10 #include "audioreader.h"
11 #include "interpolate.h"
15 #define C64_FREQUENCY 985248
16 #define SYNC_PULSE_START 1000
17 #define SYNC_PULSE_END 20000
18 #define SYNC_PULSE_LENGTH 378.0
19 #define SYNC_TEST_TOLERANCE 1.10
21 #define NUM_FILTER_COEFF 32
23 static float hysteresis_limit = 3000.0 / 32768.0;
24 static bool do_calibrate = true;
25 static bool output_cycles_plot = false;
26 static bool use_filter = false;
27 static float filter_coeff[NUM_FILTER_COEFF] = { 1.0f }; // The rest is filled with 0.
28 static bool output_filtered = false;
29 static bool quiet = false;
32 double find_zerocrossing(const std::vector<float> &pcm, int x)
37 if (pcm[x + 1] == 0) {
41 assert(pcm[x + 1] < 0);
46 while (lower - upper > 1e-3) {
47 double mid = 0.5f * (upper + lower);
48 if (lanczos_interpolate(pcm, mid) > 0) {
55 return 0.5f * (upper + lower);
59 double time; // in seconds from start
60 double len; // in seconds
63 // Calibrate on the first ~25k pulses (skip a few, just to be sure).
64 double calibrate(const std::vector<pulse> &pulses) {
65 if (pulses.size() < SYNC_PULSE_END) {
66 fprintf(stderr, "Too few pulses, not calibrating!\n");
70 int sync_pulse_end = -1;
71 double sync_pulse_stddev = -1.0;
73 // Compute the standard deviation (to check for uneven speeds).
74 // If it suddenly skyrockets, we assume that sync ended earlier
75 // than we thought (it should be 25000 cycles), and that we should
76 // calibrate on fewer cycles.
77 for (int try_end : { 2000, 4000, 5000, 7500, 10000, 15000, SYNC_PULSE_END }) {
79 for (int i = SYNC_PULSE_START; i < try_end; ++i) {
80 double cycles = pulses[i].len * C64_FREQUENCY;
81 sum2 += (cycles - SYNC_PULSE_LENGTH) * (cycles - SYNC_PULSE_LENGTH);
83 double stddev = sqrt(sum2 / (try_end - SYNC_PULSE_START - 1));
84 if (sync_pulse_end != -1 && stddev > 5.0 && stddev / sync_pulse_stddev > 1.3) {
85 fprintf(stderr, "Stopping at %d sync pulses because standard deviation would be too big (%.2f cycles); shorter-than-usual trailer?\n",
86 sync_pulse_end, stddev);
89 sync_pulse_end = try_end;
90 sync_pulse_stddev = stddev;
93 fprintf(stderr, "Sync pulse length standard deviation: %.2f cycles\n",
98 for (int i = SYNC_PULSE_START; i < sync_pulse_end; ++i) {
101 double mean_length = C64_FREQUENCY * sum / (sync_pulse_end - SYNC_PULSE_START);
102 double calibration_factor = SYNC_PULSE_LENGTH / mean_length;
104 fprintf(stderr, "Calibrated sync pulse length: %.2f -> %.2f (change %+.2f%%)\n",
105 mean_length, SYNC_PULSE_LENGTH, 100.0 * (calibration_factor - 1.0));
108 // Check for pulses outside +/- 10% (sign of misdetection).
109 for (int i = SYNC_PULSE_START; i < sync_pulse_end; ++i) {
110 double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY;
111 if (cycles < SYNC_PULSE_LENGTH / SYNC_TEST_TOLERANCE || cycles > SYNC_PULSE_LENGTH * SYNC_TEST_TOLERANCE) {
112 fprintf(stderr, "Sync cycle with downflank at %.6f was detected at %.0f cycles; misdetect?\n",
113 pulses[i].time, cycles);
117 return calibration_factor;
120 void output_tap(const std::vector<pulse>& pulses, double calibration_factor)
122 std::vector<char> tap_data;
123 for (unsigned i = 0; i < pulses.size(); ++i) {
124 double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY;
125 int len = lrintf(cycles / TAP_RESOLUTION);
126 if (i > SYNC_PULSE_END && (cycles < 100 || cycles > 800)) {
127 fprintf(stderr, "Cycle with downflank at %.6f was detected at %.0f cycles; misdetect?\n",
128 pulses[i].time, cycles);
131 tap_data.push_back(len);
133 int overflow_len = lrintf(cycles);
134 tap_data.push_back(0);
135 tap_data.push_back(overflow_len & 0xff);
136 tap_data.push_back((overflow_len >> 8) & 0xff);
137 tap_data.push_back(overflow_len >> 16);
142 memcpy(hdr.identifier, "C64-TAPE-RAW", 12);
144 hdr.reserved[0] = hdr.reserved[1] = hdr.reserved[2] = 0;
145 hdr.data_len = tap_data.size();
147 fwrite(&hdr, sizeof(hdr), 1, stdout);
148 fwrite(tap_data.data(), tap_data.size(), 1, stdout);
151 static struct option long_options[] = {
152 {"no-calibrate", 0, 0, 's' },
153 {"plot-cycles", 0, 0, 'p' },
154 {"hysteresis-limit", required_argument, 0, 'l' },
155 {"filter", required_argument, 0, 'f' },
156 {"output-filtered", 0, 0, 'F' },
157 {"quiet", 0, 0, 'q' },
158 {"help", 0, 0, 'h' },
164 fprintf(stderr, "decode [OPTIONS] AUDIO-FILE > TAP-FILE\n");
165 fprintf(stderr, "\n");
166 fprintf(stderr, " -s, --no-calibrate do not try to calibrate on sync pulse length\n");
167 fprintf(stderr, " -p, --plot-cycles output debugging info to cycles.plot\n");
168 fprintf(stderr, " -l, --hysteresis-limit VAL change amplitude threshold for ignoring pulses (0..32768)\n");
169 fprintf(stderr, " -f, --filter C1:C2:C3:... specify FIR filter (up to %d coefficients)\n", NUM_FILTER_COEFF);
170 fprintf(stderr, " -F, --output-filtered output filtered waveform to filtered.raw\n");
171 fprintf(stderr, " -q, --quiet suppress some informational messages\n");
172 fprintf(stderr, " -h, --help display this help, then exit\n");
176 void parse_options(int argc, char **argv)
179 int option_index = 0;
180 int c = getopt_long(argc, argv, "spl:f:Fqh", long_options, &option_index);
186 do_calibrate = false;
190 output_cycles_plot = true;
194 hysteresis_limit = atof(optarg) / 32768.0;
198 const char *coeffstr = strtok(optarg, ":");
200 while (coeff_index < NUM_FILTER_COEFF && coeffstr != NULL) {
201 filter_coeff[coeff_index++] = atof(coeffstr);
202 coeffstr = strtok(NULL, ":");
209 output_filtered = true;
224 // TODO: Support AVX here.
225 std::vector<float> do_filter(const std::vector<float>& pcm, const float* filter)
227 std::vector<float> filtered_pcm;
228 filtered_pcm.reserve(pcm.size());
229 for (unsigned i = NUM_FILTER_COEFF; i < pcm.size(); ++i) {
231 for (int j = 0; j < NUM_FILTER_COEFF; ++j) {
232 s += filter[j] * pcm[i - j];
234 filtered_pcm.push_back(s);
237 if (output_filtered) {
238 FILE *fp = fopen("filtered.raw", "wb");
239 fwrite(filtered_pcm.data(), filtered_pcm.size() * sizeof(filtered_pcm[0]), 1, fp);
246 std::vector<pulse> detect_pulses(const std::vector<float> &pcm, int sample_rate)
248 std::vector<pulse> pulses;
252 double last_downflank = -1;
253 for (unsigned i = 0; i < pcm.size(); ++i) {
254 int bit = (pcm[i] > 0) ? 1 : 0;
255 if (bit == 0 && last_bit == 1) {
256 // Check if we ever go up above <hysteresis_limit> before we dip down again.
257 bool true_pulse = false;
259 int min_level_after = 32767;
260 for (j = i; j < pcm.size(); ++j) {
261 min_level_after = std::min<int>(min_level_after, pcm[j]);
262 if (pcm[j] > 0) break;
263 if (pcm[j] < -hysteresis_limit) {
271 fprintf(stderr, "Ignored down-flank at %.6f seconds due to hysteresis (%d < %d).\n",
272 double(i) / sample_rate, -min_level_after, hysteresis_limit);
279 double t = find_zerocrossing(pcm, i - 1) * (1.0 / sample_rate);
280 if (last_downflank > 0) {
283 p.len = t - last_downflank;
293 int main(int argc, char **argv)
295 parse_options(argc, argv);
297 make_lanczos_weight_table();
298 std::vector<float> pcm;
300 if (!read_audio_file(argv[optind], &pcm, &sample_rate)) {
305 pcm = do_filter(pcm, filter_coeff);
309 for (int i = 0; i < LEN; ++i) {
310 in[i] += rand() % 10000;
315 for (int i = 0; i < LEN; ++i) {
316 printf("%d\n", in[i]);
320 std::vector<pulse> pulses = detect_pulses(pcm, sample_rate);
322 double calibration_factor = 1.0;
324 calibration_factor = calibrate(pulses);
327 if (output_cycles_plot) {
328 FILE *fp = fopen("cycles.plot", "w");
329 for (unsigned i = 0; i < pulses.size(); ++i) {
330 double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY;
331 fprintf(fp, "%f %f\n", pulses[i].time, cycles);
336 output_tap(pulses, calibration_factor);