X-Git-Url: https://git.sesse.net/?p=c64tapwav;a=blobdiff_plain;f=decode.cpp;h=d2d6fc89a26aeecee613f6ddb446d35571d35365;hp=f645588923bcb079f6f30b52f505f07535ab71f8;hb=e283d8b328e737d7119263010f63b7452fa05e6a;hpb=d60a4236e895ac9ff1e3aa46be89707bace121d8 diff --git a/decode.cpp b/decode.cpp index f645588..d2d6fc8 100644 --- a/decode.cpp +++ b/decode.cpp @@ -7,6 +7,9 @@ #include #include #include +#ifdef __AVX__ +#include +#endif #include #include @@ -25,6 +28,7 @@ // 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... @@ -32,7 +36,8 @@ #define GAMMA 0.166 #define ALPHA 1.0 -static float hysteresis_limit = 3000.0 / 32768.0; +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; @@ -50,37 +55,39 @@ 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; -// between [x,x+1] -double find_zerocrossing(const std::vector &pcm, int x) +// search for the value between [x,x+1] +template +double find_crossing(const std::vector &pcm, int x, float limit) { - if (pcm[x] == 0) { - return x; - } - if (pcm[x + 1] == 0) { - return x + 1; - } - - assert(pcm[x + 1] < 0); - assert(pcm[x] > 0); - - 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; + 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 { @@ -178,7 +185,9 @@ void output_tap(const std::vector& pulses, double calibration_factor) 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' }, @@ -186,6 +195,7 @@ static struct option long_options[] = { {"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 } @@ -196,11 +206,12 @@ 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..32768) for --auto-level\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 VAL change amplitude threshold for ignoring pulses (0..32768)\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"); @@ -216,7 +227,7 @@ void parse_options(int argc, char **argv) { for ( ;; ) { int option_index = 0; - int c = getopt_long(argc, argv, "aAm:spl:f:r:Fc:t:qh", long_options, &option_index); + int c = getopt_long(argc, argv, "ab:Am:spl:f:r:Fc:t:qh", long_options, &option_index); if (c == -1) break; @@ -225,12 +236,16 @@ void parse_options(int argc, char **argv) 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) / 32768.0; + min_level = atof(optarg); break; case 's': @@ -241,9 +256,17 @@ void parse_options(int argc, char **argv) output_cycles_plot = true; break; - case 'l': - hysteresis_limit = atof(optarg) / 32768.0; + 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, ": "); @@ -316,17 +339,30 @@ std::vector crop(const std::vector& pcm, float crop_start, float c return std::vector(pcm.begin() + start_sample, pcm.begin() + end_sample); } -// TODO: Support AVX here. std::vector do_fir_filter(const std::vector& pcm, const float* filter) { std::vector filtered_pcm; - filtered_pcm.reserve(pcm.size()); - for (unsigned i = NUM_FILTER_COEFF; i < pcm.size(); ++i) { + 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.push_back(s); + filtered_pcm[i] = s; } if (output_filtered) { @@ -368,49 +404,31 @@ std::vector do_rc_filter(const std::vector& pcm, float freq, int s return filtered_pcm; } -std::vector detect_pulses(const std::vector &pcm, int sample_rate) +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. - int last_bit = -1; + enum State { START, ABOVE, BELOW } state = START; 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 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; + 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; } - - 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) + 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; } - last_bit = bit; } return pulses; } @@ -499,7 +517,7 @@ void find_kmeans(const std::vector &pulses, double calibration_factor, co void spsa_train(const std::vector &pcm, int sample_rate) { - float filter[NUM_FILTER_COEFF] = { 1.0f }; // The rest is filled with 0. + 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; @@ -509,38 +527,38 @@ void spsa_train(const std::vector &pcm, int sample_rate) float c = start_c * pow(n, -GAMMA); // find a random perturbation - float p[NUM_FILTER_COEFF]; - float filter1[NUM_FILTER_COEFF], filter2[NUM_FILTER_COEFF]; - for (int i = 0; i < NUM_FILTER_COEFF; ++i) { + 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; - filter1[i] = std::max(std::min(filter[i] - c * p[i], 1.0f), -1.0f); - filter2[i] = std::max(std::min(filter[i] + c * p[i], 1.0f), -1.0f); + 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, filter1), sample_rate); - std::vector pulses2 = detect_pulses(do_fir_filter(pcm, filter2), sample_rate); + 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_FILTER_COEFF]; - for (int i = 0; i < NUM_FILTER_COEFF; ++i) { + float g[NUM_SPSA_VALS]; + for (int i = 0; i < NUM_SPSA_VALS; ++i) { g[i] = (badness2 - badness1) / (2.0 * c * p[i]); - filter[i] -= a * g[i]; - filter[i] = std::max(std::min(filter[i], 1.0f), -1.0f); + 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(filter1, filter2); + std::swap(vals1, vals2); std::swap(pulses1, pulses2); } if (badness1 < best_badness) { - printf("\nNew best filter (badness=%f):", badness1); + fprintf(stderr, "\nNew best filter (badness=%f):", badness1); for (int i = 0; i < NUM_FILTER_COEFF; ++i) { - printf(" %.5f", filter1[i]); + fprintf(stderr, " %.5f", vals1[i + 2]); } + fprintf(stderr, ", hysteresis limits = %f %f\n", vals1[0], vals1[1]); best_badness = badness1; - printf("\n"); find_kmeans(pulses1, 1.0, train_snap_points); @@ -548,8 +566,8 @@ void spsa_train(const std::vector &pcm, int sample_rate) output_cycle_plot(pulses1, 1.0); } } - printf("%d ", n); - fflush(stdout); + fprintf(stderr, "%d ", n); + fflush(stderr); } } @@ -577,7 +595,7 @@ int main(int argc, char **argv) } if (do_auto_level) { - pcm = level_samples(pcm, min_level, sample_rate); + 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); @@ -602,7 +620,7 @@ int main(int argc, char **argv) exit(0); } - std::vector pulses = detect_pulses(pcm, sample_rate); + std::vector pulses = detect_pulses(pcm, hysteresis_upper_limit, hysteresis_lower_limit, sample_rate); double calibration_factor = 1.0; if (do_calibrate) {