Improvement of Goertzel filter. Using Hamming window now. Add test routine.
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@ -84,6 +84,7 @@ src/test/test_filter
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src/test/test_sendevolumenregler
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src/test/test_compandor
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src/test/test_emphasis
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src/test/test_goertzel
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src/test/test_dtmf
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src/test/test_dms
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src/test/test_sms
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@ -114,7 +114,7 @@
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#define TACS_SAT_DEVIATION (1700.0 / TACS_SPEECH_DEVIATION) /* no emphasis (panasonic / TI) */
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#define TACS_MAX_DISPLAY (8000.0 / TACS_SPEECH_DEVIATION) /* no emphasis */
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#define TACS_BITRATE 8000
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#define SAT_DURATION 0.05 /* duration of SAT signal measurement */
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#define SAT_BANDWIDTH 30.0 /* distance between two SAT tones, also bandwidth for goertzel filter */
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#define SAT_QUALITY 0.85 /* quality needed to detect SAT signal */
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#define SAT_PRINT 10 /* print sat measurement every 0.5 seconds */
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#define DTX_LEVEL 0.50 /* SAT level needed to mute/unmute */
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@ -133,7 +133,7 @@ static double sat_freq[4] = {
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5970.0,
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6000.0,
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6030.0,
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5800.0, /* noise level to check against */
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5790.0, /* noise level to check against */
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};
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static sample_t dsp_sine_sat[65536];
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@ -243,8 +243,11 @@ int dsp_init_sender(amps_t *amps, int tolerant)
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amps->fsk_deviation = (!tacs) ? AMPS_FSK_DEVIATION : TACS_FSK_DEVIATION;
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dsp_init_ramp(amps);
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/* allocate ring buffer for SAT signal detection */
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amps->sat_samples = (int)((double)amps->sender.samplerate * SAT_DURATION + 0.5);
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/* allocate ring buffer for SAT signal detection
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* the bandwidth of the Goertzel filter is the reciprocal of the duration
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* we half our bandwidth, so that other supervisory signals will be canceled out completely by goertzel filter
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*/
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amps->sat_samples = (int)((double)amps->sender.samplerate * (1.0 / (SAT_BANDWIDTH / 2.0)) + 0.5);
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spl = calloc(sizeof(*spl), amps->sat_samples);
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if (!spl) {
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PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
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@ -732,7 +735,7 @@ static void sender_receive_frame(amps_t *amps, sample_t *samples, int length)
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/* decode SAT and signaling tone */
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/* compare supervisory signal against noise floor on 5800 Hz */
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/* compare supervisory signal against noise floor at 5790 Hz */
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static void sat_decode(amps_t *amps, sample_t *samples, int length)
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{
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double result[3], sat_quality, sig_quality, sat_level, sig_level;
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@ -742,8 +745,8 @@ static void sat_decode(amps_t *amps, sample_t *samples, int length)
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audio_goertzel(&s->sat_goertzel[4], samples, length, 0, &result[2], 1);
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/* normalize sat level and signaling tone level */
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sat_level = result[0] / ((!tacs) ? AMPS_SAT_DEVIATION : TACS_SAT_DEVIATION) / 0.63662;
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sig_level = result[2] / ((!tacs) ? AMPS_FSK_DEVIATION : TACS_FSK_DEVIATION) / 0.63662;
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sat_level = result[0] / ((!tacs) ? AMPS_SAT_DEVIATION : TACS_SAT_DEVIATION);
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sig_level = result[2] / ((!tacs) ? AMPS_FSK_DEVIATION : TACS_FSK_DEVIATION);
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/* get normalized quality of SAT and signaling tone */
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sat_quality = (result[0] - result[1]) / result[0];
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@ -41,7 +41,7 @@
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#define TX_PEAK_TONE (10500.0 / SPEECH_DEVIATION) /* 10.5 kHz, no emphasis */
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#define TX_PEAK_PAGE (15000.0 / SPEECH_DEVIATION) /* 15 kHz, no emphasis */
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#define MAX_DISPLAY (15000.0 / SPEECH_DEVIATION) /* 15 kHz, no emphasis */
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#define CHUNK_DURATION 0.010 /* 10 ms */
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#define CHUNK_DURATION 0.010 /* 10 m = 100 Hz bandwidth (-7.6 DB @ +-100 Hz) */
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#define TONE_THRESHOLD 0.05
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#define QUAL_THRESHOLD 0.5
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@ -158,15 +158,15 @@ static void fsk_decode_chunk(anetz_t *anetz, sample_t *spl, int max)
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{
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double level, result[2], quality[2];
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level = audio_level(spl, max);
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level = audio_mean_level(spl, max);
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/* convert mean (if level comes from a sine curve) to peak value */
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level = level * M_PI / 2.0 / TX_PEAK_TONE;
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audio_goertzel(anetz->fsk_tone_goertzel, spl, max, 0, result, 2);
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/* normalize quality of tones and level */
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/* calculate quality of tones */
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quality[0] = result[0] / level;
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quality[1] = result[1] / level;
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/* adjust level, so we get peak of sine curve */
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level = level / 0.63662 / TX_PEAK_TONE;
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/* show tones */
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display_measurements_update(anetz->dmp_tone_level, level * 100.0, 0.0);
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display_measurements_update(anetz->dmp_tone_quality, quality[1] * 100.0, 0.0);
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@ -35,7 +35,7 @@
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* For a perfect rectangualr wave, the result would equal the peak level.
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* For a sine wave the result would be factor (2 / PI) below peak level.
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*/
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double audio_level(sample_t *samples, int length)
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double audio_mean_level(sample_t *samples, int length)
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{
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double level, sk;
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int n;
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@ -54,8 +54,23 @@ double audio_level(sample_t *samples, int length)
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return level;
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}
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/* use hamming window */
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double window[256];
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int window_generated = 0;
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static void gen_window(void)
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{
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int i;
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for (i = 0; i < 256; i++)
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window[i] = 0.54 - 0.46 * cos(2.0 * M_PI * (double)i / 256.0);
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window_generated = 1;
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}
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void audio_goertzel_init(goertzel_t *goertzel, double freq, int samplerate)
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{
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if (!window_generated)
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gen_window();
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memset(goertzel, 0, sizeof(*goertzel));
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goertzel->coeff = 2.0 * cos(2.0 * M_PI * freq / (double)samplerate);
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}
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@ -67,10 +82,11 @@ void audio_goertzel_init(goertzel_t *goertzel, double freq, int samplerate)
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/* filter frequencies and return their levels
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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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* length: length of buffer: -7.4 dB @ +-(1 / duration) Hz and -INF @ +-(1 / duration * 2) Hz
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* -> if duration is 10 ms, we got -7.4 dB @ +-100 Hz and -INF at +-200 Hz
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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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* result: array of result levels (peak value of the target frequency)
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* k: number of frequencies to check
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*/
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void audio_goertzel(goertzel_t *goertzel, sample_t *samples, int length, int offset, double *result, int k)
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@ -87,7 +103,7 @@ void audio_goertzel(goertzel_t *goertzel, sample_t *samples, int length, int off
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cos2pik = goertzel[i].coeff;
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/* note: after 'length' cycles, offset is restored to its initial value */
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for (n = 0; n < length; n++) {
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sk = (cos2pik * sk1) - sk2 + samples[offset++];
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sk = (cos2pik * sk1) - sk2 + samples[offset++] * window[n * 256 / length];
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sk2 = sk1;
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sk1 = sk;
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if (offset == length)
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@ -98,7 +114,7 @@ void audio_goertzel(goertzel_t *goertzel, sample_t *samples, int length, int off
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(sk * sk) -
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(cos2pik * sk * sk2) +
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(sk2 * sk2)
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) / (double)length * 2.0 * 0.63662; /* 1 / (PI/2) */
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) / (double)length * 4 / 1.08;
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}
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}
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@ -1,5 +1,5 @@
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double audio_level(sample_t *samples, int length);
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double audio_mean_level(sample_t *samples, int length);
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typedef struct goertzel {
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double coeff;
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@ -64,7 +64,7 @@
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#define MAX_DISPLAY 1.4 /* something above speech level */
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#define DIALTONE_HZ 425.0 /* dial tone frequency */
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#define TX_PEAK_DIALTONE 1.0 /* dial tone peak FIXME: Not found in the specs! */
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#define SUPER_DURATION 0.25 /* duration of supervisory signal measurement */
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#define SUPER_BANDWIDTH 30.0 /* distance between two SAT tones, also bandwidth for goertzel filter */
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#define SUPER_PRINT 2 /* print supervisory signal measurement every 0.5 seconds */
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#define SUPER_LOST_COUNT 4 /* number of measures to loose supervisory signal */
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#define SUPER_DETECT_COUNT 6 /* number of measures to detect supervisory signal */
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@ -76,7 +76,7 @@ static double super_freq[5] = {
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3985.0, /* 0-Signal 2 */
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4015.0, /* 0-Signal 3 */
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4045.0, /* 0-Signal 4 */
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3900.0, /* noise level to check against */
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3895.0, /* noise level to check against */
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};
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/* table for fast sine generation */
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@ -129,8 +129,11 @@ int dsp_init_sender(nmt_t *nmt, double deviation_factor)
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return -EINVAL;
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}
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/* allocate ring buffer for supervisory signal detection */
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nmt->super_samples = (int)((double)nmt->sender.samplerate * SUPER_DURATION + 0.5);
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/* allocate ring buffer for SAT signal detection
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* the bandwidth of the Goertzel filter is the reciprocal of the duration
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* we half our bandwidth, so that other supervisory signals will be canceled out completely by goertzel filter
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*/
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nmt->super_samples = (int)((double)nmt->sender.samplerate * (1.0 / (SUPER_BANDWIDTH / 2)) + 0.5);
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spl = calloc(1, nmt->super_samples * sizeof(*spl));
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if (!spl) {
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PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
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@ -260,7 +263,7 @@ static void fsk_receive_bit(void *inst, int bit, double quality, double level)
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nmt_receive_frame(nmt, nmt->rx_frame, quality, level, frames_elapsed);
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}
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/* compare supervisory signal against noise floor on 3900 Hz */
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/* compare supervisory signal against noise floor around 3895 Hz */
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static void super_decode(nmt_t *nmt, sample_t *samples, int length)
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{
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double result[2], level, quality;
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@ -269,7 +272,7 @@ static void super_decode(nmt_t *nmt, sample_t *samples, int length)
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audio_goertzel(&nmt->super_goertzel[4], samples, length, 0, &result[1], 1); /* noise floor detection */
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/* normalize supervisory level */
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level = result[0] / 0.63662 / TX_PEAK_SUPER;
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level = result[0] / TX_PEAK_SUPER;
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quality = (result[0] - result[1]) / result[0];
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if (quality < 0)
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@ -5,6 +5,7 @@ noinst_PROGRAMS = \
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test_sendevolumenregler \
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test_compandor \
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test_emphasis \
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test_goertzel \
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test_dtmf \
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test_dms \
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test_sms \
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@ -150,3 +151,12 @@ test_v27scrambler_LDADD = \
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$(top_builddir)/src/libv27/libv27.a \
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-lm
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test_goertzel_SOURCES = test_goertzel.c dummy.c
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test_goertzel_LDADD = \
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$(COMMON_LA) \
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$(top_builddir)/src/libdebug/libdebug.a \
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$(top_builddir)/src/libgoertzel/libgoertzel.a \
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$(top_builddir)/src/liboptions/liboptions.a \
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-lm
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@ -0,0 +1,51 @@
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#include <stdio.h>
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#include <stdint.h>
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#include <math.h>
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#include <string.h>
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#include "../libsample/sample.h"
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#include "../libgoertzel/goertzel.h"
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#include "../libdebug/debug.h"
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#define level2db(level) (20 * log10(level))
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#define db2level(db) pow(10, (double)db / 20.0)
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#define SAMPLERATE 48000
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static void gen_samples(sample_t *samples, double freq)
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{
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int i;
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double value;
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for (i = 0; i < SAMPLERATE; i++) {
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value = cos(2.0 * M_PI * freq / (double)SAMPLERATE * (double)i);
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samples[i] = value;
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}
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}
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int num_kanal;
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int main(void)
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{
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goertzel_t goertzel;
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sample_t samples[SAMPLERATE];
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double frequency = 1000;
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double duration = 1.0/100.0;
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double level;
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double i;
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printf("testing goertzel with frequency %.1f and duration 1 / %.0f\n", frequency, 1.0 / duration);
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for (i = 700; i < 1301; i = i + 10) {
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gen_samples(samples, (double)i);
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audio_goertzel_init(&goertzel, frequency, SAMPLERATE);
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audio_goertzel(&goertzel, samples, SAMPLERATE * duration, 0, &level, 1);
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printf("%s%.0f Hz: %.1f dB", debug_db(level), i, level2db(level));
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if ((int)round(i) == (int)round(frequency))
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printf(" level=%.6f\n", level);
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else
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printf("\n");
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}
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return 0;
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}
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