#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.
// 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;
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]);
}
}