X-Git-Url: https://git.sesse.net/?p=c64tapwav;a=blobdiff_plain;f=decode.cpp;h=d2d6fc89a26aeecee613f6ddb446d35571d35365;hp=c5aa9669d0c473e8fe92911cc9cd348da3839159;hb=e283d8b328e737d7119263010f63b7452fa05e6a;hpb=16e71fe97173f6d8c7c1c4836eea04d460174912 diff --git a/decode.cpp b/decode.cpp index c5aa966..d2d6fc8 100644 --- a/decode.cpp +++ b/decode.cpp @@ -1,159 +1,162 @@ +// Copyright Steinar H. Gunderson +// Licensed under the GPL, v2. (See the file COPYING.) + #include #include #include #include #include +#include +#ifdef __AVX__ +#include +#endif #include #include #include "audioreader.h" #include "interpolate.h" +#include "level.h" #include "tap.h" +#include "filter.h" #define BUFSIZE 4096 -#define HYSTERESIS_LIMIT 3000 #define C64_FREQUENCY 985248 - #define SYNC_PULSE_START 1000 -#define SYNC_PULSE_END 15000 +#define SYNC_PULSE_END 20000 #define SYNC_PULSE_LENGTH 378.0 #define SYNC_TEST_TOLERANCE 1.10 -// between [x,x+1] -double find_zerocrossing(const std::vector &pcm, int x) -{ - if (pcm[x] == 0) { - return x; - } - if (pcm[x + 1] == 0) { - return x + 1; - } +// SPSA options +#define NUM_FILTER_COEFF 32 +#define NUM_SPSA_VALS (NUM_FILTER_COEFF + 2) +#define NUM_ITER 5000 +#define A NUM_ITER/10 // approx +#define INITIAL_A 0.005 // A bit of trial and error... +#define INITIAL_C 0.02 // This too. +#define GAMMA 0.166 +#define ALPHA 1.0 - assert(pcm[x + 1] < 0); - assert(pcm[x] > 0); +static float hysteresis_upper_limit = 0.1; +static float hysteresis_lower_limit = -0.1; +static bool do_calibrate = true; +static bool output_cycles_plot = false; +static bool do_crop = false; +static float crop_start = 0.0f, crop_end = HUGE_VAL; - double upper = x; - double lower = x + 1; - while (lower - upper > 1e-3) { - double mid = 0.5f * (upper + lower); - if (lanczos_interpolate(pcm, mid) > 0) { - upper = mid; - } else { - lower = mid; +static bool use_fir_filter = false; +static float filter_coeff[NUM_FILTER_COEFF] = { 1.0f }; // The rest is filled with 0. +static bool use_rc_filter = false; +static float rc_filter_freq; +static bool output_filtered = false; + +static bool quiet = false; +static bool do_auto_level = false; +static bool output_leveled = false; +static std::vector train_snap_points; +static bool do_train = false; + +// The frequency to filter on (for do_auto_level), in Hertz. +// Larger values makes the compressor react faster, but if it is too large, +// you'll ruin the waveforms themselves. +static float auto_level_freq = 200.0; + +// The minimum estimated sound level (for do_auto_level) at any given point. +// If you decrease this, you'll be able to amplify really silent signals +// by more, but you'll also increase the level of silent (ie. noise-only) segments, +// possibly caused misdetected pulses in these segments. +static float min_level = 0.05f; + +// search for the value between [x,x+1] +template +double find_crossing(const std::vector &pcm, int x, float limit) +{ + if (fast) { + // Do simple linear interpolation. + return x + (limit - pcm[x]) / (pcm[x + 1] - pcm[x]); + } else { + // Binary search for the zero crossing as given by Lanczos interpolation. + double upper = x; + double lower = x + 1; + while (lower - upper > 1e-3) { + double mid = 0.5f * (upper + lower); + if (lanczos_interpolate(pcm, mid) > limit) { + upper = mid; + } else { + lower = mid; + } } - } - return 0.5f * (upper + lower); + return 0.5f * (upper + lower); + } } struct pulse { double time; // in seconds from start double len; // in seconds }; - -int main(int argc, char **argv) -{ - make_lanczos_weight_table(); - std::vector pcm; - int sample_rate; - if (!read_audio_file(argv[1], &pcm, &sample_rate)) { - exit(1); - } - -#if 0 - for (int i = 0; i < LEN; ++i) { - in[i] += rand() % 10000; - } -#endif -#if 0 - for (int i = 0; i < LEN; ++i) { - printf("%d\n", in[i]); +// Calibrate on the first ~25k pulses (skip a few, just to be sure). +double calibrate(const std::vector &pulses) { + if (pulses.size() < SYNC_PULSE_END) { + fprintf(stderr, "Too few pulses, not calibrating!\n"); + return 1.0; } -#endif - - std::vector pulses; // in seconds - // Find the flanks. - int last_bit = -1; - double last_downflank = -1; - for (unsigned i = 0; i < pcm.size(); ++i) { - int bit = (pcm[i] > 0) ? 1 : 0; - if (bit == 0 && last_bit == 1) { - // Check if we ever go up above HYSTERESIS_LIMIT before we dip down again. - bool true_pulse = false; - unsigned j; - int min_level_after = 32767; - for (j = i; j < pcm.size(); ++j) { - min_level_after = std::min(min_level_after, pcm[j]); - if (pcm[j] > 0) break; - if (pcm[j] < -HYSTERESIS_LIMIT) { - true_pulse = true; - break; - } - } + int sync_pulse_end = -1; + double sync_pulse_stddev = -1.0; - if (!true_pulse) { -#if 0 - fprintf(stderr, "Ignored down-flank at %.6f seconds due to hysteresis (%d < %d).\n", - double(i) / sample_rate, -min_level_after, HYSTERESIS_LIMIT); -#endif - i = j; - continue; - } - - // down-flank! - double t = find_zerocrossing(pcm, i - 1) * (1.0 / sample_rate); - if (last_downflank > 0) { - pulse p; - p.time = t; - p.len = t - last_downflank; - pulses.push_back(p); - } - last_downflank = t; + // Compute the standard deviation (to check for uneven speeds). + // If it suddenly skyrockets, we assume that sync ended earlier + // than we thought (it should be 25000 cycles), and that we should + // calibrate on fewer cycles. + for (int try_end : { 2000, 4000, 5000, 7500, 10000, 15000, SYNC_PULSE_END }) { + double sum2 = 0.0; + for (int i = SYNC_PULSE_START; i < try_end; ++i) { + double cycles = pulses[i].len * C64_FREQUENCY; + sum2 += (cycles - SYNC_PULSE_LENGTH) * (cycles - SYNC_PULSE_LENGTH); } - last_bit = bit; + double stddev = sqrt(sum2 / (try_end - SYNC_PULSE_START - 1)); + if (sync_pulse_end != -1 && stddev > 5.0 && stddev / sync_pulse_stddev > 1.3) { + fprintf(stderr, "Stopping at %d sync pulses because standard deviation would be too big (%.2f cycles); shorter-than-usual trailer?\n", + sync_pulse_end, stddev); + break; + } + sync_pulse_end = try_end; + sync_pulse_stddev = stddev; + } + if (!quiet) { + fprintf(stderr, "Sync pulse length standard deviation: %.2f cycles\n", + sync_pulse_stddev); } - // Calibrate on the first ~25k pulses (skip a few, just to be sure). - double calibration_factor = 1.0f; - if (pulses.size() < SYNC_PULSE_END) { - fprintf(stderr, "Too few pulses, not calibrating!\n"); - } else { - double sum = 0.0; - for (int i = SYNC_PULSE_START; i < SYNC_PULSE_END; ++i) { - sum += pulses[i].len; - } - double mean_length = C64_FREQUENCY * sum / (SYNC_PULSE_END - SYNC_PULSE_START); - calibration_factor = SYNC_PULSE_LENGTH / mean_length; + double sum = 0.0; + for (int i = SYNC_PULSE_START; i < sync_pulse_end; ++i) { + sum += pulses[i].len; + } + double mean_length = C64_FREQUENCY * sum / (sync_pulse_end - SYNC_PULSE_START); + double calibration_factor = SYNC_PULSE_LENGTH / mean_length; + if (!quiet) { fprintf(stderr, "Calibrated sync pulse length: %.2f -> %.2f (change %+.2f%%)\n", mean_length, SYNC_PULSE_LENGTH, 100.0 * (calibration_factor - 1.0)); + } - // Check for pulses outside +/- 10% (sign of misdetection). - for (int i = SYNC_PULSE_START; i < SYNC_PULSE_END; ++i) { - double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY; - if (cycles < SYNC_PULSE_LENGTH / SYNC_TEST_TOLERANCE || cycles > SYNC_PULSE_LENGTH * SYNC_TEST_TOLERANCE) { - fprintf(stderr, "Sync cycle with downflank at %.6f was detected at %.0f cycles; misdetect?\n", - pulses[i].time, cycles); - } - } - - // Compute the standard deviation (to check for uneven speeds). - double sum2 = 0.0; - for (int i = SYNC_PULSE_START; i < SYNC_PULSE_END; ++i) { - double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY; - sum2 += (cycles - SYNC_PULSE_LENGTH) * (cycles - SYNC_PULSE_LENGTH); + // Check for pulses outside +/- 10% (sign of misdetection). + for (int i = SYNC_PULSE_START; i < sync_pulse_end; ++i) { + double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY; + if (cycles < SYNC_PULSE_LENGTH / SYNC_TEST_TOLERANCE || cycles > SYNC_PULSE_LENGTH * SYNC_TEST_TOLERANCE) { + fprintf(stderr, "Sync cycle with downflank at %.6f was detected at %.0f cycles; misdetect?\n", + pulses[i].time, cycles); } - double stddev = sqrt(sum2 / (SYNC_PULSE_END - SYNC_PULSE_START - 1)); - fprintf(stderr, "Sync pulse length standard deviation: %.2f cycles\n", - stddev); } - FILE *fp = fopen("cycles.plot", "w"); + return calibration_factor; +} + +void output_tap(const std::vector& pulses, double calibration_factor) +{ std::vector tap_data; for (unsigned i = 0; i < pulses.size(); ++i) { double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY; - fprintf(fp, "%f %f\n", pulses[i].time, cycles); int len = lrintf(cycles / TAP_RESOLUTION); if (i > SYNC_PULSE_END && (cycles < 100 || cycles > 800)) { fprintf(stderr, "Cycle with downflank at %.6f was detected at %.0f cycles; misdetect?\n", @@ -169,7 +172,6 @@ int main(int argc, char **argv) tap_data.push_back(overflow_len >> 16); } } - fclose(fp); tap_header hdr; memcpy(hdr.identifier, "C64-TAPE-RAW", 12); @@ -179,20 +181,455 @@ int main(int argc, char **argv) fwrite(&hdr, sizeof(hdr), 1, stdout); fwrite(tap_data.data(), tap_data.size(), 1, stdout); +} + +static struct option long_options[] = { + {"auto-level", 0, 0, 'a' }, + {"auto-level-freq", required_argument, 0, 'b' }, + {"output-leveled", 0, 0, 'A' }, + {"min-level", required_argument, 0, 'm' }, + {"no-calibrate", 0, 0, 's' }, + {"plot-cycles", 0, 0, 'p' }, + {"hysteresis-limit", required_argument, 0, 'l' }, + {"filter", required_argument, 0, 'f' }, + {"rc-filter", required_argument, 0, 'r' }, + {"output-filtered", 0, 0, 'F' }, + {"crop", required_argument, 0, 'c' }, + {"train", required_argument, 0, 't' }, + {"quiet", 0, 0, 'q' }, + {"help", 0, 0, 'h' }, + {0, 0, 0, 0 } +}; + +void help() +{ + fprintf(stderr, "decode [OPTIONS] AUDIO-FILE > TAP-FILE\n"); + fprintf(stderr, "\n"); + fprintf(stderr, " -a, --auto-level automatically adjust amplitude levels throughout the file\n"); + fprintf(stderr, " -b, --auto-level-freq minimum frequency in Hertz of corrected level changes (default 200 Hz)\n"); + fprintf(stderr, " -A, --output-leveled output leveled waveform to leveled.raw\n"); + fprintf(stderr, " -m, --min-level minimum estimated sound level (0..1) for --auto-level\n"); + fprintf(stderr, " -s, --no-calibrate do not try to calibrate on sync pulse length\n"); + fprintf(stderr, " -p, --plot-cycles output debugging info to cycles.plot\n"); + fprintf(stderr, " -l, --hysteresis-limit U[:L] change amplitude threshold for ignoring pulses (-1..1)\n"); + fprintf(stderr, " -f, --filter C1:C2:C3:... specify FIR filter (up to %d coefficients)\n", NUM_FILTER_COEFF); + fprintf(stderr, " -r, --rc-filter FREQ send signal through a highpass RC filter with given frequency (in Hertz)\n"); + fprintf(stderr, " -F, --output-filtered output filtered waveform to filtered.raw\n"); + fprintf(stderr, " -c, --crop START[:END] use only the given part of the file\n"); + fprintf(stderr, " -t, --train LEN1:LEN2:... train a filter for detecting any of the given number of cycles\n"); + fprintf(stderr, " (implies --no-calibrate and --quiet unless overridden)\n"); + fprintf(stderr, " -q, --quiet suppress some informational messages\n"); + fprintf(stderr, " -h, --help display this help, then exit\n"); + exit(1); +} + +void parse_options(int argc, char **argv) +{ + for ( ;; ) { + int option_index = 0; + int c = getopt_long(argc, argv, "ab:Am:spl:f:r:Fc:t:qh", long_options, &option_index); + if (c == -1) + break; + + switch (c) { + case 'a': + do_auto_level = true; + break; + + case 'b': + auto_level_freq = atof(optarg); + break; + + case 'A': + output_leveled = true; + break; + + case 'm': + min_level = atof(optarg); + break; + + case 's': + do_calibrate = false; + break; + + case 'p': + output_cycles_plot = true; + break; + + case 'l': { + const char *hyststr = strtok(optarg, ": "); + hysteresis_upper_limit = atof(hyststr); + hyststr = strtok(NULL, ": "); + if (hyststr == NULL) { + hysteresis_lower_limit = -hysteresis_upper_limit; + } else { + hysteresis_lower_limit = atof(hyststr); + } + break; + } + + case 'f': { + const char *coeffstr = strtok(optarg, ": "); + int coeff_index = 0; + while (coeff_index < NUM_FILTER_COEFF && coeffstr != NULL) { + filter_coeff[coeff_index++] = atof(coeffstr); + coeffstr = strtok(NULL, ": "); + } + use_fir_filter = true; + break; + } + + case 'r': + use_rc_filter = true; + rc_filter_freq = atof(optarg); + break; + + case 'F': + output_filtered = true; + break; + + case 'c': { + const char *cropstr = strtok(optarg, ":"); + crop_start = atof(cropstr); + cropstr = strtok(NULL, ":"); + if (cropstr == NULL) { + crop_end = HUGE_VAL; + } else { + crop_end = atof(cropstr); + } + do_crop = true; + break; + } + + case 't': { + const char *cyclestr = strtok(optarg, ":"); + while (cyclestr != NULL) { + train_snap_points.push_back(atof(cyclestr)); + cyclestr = strtok(NULL, ":"); + } + do_train = true; + + // Set reasonable defaults (can be overridden later on the command line). + do_calibrate = false; + quiet = true; + break; + } + + case 'q': + quiet = true; + break; + + case 'h': + default: + help(); + exit(1); + } + } +} + +std::vector crop(const std::vector& pcm, float crop_start, float crop_end, int sample_rate) +{ + size_t start_sample, end_sample; + if (crop_start >= 0.0f) { + start_sample = std::min(lrintf(crop_start * sample_rate), pcm.size()); + } + if (crop_end >= 0.0f) { + end_sample = std::min(lrintf(crop_end * sample_rate), pcm.size()); + } + return std::vector(pcm.begin() + start_sample, pcm.begin() + end_sample); +} + +std::vector do_fir_filter(const std::vector& pcm, const float* filter) +{ + std::vector filtered_pcm; + filtered_pcm.resize(pcm.size()); + unsigned i = NUM_FILTER_COEFF; +#ifdef __AVX__ + unsigned avx_end = i + ((pcm.size() - i) & ~7); + for ( ; i < avx_end; i += 8) { + __m256 s = _mm256_setzero_ps(); + for (int j = 0; j < NUM_FILTER_COEFF; ++j) { + __m256 f = _mm256_set1_ps(filter[j]); + s = _mm256_fmadd_ps(f, _mm256_load_ps(&pcm[i - j]), s); + } + _mm256_storeu_ps(&filtered_pcm[i], s); + } +#endif + // Do what we couldn't do with AVX (which is everything for non-AVX machines) + // as scalar code. + for (; i < pcm.size(); ++i) { + float s = 0.0f; + for (int j = 0; j < NUM_FILTER_COEFF; ++j) { + s += filter[j] * pcm[i - j]; + } + filtered_pcm[i] = s; + } + + if (output_filtered) { + FILE *fp = fopen("filtered.raw", "wb"); + fwrite(filtered_pcm.data(), filtered_pcm.size() * sizeof(filtered_pcm[0]), 1, fp); + fclose(fp); + } - // Output a debug raw file with pulse detection points. - fp = fopen("debug.raw", "wb"); - short one = 32767; - short zero = 0; - unsigned pulsenum = 0; + return filtered_pcm; +} + +std::vector do_rc_filter(const std::vector& pcm, float freq, int sample_rate) +{ + // This is only a 6 dB/oct filter, which seemingly works better + // than the Filter class, which is a standard biquad (12 dB/oct). + // The b/c calculations come from libnyquist (atone.c); + // I haven't checked, but I suppose they fall out of the bilinear + // transform of the transfer function H(s) = s/(s + w). + std::vector filtered_pcm; + filtered_pcm.resize(pcm.size()); + const float b = 2.0f - cos(2.0 * M_PI * freq / sample_rate); + const float c = b - sqrt(b * b - 1.0f); + float prev_in = 0.0f; + float prev_out = 0.0f; for (unsigned i = 0; i < pcm.size(); ++i) { - unsigned next_pulse = (pulsenum >= pulses.size()) ? INT_MAX : int(pulses[pulsenum].time * sample_rate); - if (i >= next_pulse) { - fwrite(&one, sizeof(one), 1, fp); - ++pulsenum; - } else { - fwrite(&zero, sizeof(zero), 1, fp); + float in = pcm[i]; + float out = c * (prev_out + in - prev_in); + filtered_pcm[i] = out; + prev_in = in; + prev_out = out; + } + + if (output_filtered) { + FILE *fp = fopen("filtered.raw", "wb"); + fwrite(filtered_pcm.data(), filtered_pcm.size() * sizeof(filtered_pcm[0]), 1, fp); + fclose(fp); + } + + return filtered_pcm; +} + +template +std::vector detect_pulses(const std::vector &pcm, float hysteresis_upper_limit, float hysteresis_lower_limit, int sample_rate) +{ + std::vector pulses; + + // Find the flanks. + enum State { START, ABOVE, BELOW } state = START; + double last_downflank = -1; + for (unsigned i = 0; i < pcm.size(); ++i) { + if (pcm[i] > hysteresis_upper_limit) { + state = ABOVE; + } else if (pcm[i] < hysteresis_lower_limit) { + if (state == ABOVE) { + // down-flank! + double t = find_crossing(pcm, i - 1, hysteresis_lower_limit) * (1.0 / sample_rate) + crop_start; + if (last_downflank > 0) { + pulse p; + p.time = t; + p.len = t - last_downflank; + pulses.push_back(p); + } + last_downflank = t; + } + state = BELOW; } } + return pulses; +} + +void output_cycle_plot(const std::vector &pulses, double calibration_factor) +{ + FILE *fp = fopen("cycles.plot", "w"); + for (unsigned i = 0; i < pulses.size(); ++i) { + double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY; + fprintf(fp, "%f %f\n", pulses[i].time, cycles); + } fclose(fp); } + +std::pair find_closest_point(double x, const std::vector &points) +{ + int best_point = 0; + double best_dist = (x - points[0]) * (x - points[0]); + for (unsigned j = 1; j < train_snap_points.size(); ++j) { + double dist = (x - points[j]) * (x - points[j]); + if (dist < best_dist) { + best_point = j; + best_dist = dist; + } + } + return std::make_pair(best_point, best_dist); +} + +float eval_badness(const std::vector& pulses, double calibration_factor) +{ + double sum_badness = 0.0; + for (unsigned i = 0; i < pulses.size(); ++i) { + double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY; + if (cycles > 2000.0) cycles = 2000.0; // Don't make pauses arbitrarily bad. + std::pair selected_point_and_sq_dist = find_closest_point(cycles, train_snap_points); + sum_badness += selected_point_and_sq_dist.second; + } + return sqrt(sum_badness / (pulses.size() - 1)); +} + +void find_kmeans(const std::vector &pulses, double calibration_factor, const std::vector &initial_centers) +{ + std::vector last_centers = initial_centers; + std::vector sums; + std::vector num; + sums.resize(initial_centers.size()); + num.resize(initial_centers.size()); + for ( ;; ) { + for (unsigned i = 0; i < initial_centers.size(); ++i) { + sums[i] = 0.0f; + num[i] = 0; + } + for (unsigned i = 0; i < pulses.size(); ++i) { + double cycles = pulses[i].len * calibration_factor * C64_FREQUENCY; + // Ignore heavy outliers, which are almost always long pauses. + if (cycles > 2000.0) { + continue; + } + std::pair selected_point_and_sq_dist = find_closest_point(cycles, last_centers); + int p = selected_point_and_sq_dist.first; + sums[p] += cycles; + ++num[p]; + } + bool any_moved = false; + for (unsigned i = 0; i < initial_centers.size(); ++i) { + if (num[i] == 0) { + fprintf(stderr, "K-means broke down, can't output new reference training points\n"); + return; + } + float new_center = sums[i] / num[i]; + if (fabs(new_center - last_centers[i]) > 1e-3) { + any_moved = true; + } + last_centers[i] = new_center; + } + if (!any_moved) { + break; + } + } + fprintf(stderr, "New reference training points:"); + for (unsigned i = 0; i < last_centers.size(); ++i) { + fprintf(stderr, " %.3f", last_centers[i]); + } + fprintf(stderr, "\n"); +} + +void spsa_train(const std::vector &pcm, int sample_rate) +{ + float vals[NUM_SPSA_VALS] = { hysteresis_upper_limit, hysteresis_lower_limit, 1.0f }; // The rest is filled with 0. + + float start_c = INITIAL_C; + double best_badness = HUGE_VAL; + + for (int n = 1; n < NUM_ITER; ++n) { + float a = INITIAL_A * pow(n + A, -ALPHA); + float c = start_c * pow(n, -GAMMA); + + // find a random perturbation + float p[NUM_SPSA_VALS]; + float vals1[NUM_SPSA_VALS], vals2[NUM_SPSA_VALS]; + for (int i = 0; i < NUM_SPSA_VALS; ++i) { + p[i] = (rand() % 2) ? 1.0 : -1.0; + vals1[i] = std::max(std::min(vals[i] - c * p[i], 1.0f), -1.0f); + vals2[i] = std::max(std::min(vals[i] + c * p[i], 1.0f), -1.0f); + } + + std::vector pulses1 = detect_pulses(do_fir_filter(pcm, vals1 + 2), vals1[0], vals1[1], sample_rate); + std::vector pulses2 = detect_pulses(do_fir_filter(pcm, vals2 + 2), vals2[0], vals2[1], sample_rate); + float badness1 = eval_badness(pulses1, 1.0); + float badness2 = eval_badness(pulses2, 1.0); + + // Find the gradient estimator + float g[NUM_SPSA_VALS]; + for (int i = 0; i < NUM_SPSA_VALS; ++i) { + g[i] = (badness2 - badness1) / (2.0 * c * p[i]); + vals[i] -= a * g[i]; + vals[i] = std::max(std::min(vals[i], 1.0f), -1.0f); + } + if (badness2 < badness1) { + std::swap(badness1, badness2); + std::swap(vals1, vals2); + std::swap(pulses1, pulses2); + } + if (badness1 < best_badness) { + fprintf(stderr, "\nNew best filter (badness=%f):", badness1); + for (int i = 0; i < NUM_FILTER_COEFF; ++i) { + fprintf(stderr, " %.5f", vals1[i + 2]); + } + fprintf(stderr, ", hysteresis limits = %f %f\n", vals1[0], vals1[1]); + best_badness = badness1; + + find_kmeans(pulses1, 1.0, train_snap_points); + + if (output_cycles_plot) { + output_cycle_plot(pulses1, 1.0); + } + } + fprintf(stderr, "%d ", n); + fflush(stderr); + } +} + +int main(int argc, char **argv) +{ + parse_options(argc, argv); + + make_lanczos_weight_table(); + std::vector pcm; + int sample_rate; + if (!read_audio_file(argv[optind], &pcm, &sample_rate)) { + exit(1); + } + + if (do_crop) { + pcm = crop(pcm, crop_start, crop_end, sample_rate); + } + + if (use_fir_filter) { + pcm = do_fir_filter(pcm, filter_coeff); + } + + if (use_rc_filter) { + pcm = do_rc_filter(pcm, rc_filter_freq, sample_rate); + } + + if (do_auto_level) { + pcm = level_samples(pcm, min_level, auto_level_freq, sample_rate); + if (output_leveled) { + FILE *fp = fopen("leveled.raw", "wb"); + fwrite(pcm.data(), pcm.size() * sizeof(pcm[0]), 1, fp); + fclose(fp); + } + } + +#if 0 + for (int i = 0; i < LEN; ++i) { + in[i] += rand() % 10000; + } +#endif + +#if 0 + for (int i = 0; i < LEN; ++i) { + printf("%d\n", in[i]); + } +#endif + + if (do_train) { + spsa_train(pcm, sample_rate); + exit(0); + } + + std::vector pulses = detect_pulses(pcm, hysteresis_upper_limit, hysteresis_lower_limit, sample_rate); + + double calibration_factor = 1.0; + if (do_calibrate) { + calibration_factor = calibrate(pulses); + } + + if (output_cycles_plot) { + output_cycle_plot(pulses, calibration_factor); + } + + output_tap(pulses, calibration_factor); +}