NMT: Rework on FSK process, no more requirement for special sample rates
This commit is contained in:
Andreas Eversberg
parent
2d0bc92cde
commit
d447be5e26
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@ -287,10 +287,13 @@ static void dms_encode_dt(nmt_t *nmt, uint8_t d, uint8_t s, uint8_t n, uint8_t *
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#endif
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/* render wave form */
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fsk_render_frame(nmt, frame, 127, dms->frame_spl);
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dms->frame_length = fsk_render_frame(nmt, frame, 127, dms->frame_spl);
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dms->frame_valid = 1;
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dms->frame_pos = 0;
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dms->frame_length = nmt->samples_per_bit * 127;
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if (dms->frame_length > dms->frame_size) {
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PDEBUG(DDMS, DEBUG_ERROR, "Frame exceeds buffer, please fix!\n");
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abort();
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}
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}
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/* encode RR frame and schedule for next transmission */
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@ -331,10 +334,13 @@ static void dms_encode_rr(nmt_t *nmt, uint8_t d, uint8_t s, uint8_t n)
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#endif
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/* render wave form */
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fsk_render_frame(nmt, frame, 77, dms->frame_spl);
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dms->frame_length = fsk_render_frame(nmt, frame, 77, dms->frame_spl);
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dms->frame_valid = 1;
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dms->frame_pos = 0;
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dms->frame_length = nmt->samples_per_bit * 77;
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if (dms->frame_length > dms->frame_size) {
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PDEBUG(DDMS, DEBUG_ERROR, "Frame exceeds buffer, please fix!\n");
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abort();
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}
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}
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/* check if we have to transmit a frame and render it
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@ -26,8 +26,9 @@ typedef struct dms {
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/* DMS transmission */
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int frame_valid; /* set, if there is a valid frame in sample buffer */
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int16_t *frame_spl; /* 127 * fsk_bit_length */
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int frame_size; /* total size of buffer */
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int frame_pos; /* current sample position in frame_spl */
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int frame_length; /* number of samples in frame_spl */
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int frame_length; /* number of samples currently in frame_spl */
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uint16_t rx_sync; /* shift register to detect sync */
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double rx_sync_level[256]; /* level infos */
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double rx_sync_quality[256]; /* quality infos */
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142
src/nmt/dsp.c
142
src/nmt/dsp.c
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@ -55,7 +55,7 @@
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#define MUTE_DURATION 0.280 /* a tiny bit more than two frames */
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/* two signaling tones */
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static double fsk_bits[2] = {
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static double fsk_freq[2] = {
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1800.0,
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1200.0,
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};
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@ -70,8 +70,9 @@ static double super_freq[5] = {
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};
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/* table for fast sine generation */
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int dsp_sine_super[256];
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int dsp_sine_dialtone[256];
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uint16_t dsp_tone_bit[2][2][256]; /* polarity, bit, phase */
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uint16_t dsp_sine_super[256];
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uint16_t dsp_sine_dialtone[256];
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/* global init for FSK */
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void dsp_init(void)
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@ -82,8 +83,17 @@ void dsp_init(void)
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PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for supervisory signal.\n");
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for (i = 0; i < 256; i++) {
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s = sin((double)i / 256.0 * 2.0 * PI);
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/* supervisor sine */
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dsp_sine_super[i] = (int)(s * TX_PEAK_SUPER);
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/* dialtone sine */
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dsp_sine_dialtone[i] = (int)(s * TX_PEAK_DIALTONE);
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/* bit(1) 1 cycle */
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dsp_tone_bit[0][1][i] = (int)(s * TX_PEAK_FSK);
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dsp_tone_bit[1][1][i] = (int)(-s * TX_PEAK_FSK);
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/* bit(0) 1.5 cycles */
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s = sin((double)i / 256.0 * 3.0 * PI);
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dsp_tone_bit[0][0][i] = (int)(s * TX_PEAK_FSK);
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dsp_tone_bit[1][0][i] = (int)(-s * TX_PEAK_FSK);
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}
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}
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@ -97,46 +107,24 @@ int dsp_init_sender(nmt_t *nmt)
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/* attack (3ms) and recovery time (13.5ms) according to NMT specs */
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init_compandor(&nmt->cstate, 8000, 3.0, 13.5, COMPANDOR_0DB);
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if ((nmt->sender.samplerate % (BIT_RATE * STEPS_PER_BIT))) {
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PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be a multiple of %d bits per second.\n", BIT_RATE * STEPS_PER_BIT);
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return -EINVAL;
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}
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/* this should not happen. it is implied by previous check */
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if (nmt->supervisory && nmt->sender.samplerate < 12000) {
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PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 12000 Hz to process supervisory signal.\n");
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PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 12000 Hz to process FSK+supervisory signal.\n");
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return -EINVAL;
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}
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for Transceiver.\n");
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/* allocate sample for 2 bits with 2 polarities */
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nmt->samples_per_bit = nmt->sender.samplerate / BIT_RATE;
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PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per bit duration.\n", nmt->samples_per_bit);
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nmt->fsk_filter_step = nmt->samples_per_bit / STEPS_PER_BIT;
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PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per filter step.\n", nmt->fsk_filter_step);
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nmt->fsk_sine[0][0] = calloc(4, nmt->samples_per_bit * sizeof(int16_t));
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nmt->fsk_sine[0][1] = nmt->fsk_sine[0][0] + nmt->samples_per_bit;
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nmt->fsk_sine[1][0] = nmt->fsk_sine[0][1] + nmt->samples_per_bit;
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nmt->fsk_sine[1][1] = nmt->fsk_sine[1][0] + nmt->samples_per_bit;
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if (!nmt->fsk_sine[0][0]) {
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PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
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return -ENOMEM;
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}
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PDEBUG(DDSP, DEBUG_DEBUG, "Using FSK level of %.0f (3.5 KHz deviation @ 1500 Hz)\n", TX_PEAK_FSK);
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PDEBUG(DDSP, DEBUG_DEBUG, "Using Supervisory level of %.0f (0.3 KHz deviation @ 4015 Hz)\n", TX_PEAK_SUPER);
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/* generate sines */
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for (i = 0; i < nmt->samples_per_bit; i++) {
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nmt->fsk_sine[0][0][i] = TX_PEAK_FSK * sin(3.0 * PI * (double)i / (double)nmt->samples_per_bit); /* 1.5 waves */
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nmt->fsk_sine[0][1][i] = TX_PEAK_FSK * sin(2.0 * PI * (double)i / (double)nmt->samples_per_bit); /* 1 wave */
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nmt->fsk_sine[1][0][i] = -nmt->fsk_sine[0][0][i];
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nmt->fsk_sine[1][1][i] = -nmt->fsk_sine[0][1][i];
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}
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nmt->fsk_samples_per_bit = (double)nmt->sender.samplerate / (double)BIT_RATE;
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nmt->fsk_bits_per_sample = 1.0 / nmt->fsk_samples_per_bit;
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PDEBUG(DDSP, DEBUG_DEBUG, "Use %.4f samples for full bit duration @ %d.\n", nmt->fsk_samples_per_bit, nmt->sender.samplerate);
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/* allocate ring buffers, one bit duration */
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spl = calloc(1, nmt->samples_per_bit * sizeof(*spl));
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nmt->fsk_filter_size = floor(nmt->fsk_samples_per_bit); /* buffer holds one bit (rounded down) */
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spl = calloc(1, nmt->fsk_filter_size * sizeof(*spl));
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if (!spl) {
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PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
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return -ENOMEM;
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@ -144,16 +132,18 @@ int dsp_init_sender(nmt_t *nmt)
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nmt->fsk_filter_spl = spl;
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nmt->fsk_filter_bit = -1;
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/* allocate transmit buffer for a complete frame */
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spl = calloc(166, nmt->samples_per_bit * sizeof(*spl));
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/* allocate transmit buffer for a complete frame, add 10 to be safe */
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nmt->frame_size = 166.0 * (double)nmt->fsk_samples_per_bit + 10;
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spl = calloc(nmt->frame_size, sizeof(*spl));
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if (!spl) {
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PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
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return -ENOMEM;
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}
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nmt->frame_spl = spl;
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/* allocate DMS transmit buffer for a complete frame */
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spl = calloc(127, nmt->samples_per_bit * sizeof(*spl));
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/* allocate DMS transmit buffer for a complete frame, add 10 to be safe */
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nmt->dms.frame_size = 127.0 * (double)nmt->fsk_samples_per_bit + 10;
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spl = calloc(nmt->dms.frame_size, sizeof(*spl));
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if (!spl) {
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PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
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return -ENOMEM;
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@ -171,10 +161,12 @@ int dsp_init_sender(nmt_t *nmt)
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/* count symbols */
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for (i = 0; i < 2; i++) {
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coeff = 2.0 * cos(2.0 * PI * fsk_bits[i] / (double)nmt->sender.samplerate);
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coeff = 2.0 * cos(2.0 * PI * fsk_freq[i] / (double)nmt->sender.samplerate);
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nmt->fsk_coeff[i] = coeff * 32768.0;
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PDEBUG(DDSP, DEBUG_DEBUG, "coeff[%d] = %d\n", i, (int)nmt->fsk_coeff[i]);
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PDEBUG(DDSP, DEBUG_DEBUG, "fsk_coeff[%d] = %d\n", i, (int)nmt->fsk_coeff[i]);
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}
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nmt->fsk_phaseshift256 = 256.0 / nmt->fsk_samples_per_bit;
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PDEBUG(DDSP, DEBUG_DEBUG, "fsk_phaseshift = %.4f\n", nmt->fsk_phaseshift256);
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/* count supervidory tones */
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for (i = 0; i < 5; i++) {
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@ -184,13 +176,14 @@ int dsp_init_sender(nmt_t *nmt)
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if (i < 4) {
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nmt->super_phaseshift256[i] = 256.0 / ((double)nmt->sender.samplerate / super_freq[i]);
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PDEBUG(DDSP, DEBUG_DEBUG, "phaseshift_super[%d] = %.4f\n", i, nmt->super_phaseshift256[i]);
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PDEBUG(DDSP, DEBUG_DEBUG, "super_phaseshift[%d] = %.4f\n", i, nmt->super_phaseshift256[i]);
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}
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}
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super_reset(nmt);
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/* dial tone */
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nmt->dial_phaseshift256 = 256.0 / ((double)nmt->sender.samplerate / DIALTONE_HZ);
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PDEBUG(DDSP, DEBUG_DEBUG, "dial_phaseshift = %.4f\n", nmt->dial_phaseshift256);
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/* dtmf */
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dtmf_init(&nmt->dtmf, 8000);
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@ -254,8 +247,8 @@ static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level)
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return;
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/* sync time */
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nmt->rx_sample_count_last = nmt->rx_sample_count_current;
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nmt->rx_sample_count_current = nmt->rx_sample_count - nmt->samples_per_bit * 26;
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nmt->rx_bits_count_last = nmt->rx_bits_count_current;
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nmt->rx_bits_count_current = nmt->rx_bits_count - 26.0;
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/* rest sync register */
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nmt->fsk_filter_sync = 0;
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@ -287,7 +280,7 @@ static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level)
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level /= 140.0; quality /= 140.0;
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/* send telegramm */
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frames_elapsed = (double)(nmt->rx_sample_count_current - nmt->rx_sample_count_last) / (double)(nmt->samples_per_bit * 166);
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frames_elapsed = (nmt->rx_bits_count_current - nmt->rx_bits_count_last) / 166.0;
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/* convert level so that received level at TX_PEAK_FSK results in 1.0 (100%) */
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nmt_receive_frame(nmt, nmt->fsk_filter_frame, quality, level, frames_elapsed);
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}
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@ -297,8 +290,8 @@ static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level)
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//#define DEBUG_QUALITY
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/* Filter one chunk of audio an detect tone, quality and loss of signal.
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* The chunk is a window of 10ms. This window slides over audio stream
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* and is processed every 1ms. (one step) */
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* The chunk is a window of 1/1200s. This window slides over audio stream
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* and is processed every 1/12000s. (one step) */
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static inline void fsk_decode_step(nmt_t *nmt, int pos)
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{
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double level, result[2], softbit, quality;
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@ -306,11 +299,11 @@ static inline void fsk_decode_step(nmt_t *nmt, int pos)
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int16_t *spl;
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int bit;
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max = nmt->samples_per_bit;
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max = nmt->fsk_filter_size;
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spl = nmt->fsk_filter_spl;
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/* count time in samples*/
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nmt->rx_sample_count += nmt->fsk_filter_step;
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/* count time in bits */
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nmt->rx_bits_count += 0.1;
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level = audio_level(spl, max);
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/* limit level to prevent division by zero */
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@ -430,7 +423,8 @@ void sender_receive(sender_t *sender, int16_t *samples, int length)
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{
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nmt_t *nmt = (nmt_t *) sender;
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int16_t *spl;
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int max, pos, step;
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int max, pos;
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double step, bps;
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int i;
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/* write received samples to decode buffer */
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@ -448,26 +442,32 @@ void sender_receive(sender_t *sender, int16_t *samples, int length)
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nmt->super_filter_pos = pos;
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/* write received samples to decode buffer */
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max = nmt->samples_per_bit;
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max = nmt->fsk_filter_size;
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pos = nmt->fsk_filter_pos;
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step = nmt->fsk_filter_step;
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bps = nmt->fsk_bits_per_sample;
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spl = nmt->fsk_filter_spl;
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for (i = 0; i < length; i++) {
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#ifdef DEBUG_MODULATOR
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printf("|%s|\n", debug_amplitude((double)samples[i] / TX_PEAK_FSK / 2.0));
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#endif
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/* write into ring buffer */
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spl[pos++] = samples[i];
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if (nmt->fsk_filter_mute) {
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if (pos == max)
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pos = 0;
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/* muting audio while receiving frame */
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if (nmt->fsk_filter_mute && !nmt->sender.loopback) {
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samples[i] = 0;
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nmt->fsk_filter_mute--;
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}
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if (pos == max)
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pos = 0;
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/* if filter step has been reched */
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if (!(pos % step)) {
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/* if 1/10th of a bit duration is reached, decode buffer */
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step += bps;
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if (step >= 0.1) {
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step -= 0.1;
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fsk_decode_step(nmt, pos);
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}
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}
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nmt->fsk_filter_step = step;
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nmt->fsk_filter_pos = pos;
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if ((nmt->dsp_mode == DSP_MODE_AUDIO || nmt->dsp_mode == DSP_MODE_DTMF)
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@ -495,21 +495,32 @@ void sender_receive(sender_t *sender, int16_t *samples, int length)
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}
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/* render frame */
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void fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample)
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int fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample)
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{
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int bit, polarity;
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int i;
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double phaseshift, phase;
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int count = 0, i;
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polarity = nmt->fsk_polarity;
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phaseshift = nmt->fsk_phaseshift256;
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phase = nmt->fsk_phase256;
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for (i = 0; i < length; i++) {
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bit = (frame[i] == '1');
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memcpy(sample, nmt->fsk_sine[polarity][bit], nmt->samples_per_bit * sizeof(*sample));
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sample += nmt->samples_per_bit;
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do {
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*sample++ = dsp_tone_bit[polarity][bit][((uint8_t)phase) & 0xff];
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count++;
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phase += phaseshift;
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} while (phase < 256.0);
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phase -= 256.0;
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/* flip polarity when we have 1.5 sine waves */
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if (bit == 0)
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polarity = 1 - polarity;
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}
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nmt->fsk_phase256 = phase;
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nmt->fsk_polarity = polarity;
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/* return number of samples created for frame */
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return count;
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}
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static int fsk_frame(nmt_t *nmt, int16_t *samples, int length)
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@ -520,21 +531,24 @@ static int fsk_frame(nmt_t *nmt, int16_t *samples, int length)
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int count, max;
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next_frame:
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if (!nmt->frame) {
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if (!nmt->frame_length) {
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/* request frame */
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frame = nmt_get_frame(nmt);
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if (!frame) {
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Stop sending frames.\n");
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return length;
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}
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nmt->frame = 1;
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nmt->frame_pos = 0;
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/* render frame */
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fsk_render_frame(nmt, frame, 166, nmt->frame_spl);
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nmt->frame_length = fsk_render_frame(nmt, frame, 166, nmt->frame_spl);
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nmt->frame_pos = 0;
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if (nmt->frame_length > nmt->frame_size) {
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PDEBUG_CHAN(DDSP, DEBUG_ERROR, "Frame exceeds buffer, please fix!\n");
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abort();
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}
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}
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/* send audio from frame */
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max = nmt->samples_per_bit * 166;
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max = nmt->frame_length;
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count = max - nmt->frame_pos;
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if (count > length)
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count = length;
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@ -546,7 +560,7 @@ next_frame:
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nmt->frame_pos += count;
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/* check for end of telegramm */
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if (nmt->frame_pos == max) {
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nmt->frame = 0;
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nmt->frame_length = 0;
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/* we need more ? */
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if (length)
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goto next_frame;
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@ -666,7 +680,7 @@ void nmt_set_dsp_mode(nmt_t *nmt, enum dsp_mode mode)
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{
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/* reset telegramm */
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if (mode == DSP_MODE_FRAME && nmt->dsp_mode != mode)
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nmt->frame = 0;
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nmt->frame_length = 0;
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "DSP mode %s -> %s\n", nmt_dsp_mode_name(nmt->dsp_mode), nmt_dsp_mode_name(mode));
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nmt->dsp_mode = mode;
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@ -2,7 +2,7 @@
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void dsp_init(void);
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int dsp_init_sender(nmt_t *nmt);
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void dsp_cleanup_sender(nmt_t *nmt);
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void fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample);
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int fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample);
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void nmt_set_dsp_mode(nmt_t *nmt, enum dsp_mode mode);
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void super_reset(nmt_t *nmt);
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@ -91,17 +91,17 @@ typedef struct nmt {
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/* dsp states */
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enum dsp_mode dsp_mode; /* current mode: audio, durable tone 0 or 1, paging */
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int samples_per_bit; /* number of samples for one bit (1200 Baud) */
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double fsk_samples_per_bit; /* number of samples for one bit (1200 Baud) */
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double fsk_bits_per_sample; /* fraction of a bit per sample */
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int super_samples; /* number of samples in buffer for supervisory detection */
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int fsk_coeff[2]; /* coefficient k = 2*cos(2*PI*f/samplerate), k << 15 */
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int super_coeff[5]; /* coefficient for supervisory signal */
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int16_t *fsk_sine[2][2]; /* 4 pointers to 4 precalc. sine curves */
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int fsk_polarity; /* current polarity state of bit */
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int samples_per_chunk; /* how many samples lasts one chunk */
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int16_t *fsk_filter_spl; /* array with samples_per_chunk */
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int fsk_filter_pos; /* current sample position in filter_spl */
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int fsk_filter_step; /* number of samples for each analyzation */
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int fsk_filter_bit; /* last bit, so we detect a bit change */
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int16_t *fsk_filter_spl; /* array to hold ring buffer for bit decoding */
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int fsk_filter_size; /* size of ring buffer */
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int fsk_filter_pos; /* position to write next sample */
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double fsk_filter_step; /* counts bit duration, to trigger decoding every 10th bit */
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int fsk_filter_bit; /* last bit state, so we detect a bit change */
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int fsk_filter_sample; /* count until it is time to sample bit */
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uint16_t fsk_filter_sync; /* shift register to detect sync */
|
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int fsk_filter_in_sync; /* if we are in sync and receive bits */
|
||||
|
|
@ -116,12 +116,15 @@ typedef struct nmt {
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|||
double super_phase256; /* current phase */
|
||||
double dial_phaseshift256; /* how much the phase of sine wave changes per sample */
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||||
double dial_phase256; /* current phase */
|
||||
int frame; /* set, if there is a valid frame */
|
||||
int16_t *frame_spl; /* 166 * fsk_bit_length */
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||||
double fsk_phaseshift256; /* how much the phase of fsk synbol changes per sample */
|
||||
double fsk_phase256; /* current phase */
|
||||
int16_t *frame_spl; /* samples to store a complete rendered frame */
|
||||
int frame_size; /* total size of sample buffer */
|
||||
int frame_length; /* current length of data in sample buffer */
|
||||
int frame_pos; /* current sample position in frame_spl */
|
||||
uint64_t rx_sample_count; /* sample counter */
|
||||
uint64_t rx_sample_count_current;/* sample counter of current frame */
|
||||
uint64_t rx_sample_count_last; /* sample counter of last frame */
|
||||
double rx_bits_count; /* sample counter */
|
||||
double rx_bits_count_current; /* sample counter of current frame */
|
||||
double rx_bits_count_last; /* sample counter of last frame */
|
||||
int super_detected; /* current detection state flag */
|
||||
int super_detect_count; /* current number of consecutive detections/losses */
|
||||
|
||||
|
|
|
|||
|
|
@ -55,12 +55,14 @@ void dms_all_sent(nmt_t *nmt)
|
|||
}
|
||||
|
||||
/* receive bits from DMS */
|
||||
void fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample)
|
||||
int fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample)
|
||||
{
|
||||
printf("(getting %d bits from DMS layer)\n", length);
|
||||
|
||||
memcpy(current_bits, frame, length);
|
||||
current_bit_count = length;
|
||||
|
||||
return nmt->fsk_samples_per_bit * length;
|
||||
}
|
||||
|
||||
nmt_t *alloc_nmt(void)
|
||||
|
|
@ -69,8 +71,9 @@ nmt_t *alloc_nmt(void)
|
|||
|
||||
nmt = calloc(sizeof(*nmt), 1);
|
||||
dms_init_sender(nmt);
|
||||
nmt->dms.frame_spl = calloc(1000000, 1);
|
||||
nmt->samples_per_bit = 40;
|
||||
nmt->fsk_samples_per_bit = 40;
|
||||
nmt->dms.frame_size = nmt->fsk_samples_per_bit * 127 + 10;
|
||||
nmt->dms.frame_spl = calloc(nmt->dms.frame_size, sizeof(nmt->dms.frame_spl[0]));
|
||||
|
||||
dms_reset(nmt);
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue