Compile in native by default, since AVX helps so much.

[c64tapwav] / level.cpp

diff --git a/level.cpp b/level.cpp
index e95b48e0f6d1c5e1f03a041760308745f0624090..504cf0bbf7549170600674140fff7460c14c85f9 100644 (file)
--- a/level.cpp
+++ b/level.cpp
@@ -6,10 +6,7 @@
 #include <vector>
 #include <algorithm>
 
-// 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.
-#define LPFILTER_FREQ 50.0
+#include "filter.h"
 
 // A final scalar to get the audio within approximately the right range.
 // Increase to _lower_ overall volume.
@@ -18,42 +15,7 @@
 // 6dB/oct per round.
 #define FILTER_DEPTH 4
 
-static float a1, a2, b0, b1, b2;
-static float d0, d1;
-
-static void filter_init(float cutoff_radians)
-{
-       float resonance = 1.0f / sqrt(2.0f);
-       float sn = sin(cutoff_radians), cs = cos(cutoff_radians);
-       float alpha = float(sn / (2 * resonance));
-
-       // coefficients for lowpass filter
-        float a0 = 1 + alpha;
-       b0 = (1 - cs) * 0.5f;
-       b1 = 1 - cs;
-       b2 = b0;
-        a1 = -2 * cs;
-        a2 = 1 - alpha;
-
-       b0 /= a0;
-       b1 /= a0;
-       b2 /= a0;
-       a1 /= a0;
-       a2 /= a0;
-
-       // reset filter delays
-       d0 = d1 = 0.0f;
-}
-
-static float filter_update(float in)
-{
-       float out = b0*in + d0;
-       d0 = b1 * in - a1 * out + d1;
-       d1 = b2 * in - a2 * out;
-       return out;
-}
-
-std::vector<float> level_samples(const std::vector<float> &pcm, float min_level, int sample_rate)
+std::vector<float> level_samples(const std::vector<float> &pcm, float min_level, float lpfilter_freq, int sample_rate)
 {
        // filter forwards, then backwards (perfect phase filtering)
        std::vector<float> filtered_samples, refiltered_samples, leveled_samples;
@@ -61,23 +23,23 @@ std::vector<float> level_samples(const std::vector<float> &pcm, float min_level,
        refiltered_samples.resize(pcm.size());
        leveled_samples.resize(pcm.size());
 
-       filter_init(M_PI * LPFILTER_FREQ / sample_rate);
+       Filter filter = Filter::lpf(2.0 * M_PI * lpfilter_freq / sample_rate);
        for (unsigned i = 0; i < pcm.size(); ++i) {
-               filtered_samples[i] = filter_update(fabs(pcm[i]));
+               filtered_samples[i] = filter.update(fabs(pcm[i]));
        }
-       filter_init(M_PI * LPFILTER_FREQ / sample_rate);
+       filter.reset();
        for (unsigned i = pcm.size(); i --> 0; ) {
-               refiltered_samples[i] = filter_update(filtered_samples[i]);
+               refiltered_samples[i] = filter.update(filtered_samples[i]);
        }
 
        for (int i = 1; i < FILTER_DEPTH; ++i) {
-               filter_init(M_PI * LPFILTER_FREQ / sample_rate);
+               filter.reset();
                for (unsigned i = 0; i < pcm.size(); ++i) {
-                       filtered_samples[i] = filter_update(refiltered_samples[i]);
+                       filtered_samples[i] = filter.update(refiltered_samples[i]);
                }
-               filter_init(M_PI * LPFILTER_FREQ / sample_rate);
+               filter.reset();
                for (unsigned i = pcm.size(); i --> 0; ) {
-                       refiltered_samples[i] = filter_update(filtered_samples[i]);
+                       refiltered_samples[i] = filter.update(filtered_samples[i]);
                }
        }