New common FSK implementation, replaces all individual implementations
This commit is contained in:
352
src/r2000/dsp.c
352
src/r2000/dsp.c
@@ -37,7 +37,8 @@
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*
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* Applies similar to NMT, read it there!
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*
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* I assume that the deviation at 1800 Hz (Bit 0) is +-1700 Hz.
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* I assume that the deviation at 1500 Hz is +-1425 Hz. (according to R&S)
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* This would lead to a deviation at 1800 Hz (Bit 0) about +-1700 Hz. (emphasis)
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*
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* Notes on TX_PEAK_SUPER level:
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*
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@@ -49,44 +50,32 @@
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#define MAX_MODULATION 2550.0
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#define DBM0_DEVIATION 1500.0 /* deviation of dBm0 at 1 kHz */
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#define COMPANDOR_0DB 1.0 /* A level of 0dBm (1.0) shall be unaccected */
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#define TX_PEAK_FSK (1700.0 / 1800.0 * 1000.0 / DBM0_DEVIATION) /* with emphasis */
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#define TX_PEAK_FSK (1425.0 / 1500.0 * 1000.0 / DBM0_DEVIATION) /* with emphasis */
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#define TX_PEAK_SUPER (300.0 / DBM0_DEVIATION) /* no emphasis */
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#define BIT_RATE 1200.0
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#define SUPER_RATE 50.0
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#define FSK_BIT_RATE 1200.0
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#define FSK_BIT_ADJUST 0.1 /* how much do we adjust bit clock on frequency change */
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#define FSK_F0 1800.0
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#define FSK_F1 1200.0
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#define SUPER_BIT_RATE 50.0
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#define SUPER_BIT_ADJUST 0.5 /* how much do we adjust bit clock on frequency change */
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#define SUPER_F0 136.0
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#define SUPER_F1 164.0
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#define FILTER_STEP 0.002 /* step every 2 ms */
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#define MAX_DISPLAY 1.4 /* something above dBm0 */
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/* two signaling tones */
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static double super_bits[2] = {
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136.0,
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164.0,
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};
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/* table for fast sine generation */
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static sample_t super_sine[65536];
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/* global init for FFSK */
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/* global init for FSK */
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void dsp_init(void)
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{
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int i;
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ffsk_global_init(TX_PEAK_FSK);
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PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table.\n");
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for (i = 0; i < 65536; i++) {
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super_sine[i] = sin((double)i / 65536.0 * 2.0 * PI) * TX_PEAK_SUPER;
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}
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}
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static int fsk_send_bit(void *inst);
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static void fsk_receive_bit(void *inst, int bit, double quality, double level);
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static int super_send_bit(void *inst);
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static void super_receive_bit(void *inst, int bit, double quality, double level);
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/* Init FSK of transceiver */
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int dsp_init_sender(r2000_t *r2000)
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{
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sample_t *spl;
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double fsk_samples_per_bit;
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int i;
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/* attack (3ms) and recovery time (13.5ms) according to NMT specs */
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init_compandor(&r2000->cstate, 8000, 3.0, 13.5, COMPANDOR_0DB);
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@@ -97,9 +86,9 @@ int dsp_init_sender(r2000_t *r2000)
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PDEBUG(DDSP, DEBUG_DEBUG, "Using FSK level of %.3f\n", TX_PEAK_FSK);
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/* init ffsk */
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if (ffsk_init(&r2000->ffsk, r2000, fsk_receive_bit, r2000->sender.kanal, r2000->sender.samplerate) < 0) {
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "FFSK init failed!\n");
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/* init fsk */
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if (fsk_init(&r2000->fsk, r2000, fsk_send_bit, fsk_receive_bit, r2000->sender.samplerate, FSK_BIT_RATE, FSK_F0, FSK_F1, TX_PEAK_FSK, 1, FSK_BIT_ADJUST) < 0) {
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "FSK init failed!\n");
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return -EINVAL;
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}
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if (r2000->sender.loopback)
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@@ -107,43 +96,11 @@ int dsp_init_sender(r2000_t *r2000)
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else
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r2000->rx_max = 144;
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/* allocate transmit buffer for a complete frame, add 10 to be safe */
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fsk_samples_per_bit = (double)r2000->sender.samplerate / BIT_RATE;
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r2000->frame_size = 208.0 * fsk_samples_per_bit + 10;
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spl = calloc(r2000->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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/* init supervisorty fsk */
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if (fsk_init(&r2000->super_fsk, r2000, super_send_bit, super_receive_bit, r2000->sender.samplerate, SUPER_BIT_RATE, SUPER_F0, SUPER_F1, TX_PEAK_SUPER, 0, SUPER_BIT_ADJUST) < 0) {
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "FSK init failed!\n");
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return -EINVAL;
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}
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r2000->frame_spl = spl;
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/* strange: better quality with window size of two bits */
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r2000->super_samples_per_window = (double)r2000->sender.samplerate / SUPER_RATE * 2.0;
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r2000->super_filter_step = (double)r2000->sender.samplerate * FILTER_STEP;
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r2000->super_size = 20.0 * r2000->super_samples_per_window + 10;
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PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per filter step for supervisory signal.\n", r2000->super_filter_step);
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spl = calloc(r2000->super_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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r2000->super_spl = spl;
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spl = calloc(1, r2000->super_samples_per_window * 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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r2000->super_filter_spl = spl;
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r2000->super_filter_bit = -1;
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/* count supervisory symbols */
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for (i = 0; i < 2; i++) {
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audio_goertzel_init(&r2000->super_goertzel[i], super_bits[i], r2000->sender.samplerate);
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r2000->super_phaseshift65536[i] = 65536.0 / ((double)r2000->sender.samplerate / super_bits[i]);
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PDEBUG(DDSP, DEBUG_DEBUG, "phaseshift[%d] = %.4f\n", i, r2000->super_phaseshift65536[i]);
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}
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r2000->super_bittime = SUPER_RATE / (double)r2000->sender.samplerate;
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return 0;
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}
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@@ -153,20 +110,8 @@ void dsp_cleanup_sender(r2000_t *r2000)
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{
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup DSP for Transceiver.\n");
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ffsk_cleanup(&r2000->ffsk);
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if (r2000->frame_spl) {
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free(r2000->frame_spl);
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r2000->frame_spl = NULL;
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}
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if (r2000->super_spl) {
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free(r2000->super_spl);
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r2000->super_spl = NULL;
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}
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if (r2000->super_filter_spl) {
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free(r2000->super_filter_spl);
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r2000->super_filter_spl = NULL;
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}
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fsk_cleanup(&r2000->fsk);
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fsk_cleanup(&r2000->super_fsk);
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}
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/* Check for SYNC bits, then collect data bits */
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@@ -242,8 +187,9 @@ static void fsk_receive_bit(void *inst, int bit, double quality, double level)
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r2000_receive_frame(r2000, r2000->rx_frame, quality, level);
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}
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static void super_receive_bit(r2000_t *r2000, int bit, double level, double quality)
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static void super_receive_bit(void *inst, int bit, double quality, double level)
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{
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r2000_t *r2000 = (r2000_t *)inst;
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int i;
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/* normalize supervisory level */
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@@ -272,108 +218,6 @@ static void super_receive_bit(r2000_t *r2000, int bit, double level, double qual
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r2000_receive_super(r2000, (r2000->super_rx_word >> 1) & 0x7f, quality, level);
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}
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//#define DEBUG_FILTER
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//#define DEBUG_QUALITY
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/* demodulate supervisory signal
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* filter one chunk, that is 2ms long (1/10th of a bit) */
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static inline void super_decode_step(r2000_t *r2000, int pos)
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{
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double level, result[2], softbit, quality;
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int max;
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sample_t *spl;
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int bit;
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max = r2000->super_samples_per_window;
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spl = r2000->super_filter_spl;
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level = audio_level(spl, max);
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audio_goertzel(r2000->super_goertzel, spl, max, pos, result, 2);
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/* calculate soft bit from both frequencies */
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softbit = (result[1] / level - result[0] / level + 1.0) / 2.0;
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// /* scale it, since both filters overlap by some percent */
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//#define MIN_QUALITY 0.08
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// softbit = (softbit - MIN_QUALITY) / (0.850 - MIN_QUALITY - MIN_QUALITY);
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if (softbit > 1)
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softbit = 1;
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if (softbit < 0)
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softbit = 0;
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#ifdef DEBUG_FILTER
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printf("|%s", debug_amplitude(result[0]/level));
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printf("|%s| low=%.3f high=%.3f level=%d\n", debug_amplitude(result[1]/level), result[0]/level, result[1]/level, (int)level);
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#endif
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if (softbit > 0.5)
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bit = 1;
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else
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bit = 0;
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// quality = result[bit] / level;
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if (softbit > 0.5)
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quality = softbit * 2.0 - 1.0;
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else
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quality = 1.0 - softbit * 2.0;
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/* scale quality, because filters overlap */
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quality /= 0.80;
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if (r2000->super_filter_bit != bit) {
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#ifdef DEBUG_FILTER
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puts("bit change");
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#endif
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r2000->super_filter_bit = bit;
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#if 0
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/* If we have a bit change, move sample counter towards one half bit duration.
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* We may have noise, so the bit change may be wrong or not at the correct place.
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* This can cause bit slips.
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* Therefore we change the sample counter only slightly, so bit slips may not
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* happen so quickly.
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*/
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if (r2000->super_filter_sample < 5)
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r2000->super_filter_sample++;
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if (r2000->super_filter_sample > 5)
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r2000->super_filter_sample--;
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#else
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/* directly center the sample position, because we don't have any sync sequence */
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r2000->super_filter_sample = 5;
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#endif
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} else if (--r2000->super_filter_sample == 0) {
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/* if sample counter bit reaches 0, we reset sample counter to one bit duration */
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#ifdef DEBUG_QUALITY
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printf("|%s| quality=%.2f ", debug_amplitude(softbit), quality);
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printf("|%s|\n", debug_amplitude(quality);
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#endif
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/* adjust level, so we get peak of sine curve */
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super_receive_bit(r2000, bit, level / 0.63662, quality);
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r2000->super_filter_sample = 10;
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}
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}
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/* get audio chunk out of received stream */
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void super_receive(r2000_t *r2000, sample_t *samples, int length)
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{
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sample_t *spl;
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int max, pos, step;
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int i;
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/* write received samples to decode buffer */
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max = r2000->super_samples_per_window;
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pos = r2000->super_filter_pos;
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step = r2000->super_filter_step;
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spl = r2000->super_filter_spl;
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for (i = 0; i < length; i++) {
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spl[pos++] = samples[i];
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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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super_decode_step(r2000, pos);
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}
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}
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r2000->super_filter_pos = pos;
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}
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/* Process received audio stream from radio unit. */
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void sender_receive(sender_t *sender, sample_t *samples, int length)
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{
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@@ -390,14 +234,14 @@ void sender_receive(sender_t *sender, sample_t *samples, int length)
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if (r2000->dsp_mode == DSP_MODE_AUDIO_TX
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|| r2000->dsp_mode == DSP_MODE_AUDIO_TX_RX
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|| r2000->sender.loopback)
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super_receive(r2000, samples, length);
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fsk_receive(&r2000->super_fsk, samples, length);
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/* do de-emphasis */
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if (r2000->de_emphasis)
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de_emphasis(&r2000->estate, samples, length);
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/* fsk signal */
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ffsk_receive(&r2000->ffsk, samples, length);
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fsk_receive(&r2000->fsk, samples, length);
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/* we must have audio mode for both ways and a call */
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if (r2000->dsp_mode == DSP_MODE_AUDIO_TX_RX
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@@ -424,125 +268,43 @@ void sender_receive(sender_t *sender, sample_t *samples, int length)
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r2000->sender.rxbuf_pos = 0;
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}
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static int fsk_frame(r2000_t *r2000, sample_t *samples, int length)
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static int fsk_send_bit(void *inst)
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{
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r2000_t *r2000 = (r2000_t *)inst;
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const char *frame;
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sample_t *spl;
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int i;
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int count, max;
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next_frame:
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if (!r2000->frame_length) {
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/* request frame */
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if (!r2000->tx_frame_length || r2000->tx_frame_pos == r2000->tx_frame_length) {
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frame = r2000_get_frame(r2000);
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if (!frame) {
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r2000->tx_frame_length = 0;
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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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/* render frame */
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r2000->frame_length = ffsk_render_frame(&r2000->ffsk, frame, 208, r2000->frame_spl);
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r2000->frame_pos = 0;
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if (r2000->frame_length > r2000->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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return -1;
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}
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memcpy(r2000->tx_frame, frame, 208);
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r2000->tx_frame_length = 208;
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r2000->tx_frame_pos = 0;
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}
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/* send audio from frame */
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max = r2000->frame_length;
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count = max - r2000->frame_pos;
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if (count > length)
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count = length;
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spl = r2000->frame_spl + r2000->frame_pos;
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for (i = 0; i < count; i++) {
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*samples++ = *spl++;
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}
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length -= count;
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r2000->frame_pos += count;
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/* check for end of telegramm */
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if (r2000->frame_pos == max) {
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r2000->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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}
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return length;
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return r2000->tx_frame[r2000->tx_frame_pos++];
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}
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static int super_render_frame(r2000_t *r2000, uint32_t word, sample_t *sample)
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static int super_send_bit(void *inst)
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{
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double phaseshift, phase, bittime, bitpos;
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int count = 0, i;
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r2000_t *r2000 = (r2000_t *)inst;
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phase = r2000->super_phase65536;
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bittime = r2000->super_bittime;
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bitpos = r2000->super_bitpos;
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for (i = 0; i < 20; i++) {
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phaseshift = r2000->super_phaseshift65536[(word >> 19) & 1];
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do {
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*sample++ = super_sine[(uint16_t)phase];
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count++;
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phase += phaseshift;
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if (phase >= 65536.0)
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phase -= 65536.0;
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bitpos += bittime;
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} while (bitpos < 1.0);
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bitpos -= 1.0;
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word <<= 1;
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}
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r2000->super_phase65536 = phase;
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bitpos = r2000->super_bitpos;
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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 super_frame(r2000_t *r2000, sample_t *samples, int length)
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{
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sample_t *spl;
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int i;
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int count, max;
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next_frame:
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if (!r2000->super_length) {
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/* render supervisory rame */
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "render word 0x%05x\n", r2000->super_tx_word);
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r2000->super_length = super_render_frame(r2000, r2000->super_tx_word, r2000->super_spl);
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r2000->super_pos = 0;
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if (r2000->super_length > r2000->super_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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if (!r2000->super_tx_word_length || r2000->super_tx_word_pos == r2000->super_tx_word_length) {
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r2000->super_tx_word_length = 20;
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r2000->super_tx_word_pos = 0;
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}
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/* send audio from frame */
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max = r2000->super_length;
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count = max - r2000->super_pos;
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if (count > length)
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count = length;
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spl = r2000->super_spl + r2000->super_pos;
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for (i = 0; i < count; i++) {
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*samples++ += *spl++;
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}
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length -= count;
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r2000->super_pos += count;
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/* check for end of telegramm */
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if (r2000->super_pos == max) {
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r2000->super_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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}
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return length;
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return (r2000->super_tx_word >> (r2000->super_tx_word_length - (++r2000->super_tx_word_pos))) & 1;
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}
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/* Provide stream of audio toward radio unit */
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void sender_send(sender_t *sender, sample_t *samples, int length)
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{
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r2000_t *r2000 = (r2000_t *) sender;
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int len;
|
||||
int count;
|
||||
|
||||
again:
|
||||
switch (r2000->dsp_mode) {
|
||||
@@ -555,20 +317,25 @@ again:
|
||||
/* do pre-emphasis */
|
||||
if (r2000->pre_emphasis)
|
||||
pre_emphasis(&r2000->estate, samples, length);
|
||||
super_frame(r2000, samples, length);
|
||||
/* add supervisory to sample buffer */
|
||||
fsk_send(&r2000->super_fsk, samples, length, 1);
|
||||
break;
|
||||
case DSP_MODE_FRAME:
|
||||
/* Encode frame into audio stream. If frames have
|
||||
* stopped, process again for rest of stream. */
|
||||
len = fsk_frame(r2000, samples, length);
|
||||
count = fsk_send(&r2000->fsk, samples, length, 0);
|
||||
/* do pre-emphasis */
|
||||
if (r2000->pre_emphasis)
|
||||
pre_emphasis(&r2000->estate, samples, length - len);
|
||||
if (len) {
|
||||
samples += length - len;
|
||||
length = len;
|
||||
goto again;
|
||||
pre_emphasis(&r2000->estate, samples, count);
|
||||
/* special case: add supervisory signal to frame at loop test */
|
||||
if (r2000->sender.loopback) {
|
||||
/* add supervisory to sample buffer */
|
||||
fsk_send(&r2000->super_fsk, samples, count, 1);
|
||||
}
|
||||
samples += count;
|
||||
length -= count;
|
||||
if (length)
|
||||
goto again;
|
||||
break;
|
||||
}
|
||||
}
|
||||
@@ -596,11 +363,13 @@ void r2000_set_dsp_mode(r2000_t *r2000, enum dsp_mode mode, int super)
|
||||
{
|
||||
/* reset telegramm */
|
||||
if (mode == DSP_MODE_FRAME && r2000->dsp_mode != mode) {
|
||||
r2000->frame_length = 0;
|
||||
r2000->tx_frame_length = 0;
|
||||
fsk_tx_reset(&r2000->fsk);
|
||||
}
|
||||
if ((mode == DSP_MODE_AUDIO_TX || mode == DSP_MODE_AUDIO_TX_RX)
|
||||
&& (r2000->dsp_mode != DSP_MODE_AUDIO_TX && r2000->dsp_mode != DSP_MODE_AUDIO_TX_RX)) {
|
||||
r2000->super_length = 0;
|
||||
r2000->super_tx_word_length = 0;
|
||||
fsk_tx_reset(&r2000->super_fsk);
|
||||
}
|
||||
|
||||
if (super >= 0) {
|
||||
@@ -615,4 +384,3 @@ void r2000_set_dsp_mode(r2000_t *r2000, enum dsp_mode mode, int super)
|
||||
r2000->dsp_mode = mode;
|
||||
}
|
||||
|
||||
#warning fixme: high pass filter on tx side to prevent desturbance of supervisory signal
|
||||
|
Reference in New Issue
Block a user