From 75765d49b735f3a998a5c0ee4b372b4d1da7576c Mon Sep 17 00:00:00 2001 From: Martin Hauke Date: Mon, 15 Jul 2019 21:14:52 +0200 Subject: Fixed many typos in output and source code comments --- src/amps/amps.c | 4 ++-- src/amps/dsp.c | 4 ++-- src/amps/frame.c | 4 ++-- src/bnetz/bnetz.h | 2 +- src/bnetz/dsp.c | 4 ++-- src/cnetz/cnetz.c | 2 +- src/cnetz/fsk_demod.c | 4 ++-- src/cnetz/main.c | 2 +- src/jolly/dsp.c | 4 ++-- src/jolly/dsp.h | 2 +- src/jolly/jolly.c | 10 +++++----- src/jolly/main.c | 2 +- src/libdisplay/display_wave.c | 2 +- src/libfsk/fsk.c | 4 ++-- src/libgoertzel/goertzel.c | 2 +- src/libsample/sample.c | 4 ++-- src/libsdr/sdr.c | 2 +- src/libsdr/sdr_config.c | 2 +- src/libsdr/soapy.c | 2 +- src/libsquelch/squelch.c | 2 +- src/nmt/main.c | 2 +- src/nmt/sms.c | 2 +- src/nmt/sms.h | 4 ++-- src/r2000/main.c | 2 +- src/r2000/r2000.c | 2 +- src/radio/radio.c | 2 +- src/test/test_performance.c | 2 +- src/tv/bas.c | 4 ++-- 28 files changed, 42 insertions(+), 42 deletions(-) (limited to 'src') diff --git a/src/amps/amps.c b/src/amps/amps.c index 7ecf76f..0585823 100644 --- a/src/amps/amps.c +++ b/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(&s->estate, samplerate, CUT_OFF_EMPHASIS_DEFAULT, CUT_OFF_HIGHPASS_DEFAULT, CUT_OFF_LOWPASS_DEFAULT); if (rc < 0) goto error; diff --git a/src/amps/dsp.c b/src/amps/dsp.c index 1bf8c43..f08d023 100644 --- a/src/amps/dsp.c +++ b/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) */ diff --git a/src/amps/frame.c b/src/amps/frame.c index d22dcd3..496241b 100644 --- a/src/amps/frame.c +++ b/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; diff --git a/src/bnetz/bnetz.h b/src/bnetz/bnetz.h index dcee8c7..71acab9 100644 --- a/src/bnetz/bnetz.h +++ b/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 */ diff --git a/src/bnetz/dsp.c b/src/bnetz/dsp.c index d95fb4e..1773dc6 100644 --- a/src/bnetz/dsp.c +++ b/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) diff --git a/src/cnetz/cnetz.c b/src/cnetz/cnetz.c index 44dd635..7ccfc3b 100644 --- a/src/cnetz/cnetz.c +++ b/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. */ /* diff --git a/src/cnetz/fsk_demod.c b/src/cnetz/fsk_demod.c index 54ad41f..6362d9e 100644 --- a/src/cnetz/fsk_demod.c +++ b/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 diff --git a/src/cnetz/main.c b/src/cnetz/main.c index 50f7e9b..7acca67 100644 --- a/src/cnetz/main.c +++ b/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); diff --git a/src/jolly/dsp.c b/src/jolly/dsp.c index 00407eb..e85d37b 100644 --- a/src/jolly/dsp.c +++ b/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; diff --git a/src/jolly/dsp.h b/src/jolly/dsp.h index c27e225..6afa3ce 100644 --- a/src/jolly/dsp.h +++ b/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); diff --git a/src/jolly/jolly.c b/src/jolly/jolly.c index 51b06a5..d235903 100644 --- a/src/jolly/jolly.c +++ b/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 * | | */ diff --git a/src/jolly/main.c b/src/jolly/main.c index 3ec19b7..4fcbb6b 100644 --- a/src/jolly/main.c +++ b/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); diff --git a/src/libdisplay/display_wave.c b/src/libdisplay/display_wave.c index d5f285b..e89fb3b 100644 --- a/src/libdisplay/display_wave.c +++ b/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). diff --git a/src/libfsk/fsk.c b/src/libfsk/fsk.c index 726f3be..6b5fe03 100644 --- a/src/libfsk/fsk.c +++ b/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 */ diff --git a/src/libgoertzel/goertzel.c b/src/libgoertzel/goertzel.c index d4038d6..cb06f99 100644 --- a/src/libgoertzel/goertzel.c +++ b/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 diff --git a/src/libsample/sample.c b/src/libsample/sample.c index 72ba941..a084b66 100644 --- a/src/libsample/sample.c +++ b/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) diff --git a/src/libsdr/sdr.c b/src/libsdr/sdr.c index dd89cea..483d4ff 100644 --- a/src/libsdr/sdr.c +++ b/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; diff --git a/src/libsdr/sdr_config.c b/src/libsdr/sdr_config.c index 0ff021d..960d718 100644 --- a/src/libsdr/sdr_config.c +++ b/src/libsdr/sdr_config.c @@ -62,7 +62,7 @@ void sdr_config_print_help(void) printf(" --sdr-device-args \n"); printf(" --sdr-stream-args \n"); printf(" --sdr-tune-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 =[,=[,...]]\n"); printf(" --sdr-samplerate \n"); printf(" Sample rate to use with SDR. By default it equals the regular sample\n"); diff --git a/src/libsdr/soapy.c b/src/libsdr/soapy.c index 0919060..3306af1 100644 --- a/src/libsdr/soapy.c +++ b/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; diff --git a/src/libsquelch/squelch.c b/src/libsquelch/squelch.c index cd6afa3..fc05003 100644 --- a/src/libsquelch/squelch.c +++ b/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) { diff --git a/src/nmt/main.c b/src/nmt/main.c index 4ec931b..e505aed 100644 --- a/src/nmt/main.c +++ b/src/nmt/main.c @@ -76,7 +76,7 @@ void print_help(const char *arg0) printf(" -Y --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"); diff --git a/src/nmt/sms.c b/src/nmt/sms.c index 92dc09c..04488ec 100644 --- a/src/nmt/sms.c +++ b/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); diff --git a/src/nmt/sms.h b/src/nmt/sms.h index 2e322dc..0aaa011 100644 --- a/src/nmt/sms.h +++ b/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 diff --git a/src/r2000/main.c b/src/r2000/main.c index 926ff21..efce6cb 100644 --- a/src/r2000/main.c +++ b/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"); } diff --git a/src/r2000/r2000.c b/src/r2000/r2000.c index 8baf8da..a625467 100644 --- a/src/r2000/r2000.c +++ b/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 { diff --git a/src/radio/radio.c b/src/radio/radio.c index 4131ccc..ee85805 100644 --- a/src/radio/radio.c +++ b/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) diff --git a/src/test/test_performance.c b/src/test/test_performance.c index 2f9b062..a045eee 100644 --- a/src/test/test_performance.c +++ b/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(); diff --git a/src/tv/bas.c b/src/tv/bas.c index b7d8381..55ffa58 100644 --- a/src/tv/bas.c +++ b/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); } -- cgit v1.2.3