Make leveler minimum level configurable, since it is connected to hysteresis limit.

diff --git a/decode.cpp b/decode.cpp
index e8263eab4757ca30a903f96aa6c32cfbc3d1997e..b3b8aa3376f355b9fcad9e8cd3aa4fc9b1bf2b4b 100644 (file)
--- a/decode.cpp
+++ b/decode.cpp
@@ -42,6 +42,12 @@ static bool output_leveled = false;
 static std::vector<float> train_snap_points;
 static bool do_train = false;
 
+// 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<float> &pcm, int x)
 {
@@ -182,6 +188,7 @@ void help()
        fprintf(stderr, "\n");
        fprintf(stderr, "  -a, --auto-level             automatically adjust amplitude levels throughout the file\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, "  -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");
@@ -199,7 +206,7 @@ void parse_options(int argc, char **argv)
 {
        for ( ;; ) {
                int option_index = 0;
-               int c = getopt_long(argc, argv, "aAspl:f:Fc:t:qh", long_options, &option_index);
+               int c = getopt_long(argc, argv, "aAm:spl:f:Fc:t:qh", long_options, &option_index);
                if (c == -1)
                        break;
 
@@ -212,6 +219,10 @@ void parse_options(int argc, char **argv)
                        output_leveled = true;
                        break;
 
+               case 'm':
+                       min_level = atof(optarg) / 32768.0;
+                       break;
+
                case 's':
                        do_calibrate = false;
                        break;
@@ -459,7 +470,7 @@ int main(int argc, char **argv)
        }
 
        if (do_auto_level) {
-               pcm = level_samples(pcm, sample_rate);
+               pcm = level_samples(pcm, min_level, sample_rate);
                if (output_leveled) {
                        FILE *fp = fopen("leveled.raw", "wb");
                        fwrite(pcm.data(), pcm.size() * sizeof(pcm[0]), 1, fp);

diff --git a/level.cpp b/level.cpp
index fea59261d3c585d168081bab12edf5a778c352ed..e95b48e0f6d1c5e1f03a041760308745f0624090 100644 (file)
--- a/level.cpp
+++ b/level.cpp
@@ -11,12 +11,6 @@
 // ruin the waveforms themselves.
 #define LPFILTER_FREQ 50.0
 
-// The minimum estimated sound 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.
-#define MIN_LEVEL 0.05
-
 // A final scalar to get the audio within approximately the right range.
 // Increase to _lower_ overall volume.
 #define DAMPENING_FACTOR 5.0
@@ -59,7 +53,7 @@ static float filter_update(float in)
        return out;
 }
 
-std::vector<float> level_samples(const std::vector<float> &pcm, int sample_rate)
+std::vector<float> level_samples(const std::vector<float> &pcm, float min_level, int sample_rate)
 {
        // filter forwards, then backwards (perfect phase filtering)
        std::vector<float> filtered_samples, refiltered_samples, leveled_samples;
@@ -88,7 +82,7 @@ std::vector<float> level_samples(const std::vector<float> &pcm, int sample_rate)
        }
 
        for (unsigned i = 0; i < pcm.size(); ++i) {
-               float f = DAMPENING_FACTOR * std::max<float>(refiltered_samples[i], MIN_LEVEL);
+               float f = DAMPENING_FACTOR * std::max<float>(refiltered_samples[i], min_level);
                leveled_samples[i] = pcm[i] / f;
        }
 

diff --git a/level.h b/level.h
index 3f830d77f0a3718049863e4e929eff0c17bd4d6c..763e617958143c21bbf8c5ee31c84cb5b5751189 100644 (file)
--- a/level.h
+++ b/level.h
@@ -3,6 +3,6 @@
 
 #include <vector>
 
-std::vector<float> level_samples(const std::vector<float> &pcm, int sample_rate);
+std::vector<float> level_samples(const std::vector<float> &pcm, float min_level, int sample_rate);
 
 #endif  // !defined(_LEVEL_H)