and other sources) */
 #define TACS_MAX_MODULATION	9500.0	/* (according to panasonic manual) */
 #define TACS_FSK_DEVIATION	(6400.0 / TACS_DBM0_DEVIATION)	/* no emphasis */
 #define TACS_SAT_DEVIATION	(1700.0 / TACS_DBM0_DEVIATION)	/* no emphasis (panasonic / TI) */

src/amps/frame.c

@@ -2221,7 +2221,7 @@ struct amps_ie_desc amps_ie_desc[] = {
 	{ AMPS_IE_DMAC,			"DMAC",			"Digital mobile attenuation code field", ie_cmac },
 	{ AMPS_IE_DTX,			"DTX",			"Discontinuous-Transmission field", ie_yes },
 	{ AMPS_IE_DTX_Support,		"DTX Support",		"Indicates the nature of DTX supported on an analog voice", ie_dtx_support },
-	{ AMPS_IE_DVCC,			"DVCC",			"Digital Verfication Color Code", NULL},
+	{ AMPS_IE_DVCC,			"DVCC",			"Digital Verification Color Code", NULL},
 	{ AMPS_IE_Data_Part,		"Data Part",		"Identifies the Data Port associated with a data/fax call", ie_data_part },
 	{ AMPS_IE_Data_Privacy,		"Data Privacy",		"This field indicates whether or not Data Privacy is supported", ie_yes },
 	{ AMPS_IE_E,			"E",			"Extended address field", ie_yes },
@@ -3699,7 +3699,7 @@ int amps_decode_frame(amps_t *amps, const char *bits, int count, double level, d
 	} else if (count == 240) {
 		more = amps_decode_bits_recc(amps, bits, 0);
 	} else {
-		PDEBUG_CHAN(DFRAME, DEBUG_ERROR, "Frame with unknown lenght = %d, please fix!\n", count);
+		PDEBUG_CHAN(DFRAME, DEBUG_ERROR, "Frame with unknown length = %d, please fix!\n", count);
 	}
 
 	return more;

src/bnetz/bnetz.h

@@ -92,7 +92,7 @@ typedef struct bnetz {
 	double			rx_telegramm_quality[16];/* quality of each bit in telegramm */
 	double			rx_telegramm_level[16];	/* level of each bit in telegramm */
 	int			rx_telegramm_qualidx;	/* index of quality array above */
-	uint16_t		rx_tone;		/* rx shift register for receiveing continous tone */
+	uint16_t		rx_tone;		/* rx shift register for receiveing continuous tone */
 	double			rx_tone_quality[16];	/* quality of tone fragment (100th of second) */
 	double			rx_tone_level[16];	/* level of tone fragment (100th of second) */
 	int			rx_tone_qualidx;	/* index of quality array above */

src/bnetz/dsp.c

@@ -152,7 +152,7 @@ static void fsk_receive_tone(bnetz_t *bnetz, int tone, int goodtone, double leve
 	if (!goodtone && bnetz->tone_detected > -1) {
 		bnetz->tone_count++;
 		if (bnetz->tone_count == TONE_LOST_CNT) {
-			/* substract TONE_LOST_CNT from duration, because it took that long to detect loss of tone */
+			/* subtract TONE_LOST_CNT from duration, because it took that long to detect loss of tone */
 			PDEBUG_CHAN(DDSP, DEBUG_INFO, "Lost F%d tone after %.2f seconds.\n", bnetz->tone_detected, (double)(bnetz->tone_duration - TONE_LOST_CNT) / 100.0);
 			bnetz->tone_detected = -1;
 			bnetz_receive_tone(bnetz, -1);
@@ -194,7 +194,7 @@ static void fsk_receive_bit(void *inst, int bit, double quality, double level)
 	display_measurements_update(bnetz->dmp_tone_stddev, level_stddev / level_avg * 100.0, 0.0);
 	display_measurements_update(bnetz->dmp_tone_quality, quality_avg * 100.0, 0.0);
 
-	/* collect bits, and check for level and continous tone */
+	/* collect bits, and check for level and continuous tone */
 	bnetz->rx_tone = (bnetz->rx_tone << 1) | bit;
 	for (i = 0; i < TONE_DETECT_CNT; i++) {
 		if (((bnetz->rx_tone >> i) & 1) != bit)

src/cnetz/cnetz.c

@@ -96,7 +96,7 @@
  *
  * In case of a combined OgK+SpK, the channel stays the same, but will change.
  *
- * See below for detailled processing.
+ * See below for detailed processing.
  */
 
 /*

src/cnetz/fsk_demod.c

@@ -81,7 +81,7 @@
  * When we are synced:
  *
  * After we recorded the time of all level changes during the sync sequence, we
- * calulate an average and use it as a time base for sampling the subsequent 150
+ * calculate an average and use it as a time base for sampling the subsequent 150
  * bit of a message.  From now on, a bit change does not cause any resync. We
  * just remember what change we received. Later we use it for sampling the 150
  * bits.
@@ -100,7 +100,7 @@
  * since the mobile phone is perfectly synced to us.
  *
  * After receiving and decoding of a frame, we use the time of received sync
- * sequence to synchronize the reciever to the mobile phone. If we receive a
+ * sequence to synchronize the receiver to the mobile phone. If we receive a
  * message on the OgK (control channel), we know that this is a response to a
  * message of a specific time slot we recently sent. Then we can fully sync the
  * receiver's clock.  For any other frame, we cannot determine the absolute

src/cnetz/main.c

@@ -280,7 +280,7 @@ static int handle_options(int short_option, int argi, char **argv)
 	case 'C':
 		p = strchr(argv[argi], ',');
 		if (!p) {
-			fprintf(stderr, "Illegal clock speed, use two values, seperated by comma and no spaces!\n");
+			fprintf(stderr, "Illegal clock speed, use two values, separated by comma and no spaces!\n");
 			return -EINVAL;
 		}
 		clock_speed[0] = strtold(argv[argi], NULL);

src/jolly/dsp.c

@@ -143,9 +143,9 @@ void dsp_cleanup_sender(jolly_t *jolly)
 	}
 }
 
-void set_speech_string(jolly_t *jolly, char anouncement, const char *number)
+void set_speech_string(jolly_t *jolly, char announcement, const char *number)
 {
-	jolly->speech_string[0] = anouncement;
+	jolly->speech_string[0] = announcement;
 	jolly->speech_string[1] = '\0';
 	strncat(jolly->speech_string, number, sizeof(jolly->speech_string) - 1);
 	jolly->speech_digit = 0;

src/jolly/dsp.h

@@ -2,6 +2,6 @@
 void dsp_init(void);
 int dsp_init_sender(jolly_t *jolly, int nbfm, double squelch_db, int repeater);
 void dsp_cleanup_sender(jolly_t *jolly);
-void set_speech_string(jolly_t *jolly, char anouncement, const char *number);
+void set_speech_string(jolly_t *jolly, char announcement, const char *number);
 void reset_speech_string(jolly_t *jolly);
 

src/jolly/jolly.c

@@ -88,7 +88,7 @@
  * 		| '#' received		| stop timer
  * 		|			| call setup
  * 		|			| if call setup fails:
- * 		|			| 	play release anouncement
+ * 		|			| 	play release announcement
  * 		|			| 	go to state RELEASED
  * 		|			| go to state CALL
  * 		|			|
@@ -98,18 +98,18 @@
  * CALL		| '*' received		| start timer T-DIAL2
  * 		|			| go to state CALL-DIALING
  * 		|			|
- * 		| call release		| play release anouncement
+ * 		| call release		| play release announcement
  * 		|			| go to state RELEASED
  * 		|			|
  * -------------+-----------------------+--------------------------------------
  * CALL-DIALING	| '#' received		| stop timer
  * 		|			| call release
- * 		|			| play release anouncement
+ * 		|			| play release announcement
  * 		|			| go to state RELEASED
  * 		|			|
  * 		| timeout		| go state CALL
  * 		|			|
- * 		| call release		| play release anouncement
+ * 		| call release		| play release announcement
  * 		|			| go to state RELEASED
  * 		|			|
  * -------------+-----------------------+--------------------------------------
@@ -121,7 +121,7 @@
  * 		| call release		| go to state IDLE
  * 		|			|
  * -------------+-----------------------+--------------------------------------
- * RELEASED	| end of anouncement	| go to state IDLE
+ * RELEASED	| end of announcement	| go to state IDLE
  * 		|			|
  */
 

src/jolly/main.c

@@ -93,7 +93,7 @@ static int handle_options(int short_option, int argi, char **argv)
 		string_ul = strsep(&string, ",");
 		string_step = strsep(&string, ",");
 		if (!string_dl || !string_ul || !string_step) {
-			fprintf(stderr, "Please give 3 values for --frequency, seperated by comma and no space!\n");
+			fprintf(stderr, "Please give 3 values for --frequency, separated by comma and no space!\n");
 			exit(0);
 		}
 		dl_freq = atof(string_dl);

src/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).

src/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 */

src/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

src/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)

src/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;

src/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");

src/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;

src/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) {

src/nmt/main.c

@@ -76,7 +76,7 @@ void print_help(const char *arg0)
 	printf(" -Y --traffic-area <traffic area> | list\n");
 	printf("        NOTE: MUST MATCH WITH YOUR ROAMING SETTINGS IN THE PHONE!\n");
 	printf("              Your phone will not connect, if country code is different!\n");
-	printf("        Give short country code and traffic area seperated by comma.\n");
+	printf("        Give short country code and traffic area separated by comma.\n");
 	printf("        (Example: Give 'SE,1' for Sweden, traffic area 1)\n");
 	printf("        Add '!' to force traffic area that is not supported by country.\n");
 	printf("        (Example: Give 'B,12!' for Belgium, traffic area 12)\n");

src/nmt/sms.c

@@ -326,7 +326,7 @@ int sms_deliver(nmt_t *nmt, uint8_t ref, const char *orig_address, uint8_t orig_
 
 	/* RP length */
 	*tpdu_length = length - (uint8_t)(tpdu_length - data) - 1;
-	PDEBUG(DSMS, DEBUG_DEBUG, " -> TPDU lenght = %d\n", *tpdu_length);
+	PDEBUG(DSMS, DEBUG_DEBUG, " -> TPDU length = %d\n", *tpdu_length);
 
 	nmt->sms.mt = 1;
 	dms_send(nmt, data, length, 1);

src/nmt/sms.h

@@ -1,5 +1,5 @@
 
-#define SMS_TYPE_UKNOWN		0x0
+#define SMS_TYPE_UNKNOWN	0x0
 #define SMS_TYPE_INTERNATIONAL	0x1
 #define SMS_TYPE_NATIONAL	0x2
 #define SMS_TYPE_NETWORK	0x3
@@ -8,7 +8,7 @@
 #define SMS_TYPE_ABBREVIATED	0x6
 #define SMS_TYPE_RESERVED	0x7
 
-#define SMS_PLAN_UNKOWN		0x0
+#define SMS_PLAN_UNKNOWN	0x0
 #define SMS_PLAN_ISDN_TEL	0x1
 #define SMS_PLAN_DATA		0x3
 #define SMS_PLAN_TELEX		0x4

src/r2000/main.c

@@ -345,7 +345,7 @@ int main(int argc, char *argv[])
 		fprintf(stderr, "*******************************************************************************\n");
 		fprintf(stderr, "I strongly suggest to let me do pre- and de-emphasis (options -p -d)!\n");
 		fprintf(stderr, "Use a transmitter/receiver without emphasis and let me do that!\n");
-		fprintf(stderr, "Because 50 baud supervisory signalling arround 150 Hz will not be tranmitted by\n");
+		fprintf(stderr, "Because 50 baud supervisory signalling around 150 Hz will not be tranmitted by\n");
 		fprintf(stderr, "regular radio, use direct input to the PLL of your transmitter (or use SDR).\n");
 		fprintf(stderr, "*******************************************************************************\n");
 	}

src/r2000/r2000.c

@@ -49,7 +49,7 @@
    This offset of 0x400000000 is required for MNCC interface. */
 static int new_callref = 0x40000000;
 
-/* definiton of bands and channels */
+/* definition of bands and channels */
 #define CHANNEL_SPACING	0.0125
 
 static struct r2000_bands {

src/radio/radio.c

@@ -633,7 +633,7 @@ int radio_rx(radio_t *radio, float *baseband, int signal_num)
 			/* mix pilot tone (double phase) with differential signal */
 			for (i = 0; i < signal_num; i++) {
 				p = atan2(samples[2][i], samples[1][i]);
-				/* substract measured phase difference (use double amplitude, because we filter later) */
+				/* subtract measured phase difference (use double amplitude, because we filter later) */
 			        samples[1][i] = samples[0][i] * sin((radio->rx_pilot_phase - p) * 2.0) * 2.0;
 				radio->rx_pilot_phase += radio->pilot_phasestep;
 				if (radio->rx_pilot_phase >= 2.0 * M_PI)

src/test/test_performance.c

@@ -82,7 +82,7 @@ int main(void)
 	iir_lowpass_init(&lp, 10000.0 / 2.0, 50000, 4);
 	T_START()
 	iir_process(&lp, samples, SAMPLES);
-	T_STOP("low-pass filter (eigth order)", SAMPLES)
+	T_STOP("low-pass filter (eighth order)", SAMPLES)
 
 	fm_exit();
 

src/tv/bas.c

@@ -42,7 +42,7 @@
 #define H_SYNC2_START	(H_SYNC_START + H_LINE_END/2.0)
 #define H_SYNC2_STOP	(H_SYNC_STOP + H_LINE_END/2.0)
 #define V_SYNC_STOP	(H_SYNC2_START - (H_SYNC_STOP - H_SYNC_START))
-#define V_SYNC2_STOP	(H_SYNC_START - (H_SYNC_STOP - H_SYNC_START) + H_LINE_END) // wraps, so we substract H_LINE_END
+#define V_SYNC2_STOP	(H_SYNC_START - (H_SYNC_STOP - H_SYNC_START) + H_LINE_END) // wraps, so we subtract H_LINE_END
 #define SYNC_RAMP	0.0000003
 #define IMAGE_RAMP	0.0000002
 #define H_CBURST_START	0.0000068
@@ -71,7 +71,7 @@ void bas_init(bas_t *bas, double samplerate, enum bas_type type, int fbas, doubl
 	/* filter color signal */
 	iir_lowpass_init(&bas->lp_u, 1300000.0, samplerate, COLOR_FILTER_ITER);
 	iir_lowpass_init(&bas->lp_v, 1300000.0, samplerate, COLOR_FILTER_ITER);
-	/* filter final FBAS, so we prevent from beeing in the audio carrier spectrum */
+	/* filter final FBAS, so we prevent from being in the audio carrier spectrum */
 	iir_lowpass_init(&bas->lp_y, 4500000.0, samplerate, COLOR_FILTER_ITER);
 }