1 // Copyright Steinar H. Gunderson <sgunderson@bigfoot.com>
2 // Licensed under the GPL, v2. (See the file COPYING.)
13 #include "audioreader.h"
14 #include "interpolate.h"
20 #define C64_FREQUENCY 985248
21 #define SYNC_PULSE_START 1000
22 #define SYNC_PULSE_END 20000
23 #define SYNC_PULSE_LENGTH 378.0
24 #define SYNC_TEST_TOLERANCE 1.10
27 #define NUM_FILTER_COEFF 32
28 #define NUM_SPSA_VALS (NUM_FILTER_COEFF + 2)
30 #define A NUM_ITER/10 // approx
31 #define INITIAL_A 0.005 // A bit of trial and error...
32 #define INITIAL_C 0.02 // This too.
36 static float hysteresis_upper_limit = 0.1;
37 static float hysteresis_lower_limit = -0.1;
38 static bool do_calibrate = true;
39 static bool output_cycles_plot = false;
40 static bool do_crop = false;
41 static float crop_start = 0.0f, crop_end = HUGE_VAL;
43 static bool use_fir_filter = false;
44 static float filter_coeff[NUM_FILTER_COEFF] = { 1.0f }; // The rest is filled with 0.
45 static bool use_rc_filter = false;
46 static float rc_filter_freq;
47 static bool output_filtered = false;
49 static bool quiet = false;
50 static bool do_auto_level = false;
51 static bool output_leveled = false;
52 static std::vector<float> train_snap_points;
53 static bool do_train = false;
55 // The frequency to filter on (for do_auto_level), in Hertz.
56 // Larger values makes the compressor react faster, but if it is too large,
57 // you'll ruin the waveforms themselves.
58 static float auto_level_freq = 200.0;
60 // The minimum estimated sound level (for do_auto_level) at any given point.
61 // If you decrease this, you'll be able to amplify really silent signals
62 // by more, but you'll also increase the level of silent (ie. noise-only) segments,
63 // possibly caused misdetected pulses in these segments.
64 static float min_level = 0.05f;
66 // search for the value <limit> between [x,x+1]
67 double find_crossing(const std::vector<float> &pcm, int x, float limit)
71 while (lower - upper > 1e-3) {
72 double mid = 0.5f * (upper + lower);
73 if (lanczos_interpolate(pcm, mid) > limit) {
80 return 0.5f * (upper + lower);
84 double time; // in seconds from start
85 double len; // in seconds
88 // Calibrate on the first ~25k pulses (skip a few, just to be sure).
89 double calibrate(const std::vector<pulse> &pulses) {
90 if (pulses.size() < SYNC_PULSE_END) {
91 fprintf(stderr, "Too few pulses, not calibrating!\n");
95 int sync_pulse_end = -1;
96 double sync_pulse_stddev = -1.0;
98 // Compute the standard deviation (to check for uneven speeds).
99 // If it suddenly skyrockets, we assume that sync ended earlier
100 // than we thought (it should be 25000 cycles), and that we should
101 // calibrate on fewer cycles.
102 for (int try_end : { 2000, 4000, 5000, 7500, 10000, 15000, SYNC_PULSE_END }) {
104 for (int i = SYNC_PULSE_START; i < try_end; ++i) {
105 double cycles = pulses[i].len * C64_FREQUENCY;
106 sum2 += (cycles - SYNC_PULSE_LENGTH) * (cycles - SYNC_PULSE_LENGTH);
108 double stddev = sqrt(sum2 / (try_end - SYNC_PULSE_START - 1));
109 if (sync_pulse_end != -1 && stddev > 5.0 && stddev / sync_pulse_stddev > 1.3) {
110 fprintf(stderr, "Stopping at %d sync pulses because standard deviation would be too big (%.2f cycles); shorter-than-usual trailer?\n",
111 sync_pulse_end, stddev);
114 sync_pulse_end = try_end;
115 sync_pulse_stddev = stddev;
118 fprintf(stderr, "Sync pulse length standard deviation: %.2f cycles\n",
123 for (int i = SYNC_PULSE_START; i < sync_pulse_end; ++i) {
124 sum += pulses[i].len;
126 double mean_length = C64_FREQUENCY * sum / (sync_pulse_end - SYNC_PULSE_START);
127 double calibration_factor = SYNC_PULSE_LENGTH / mean_length;
129 fprintf(stderr, "Calibrated sync pulse length: %.2f -> %.2f (change %+.2f%%)\n",
130 mean_length, SYNC_PULSE_LENGTH, 100.0 * (calibration_factor - 1.0));
133 // Check for pulses outside +/- 10% (sign of misdetection).
134 for (int i = SYNC_PULSE_START; i < sync_pulse_end; ++i) {
135 double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY;
136 if (cycles < SYNC_PULSE_LENGTH / SYNC_TEST_TOLERANCE || cycles > SYNC_PULSE_LENGTH * SYNC_TEST_TOLERANCE) {
137 fprintf(stderr, "Sync cycle with downflank at %.6f was detected at %.0f cycles; misdetect?\n",
138 pulses[i].time, cycles);
142 return calibration_factor;
145 void output_tap(const std::vector<pulse>& pulses, double calibration_factor)
147 std::vector<char> tap_data;
148 for (unsigned i = 0; i < pulses.size(); ++i) {
149 double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY;
150 int len = lrintf(cycles / TAP_RESOLUTION);
151 if (i > SYNC_PULSE_END && (cycles < 100 || cycles > 800)) {
152 fprintf(stderr, "Cycle with downflank at %.6f was detected at %.0f cycles; misdetect?\n",
153 pulses[i].time, cycles);
156 tap_data.push_back(len);
158 int overflow_len = lrintf(cycles);
159 tap_data.push_back(0);
160 tap_data.push_back(overflow_len & 0xff);
161 tap_data.push_back((overflow_len >> 8) & 0xff);
162 tap_data.push_back(overflow_len >> 16);
167 memcpy(hdr.identifier, "C64-TAPE-RAW", 12);
169 hdr.reserved[0] = hdr.reserved[1] = hdr.reserved[2] = 0;
170 hdr.data_len = tap_data.size();
172 fwrite(&hdr, sizeof(hdr), 1, stdout);
173 fwrite(tap_data.data(), tap_data.size(), 1, stdout);
176 static struct option long_options[] = {
177 {"auto-level", 0, 0, 'a' },
178 {"auto-level-freq", required_argument, 0, 'b' },
179 {"output-leveled", 0, 0, 'A' },
180 {"min-level", required_argument, 0, 'm' },
181 {"no-calibrate", 0, 0, 's' },
182 {"plot-cycles", 0, 0, 'p' },
183 {"hysteresis-limit", required_argument, 0, 'l' },
184 {"filter", required_argument, 0, 'f' },
185 {"rc-filter", required_argument, 0, 'r' },
186 {"output-filtered", 0, 0, 'F' },
187 {"crop", required_argument, 0, 'c' },
188 {"quiet", 0, 0, 'q' },
189 {"help", 0, 0, 'h' },
195 fprintf(stderr, "decode [OPTIONS] AUDIO-FILE > TAP-FILE\n");
196 fprintf(stderr, "\n");
197 fprintf(stderr, " -a, --auto-level automatically adjust amplitude levels throughout the file\n");
198 fprintf(stderr, " -b, --auto-level-freq minimum frequency in Hertz of corrected level changes (default 200 Hz)\n");
199 fprintf(stderr, " -A, --output-leveled output leveled waveform to leveled.raw\n");
200 fprintf(stderr, " -m, --min-level minimum estimated sound level (0..1) for --auto-level\n");
201 fprintf(stderr, " -s, --no-calibrate do not try to calibrate on sync pulse length\n");
202 fprintf(stderr, " -p, --plot-cycles output debugging info to cycles.plot\n");
203 fprintf(stderr, " -l, --hysteresis-limit U[:L] change amplitude threshold for ignoring pulses (-1..1)\n");
204 fprintf(stderr, " -f, --filter C1:C2:C3:... specify FIR filter (up to %d coefficients)\n", NUM_FILTER_COEFF);
205 fprintf(stderr, " -r, --rc-filter FREQ send signal through a highpass RC filter with given frequency (in Hertz)\n");
206 fprintf(stderr, " -F, --output-filtered output filtered waveform to filtered.raw\n");
207 fprintf(stderr, " -c, --crop START[:END] use only the given part of the file\n");
208 fprintf(stderr, " -t, --train LEN1:LEN2:... train a filter for detecting any of the given number of cycles\n");
209 fprintf(stderr, " (implies --no-calibrate and --quiet unless overridden)\n");
210 fprintf(stderr, " -q, --quiet suppress some informational messages\n");
211 fprintf(stderr, " -h, --help display this help, then exit\n");
215 void parse_options(int argc, char **argv)
218 int option_index = 0;
219 int c = getopt_long(argc, argv, "ab:Am:spl:f:r:Fc:t:qh", long_options, &option_index);
225 do_auto_level = true;
229 auto_level_freq = atof(optarg);
233 output_leveled = true;
237 min_level = atof(optarg);
241 do_calibrate = false;
245 output_cycles_plot = true;
249 const char *hyststr = strtok(optarg, ": ");
250 hysteresis_upper_limit = atof(hyststr);
251 hyststr = strtok(NULL, ": ");
252 if (hyststr == NULL) {
253 hysteresis_lower_limit = -hysteresis_upper_limit;
255 hysteresis_lower_limit = atof(hyststr);
261 const char *coeffstr = strtok(optarg, ": ");
263 while (coeff_index < NUM_FILTER_COEFF && coeffstr != NULL) {
264 filter_coeff[coeff_index++] = atof(coeffstr);
265 coeffstr = strtok(NULL, ": ");
267 use_fir_filter = true;
272 use_rc_filter = true;
273 rc_filter_freq = atof(optarg);
277 output_filtered = true;
281 const char *cropstr = strtok(optarg, ":");
282 crop_start = atof(cropstr);
283 cropstr = strtok(NULL, ":");
284 if (cropstr == NULL) {
287 crop_end = atof(cropstr);
294 const char *cyclestr = strtok(optarg, ":");
295 while (cyclestr != NULL) {
296 train_snap_points.push_back(atof(cyclestr));
297 cyclestr = strtok(NULL, ":");
301 // Set reasonable defaults (can be overridden later on the command line).
302 do_calibrate = false;
319 std::vector<float> crop(const std::vector<float>& pcm, float crop_start, float crop_end, int sample_rate)
321 size_t start_sample, end_sample;
322 if (crop_start >= 0.0f) {
323 start_sample = std::min<size_t>(lrintf(crop_start * sample_rate), pcm.size());
325 if (crop_end >= 0.0f) {
326 end_sample = std::min<size_t>(lrintf(crop_end * sample_rate), pcm.size());
328 return std::vector<float>(pcm.begin() + start_sample, pcm.begin() + end_sample);
331 // TODO: Support AVX here.
332 std::vector<float> do_fir_filter(const std::vector<float>& pcm, const float* filter)
334 std::vector<float> filtered_pcm;
335 filtered_pcm.reserve(pcm.size());
336 for (unsigned i = NUM_FILTER_COEFF; i < pcm.size(); ++i) {
338 for (int j = 0; j < NUM_FILTER_COEFF; ++j) {
339 s += filter[j] * pcm[i - j];
341 filtered_pcm.push_back(s);
344 if (output_filtered) {
345 FILE *fp = fopen("filtered.raw", "wb");
346 fwrite(filtered_pcm.data(), filtered_pcm.size() * sizeof(filtered_pcm[0]), 1, fp);
353 std::vector<float> do_rc_filter(const std::vector<float>& pcm, float freq, int sample_rate)
355 // This is only a 6 dB/oct filter, which seemingly works better
356 // than the Filter class, which is a standard biquad (12 dB/oct).
357 // The b/c calculations come from libnyquist (atone.c);
358 // I haven't checked, but I suppose they fall out of the bilinear
359 // transform of the transfer function H(s) = s/(s + w).
360 std::vector<float> filtered_pcm;
361 filtered_pcm.resize(pcm.size());
362 const float b = 2.0f - cos(2.0 * M_PI * freq / sample_rate);
363 const float c = b - sqrt(b * b - 1.0f);
364 float prev_in = 0.0f;
365 float prev_out = 0.0f;
366 for (unsigned i = 0; i < pcm.size(); ++i) {
368 float out = c * (prev_out + in - prev_in);
369 filtered_pcm[i] = out;
374 if (output_filtered) {
375 FILE *fp = fopen("filtered.raw", "wb");
376 fwrite(filtered_pcm.data(), filtered_pcm.size() * sizeof(filtered_pcm[0]), 1, fp);
383 std::vector<pulse> detect_pulses(const std::vector<float> &pcm, float hysteresis_upper_limit, float hysteresis_lower_limit, int sample_rate)
385 std::vector<pulse> pulses;
388 enum State { START, ABOVE, BELOW } state = START;
389 double last_downflank = -1;
390 for (unsigned i = 0; i < pcm.size(); ++i) {
391 if (pcm[i] > hysteresis_upper_limit) {
393 } else if (pcm[i] < hysteresis_lower_limit) {
394 if (state == ABOVE) {
396 double t = find_crossing(pcm, i - 1, hysteresis_lower_limit) * (1.0 / sample_rate) + crop_start;
397 if (last_downflank > 0) {
400 p.len = t - last_downflank;
411 void output_cycle_plot(const std::vector<pulse> &pulses, double calibration_factor)
413 FILE *fp = fopen("cycles.plot", "w");
414 for (unsigned i = 0; i < pulses.size(); ++i) {
415 double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY;
416 fprintf(fp, "%f %f\n", pulses[i].time, cycles);
421 std::pair<int, double> find_closest_point(double x, const std::vector<float> &points)
424 double best_dist = (x - points[0]) * (x - points[0]);
425 for (unsigned j = 1; j < train_snap_points.size(); ++j) {
426 double dist = (x - points[j]) * (x - points[j]);
427 if (dist < best_dist) {
432 return std::make_pair(best_point, best_dist);
435 float eval_badness(const std::vector<pulse>& pulses, double calibration_factor)
437 double sum_badness = 0.0;
438 for (unsigned i = 0; i < pulses.size(); ++i) {
439 double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY;
440 if (cycles > 2000.0) cycles = 2000.0; // Don't make pauses arbitrarily bad.
441 std::pair<int, double> selected_point_and_sq_dist = find_closest_point(cycles, train_snap_points);
442 sum_badness += selected_point_and_sq_dist.second;
444 return sqrt(sum_badness / (pulses.size() - 1));
447 void find_kmeans(const std::vector<pulse> &pulses, double calibration_factor, const std::vector<float> &initial_centers)
449 std::vector<float> last_centers = initial_centers;
450 std::vector<float> sums;
451 std::vector<float> num;
452 sums.resize(initial_centers.size());
453 num.resize(initial_centers.size());
455 for (unsigned i = 0; i < initial_centers.size(); ++i) {
459 for (unsigned i = 0; i < pulses.size(); ++i) {
460 double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY;
461 // Ignore heavy outliers, which are almost always long pauses.
462 if (cycles > 2000.0) {
465 std::pair<int, double> selected_point_and_sq_dist = find_closest_point(cycles, last_centers);
466 int p = selected_point_and_sq_dist.first;
470 bool any_moved = false;
471 for (unsigned i = 0; i < initial_centers.size(); ++i) {
473 fprintf(stderr, "K-means broke down, can't output new reference training points\n");
476 float new_center = sums[i] / num[i];
477 if (fabs(new_center - last_centers[i]) > 1e-3) {
480 last_centers[i] = new_center;
486 fprintf(stderr, "New reference training points:");
487 for (unsigned i = 0; i < last_centers.size(); ++i) {
488 fprintf(stderr, " %.3f", last_centers[i]);
490 fprintf(stderr, "\n");
493 void spsa_train(const std::vector<float> &pcm, int sample_rate)
495 float vals[NUM_SPSA_VALS] = { hysteresis_upper_limit, hysteresis_lower_limit, 1.0f }; // The rest is filled with 0.
497 float start_c = INITIAL_C;
498 double best_badness = HUGE_VAL;
500 for (int n = 1; n < NUM_ITER; ++n) {
501 float a = INITIAL_A * pow(n + A, -ALPHA);
502 float c = start_c * pow(n, -GAMMA);
504 // find a random perturbation
505 float p[NUM_SPSA_VALS];
506 float vals1[NUM_SPSA_VALS], vals2[NUM_SPSA_VALS];
507 for (int i = 0; i < NUM_SPSA_VALS; ++i) {
508 p[i] = (rand() % 2) ? 1.0 : -1.0;
509 vals1[i] = std::max(std::min(vals[i] - c * p[i], 1.0f), -1.0f);
510 vals2[i] = std::max(std::min(vals[i] + c * p[i], 1.0f), -1.0f);
513 std::vector<pulse> pulses1 = detect_pulses(do_fir_filter(pcm, vals1 + 2), vals1[0], vals1[1], sample_rate);
514 std::vector<pulse> pulses2 = detect_pulses(do_fir_filter(pcm, vals2 + 2), vals2[0], vals2[1], sample_rate);
515 float badness1 = eval_badness(pulses1, 1.0);
516 float badness2 = eval_badness(pulses2, 1.0);
518 // Find the gradient estimator
519 float g[NUM_SPSA_VALS];
520 for (int i = 0; i < NUM_SPSA_VALS; ++i) {
521 g[i] = (badness2 - badness1) / (2.0 * c * p[i]);
523 vals[i] = std::max(std::min(vals[i], 1.0f), -1.0f);
525 if (badness2 < badness1) {
526 std::swap(badness1, badness2);
527 std::swap(vals1, vals2);
528 std::swap(pulses1, pulses2);
530 if (badness1 < best_badness) {
531 printf("\nNew best filter (badness=%f):", badness1);
532 for (int i = 0; i < NUM_FILTER_COEFF; ++i) {
533 printf(" %.5f", vals1[i + 2]);
535 printf(", hysteresis limits = %f %f\n", vals1[0], vals1[1]);
536 best_badness = badness1;
538 find_kmeans(pulses1, 1.0, train_snap_points);
540 if (output_cycles_plot) {
541 output_cycle_plot(pulses1, 1.0);
549 int main(int argc, char **argv)
551 parse_options(argc, argv);
553 make_lanczos_weight_table();
554 std::vector<float> pcm;
556 if (!read_audio_file(argv[optind], &pcm, &sample_rate)) {
561 pcm = crop(pcm, crop_start, crop_end, sample_rate);
564 if (use_fir_filter) {
565 pcm = do_fir_filter(pcm, filter_coeff);
569 pcm = do_rc_filter(pcm, rc_filter_freq, sample_rate);
573 pcm = level_samples(pcm, min_level, auto_level_freq, sample_rate);
574 if (output_leveled) {
575 FILE *fp = fopen("leveled.raw", "wb");
576 fwrite(pcm.data(), pcm.size() * sizeof(pcm[0]), 1, fp);
582 for (int i = 0; i < LEN; ++i) {
583 in[i] += rand() % 10000;
588 for (int i = 0; i < LEN; ++i) {
589 printf("%d\n", in[i]);
594 spsa_train(pcm, sample_rate);
598 std::vector<pulse> pulses = detect_pulses(pcm, hysteresis_upper_limit, hysteresis_lower_limit, sample_rate);
600 double calibration_factor = 1.0;
602 calibration_factor = calibrate(pulses);
605 if (output_cycles_plot) {
606 output_cycle_plot(pulses, calibration_factor);
609 output_tap(pulses, calibration_factor);