diff --git a/src/nmt/dms.c b/src/nmt/dms.c index c871b5a..1027e34 100644 --- a/src/nmt/dms.c +++ b/src/nmt/dms.c @@ -287,10 +287,13 @@ static void dms_encode_dt(nmt_t *nmt, uint8_t d, uint8_t s, uint8_t n, uint8_t * #endif /* render wave form */ - fsk_render_frame(nmt, frame, 127, dms->frame_spl); + dms->frame_length = fsk_render_frame(nmt, frame, 127, dms->frame_spl); dms->frame_valid = 1; dms->frame_pos = 0; - dms->frame_length = nmt->samples_per_bit * 127; + if (dms->frame_length > dms->frame_size) { + PDEBUG(DDMS, DEBUG_ERROR, "Frame exceeds buffer, please fix!\n"); + abort(); + } } /* encode RR frame and schedule for next transmission */ @@ -331,10 +334,13 @@ static void dms_encode_rr(nmt_t *nmt, uint8_t d, uint8_t s, uint8_t n) #endif /* render wave form */ - fsk_render_frame(nmt, frame, 77, dms->frame_spl); + dms->frame_length = fsk_render_frame(nmt, frame, 77, dms->frame_spl); dms->frame_valid = 1; dms->frame_pos = 0; - dms->frame_length = nmt->samples_per_bit * 77; + if (dms->frame_length > dms->frame_size) { + PDEBUG(DDMS, DEBUG_ERROR, "Frame exceeds buffer, please fix!\n"); + abort(); + } } /* check if we have to transmit a frame and render it diff --git a/src/nmt/dms.h b/src/nmt/dms.h index d72af8e..b03b91e 100644 --- a/src/nmt/dms.h +++ b/src/nmt/dms.h @@ -26,8 +26,9 @@ typedef struct dms { /* DMS transmission */ int frame_valid; /* set, if there is a valid frame in sample buffer */ int16_t *frame_spl; /* 127 * fsk_bit_length */ + int frame_size; /* total size of buffer */ int frame_pos; /* current sample position in frame_spl */ - int frame_length; /* number of samples in frame_spl */ + int frame_length; /* number of samples currently in frame_spl */ uint16_t rx_sync; /* shift register to detect sync */ double rx_sync_level[256]; /* level infos */ double rx_sync_quality[256]; /* quality infos */ diff --git a/src/nmt/dsp.c b/src/nmt/dsp.c index 9929a32..f0069e8 100644 --- a/src/nmt/dsp.c +++ b/src/nmt/dsp.c @@ -55,7 +55,7 @@ #define MUTE_DURATION 0.280 /* a tiny bit more than two frames */ /* two signaling tones */ -static double fsk_bits[2] = { +static double fsk_freq[2] = { 1800.0, 1200.0, }; @@ -70,8 +70,9 @@ static double super_freq[5] = { }; /* table for fast sine generation */ -int dsp_sine_super[256]; -int dsp_sine_dialtone[256]; +uint16_t dsp_tone_bit[2][2][256]; /* polarity, bit, phase */ +uint16_t dsp_sine_super[256]; +uint16_t dsp_sine_dialtone[256]; /* global init for FSK */ void dsp_init(void) @@ -82,8 +83,17 @@ void dsp_init(void) PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for supervisory signal.\n"); for (i = 0; i < 256; i++) { s = sin((double)i / 256.0 * 2.0 * PI); + /* supervisor sine */ dsp_sine_super[i] = (int)(s * TX_PEAK_SUPER); + /* dialtone sine */ dsp_sine_dialtone[i] = (int)(s * TX_PEAK_DIALTONE); + /* bit(1) 1 cycle */ + dsp_tone_bit[0][1][i] = (int)(s * TX_PEAK_FSK); + dsp_tone_bit[1][1][i] = (int)(-s * TX_PEAK_FSK); + /* bit(0) 1.5 cycles */ + s = sin((double)i / 256.0 * 3.0 * PI); + dsp_tone_bit[0][0][i] = (int)(s * TX_PEAK_FSK); + dsp_tone_bit[1][0][i] = (int)(-s * TX_PEAK_FSK); } } @@ -97,46 +107,24 @@ int dsp_init_sender(nmt_t *nmt) /* attack (3ms) and recovery time (13.5ms) according to NMT specs */ init_compandor(&nmt->cstate, 8000, 3.0, 13.5, COMPANDOR_0DB); - if ((nmt->sender.samplerate % (BIT_RATE * STEPS_PER_BIT))) { - PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be a multiple of %d bits per second.\n", BIT_RATE * STEPS_PER_BIT); - return -EINVAL; - } - /* this should not happen. it is implied by previous check */ if (nmt->supervisory && nmt->sender.samplerate < 12000) { - PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 12000 Hz to process supervisory signal.\n"); + PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 12000 Hz to process FSK+supervisory signal.\n"); return -EINVAL; } PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for Transceiver.\n"); - /* allocate sample for 2 bits with 2 polarities */ - nmt->samples_per_bit = nmt->sender.samplerate / BIT_RATE; - PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per bit duration.\n", nmt->samples_per_bit); - nmt->fsk_filter_step = nmt->samples_per_bit / STEPS_PER_BIT; - PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per filter step.\n", nmt->fsk_filter_step); - nmt->fsk_sine[0][0] = calloc(4, nmt->samples_per_bit * sizeof(int16_t)); - nmt->fsk_sine[0][1] = nmt->fsk_sine[0][0] + nmt->samples_per_bit; - nmt->fsk_sine[1][0] = nmt->fsk_sine[0][1] + nmt->samples_per_bit; - nmt->fsk_sine[1][1] = nmt->fsk_sine[1][0] + nmt->samples_per_bit; - if (!nmt->fsk_sine[0][0]) { - PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n"); - return -ENOMEM; - } - PDEBUG(DDSP, DEBUG_DEBUG, "Using FSK level of %.0f (3.5 KHz deviation @ 1500 Hz)\n", TX_PEAK_FSK); PDEBUG(DDSP, DEBUG_DEBUG, "Using Supervisory level of %.0f (0.3 KHz deviation @ 4015 Hz)\n", TX_PEAK_SUPER); - /* generate sines */ - for (i = 0; i < nmt->samples_per_bit; i++) { - nmt->fsk_sine[0][0][i] = TX_PEAK_FSK * sin(3.0 * PI * (double)i / (double)nmt->samples_per_bit); /* 1.5 waves */ - nmt->fsk_sine[0][1][i] = TX_PEAK_FSK * sin(2.0 * PI * (double)i / (double)nmt->samples_per_bit); /* 1 wave */ - nmt->fsk_sine[1][0][i] = -nmt->fsk_sine[0][0][i]; - nmt->fsk_sine[1][1][i] = -nmt->fsk_sine[0][1][i]; - } + nmt->fsk_samples_per_bit = (double)nmt->sender.samplerate / (double)BIT_RATE; + nmt->fsk_bits_per_sample = 1.0 / nmt->fsk_samples_per_bit; + PDEBUG(DDSP, DEBUG_DEBUG, "Use %.4f samples for full bit duration @ %d.\n", nmt->fsk_samples_per_bit, nmt->sender.samplerate); /* allocate ring buffers, one bit duration */ - spl = calloc(1, nmt->samples_per_bit * sizeof(*spl)); + nmt->fsk_filter_size = floor(nmt->fsk_samples_per_bit); /* buffer holds one bit (rounded down) */ + spl = calloc(1, nmt->fsk_filter_size * sizeof(*spl)); if (!spl) { PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n"); return -ENOMEM; @@ -144,16 +132,18 @@ int dsp_init_sender(nmt_t *nmt) nmt->fsk_filter_spl = spl; nmt->fsk_filter_bit = -1; - /* allocate transmit buffer for a complete frame */ - spl = calloc(166, nmt->samples_per_bit * sizeof(*spl)); + /* allocate transmit buffer for a complete frame, add 10 to be safe */ + nmt->frame_size = 166.0 * (double)nmt->fsk_samples_per_bit + 10; + spl = calloc(nmt->frame_size, sizeof(*spl)); if (!spl) { PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n"); return -ENOMEM; } nmt->frame_spl = spl; - /* allocate DMS transmit buffer for a complete frame */ - spl = calloc(127, nmt->samples_per_bit * sizeof(*spl)); + /* allocate DMS transmit buffer for a complete frame, add 10 to be safe */ + nmt->dms.frame_size = 127.0 * (double)nmt->fsk_samples_per_bit + 10; + spl = calloc(nmt->dms.frame_size, sizeof(*spl)); if (!spl) { PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n"); return -ENOMEM; @@ -171,10 +161,12 @@ int dsp_init_sender(nmt_t *nmt) /* count symbols */ for (i = 0; i < 2; i++) { - coeff = 2.0 * cos(2.0 * PI * fsk_bits[i] / (double)nmt->sender.samplerate); + coeff = 2.0 * cos(2.0 * PI * fsk_freq[i] / (double)nmt->sender.samplerate); nmt->fsk_coeff[i] = coeff * 32768.0; - PDEBUG(DDSP, DEBUG_DEBUG, "coeff[%d] = %d\n", i, (int)nmt->fsk_coeff[i]); + PDEBUG(DDSP, DEBUG_DEBUG, "fsk_coeff[%d] = %d\n", i, (int)nmt->fsk_coeff[i]); } + nmt->fsk_phaseshift256 = 256.0 / nmt->fsk_samples_per_bit; + PDEBUG(DDSP, DEBUG_DEBUG, "fsk_phaseshift = %.4f\n", nmt->fsk_phaseshift256); /* count supervidory tones */ for (i = 0; i < 5; i++) { @@ -184,13 +176,14 @@ int dsp_init_sender(nmt_t *nmt) if (i < 4) { nmt->super_phaseshift256[i] = 256.0 / ((double)nmt->sender.samplerate / super_freq[i]); - PDEBUG(DDSP, DEBUG_DEBUG, "phaseshift_super[%d] = %.4f\n", i, nmt->super_phaseshift256[i]); + PDEBUG(DDSP, DEBUG_DEBUG, "super_phaseshift[%d] = %.4f\n", i, nmt->super_phaseshift256[i]); } } super_reset(nmt); /* dial tone */ nmt->dial_phaseshift256 = 256.0 / ((double)nmt->sender.samplerate / DIALTONE_HZ); + PDEBUG(DDSP, DEBUG_DEBUG, "dial_phaseshift = %.4f\n", nmt->dial_phaseshift256); /* dtmf */ dtmf_init(&nmt->dtmf, 8000); @@ -254,8 +247,8 @@ static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level) return; /* sync time */ - nmt->rx_sample_count_last = nmt->rx_sample_count_current; - nmt->rx_sample_count_current = nmt->rx_sample_count - nmt->samples_per_bit * 26; + nmt->rx_bits_count_last = nmt->rx_bits_count_current; + nmt->rx_bits_count_current = nmt->rx_bits_count - 26.0; /* rest sync register */ nmt->fsk_filter_sync = 0; @@ -287,7 +280,7 @@ static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level) level /= 140.0; quality /= 140.0; /* send telegramm */ - frames_elapsed = (double)(nmt->rx_sample_count_current - nmt->rx_sample_count_last) / (double)(nmt->samples_per_bit * 166); + frames_elapsed = (nmt->rx_bits_count_current - nmt->rx_bits_count_last) / 166.0; /* convert level so that received level at TX_PEAK_FSK results in 1.0 (100%) */ nmt_receive_frame(nmt, nmt->fsk_filter_frame, quality, level, frames_elapsed); } @@ -297,8 +290,8 @@ static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level) //#define DEBUG_QUALITY /* Filter one chunk of audio an detect tone, quality and loss of signal. - * The chunk is a window of 10ms. This window slides over audio stream - * and is processed every 1ms. (one step) */ + * The chunk is a window of 1/1200s. This window slides over audio stream + * and is processed every 1/12000s. (one step) */ static inline void fsk_decode_step(nmt_t *nmt, int pos) { double level, result[2], softbit, quality; @@ -306,11 +299,11 @@ static inline void fsk_decode_step(nmt_t *nmt, int pos) int16_t *spl; int bit; - max = nmt->samples_per_bit; + max = nmt->fsk_filter_size; spl = nmt->fsk_filter_spl; - /* count time in samples*/ - nmt->rx_sample_count += nmt->fsk_filter_step; + /* count time in bits */ + nmt->rx_bits_count += 0.1; level = audio_level(spl, max); /* limit level to prevent division by zero */ @@ -430,7 +423,8 @@ void sender_receive(sender_t *sender, int16_t *samples, int length) { nmt_t *nmt = (nmt_t *) sender; int16_t *spl; - int max, pos, step; + int max, pos; + double step, bps; int i; /* write received samples to decode buffer */ @@ -448,26 +442,32 @@ void sender_receive(sender_t *sender, int16_t *samples, int length) nmt->super_filter_pos = pos; /* write received samples to decode buffer */ - max = nmt->samples_per_bit; + max = nmt->fsk_filter_size; pos = nmt->fsk_filter_pos; step = nmt->fsk_filter_step; + bps = nmt->fsk_bits_per_sample; spl = nmt->fsk_filter_spl; for (i = 0; i < length; i++) { #ifdef DEBUG_MODULATOR printf("|%s|\n", debug_amplitude((double)samples[i] / TX_PEAK_FSK / 2.0)); #endif + /* write into ring buffer */ spl[pos++] = samples[i]; - if (nmt->fsk_filter_mute) { + if (pos == max) + pos = 0; + /* muting audio while receiving frame */ + if (nmt->fsk_filter_mute && !nmt->sender.loopback) { samples[i] = 0; nmt->fsk_filter_mute--; } - if (pos == max) - pos = 0; - /* if filter step has been reched */ - if (!(pos % step)) { + /* if 1/10th of a bit duration is reached, decode buffer */ + step += bps; + if (step >= 0.1) { + step -= 0.1; fsk_decode_step(nmt, pos); } } + nmt->fsk_filter_step = step; nmt->fsk_filter_pos = pos; if ((nmt->dsp_mode == DSP_MODE_AUDIO || nmt->dsp_mode == DSP_MODE_DTMF) @@ -495,21 +495,32 @@ void sender_receive(sender_t *sender, int16_t *samples, int length) } /* render frame */ -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) { int bit, polarity; - int i; + double phaseshift, phase; + int count = 0, i; polarity = nmt->fsk_polarity; + phaseshift = nmt->fsk_phaseshift256; + phase = nmt->fsk_phase256; for (i = 0; i < length; i++) { bit = (frame[i] == '1'); - memcpy(sample, nmt->fsk_sine[polarity][bit], nmt->samples_per_bit * sizeof(*sample)); - sample += nmt->samples_per_bit; + do { + *sample++ = dsp_tone_bit[polarity][bit][((uint8_t)phase) & 0xff]; + count++; + phase += phaseshift; + } while (phase < 256.0); + phase -= 256.0; /* flip polarity when we have 1.5 sine waves */ if (bit == 0) polarity = 1 - polarity; } + nmt->fsk_phase256 = phase; nmt->fsk_polarity = polarity; + + /* return number of samples created for frame */ + return count; } static int fsk_frame(nmt_t *nmt, int16_t *samples, int length) @@ -520,21 +531,24 @@ static int fsk_frame(nmt_t *nmt, int16_t *samples, int length) int count, max; next_frame: - if (!nmt->frame) { + if (!nmt->frame_length) { /* request frame */ frame = nmt_get_frame(nmt); if (!frame) { PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Stop sending frames.\n"); return length; } - nmt->frame = 1; - nmt->frame_pos = 0; /* render frame */ - fsk_render_frame(nmt, frame, 166, nmt->frame_spl); + nmt->frame_length = fsk_render_frame(nmt, frame, 166, nmt->frame_spl); + nmt->frame_pos = 0; + if (nmt->frame_length > nmt->frame_size) { + PDEBUG_CHAN(DDSP, DEBUG_ERROR, "Frame exceeds buffer, please fix!\n"); + abort(); + } } /* send audio from frame */ - max = nmt->samples_per_bit * 166; + max = nmt->frame_length; count = max - nmt->frame_pos; if (count > length) count = length; @@ -546,7 +560,7 @@ next_frame: nmt->frame_pos += count; /* check for end of telegramm */ if (nmt->frame_pos == max) { - nmt->frame = 0; + nmt->frame_length = 0; /* we need more ? */ if (length) goto next_frame; @@ -666,7 +680,7 @@ void nmt_set_dsp_mode(nmt_t *nmt, enum dsp_mode mode) { /* reset telegramm */ if (mode == DSP_MODE_FRAME && nmt->dsp_mode != mode) - nmt->frame = 0; + nmt->frame_length = 0; PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "DSP mode %s -> %s\n", nmt_dsp_mode_name(nmt->dsp_mode), nmt_dsp_mode_name(mode)); nmt->dsp_mode = mode; diff --git a/src/nmt/dsp.h b/src/nmt/dsp.h index b6fbc60..de5dcf4 100644 --- a/src/nmt/dsp.h +++ b/src/nmt/dsp.h @@ -2,7 +2,7 @@ void dsp_init(void); int dsp_init_sender(nmt_t *nmt); void dsp_cleanup_sender(nmt_t *nmt); -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); void nmt_set_dsp_mode(nmt_t *nmt, enum dsp_mode mode); void super_reset(nmt_t *nmt); diff --git a/src/nmt/nmt.h b/src/nmt/nmt.h index d1af0f8..919e0aa 100644 --- a/src/nmt/nmt.h +++ b/src/nmt/nmt.h @@ -91,17 +91,17 @@ typedef struct nmt { /* dsp states */ enum dsp_mode dsp_mode; /* current mode: audio, durable tone 0 or 1, paging */ - int samples_per_bit; /* number of samples for one bit (1200 Baud) */ + double fsk_samples_per_bit; /* number of samples for one bit (1200 Baud) */ + double fsk_bits_per_sample; /* fraction of a bit per sample */ int super_samples; /* number of samples in buffer for supervisory detection */ int fsk_coeff[2]; /* coefficient k = 2*cos(2*PI*f/samplerate), k << 15 */ int super_coeff[5]; /* coefficient for supervisory signal */ - int16_t *fsk_sine[2][2]; /* 4 pointers to 4 precalc. sine curves */ int fsk_polarity; /* current polarity state of bit */ - int samples_per_chunk; /* how many samples lasts one chunk */ - int16_t *fsk_filter_spl; /* array with samples_per_chunk */ - int fsk_filter_pos; /* current sample position in filter_spl */ - int fsk_filter_step; /* number of samples for each analyzation */ - int fsk_filter_bit; /* last bit, so we detect a bit change */ + int16_t *fsk_filter_spl; /* array to hold ring buffer for bit decoding */ + int fsk_filter_size; /* size of ring buffer */ + int fsk_filter_pos; /* position to write next sample */ + double fsk_filter_step; /* counts bit duration, to trigger decoding every 10th bit */ + int fsk_filter_bit; /* last bit state, so we detect a bit change */ int fsk_filter_sample; /* count until it is time to sample bit */ uint16_t fsk_filter_sync; /* shift register to detect sync */ int fsk_filter_in_sync; /* if we are in sync and receive bits */ @@ -116,12 +116,15 @@ typedef struct nmt { double super_phase256; /* current phase */ double dial_phaseshift256; /* how much the phase of sine wave changes per sample */ double dial_phase256; /* current phase */ - int frame; /* set, if there is a valid frame */ - int16_t *frame_spl; /* 166 * fsk_bit_length */ + 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 */ diff --git a/src/test/test_dms.c b/src/test/test_dms.c index 677b78e..c047e5a 100644 --- a/src/test/test_dms.c +++ b/src/test/test_dms.c @@ -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);