Fixed many typos in output and source code comments
Andreas Eversberg
parent
c357ab5ad5
commit
75765d49b7
2
INSTALL
2
INSTALL
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@ -1,4 +1,4 @@
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You need to have the folowing packages install:
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You need to have the following packages install:
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* automake, gcc for compilation
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* libasound2 for sound card support, which is what you need!
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@ -558,7 +558,7 @@ bnetz.c:351 info : Entering IDLE state, sending 'Gruppenfreisignal' 2.
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Base station for channel 1 ready, please tune transmitter to 153.010 MHz and receiver to 148.410 MHz.
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To call phone, switch transmitter (using paging signal) to 153.370 MHz.
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mncc_sock.c:137 notice : MNCC socket connected.
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dsp.c:159 info : Detecting continous tone: 2070:Level= 80% Quality=100%
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dsp.c:159 info : Detecting continuous tone: 2070:Level= 80% Quality=100%
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bnetz.c:470 info : Received signal 'Kanalbelegung' from mobile station, sending signal 'Wahlabruf'.
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bnetz.c:509 info : Digit RX Level: 80% Quality=85
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bnetz.c:524 info : Received telegramm digit 'Funkwahl ohne Gebuehrenuebermittlung'.
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@ -692,15 +692,15 @@ call.c:859 info : Outgoing call from to '50993'
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bnetz.c:757 info : Call to mobile station, paging station id '50993'
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bnetz.c:374 info : Entering paging state (try 1), sending 'Selektivruf' to '50993'.
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bnetz.c:410 info : Paging mobile station 50993 complete, waiting for answer.
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dsp.c:159 info : Detecting continous tone: 1950:Level= 105% Quality=100%
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dsp.c:159 info : Detecting continuous tone: 1950:Level= 105% Quality=100%
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bnetz.c:480 info : Received signal 'Rufbestaetigung' from mobile station, sending signal 'Rufhaltung'.
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(call is ringing)
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call.c:641 info : Call is alerting
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...
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dsp.c:159 info : Detecting continous tone: 2070:Level= 102% Quality=99%
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dsp.c:159 info : Detecting continuous tone: 2070:Level= 102% Quality=99%
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bnetz.c:491 info : Received signal 'Beginnsignal' from mobile station, call establised.
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call.c:684 info : Call has been answered by '50993'
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dsp.c:159 info : Detecting continous tone: 2070:Level= 104% Quality=100%
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dsp.c:159 info : Detecting continuous tone: 2070:Level= 104% Quality=100%
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</pre>
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@ -727,7 +727,7 @@ squelch.c:94 info : RF signal measurement: -69.2 dB noise floor, using thresho
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</pre>
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<p>
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In the following example I cut off the power of the phone beeing in a call and waited 12 seconds for the base station to time out:
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In the following example I cut off the power of the phone being in a call and waited 12 seconds for the base station to time out:
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</p>
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<pre>
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@ -749,7 +749,7 @@ Kennungsspeicher (The Security Module)
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<p>
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Older phones used soldered jumpers to set the phone number (ID) of the phone.
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Just by soldering a different number, the network could be used without paying.
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So simple was hacking back then - if you could affort an expensive B-Netz phone.
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So simple was hacking back then - if you could afford an expensive B-Netz phone.
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The security module "Kennungsspeicher" was introduced to prevent using the phone, if it is not inserted into the internal socket.
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The idea was to disable unsubscribed phones, just by removing the module.
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This module was owned by the German post office and I got a phone without it.
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@ -184,7 +184,7 @@ void amps_number2min(const char *number, uint32_t *min1, uint16_t *min2)
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int i;
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if (nlen != 10) {
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fprintf(stderr, "illegal lenght %d. Must be 10, aborting!", nlen);
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fprintf(stderr, "illegal length %d. Must be 10, aborting!", nlen);
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abort();
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}
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@ -580,7 +580,7 @@ int amps_create(int channel, enum amps_chan_type chan_type, const char *audiodev
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amps->pre_emphasis = pre_emphasis;
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amps->de_emphasis = de_emphasis;
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/* 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 */
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/* 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 */
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rc = init_emphasis(&s->estate, samplerate, CUT_OFF_EMPHASIS_DEFAULT, CUT_OFF_HIGHPASS_DEFAULT, CUT_OFF_LOWPASS_DEFAULT);
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if (rc < 0)
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goto error;
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@ -39,7 +39,7 @@
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* The average level change offsets of the dotting sequence is used to set the
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* window for the first bit. When all samples for the window are received, a
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* raise in level is detected as 1, fall in level is detected as 0. This is done
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* by substracting the average sample value of the left side of the window by
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* by subtracting the average sample value of the left side of the window by
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* the average sample value of the right side. After the bit has been detected,
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* the samples for the next window will be received and detected.
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*
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@ -108,7 +108,7 @@
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#define AMPS_BITRATE 10000
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/* for some reason, 4000 Hz deviation works better */
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#define TACS_DBM0_DEVIATION 4000.0 /* 2300 Hz deviation at 1 kHz (according to panasonic manual) */
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#define TACS_MAX_DEVIATION 6400.0 /* (according to texas intruments and other sources) */
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#define TACS_MAX_DEVIATION 6400.0 /* (according to texas instruments and other sources) */
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#define TACS_MAX_MODULATION 9500.0 /* (according to panasonic manual) */
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#define TACS_FSK_DEVIATION (6400.0 / TACS_DBM0_DEVIATION) /* no emphasis */
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#define TACS_SAT_DEVIATION (1700.0 / TACS_DBM0_DEVIATION) /* no emphasis (panasonic / TI) */
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@ -2221,7 +2221,7 @@ struct amps_ie_desc amps_ie_desc[] = {
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{ AMPS_IE_DMAC, "DMAC", "Digital mobile attenuation code field", ie_cmac },
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{ AMPS_IE_DTX, "DTX", "Discontinuous-Transmission field", ie_yes },
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{ AMPS_IE_DTX_Support, "DTX Support", "Indicates the nature of DTX supported on an analog voice", ie_dtx_support },
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{ AMPS_IE_DVCC, "DVCC", "Digital Verfication Color Code", NULL},
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{ AMPS_IE_DVCC, "DVCC", "Digital Verification Color Code", NULL},
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{ AMPS_IE_Data_Part, "Data Part", "Identifies the Data Port associated with a data/fax call", ie_data_part },
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{ AMPS_IE_Data_Privacy, "Data Privacy", "This field indicates whether or not Data Privacy is supported", ie_yes },
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{ AMPS_IE_E, "E", "Extended address field", ie_yes },
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@ -3699,7 +3699,7 @@ int amps_decode_frame(amps_t *amps, const char *bits, int count, double level, d
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} else if (count == 240) {
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more = amps_decode_bits_recc(amps, bits, 0);
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} else {
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PDEBUG_CHAN(DFRAME, DEBUG_ERROR, "Frame with unknown lenght = %d, please fix!\n", count);
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PDEBUG_CHAN(DFRAME, DEBUG_ERROR, "Frame with unknown length = %d, please fix!\n", count);
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}
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return more;
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@ -92,7 +92,7 @@ typedef struct bnetz {
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double rx_telegramm_quality[16];/* quality of each bit in telegramm */
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double rx_telegramm_level[16]; /* level of each bit in telegramm */
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int rx_telegramm_qualidx; /* index of quality array above */
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uint16_t rx_tone; /* rx shift register for receiveing continous tone */
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uint16_t rx_tone; /* rx shift register for receiveing continuous tone */
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double rx_tone_quality[16]; /* quality of tone fragment (100th of second) */
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double rx_tone_level[16]; /* level of tone fragment (100th of second) */
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int rx_tone_qualidx; /* index of quality array above */
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@ -152,7 +152,7 @@ static void fsk_receive_tone(bnetz_t *bnetz, int tone, int goodtone, double leve
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if (!goodtone && bnetz->tone_detected > -1) {
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bnetz->tone_count++;
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if (bnetz->tone_count == TONE_LOST_CNT) {
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/* substract TONE_LOST_CNT from duration, because it took that long to detect loss of tone */
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/* subtract TONE_LOST_CNT from duration, because it took that long to detect loss of tone */
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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);
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bnetz->tone_detected = -1;
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bnetz_receive_tone(bnetz, -1);
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@ -194,7 +194,7 @@ static void fsk_receive_bit(void *inst, int bit, double quality, double level)
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display_measurements_update(bnetz->dmp_tone_stddev, level_stddev / level_avg * 100.0, 0.0);
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display_measurements_update(bnetz->dmp_tone_quality, quality_avg * 100.0, 0.0);
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/* collect bits, and check for level and continous tone */
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/* collect bits, and check for level and continuous tone */
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bnetz->rx_tone = (bnetz->rx_tone << 1) | bit;
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for (i = 0; i < TONE_DETECT_CNT; i++) {
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if (((bnetz->rx_tone >> i) & 1) != bit)
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@ -96,7 +96,7 @@
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*
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* In case of a combined OgK+SpK, the channel stays the same, but will change.
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*
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* See below for detailled processing.
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* See below for detailed processing.
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*/
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/*
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@ -81,7 +81,7 @@
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* When we are synced:
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*
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* After we recorded the time of all level changes during the sync sequence, we
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* calulate an average and use it as a time base for sampling the subsequent 150
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* calculate an average and use it as a time base for sampling the subsequent 150
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* bit of a message. From now on, a bit change does not cause any resync. We
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* just remember what change we received. Later we use it for sampling the 150
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* bits.
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@ -100,7 +100,7 @@
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* since the mobile phone is perfectly synced to us.
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*
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* After receiving and decoding of a frame, we use the time of received sync
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* sequence to synchronize the reciever to the mobile phone. If we receive a
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* sequence to synchronize the receiver to the mobile phone. If we receive a
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* message on the OgK (control channel), we know that this is a response to a
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* message of a specific time slot we recently sent. Then we can fully sync the
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* receiver's clock. For any other frame, we cannot determine the absolute
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@ -280,7 +280,7 @@ static int handle_options(int short_option, int argi, char **argv)
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case 'C':
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p = strchr(argv[argi], ',');
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if (!p) {
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fprintf(stderr, "Illegal clock speed, use two values, seperated by comma and no spaces!\n");
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fprintf(stderr, "Illegal clock speed, use two values, separated by comma and no spaces!\n");
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return -EINVAL;
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}
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clock_speed[0] = strtold(argv[argi], NULL);
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@ -143,9 +143,9 @@ void dsp_cleanup_sender(jolly_t *jolly)
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}
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}
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void set_speech_string(jolly_t *jolly, char anouncement, const char *number)
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void set_speech_string(jolly_t *jolly, char announcement, const char *number)
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{
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jolly->speech_string[0] = anouncement;
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jolly->speech_string[0] = announcement;
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jolly->speech_string[1] = '\0';
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strncat(jolly->speech_string, number, sizeof(jolly->speech_string) - 1);
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jolly->speech_digit = 0;
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@ -2,6 +2,6 @@
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void dsp_init(void);
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int dsp_init_sender(jolly_t *jolly, int nbfm, double squelch_db, int repeater);
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void dsp_cleanup_sender(jolly_t *jolly);
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void set_speech_string(jolly_t *jolly, char anouncement, const char *number);
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void set_speech_string(jolly_t *jolly, char announcement, const char *number);
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void reset_speech_string(jolly_t *jolly);
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@ -88,7 +88,7 @@
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* | '#' received | stop timer
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* | | call setup
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* | | if call setup fails:
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* | | play release anouncement
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* | | play release announcement
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* | | go to state RELEASED
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* | | go to state CALL
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* | |
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@ -98,18 +98,18 @@
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* CALL | '*' received | start timer T-DIAL2
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* | | go to state CALL-DIALING
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* | |
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* | call release | play release anouncement
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* | call release | play release announcement
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* | | go to state RELEASED
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* | |
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* -------------+-----------------------+--------------------------------------
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* CALL-DIALING | '#' received | stop timer
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* | | call release
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* | | play release anouncement
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* | | play release announcement
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* | | go to state RELEASED
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* | |
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* | timeout | go state CALL
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* | |
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* | call release | play release anouncement
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* | call release | play release announcement
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* | | go to state RELEASED
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* | |
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* -------------+-----------------------+--------------------------------------
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@ -121,7 +121,7 @@
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* | call release | go to state IDLE
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* | |
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* -------------+-----------------------+--------------------------------------
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* RELEASED | end of anouncement | go to state IDLE
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* RELEASED | end of announcement | go to state IDLE
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* | |
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*/
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@ -93,7 +93,7 @@ static int handle_options(int short_option, int argi, char **argv)
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string_ul = strsep(&string, ",");
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string_step = strsep(&string, ",");
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if (!string_dl || !string_ul || !string_step) {
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fprintf(stderr, "Please give 3 values for --frequency, seperated by comma and no space!\n");
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fprintf(stderr, "Please give 3 values for --frequency, separated by comma and no space!\n");
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exit(0);
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}
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dl_freq = atof(string_dl);
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@ -120,7 +120,7 @@ void display_wave(dispwav_t *disp, sample_t *samples, int length, double range)
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if (pos == width + 2) {
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memset(&screen, ' ', sizeof(screen));
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for (j = 0; j < width; j++) {
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/* Input value is scaled to range -1 .. 1 and then substracted from 1,
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/* Input value is scaled to range -1 .. 1 and then subtracted from 1,
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* so the result ranges from 0 .. 2.
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* HEIGHT-1 is multiplied with the range, so a HEIGHT of 3 would allow
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* 0..4 (5 steps) and a HEIGHT of 11 would allow 0..20 (21 steps).
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@ -157,7 +157,7 @@ void fsk_receive(fsk_t *fsk, sample_t *sample, int length)
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int bit;
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double level, quality;
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/* demod samples to offset arround center frequency */
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/* demod samples to offset around center frequency */
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fm_demodulate_real(&fsk->demod, frequency, length, sample, I, Q);
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for (i = 0; i < length; i++) {
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@ -187,7 +187,7 @@ void fsk_receive(fsk_t *fsk, sample_t *sample, int length)
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fsk->rx_bitpos = 0.5;
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}
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}
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/* if bit counter reaches 1, we substract 1 and sample the bit */
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/* if bit counter reaches 1, we subtract 1 and sample the bit */
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if (fsk->rx_bitpos >= 1.0) {
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/* peak level is the length of I/Q vector
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* since we filter out the unwanted modulation product, the vector is only half of length */
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@ -68,7 +68,7 @@ void audio_goertzel_init(goertzel_t *goertzel, double freq, int samplerate)
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*
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* samples: pointer to sample buffer
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* length: length of buffer
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* offset: for ring buffer, start here and wrap arround to 0 when length has been hit
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* offset: for ring buffer, start here and wrap around to 0 when length has been hit
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* coeff: array of coefficients (coeff << 15)
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* result: array of result levels (average value of the sine, that is 1 / (PI/2) of the sine's peak)
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* k: number of frequencies to check
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@ -30,14 +30,14 @@ static double int_16_speech_level = SPEECH_LEVEL * 0.7079; /* 16 dBm below dBm0,
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* support high numbers. 'double' or 'float' types are sufficient.
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*
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* When using sample_t inside signal processing of each base station, the
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* level of +- 1 is relative to the normal speach evenlope.
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* level of +- 1 is relative to the normal speech evenlope.
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*
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* When converting sample_t to int16_t, the level of +- 1 is reduced by factor.
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* This way the speech may be louder before clipping happens.
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*
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* When using sample_t to modulate (SDR or sound card), the level is changed,
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* so it represents the frequency deviation in Hz. The deviation of speech
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* envelope is network dependant.
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* envelope is network dependent.
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*/
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void samples_to_int16(int16_t *spl, sample_t *samples, int length)
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@ -967,7 +967,7 @@ int sdr_get_tosend(void *inst, int latspl)
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count /= sdr->oversample;
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if (sdr->threads) {
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/* substract what we have in write buffer, because this is not jent sent to the SDR */
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/* subtract what we have in write buffer, because this is not jent sent to the SDR */
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int fill;
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fill = (sdr->thread_write.in - sdr->thread_write.out + sdr->thread_write.buffer_size) % sdr->thread_write.buffer_size;
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@ -62,7 +62,7 @@ void sdr_config_print_help(void)
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printf(" --sdr-device-args <args>\n");
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printf(" --sdr-stream-args <args>\n");
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printf(" --sdr-tune-args <args>\n");
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printf(" Optional SDR device arguments, seperated by comma\n");
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printf(" Optional SDR device arguments, separated by comma\n");
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printf(" e.g. --sdr-device-args <key>=<value>[,<key>=<value>[,...]]\n");
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printf(" --sdr-samplerate <samplerate>\n");
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printf(" Sample rate to use with SDR. By default it equals the regular sample\n");
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@ -501,7 +501,7 @@ int soapy_get_tosend(int latspl)
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tosend = latspl - (tx_count - rx_count);
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/* in case of underrun: */
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if (tosend > latspl) {
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// It is normal that we have underruns, prior inital filling of buffer.
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// It is normal that we have underruns, prior initial filling of buffer.
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// FIXME: better solution to detect underrun
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// PDEBUG(DSOAPY, DEBUG_ERROR, "SDR TX underrun!\n");
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tosend = 0;
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@ -104,7 +104,7 @@ enum squelch_result squelch(squelch_t *squelch, double rf_level_db, double durat
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}
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}
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/* enough RF level, so we unmute when mute_count reched 0 */
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/* 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) {
|
||||
|
|
|
|||
|
|
@ -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");
|
||||
|
|
|
|||
|
|
@ -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);
|
||||
|
|
|
|||
|
|
@ -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
|
||||
|
|
|
|||
|
|
@ -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");
|
||||
}
|
||||
|
|
|
|||
|
|
@ -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 {
|
||||
|
|
|
|||
|
|
@ -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)
|
||||
|
|
|
|||
|
|
@ -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();
|
||||
|
||||
|
|
|
|||
|
|
@ -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);
|
||||
}
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue