+// Copyright Steinar H. Gunderson <sgunderson@bigfoot.com>
+// Licensed under the GPL, v2. (See the file COPYING.)
+
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <assert.h>
#include <limits.h>
+#include <getopt.h>
+#ifdef __AVX__
+#include <immintrin.h>
+#endif
#include <vector>
#include <algorithm>
#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 20000
#define SYNC_PULSE_LENGTH 378.0
#define SYNC_TEST_TOLERANCE 1.10
-// between [x,x+1]
-double find_zerocrossing(const std::vector<float> &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<float> 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 <limit> between [x,x+1]
+template<bool fast>
+double find_crossing(const std::vector<float> &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 {
sync_pulse_end = try_end;
sync_pulse_stddev = stddev;
}
- fprintf(stderr, "Sync pulse length standard deviation: %.2f cycles\n",
- sync_pulse_stddev);
+ if (!quiet) {
+ fprintf(stderr, "Sync pulse length standard deviation: %.2f cycles\n",
+ sync_pulse_stddev);
+ }
double sum = 0.0;
for (int i = SYNC_PULSE_START; i < sync_pulse_end; ++i) {
}
double mean_length = C64_FREQUENCY * sum / (sync_pulse_end - SYNC_PULSE_START);
double calibration_factor = SYNC_PULSE_LENGTH / mean_length;
- fprintf(stderr, "Calibrated sync pulse length: %.2f -> %.2f (change %+.2f%%)\n",
- mean_length, SYNC_PULSE_LENGTH, 100.0 * (calibration_factor - 1.0));
+ 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) {
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<float> crop(const std::vector<float>& 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<size_t>(lrintf(crop_start * sample_rate), pcm.size());
+ }
+ if (crop_end >= 0.0f) {
+ end_sample = std::min<size_t>(lrintf(crop_end * sample_rate), pcm.size());
+ }
+ return std::vector<float>(pcm.begin() + start_sample, pcm.begin() + end_sample);
+}
+
+std::vector<float> do_fir_filter(const std::vector<float>& pcm, const float* filter)
+{
+ std::vector<float> 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);
+ }
+
+ return filtered_pcm;
+}
+
+std::vector<float> do_rc_filter(const std::vector<float>& 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<float> 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) {
+ 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<bool fast>
+std::vector<pulse> detect_pulses(const std::vector<float> &pcm, float hysteresis_upper_limit, float hysteresis_lower_limit, int sample_rate)
+{
+ std::vector<pulse> 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<fast>(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<pulse> &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<int, double> find_closest_point(double x, const std::vector<float> &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<pulse>& 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<int, double> 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<pulse> &pulses, double calibration_factor, const std::vector<float> &initial_centers)
+{
+ std::vector<float> last_centers = initial_centers;
+ std::vector<float> sums;
+ std::vector<float> 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<int, double> 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<float> &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<pulse> pulses1 = detect_pulses<true>(do_fir_filter(pcm, vals1 + 2), vals1[0], vals1[1], sample_rate);
+ std::vector<pulse> pulses2 = detect_pulses<true>(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<float> pcm;
int sample_rate;
- if (!read_audio_file(argv[1], &pcm, &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
- std::vector<pulse> pulses; // in seconds
+ if (do_train) {
+ spsa_train(pcm, sample_rate);
+ exit(0);
+ }
- // 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<int>(min_level_after, pcm[j]);
- if (pcm[j] > 0) break;
- if (pcm[j] < -HYSTERESIS_LIMIT) {
- true_pulse = true;
- break;
- }
- }
+ std::vector<pulse> pulses = detect_pulses<false>(pcm, hysteresis_upper_limit, hysteresis_lower_limit, sample_rate);
- 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;
- }
- last_bit = bit;
+ double calibration_factor = 1.0;
+ if (do_calibrate) {
+ calibration_factor = calibrate(pulses);
}
- double calibration_factor = calibrate(pulses);
-
- FILE *fp = fopen("cycles.plot", "w");
- std::vector<char> 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);
+ if (output_cycles_plot) {
+ output_cycle_plot(pulses, calibration_factor);
}
- fclose(fp);
output_tap(pulses, calibration_factor);
}
projects / c64tapwav / blobdiff