Add support for decoding only parts of the sample.

[c64tapwav] / decode.cpp

diff --git a/decode.cpp b/decode.cpp
index 94bc1dd1668abe0482d6071e3de40082741806e1..43d6766bfd38774c58f4b6fc4601f7c18f598bb0 100644 (file)
--- a/decode.cpp
+++ b/decode.cpp
@@ -24,8 +24,11 @@ static float hysteresis_limit = 3000.0 / 32768.0;
 static bool do_calibrate = true;
 static bool output_cycles_plot = false;
 static bool use_filter = false;
+static bool do_crop = false;
+static float crop_start = 0.0f, crop_end = HUGE_VAL;
 static float filter_coeff[NUM_FILTER_COEFF] = { 1.0f };  // The rest is filled with 0.
 static bool output_filtered = false;
+static bool quiet = false;
 
 // between [x,x+1]
 double find_zerocrossing(const std::vector<float> &pcm, int x)
@@ -88,8 +91,10 @@ double calibrate(const std::vector<pulse> &pulses) {
                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) {
@@ -97,8 +102,10 @@ double calibrate(const std::vector<pulse> &pulses) {
        }
        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) {
@@ -149,6 +156,8 @@ static struct option long_options[] = {
        {"hysteresis-limit", required_argument, 0, 'l' },
        {"filter",           required_argument, 0, 'f' },
        {"output-filtered",  0,                 0, 'F' },
+       {"crop",             required_argument, 0, 'c' },
+       {"quiet",            0,                 0, 'q' },
        {"help",             0,                 0, 'h' },
        {0,                  0,                 0, 0   }
 };
@@ -162,6 +171,8 @@ void help()
        fprintf(stderr, "  -l, --hysteresis-limit VAL   change amplitude threshold for ignoring pulses (0..32768)\n");
        fprintf(stderr, "  -f, --filter C1:C2:C3:...    specify FIR filter (up to %d coefficients)\n", NUM_FILTER_COEFF);
        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, "  -q, --quiet                  suppress some informational messages\n");
        fprintf(stderr, "  -h, --help                   display this help, then exit\n");
        exit(1);
 }
@@ -170,7 +181,7 @@ void parse_options(int argc, char **argv)
 {
        for ( ;; ) {
                int option_index = 0;
-               int c = getopt_long(argc, argv, "spl:f:Fh", long_options, &option_index);
+               int c = getopt_long(argc, argv, "spl:f:Fc:qh", long_options, &option_index);
                if (c == -1)
                        break;
 
@@ -182,6 +193,7 @@ void parse_options(int argc, char **argv)
                case 'p':
                        output_cycles_plot = true;
                        break;
+
                case 'l':
                        hysteresis_limit = atof(optarg) / 32768.0;
                        break;
@@ -201,6 +213,23 @@ void parse_options(int argc, char **argv)
                        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 'q':
+                       quiet = true;
+                       break;
+
                case 'h':
                default:
                        help();
@@ -209,6 +238,18 @@ void parse_options(int argc, char **argv)
        }
 }
 
+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);
+}
+
 // TODO: Support AVX here.
 std::vector<float> do_filter(const std::vector<float>& pcm, const float* filter)
 {
@@ -231,34 +272,9 @@ std::vector<float> do_filter(const std::vector<float>& pcm, const float* filter)
        return filtered_pcm;
 }
 
-int main(int argc, char **argv)
+std::vector<pulse> detect_pulses(const std::vector<float> &pcm, int sample_rate)
 {
-       parse_options(argc, argv);
-
-       make_lanczos_weight_table();
-       std::vector<float> pcm;
-       int sample_rate;
-       if (!read_audio_file(argv[optind], &pcm, &sample_rate)) {
-               exit(1);
-       }
-
-       if (use_filter) {
-               pcm = do_filter(pcm, filter_coeff);
-       }
-
-#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
-
-       std::vector<pulse> pulses;  // in seconds
+       std::vector<pulse> pulses;
 
        // Find the flanks.
        int last_bit = -1;
@@ -289,7 +305,7 @@ int main(int argc, char **argv)
                        } 
 
                        // down-flank!
-                       double t = find_zerocrossing(pcm, i - 1) * (1.0 / sample_rate);
+                       double t = find_zerocrossing(pcm, i - 1) * (1.0 / sample_rate) + crop_start;
                        if (last_downflank > 0) {
                                pulse p;
                                p.time = t;
@@ -300,6 +316,41 @@ int main(int argc, char **argv)
                }
                last_bit = bit;
        }
+       return pulses;
+}
+
+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[optind], &pcm, &sample_rate)) {
+               exit(1);
+       }
+
+       if (do_crop) {
+               pcm = crop(pcm, crop_start, crop_end, sample_rate);
+       }
+
+       if (use_filter) {
+               pcm = do_filter(pcm, filter_coeff);
+       }
+
+#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
+
+       std::vector<pulse> pulses = detect_pulses(pcm, sample_rate);
 
        double calibration_factor = 1.0;
        if (do_calibrate) {