10 #include "audioreader.h"
11 #include "interpolate.h"
16 #define C64_FREQUENCY 985248
17 #define SYNC_PULSE_START 1000
18 #define SYNC_PULSE_END 20000
19 #define SYNC_PULSE_LENGTH 378.0
20 #define SYNC_TEST_TOLERANCE 1.10
22 #define NUM_FILTER_COEFF 32
24 static float hysteresis_limit = 3000.0 / 32768.0;
25 static bool do_calibrate = true;
26 static bool output_cycles_plot = false;
27 static bool use_filter = false;
28 static bool do_crop = false;
29 static float crop_start = 0.0f, crop_end = HUGE_VAL;
30 static float filter_coeff[NUM_FILTER_COEFF] = { 1.0f }; // The rest is filled with 0.
31 static bool output_filtered = false;
32 static bool quiet = false;
33 static bool do_auto_level = false;
34 static bool output_leveled = false;
37 double find_zerocrossing(const std::vector<float> &pcm, int x)
42 if (pcm[x + 1] == 0) {
46 assert(pcm[x + 1] < 0);
51 while (lower - upper > 1e-3) {
52 double mid = 0.5f * (upper + lower);
53 if (lanczos_interpolate(pcm, mid) > 0) {
60 return 0.5f * (upper + lower);
64 double time; // in seconds from start
65 double len; // in seconds
68 // Calibrate on the first ~25k pulses (skip a few, just to be sure).
69 double calibrate(const std::vector<pulse> &pulses) {
70 if (pulses.size() < SYNC_PULSE_END) {
71 fprintf(stderr, "Too few pulses, not calibrating!\n");
75 int sync_pulse_end = -1;
76 double sync_pulse_stddev = -1.0;
78 // Compute the standard deviation (to check for uneven speeds).
79 // If it suddenly skyrockets, we assume that sync ended earlier
80 // than we thought (it should be 25000 cycles), and that we should
81 // calibrate on fewer cycles.
82 for (int try_end : { 2000, 4000, 5000, 7500, 10000, 15000, SYNC_PULSE_END }) {
84 for (int i = SYNC_PULSE_START; i < try_end; ++i) {
85 double cycles = pulses[i].len * C64_FREQUENCY;
86 sum2 += (cycles - SYNC_PULSE_LENGTH) * (cycles - SYNC_PULSE_LENGTH);
88 double stddev = sqrt(sum2 / (try_end - SYNC_PULSE_START - 1));
89 if (sync_pulse_end != -1 && stddev > 5.0 && stddev / sync_pulse_stddev > 1.3) {
90 fprintf(stderr, "Stopping at %d sync pulses because standard deviation would be too big (%.2f cycles); shorter-than-usual trailer?\n",
91 sync_pulse_end, stddev);
94 sync_pulse_end = try_end;
95 sync_pulse_stddev = stddev;
98 fprintf(stderr, "Sync pulse length standard deviation: %.2f cycles\n",
103 for (int i = SYNC_PULSE_START; i < sync_pulse_end; ++i) {
104 sum += pulses[i].len;
106 double mean_length = C64_FREQUENCY * sum / (sync_pulse_end - SYNC_PULSE_START);
107 double calibration_factor = SYNC_PULSE_LENGTH / mean_length;
109 fprintf(stderr, "Calibrated sync pulse length: %.2f -> %.2f (change %+.2f%%)\n",
110 mean_length, SYNC_PULSE_LENGTH, 100.0 * (calibration_factor - 1.0));
113 // Check for pulses outside +/- 10% (sign of misdetection).
114 for (int i = SYNC_PULSE_START; i < sync_pulse_end; ++i) {
115 double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY;
116 if (cycles < SYNC_PULSE_LENGTH / SYNC_TEST_TOLERANCE || cycles > SYNC_PULSE_LENGTH * SYNC_TEST_TOLERANCE) {
117 fprintf(stderr, "Sync cycle with downflank at %.6f was detected at %.0f cycles; misdetect?\n",
118 pulses[i].time, cycles);
122 return calibration_factor;
125 void output_tap(const std::vector<pulse>& pulses, double calibration_factor)
127 std::vector<char> tap_data;
128 for (unsigned i = 0; i < pulses.size(); ++i) {
129 double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY;
130 int len = lrintf(cycles / TAP_RESOLUTION);
131 if (i > SYNC_PULSE_END && (cycles < 100 || cycles > 800)) {
132 fprintf(stderr, "Cycle with downflank at %.6f was detected at %.0f cycles; misdetect?\n",
133 pulses[i].time, cycles);
136 tap_data.push_back(len);
138 int overflow_len = lrintf(cycles);
139 tap_data.push_back(0);
140 tap_data.push_back(overflow_len & 0xff);
141 tap_data.push_back((overflow_len >> 8) & 0xff);
142 tap_data.push_back(overflow_len >> 16);
147 memcpy(hdr.identifier, "C64-TAPE-RAW", 12);
149 hdr.reserved[0] = hdr.reserved[1] = hdr.reserved[2] = 0;
150 hdr.data_len = tap_data.size();
152 fwrite(&hdr, sizeof(hdr), 1, stdout);
153 fwrite(tap_data.data(), tap_data.size(), 1, stdout);
156 static struct option long_options[] = {
157 {"auto-level", 0, 0, 'a' },
158 {"no-calibrate", 0, 0, 's' },
159 {"plot-cycles", 0, 0, 'p' },
160 {"hysteresis-limit", required_argument, 0, 'l' },
161 {"filter", required_argument, 0, 'f' },
162 {"output-filtered", 0, 0, 'F' },
163 {"crop", required_argument, 0, 'c' },
164 {"quiet", 0, 0, 'q' },
165 {"help", 0, 0, 'h' },
171 fprintf(stderr, "decode [OPTIONS] AUDIO-FILE > TAP-FILE\n");
172 fprintf(stderr, "\n");
173 fprintf(stderr, " -a, --auto-level automatically adjust amplitude levels throughout the file\n");
174 fprintf(stderr, " -A, --output-leveled output leveled waveform to leveled.raw\n");
175 fprintf(stderr, " -s, --no-calibrate do not try to calibrate on sync pulse length\n");
176 fprintf(stderr, " -p, --plot-cycles output debugging info to cycles.plot\n");
177 fprintf(stderr, " -l, --hysteresis-limit VAL change amplitude threshold for ignoring pulses (0..32768)\n");
178 fprintf(stderr, " -f, --filter C1:C2:C3:... specify FIR filter (up to %d coefficients)\n", NUM_FILTER_COEFF);
179 fprintf(stderr, " -F, --output-filtered output filtered waveform to filtered.raw\n");
180 fprintf(stderr, " -c, --crop START[:END] use only the given part of the file\n");
181 fprintf(stderr, " -q, --quiet suppress some informational messages\n");
182 fprintf(stderr, " -h, --help display this help, then exit\n");
186 void parse_options(int argc, char **argv)
189 int option_index = 0;
190 int c = getopt_long(argc, argv, "aAspl:f:Fc:qh", long_options, &option_index);
196 do_auto_level = true;
200 output_leveled = true;
204 do_calibrate = false;
208 output_cycles_plot = true;
212 hysteresis_limit = atof(optarg) / 32768.0;
216 const char *coeffstr = strtok(optarg, ":");
218 while (coeff_index < NUM_FILTER_COEFF && coeffstr != NULL) {
219 filter_coeff[coeff_index++] = atof(coeffstr);
220 coeffstr = strtok(NULL, ":");
227 output_filtered = true;
231 const char *cropstr = strtok(optarg, ":");
232 crop_start = atof(cropstr);
233 cropstr = strtok(NULL, ":");
234 if (cropstr == NULL) {
237 crop_end = atof(cropstr);
255 std::vector<float> crop(const std::vector<float>& pcm, float crop_start, float crop_end, int sample_rate)
257 size_t start_sample, end_sample;
258 if (crop_start >= 0.0f) {
259 start_sample = std::min<size_t>(lrintf(crop_start * sample_rate), pcm.size());
261 if (crop_end >= 0.0f) {
262 end_sample = std::min<size_t>(lrintf(crop_end * sample_rate), pcm.size());
264 return std::vector<float>(pcm.begin() + start_sample, pcm.begin() + end_sample);
267 // TODO: Support AVX here.
268 std::vector<float> do_filter(const std::vector<float>& pcm, const float* filter)
270 std::vector<float> filtered_pcm;
271 filtered_pcm.reserve(pcm.size());
272 for (unsigned i = NUM_FILTER_COEFF; i < pcm.size(); ++i) {
274 for (int j = 0; j < NUM_FILTER_COEFF; ++j) {
275 s += filter[j] * pcm[i - j];
277 filtered_pcm.push_back(s);
280 if (output_filtered) {
281 FILE *fp = fopen("filtered.raw", "wb");
282 fwrite(filtered_pcm.data(), filtered_pcm.size() * sizeof(filtered_pcm[0]), 1, fp);
289 std::vector<pulse> detect_pulses(const std::vector<float> &pcm, int sample_rate)
291 std::vector<pulse> pulses;
295 double last_downflank = -1;
296 for (unsigned i = 0; i < pcm.size(); ++i) {
297 int bit = (pcm[i] > 0) ? 1 : 0;
298 if (bit == 0 && last_bit == 1) {
299 // Check if we ever go up above <hysteresis_limit> before we dip down again.
300 bool true_pulse = false;
302 int min_level_after = 32767;
303 for (j = i; j < pcm.size(); ++j) {
304 min_level_after = std::min<int>(min_level_after, pcm[j]);
305 if (pcm[j] > 0) break;
306 if (pcm[j] < -hysteresis_limit) {
314 fprintf(stderr, "Ignored down-flank at %.6f seconds due to hysteresis (%d < %d).\n",
315 double(i) / sample_rate, -min_level_after, hysteresis_limit);
322 double t = find_zerocrossing(pcm, i - 1) * (1.0 / sample_rate) + crop_start;
323 if (last_downflank > 0) {
326 p.len = t - last_downflank;
336 void output_cycle_plot(const std::vector<pulse> &pulses, double calibration_factor)
338 FILE *fp = fopen("cycles.plot", "w");
339 for (unsigned i = 0; i < pulses.size(); ++i) {
340 double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY;
341 fprintf(fp, "%f %f\n", pulses[i].time, cycles);
346 int main(int argc, char **argv)
348 parse_options(argc, argv);
350 make_lanczos_weight_table();
351 std::vector<float> pcm;
353 if (!read_audio_file(argv[optind], &pcm, &sample_rate)) {
358 pcm = crop(pcm, crop_start, crop_end, sample_rate);
362 pcm = do_filter(pcm, filter_coeff);
366 pcm = level_samples(pcm, sample_rate);
367 if (output_leveled) {
368 FILE *fp = fopen("leveled.raw", "wb");
369 fwrite(pcm.data(), pcm.size() * sizeof(pcm[0]), 1, fp);
375 for (int i = 0; i < LEN; ++i) {
376 in[i] += rand() % 10000;
381 for (int i = 0; i < LEN; ++i) {
382 printf("%d\n", in[i]);
386 std::vector<pulse> pulses = detect_pulses(pcm, sample_rate);
388 double calibration_factor = 1.0;
390 calibration_factor = calibrate(pulses);
393 if (output_cycles_plot) {
394 output_cycle_plot(pulses, calibration_factor);
397 output_tap(pulses, calibration_factor);