*/
-#include <cmath>
#include "audio_filter.h"
#include "audio_buffers.h"
+#include <cmath>
using std::vector;
using std::min;
AudioFilter::sinc_blackman (float cutoff, bool invert) const
{
vector<float> ir (_M + 1);
-
+
/* Impulse response */
-
+
for (int i = 0; i <= _M; ++i) {
if (i == (_M / 2)) {
ir[i] = 2 * M_PI * cutoff;
ir[i] *= (0.42 - 0.5 * cos (2 * M_PI * i / _M) + 0.08 * cos (4 * M_PI * i / _M));
}
}
-
+
/* Normalise */
-
+
float sum = 0;
for (int i = 0; i <= _M; ++i) {
sum += ir[i];
}
-
+
for (int i = 0; i <= _M; ++i) {
ir[i] /= sum;
}
-
+
/* Frequency inversion (swapping low-pass for high-pass, or whatever) */
-
+
if (invert) {
for (int i = 0; i <= _M; ++i) {
ir[i] = -ir[i];
}
ir[_M / 2] += 1;
}
-
+
return ir;
}
AudioFilter::run (shared_ptr<AudioBuffers> in)
{
shared_ptr<AudioBuffers> out (new AudioBuffers (in->channels(), in->frames()));
-
+
if (!_tail) {
_tail.reset (new AudioBuffers (in->channels(), _M + 1));
_tail->make_silent ();
}
-
+
for (int i = 0; i < in->channels(); ++i) {
for (int j = 0; j < in->frames(); ++j) {
float s = 0;
s += in->data(i)[j - k] * _ir[k];
}
}
-
+
out->data(i)[j] = s;
}
}
-
+
int const amount = min (in->frames(), _tail->frames());
if (amount < _tail->frames ()) {
_tail->move (amount, 0, _tail->frames() - amount);
}
_tail->copy_from (in.get(), amount, in->frames() - amount, _tail->frames () - amount);
-
+
return out;
}
{
vector<float> lpf = sinc_blackman (lower, false);
vector<float> hpf = sinc_blackman (higher, true);
-
+
_ir.resize (_M + 1);
for (int i = 0; i <= _M; ++i) {
_ir[i] = lpf[i] + hpf[i];
}
-
+
/* We now have a band-stop, so invert for band-pass */
for (int i = 0; i <= _M; ++i) {
_ir[i] = -_ir[i];
}
-
+
_ir[_M / 2] += 1;
}