mirror of https://github.com/wb2osz/direwolf.git
Clean out old obsolete demodulators.
This commit is contained in:
wb2osz
parent
747224ce57
commit
249f5bd471
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@ -173,9 +173,6 @@ struct audio_s {
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int upsample; /* Upsample by this factor for G3RUH. */
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int interleave; /* If > 1, interleave samples among multiple decoders. */
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/* Quick hack for experiment. */
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int mark_freq; /* Two tones for AFSK modulation, in Hz. */
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int space_freq; /* Standard tones are 1200 and 2200 for 1200 baud. */
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45
src/demod.c
45
src/demod.c
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@ -120,7 +120,6 @@ int demod_init (struct audio_s *pa)
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* This can be increased by:
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* Multiple frequencies.
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* Multiple letters (not sure if I will continue this).
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* New interleaved decoders.
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*
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* num_slicers is set to max by the "+" option.
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*/
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@ -313,48 +312,6 @@ int demod_init (struct audio_s *pa)
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save_audio_config_p->achan[chan].num_subchan = num_letters;
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/*
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* Quick hack with special case for another experiment.
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* Do this in a more general way if it turns out to be useful.
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*/
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save_audio_config_p->achan[chan].interleave = 1;
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if (strcasecmp(save_audio_config_p->achan[chan].profiles, "EE") == 0) {
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save_audio_config_p->achan[chan].interleave = 2;
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save_audio_config_p->achan[chan].decimate = 1;
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}
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else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "EEE") == 0) {
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save_audio_config_p->achan[chan].interleave = 3;
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save_audio_config_p->achan[chan].decimate = 1;
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}
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else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "EEEE") == 0) {
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save_audio_config_p->achan[chan].interleave = 4;
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save_audio_config_p->achan[chan].decimate = 1;
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}
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else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "EEEEE") == 0) {
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save_audio_config_p->achan[chan].interleave = 5;
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save_audio_config_p->achan[chan].decimate = 1;
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}
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else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GG") == 0) {
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save_audio_config_p->achan[chan].interleave = 2;
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save_audio_config_p->achan[chan].decimate = 1;
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}
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else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GGG") == 0) {
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save_audio_config_p->achan[chan].interleave = 3;
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save_audio_config_p->achan[chan].decimate = 1;
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}
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else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GGG+") == 0) {
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save_audio_config_p->achan[chan].interleave = 3;
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save_audio_config_p->achan[chan].decimate = 1;
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}
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else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GGGG") == 0) {
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save_audio_config_p->achan[chan].interleave = 4;
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save_audio_config_p->achan[chan].decimate = 1;
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}
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else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GGGGG") == 0) {
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save_audio_config_p->achan[chan].interleave = 5;
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save_audio_config_p->achan[chan].decimate = 1;
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}
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if (save_audio_config_p->achan[chan].num_subchan != num_letters) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("INTERNAL ERROR, %s:%d, chan=%d, num_subchan(%d) != strlen(\"%s\")\n",
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@ -383,7 +340,7 @@ int demod_init (struct audio_s *pa)
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dw_printf (" %d.%d: %c %d & %d\n", chan, d, profile, mark, space);
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}
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demod_afsk_init (save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec / (save_audio_config_p->achan[chan].decimate * save_audio_config_p->achan[chan].interleave),
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demod_afsk_init (save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec / save_audio_config_p->achan[chan].decimate,
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save_audio_config_p->achan[chan].baud,
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mark,
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space,
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@ -159,12 +159,6 @@ void demod_9600_init (int samples_per_sec, int baud, struct demodulator_state_s
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D->pll_locked_inertia = 0.89;
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D->pll_searching_inertia = 0.67;
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D->play_it_again_sample = 0; // TODO: 1.6 experiment.
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// assuming lp_filter_size > lp2_filter_size
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D->lp2_filter_size = samples_per_sec / baud; // samples for 1 bit
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// break;
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// }
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@ -216,12 +210,6 @@ void demod_9600_init (int samples_per_sec, int baud, struct demodulator_state_s
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(void)gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window, 0);
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// Go back and resample where bit is expected.
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fc = (float)baud * 1 / (float)samples_per_sec;
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(void)gen_lowpass (fc, D->lp2_filter, D->lp2_filter_size, D->lp_window, 0);
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/* Version 1.2: Experiment with different slicing levels. */
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for (j = 0; j < MAX_SUBCHANS; j++) {
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@ -524,31 +512,7 @@ inline static void nudge_pll (int chan, int subchan, int slice, float demod_out_
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/* Overflow. Was large positive, wrapped around, now large negative. */
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if (D->play_it_again_sample) { // New experiment in 1.6.
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// FIXME: double check position and draw picture.
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int offset = ( D->lp_filter_size - D->lp2_filter_size ) / 2;
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float amp = convolve (D->raw_cb + offset, D->lp2_filter, D->lp2_filter_size);
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int resampled;
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if (D->num_slicers > 1) {
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resampled = amp - slice_point[slice] > 0;;
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}
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else {
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resampled = amp > 0;
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}
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hdlc_rec_bit (chan, subchan, slice, resampled, 1, D->slicer[slice].lfsr);
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}
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else {
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// traditional
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hdlc_rec_bit (chan, subchan, slice, demod_out_f > 0, 1, D->slicer[slice].lfsr);
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}
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hdlc_rec_bit (chan, subchan, slice, demod_out_f > 0, 1, D->slicer[slice].lfsr);
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}
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/*
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248
src/demod_afsk.c
248
src/demod_afsk.c
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@ -197,79 +197,6 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
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switch (profile) {
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case 'A':
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/* Original. 52 taps, truncated bandpass, IIR lowpass */
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/* 'F' is the fast version for low end processors. */
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/* It is a special case that works only for a particular */
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/* baud rate, tone pair, and sampling rate. */
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D->use_prefilter = 0;
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D->ms_filter_len_bits = 1.415; /* 52 @ 44100, 1200 */
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D->ms_window = BP_WINDOW_TRUNCATED;
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//D->bp_window = BP_WINDOW_TRUNCATED;
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D->lpf_use_fir = 0;
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D->lpf_iir = 0.195;
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D->agc_fast_attack = 0.250;
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D->agc_slow_decay = 0.00012;
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D->hysteresis = 0.005;
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D->pll_locked_inertia = 0.700;
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D->pll_searching_inertia = 0.580;
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break;
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case 'B':
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/* Original bandpass. Use FIR lowpass instead. */
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D->use_prefilter = 0;
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D->ms_filter_len_bits = 1.415; /* 52 @ 44100, 1200 */
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D->ms_window = BP_WINDOW_TRUNCATED;
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//D->bp_window = BP_WINDOW_TRUNCATED;
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D->lpf_use_fir = 1;
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D->lpf_baud = 1.09;
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D->lp_filter_len_bits = D->ms_filter_len_bits;
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D->lp_window = BP_WINDOW_TRUNCATED;
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D->agc_fast_attack = 0.370;
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D->agc_slow_decay = 0.00014;
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D->hysteresis = 0.003;
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D->pll_locked_inertia = 0.620;
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D->pll_searching_inertia = 0.350;
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break;
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case 'C':
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/* Cosine window, 76 taps for bandpass, FIR lowpass. */
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D->use_prefilter = 0;
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D->ms_filter_len_bits = 2.068; /* 76 @ 44100, 1200 */
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D->ms_window = BP_WINDOW_COSINE;
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//D->bp_window = BP_WINDOW_COSINE;
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D->lpf_use_fir = 1;
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D->lpf_baud = 1.09;
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D->lp_filter_len_bits = D->ms_filter_len_bits;
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D->lp_window = BP_WINDOW_TRUNCATED;
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D->agc_fast_attack = 0.495;
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D->agc_slow_decay = 0.00022;
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D->hysteresis = 0.005;
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D->pll_locked_inertia = 0.620;
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D->pll_searching_inertia = 0.350;
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break;
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case 'D':
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/* Prefilter, Cosine window, FIR lowpass. Tweeked for 300 baud. */
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@ -335,73 +262,6 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
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D->pll_searching_inertia = 0.50;
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break;
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case 'G':
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/* 1200 baud - Started out same as E but add 3 way interleave. */
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/* Version 1.3 - EXPERIMENTAL - Needs more fine tuning. */
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//D->bp_window = BP_WINDOW_COSINE; /* The name says BP but it is used for all of them. */
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D->use_prefilter = 1; /* first, a bandpass filter. */
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D->prefilter_baud = 0.15;
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D->pre_filter_len_bits = 128 * 1200. / (44100. / 3.);
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D->pre_window = BP_WINDOW_TRUNCATED;
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D->ms_filter_len_bits = 25 * 1200. / (44100. / 3.);
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D->ms_window = BP_WINDOW_COSINE;
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D->lpf_use_fir = 1;
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D->lpf_baud = 1.16;
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D->lp_filter_len_bits = 21 * 1200. / (44100. / 3.);
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D->lp_window = BP_WINDOW_TRUNCATED;
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D->agc_fast_attack = 0.130;
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D->agc_slow_decay = 0.00013;
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D->hysteresis = 0.01;
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D->pll_locked_inertia = 0.73;
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D->pll_searching_inertia = 0.64;
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break;
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case 'H':
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/* Experiment in Version 1.6 */
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/* 1200 baud - Started out as a copy of E but */
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/* will probably have little tweaks after the */
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/* major experiment. */
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/* Enhancements: */
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/* + Look back and sample the bit position. */
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/* + Avoid smearing by long filter and low pass. */
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D->use_prefilter = 1; /* first, a bandpass filter. */
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D->prefilter_baud = 0.21;
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D->pre_filter_len_bits = 184 * 1200. / 44100.;
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D->pre_filter_len_bits = 235 * 1200. / 44100.;
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D->pre_window = BP_WINDOW_TRUNCATED;
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D->ms_filter_len_bits = 65 * 1200. / 44100.; // Just over 2 bit times.
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D->ms_window = BP_WINDOW_COSINE;
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/* New for synchronous re-demod in 1.6. */
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D->play_it_again_sample = 1;
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D->m2_filter_len_bits = 44 * 1200. / 44100.; // a little more than 1 bit time.
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D->s2_filter_len_bits = 48 * 1200. / 44100.; // a little more than 1 bit time.
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D->ms2_window = BP_WINDOW_TRUNCATED;
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D->lp_delay_fract = 0.53; // FIXME: This is backwards.
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D->lpf_use_fir = 1;
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D->lpf_baud = 1.10;
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D->lp_filter_len_bits = 64 * 1200. / 44100.;
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D->lp_window = BP_WINDOW_TRUNCATED;
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D->agc_fast_attack = 0.820;
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D->agc_slow_decay = 0.000214;
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D->hysteresis = 0.001;
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D->pll_locked_inertia = 0.765;
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D->pll_searching_inertia = 0.44;
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break;
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default:
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text_color_set(DW_COLOR_ERROR);
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@ -457,8 +317,6 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
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D->pre_filter_size = (int) round( D->pre_filter_len_bits * (float)samples_per_sec / (float)baud );
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D->ms_filter_size = (int) round( D->ms_filter_len_bits * (float)samples_per_sec / (float)baud );
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D->m2_filter_size = (int) round( D->m2_filter_len_bits * (float)samples_per_sec / (float)baud );
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D->s2_filter_size = (int) round( D->s2_filter_len_bits * (float)samples_per_sec / (float)baud );
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D->lp_filter_size = (int) round( D->lp_filter_len_bits * (float)samples_per_sec / (float)baud );
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/* Experiment with other sizes. */
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@ -469,12 +327,6 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
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#ifdef TUNE_MS_FILTER_SIZE
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D->ms_filter_size = TUNE_MS_FILTER_SIZE;
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#endif
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#ifdef TUNE_M2_FILTER_SIZE
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D->m2_filter_size = TUNE_M2_FILTER_SIZE;
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#endif
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#ifdef TUNE_S2_FILTER_SIZE
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D->s2_filter_size = TUNE_S2_FILTER_SIZE;
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#endif
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#ifdef TUNE_LP_FILTER_SIZE
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D->lp_filter_size = TUNE_LP_FILTER_SIZE;
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#endif
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@ -503,24 +355,7 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
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exit (1);
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}
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if (D->m2_filter_size * 2 > MAX_FILTER_SIZE)
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{
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text_color_set (DW_COLOR_ERROR);
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dw_printf ("Calculated filter size of %d is too large.\n", D->m2_filter_size);
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dw_printf ("Decrease the audio sample rate or increase the baud rate or\n");
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dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n",
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MAX_FILTER_SIZE);
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exit (1);
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}
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if (D->s2_filter_size * 2 > MAX_FILTER_SIZE)
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{
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text_color_set (DW_COLOR_ERROR);
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dw_printf ("Calculated filter size of %d is too large.\n", D->s2_filter_size);
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dw_printf ("Decrease the audio sample rate or increase the baud rate or\n");
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dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n",
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MAX_FILTER_SIZE);
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exit (1);
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}
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if (D->lp_filter_size > MAX_FILTER_SIZE)
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{
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@ -585,13 +420,6 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
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gen_ms (space_freq, samples_per_sec, D->s_sin_table, D->s_cos_table, D->ms_filter_size, D->ms_window);
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// Note that these are twice as long so we can try matching different phases.
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if (D->play_it_again_sample) {
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gen_ms (mark_freq, samples_per_sec, D->m2_sin_table, D->m2_cos_table, D->m2_filter_size * 2, D->ms2_window);
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gen_ms (space_freq, samples_per_sec, D->s2_sin_table, D->s2_cos_table, D->s2_filter_size * 2, D->ms2_window);
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}
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/*
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* Now the lowpass filter.
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* I thought we'd want a cutoff of about 0.5 the baud rate
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@ -764,9 +592,6 @@ void demod_afsk_process_sample (int chan, int subchan, int sam, struct demodulat
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// FIXME: calculate how much we really need.
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int extra = 0;
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if (D->play_it_again_sample) {
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extra = D->lp_filter_size;
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}
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if (D->use_prefilter) {
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float cleaner;
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@ -1015,76 +840,7 @@ inline static void nudge_pll (int chan, int subchan, int slice, int demod_data,
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if (D->slicer[slice].data_clock_pll < 0 && D->slicer[slice].prev_d_c_pll > 0) {
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/* Overflow. */
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// In version 1.6 we will try a new experiment.
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// The tone filters are about 2 bit times wide so this smears the signal.
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// I originally expected this to be about 1 bit time but 2 turned out
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// to give the best results after extensive experimentation.
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// We are looking at half of each adjacent bit, not just the one we want.
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// The low pass filter also smears the signal.
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//
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// We will try to look back in time and re-demodulate only the time period of the bit.
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//
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if (D->play_it_again_sample) { // New in 1.6. Currently 'H' demod profile.
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// FIXME: double check position and draw picture.
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#if 0
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// This provided a slight benefit.
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int offset = ( D->ms_filter_size - D->m2_filter_size ) / 2 + D->lp_filter_delay;
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float m_sum1 = convolve (D->ms_in_cb + offset, D->m2_sin_table, D->m2_filter_size);
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float m_sum2 = convolve (D->ms_in_cb + offset, D->m2_cos_table, D->m2_filter_size);
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float m_amp = sqrtf(m_sum1 * m_sum1 + m_sum2 * m_sum2);
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offset = ( D->ms_filter_size - D->s2_filter_size ) / 2 + D->lp_filter_delay;
|
||||
|
||||
float s_sum1 = convolve (D->ms_in_cb + offset, D->s2_sin_table, D->s2_filter_size);
|
||||
float s_sum2 = convolve (D->ms_in_cb + offset, D->s2_cos_table, D->s2_filter_size);
|
||||
float s_amp = sqrtf(s_sum1 * s_sum1 + s_sum2 * s_sum2);
|
||||
#else
|
||||
// Here we try something completely new.
|
||||
// Rather than taking the vector magnitude of the I+Q components,
|
||||
// correlate it with only a sine wave. We don't know the phase so
|
||||
// we will try matching with a bunch of different phases and take the best match.
|
||||
|
||||
// Trying match with cosine as well could be beneficial for lower sample rates.
|
||||
|
||||
int j;
|
||||
float m_amp = 0;
|
||||
float s_amp = 0;
|
||||
|
||||
int offset = ( D->ms_filter_size - D->m2_filter_size ) / 2 + D->lp_filter_delay;
|
||||
|
||||
for (j = 0; j <= D->m2_filter_size; j++) {
|
||||
float match = fabsf(convolve (D->ms_in_cb + offset, D->m2_sin_table + j, D->m2_filter_size));
|
||||
if (match > m_amp) m_amp = match;
|
||||
}
|
||||
|
||||
offset = ( D->ms_filter_size - D->s2_filter_size ) / 2 + D->lp_filter_delay;
|
||||
|
||||
for (j = 0; j <= D->s2_filter_size; j++) {
|
||||
float match = fabsf(convolve (D->ms_in_cb + offset, D->s2_sin_table + j, D->s2_filter_size));
|
||||
if (match > s_amp) s_amp = match;
|
||||
}
|
||||
#endif
|
||||
|
||||
int resampled;
|
||||
if (D->num_slicers > 1) {
|
||||
resampled = m_amp > s_amp * space_gain[slice];
|
||||
}
|
||||
else {
|
||||
resampled = m_amp > s_amp;
|
||||
}
|
||||
|
||||
hdlc_rec_bit (chan, subchan, slice, resampled, 0, -1);
|
||||
}
|
||||
else {
|
||||
// Traditional way, after the low pass filter.
|
||||
hdlc_rec_bit (chan, subchan, slice, demod_data, 0, -1);
|
||||
}
|
||||
hdlc_rec_bit (chan, subchan, slice, demod_data, 0, -1);
|
||||
}
|
||||
|
||||
// Even if we used alternative method to extract the data bit,
|
||||
|
|
|
|||
|
|
@ -33,8 +33,6 @@ struct demodulator_state_s
|
|||
char profile; // 'A', 'B', etc. Upper case.
|
||||
// Only needed to see if we are using 'F' to take fast path.
|
||||
|
||||
int play_it_again_sample; // Enable new synchronous demod in version 1.6.
|
||||
|
||||
#define TICKS_PER_PLL_CYCLE ( 256.0 * 256.0 * 256.0 * 256.0 )
|
||||
|
||||
int pll_step_per_sample; // PLL is advanced by this much each audio sample.
|
||||
|
|
@ -46,13 +44,6 @@ struct demodulator_state_s
|
|||
/* but about 2 bit times turned out to be better. */
|
||||
/* Currently using same size for any prefilter. */
|
||||
|
||||
int m2_filter_size;
|
||||
int s2_filter_size; /* Size of mark & space filters, in audio samples */
|
||||
/* for the synchronous demodulator. I'm expecting */
|
||||
/* smaller, perhaps just over 1 bit time here. */
|
||||
|
||||
int lp2_filter_size; /* FSK resampling - Size of Low Pass filter, in audio samples. */
|
||||
|
||||
|
||||
#define MAX_FILTER_SIZE 320 /* 304 is needed for profile C, 300 baud & 44100. */
|
||||
|
||||
|
|
@ -61,8 +52,6 @@ struct demodulator_state_s
|
|||
* e.g. 1 means 1/1200 second for 1200 baud.
|
||||
*/
|
||||
float ms_filter_len_bits;
|
||||
float m2_filter_len_bits;
|
||||
float s2_filter_len_bits;
|
||||
float lp_delay_fract;
|
||||
|
||||
/*
|
||||
|
|
@ -72,7 +61,6 @@ struct demodulator_state_s
|
|||
bp_window_t pre_window;
|
||||
bp_window_t ms_window;
|
||||
bp_window_t lp_window;
|
||||
bp_window_t ms2_window; /* New in 1.6. */
|
||||
|
||||
|
||||
/*
|
||||
|
|
@ -158,17 +146,6 @@ struct demodulator_state_s
|
|||
float s_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
|
||||
float s_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
|
||||
|
||||
/*
|
||||
* Same for the synchronous re-demodulator.
|
||||
*/
|
||||
|
||||
float m2_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
|
||||
float m2_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
|
||||
|
||||
float s2_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
|
||||
float s2_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
|
||||
|
||||
float lp2_filter[MAX_FILTER_SIZE] __attribute__((aligned(16)));
|
||||
|
||||
/*
|
||||
* These are for PSK only.
|
||||
|
|
|
|||
|
|
@ -306,7 +306,6 @@ void multi_modem_process_sample (int chan, int audio_sample)
|
|||
{
|
||||
int d;
|
||||
int subchan;
|
||||
static int i = 0; /* for interleaving among multiple demodulators. */
|
||||
|
||||
// Accumulate an average DC bias level.
|
||||
// Shouldn't happen with a soundcard but could with mistuned SDR.
|
||||
|
|
@ -334,21 +333,9 @@ void multi_modem_process_sample (int chan, int audio_sample)
|
|||
/* Formerly one loop. */
|
||||
/* 1.2: We can feed one demodulator but end up with multiple outputs. */
|
||||
|
||||
|
||||
if (save_audio_config_p->achan[chan].interleave > 1) {
|
||||
|
||||
// TODO: temp debug, remove this.
|
||||
|
||||
assert (save_audio_config_p->achan[chan].interleave == save_audio_config_p->achan[chan].num_subchan);
|
||||
demod_process_sample(chan, i, audio_sample);
|
||||
i++;
|
||||
if (i >= save_audio_config_p->achan[chan].interleave) i = 0;
|
||||
}
|
||||
else {
|
||||
/* Send same thing to all. */
|
||||
for (d = 0; d < save_audio_config_p->achan[chan].num_subchan; d++) {
|
||||
demod_process_sample(chan, d, audio_sample);
|
||||
}
|
||||
/* Send same thing to all. */
|
||||
for (d = 0; d < save_audio_config_p->achan[chan].num_subchan; d++) {
|
||||
demod_process_sample(chan, d, audio_sample);
|
||||
}
|
||||
|
||||
for (subchan = 0; subchan < save_audio_config_p->achan[chan].num_subchan; subchan++) {
|
||||
|
|
|
|||
Loading…
Reference in New Issue