Integrate the leveler into decode.

[c64tapwav] / level.cpp

diff --git a/level.cpp b/level.cpp
index 36377991764e0daae030ade42a95b688f5421041..fea59261d3c585d168081bab12edf5a778c352ed 100644 (file)
--- a/level.cpp
+++ b/level.cpp
@@ -6,9 +6,6 @@
 #include <vector>
 #include <algorithm>
 
-#define BUFSIZE 4096
-#define WAVE_FREQ 44100.0
-
 // The frequency to filter on, in Hertz. Larger values makes the
 // compressor react faster, but if it is too large, you'll
 // ruin the waveforms themselves.
@@ -27,21 +24,6 @@
 // 6dB/oct per round.
 #define FILTER_DEPTH 4
 
-struct stereo_sample {
-       short left, right;
-};
-
-inline short clip(int x)
-{
-       if (x < -32768) {
-               return x;
-       } else if (x > 32767) {
-               return 32767;
-       } else {
-               return short(x);
-       }
-}
-
 static float a1, a2, b0, b1, b2;
 static float d0, d1;
 
@@ -77,46 +59,38 @@ static float filter_update(float in)
        return out;
 }
 
-int main(int argc, char **argv)
+std::vector<float> level_samples(const std::vector<float> &pcm, int sample_rate)
 {
-       std::vector<short> pcm;
-
-       while (!feof(stdin)) {
-               short buf[BUFSIZE];
-               ssize_t ret = fread(buf, sizeof(short), BUFSIZE, stdin);
-               if (ret >= 0) {
-                       pcm.insert(pcm.end(), buf, buf + ret);
-               }
-       }
-       
        // filter forwards, then backwards (perfect phase filtering)
-       std::vector<float> filtered_samples, refiltered_samples;
+       std::vector<float> filtered_samples, refiltered_samples, leveled_samples;
        filtered_samples.resize(pcm.size());
        refiltered_samples.resize(pcm.size());
+       leveled_samples.resize(pcm.size());
 
-       filter_init(M_PI * LPFILTER_FREQ / WAVE_FREQ);
+       filter_init(M_PI * LPFILTER_FREQ / sample_rate);
        for (unsigned i = 0; i < pcm.size(); ++i) {
                filtered_samples[i] = filter_update(fabs(pcm[i]));
        }
-       filter_init(M_PI * LPFILTER_FREQ / WAVE_FREQ);
+       filter_init(M_PI * LPFILTER_FREQ / sample_rate);
        for (unsigned i = pcm.size(); i --> 0; ) {
                refiltered_samples[i] = filter_update(filtered_samples[i]);
        }
 
        for (int i = 1; i < FILTER_DEPTH; ++i) {
-               filter_init(M_PI * LPFILTER_FREQ / WAVE_FREQ);
+               filter_init(M_PI * LPFILTER_FREQ / sample_rate);
                for (unsigned i = 0; i < pcm.size(); ++i) {
                        filtered_samples[i] = filter_update(refiltered_samples[i]);
                }
-               filter_init(M_PI * LPFILTER_FREQ / WAVE_FREQ);
+               filter_init(M_PI * LPFILTER_FREQ / sample_rate);
                for (unsigned i = pcm.size(); i --> 0; ) {
                        refiltered_samples[i] = filter_update(filtered_samples[i]);
                }
        }
 
        for (unsigned i = 0; i < pcm.size(); ++i) {
-               float f = DAMPENING_FACTOR * std::max(refiltered_samples[i] * (1.0 / 32768.0), MIN_LEVEL);
-               short s = clip(lrintf(pcm[i] / f));
-               fwrite(&s, sizeof(s), 1, stdout);
+               float f = DAMPENING_FACTOR * std::max<float>(refiltered_samples[i], MIN_LEVEL);
+               leveled_samples[i] = pcm[i] / f;
        }
+
+       return leveled_samples;
 }