/*
- Copyright (C) 2014 Carl Hetherington <cth@carlh.net>
+ Copyright (C) 2014-2021 Carl Hetherington <cth@carlh.net>
- This program is free software; you can redistribute it and/or modify
+ This file is part of DCP-o-matic.
+
+ DCP-o-matic is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
- This program is distributed in the hope that it will be useful,
+ DCP-o-matic is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
- along with this program; if not, write to the Free Software
- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ along with DCP-o-matic. If not, see <http://www.gnu.org/licenses/>.
*/
+
#include "audio_filter.h"
#include "audio_buffers.h"
+#include "maths_util.h"
+#include "util.h"
#include <cmath>
+
+using std::make_shared;
using std::min;
-using boost::shared_ptr;
+using std::shared_ptr;
-/** @return array of floats which the caller must destroy with delete[] */
-float *
+
+std::vector<float>
AudioFilter::sinc_blackman (float cutoff, bool invert) const
{
- float* ir = new float[_M + 1];
+ auto ir = std::vector<float>();
+ ir.resize(_M + 1);
/* Impulse response */
return ir;
}
-AudioFilter::~AudioFilter ()
-{
- delete[] _ir;
-}
shared_ptr<AudioBuffers>
AudioFilter::run (shared_ptr<const AudioBuffers> in)
{
- shared_ptr<AudioBuffers> out (new AudioBuffers (in->channels(), in->frames()));
+ auto out = make_shared<AudioBuffers>(in->channels(), in->frames());
if (!_tail) {
- _tail.reset (new AudioBuffers (in->channels(), _M + 1));
+ _tail = make_shared<AudioBuffers>(in->channels(), _M + 1);
_tail->make_silent ();
}
int const frames = in->frames ();
for (int i = 0; i < channels; ++i) {
- float* tail_p = _tail->data (i);
- float* in_p = in->data (i);
- float* out_p = out->data (i);
+ auto tail_p = _tail->data (i);
+ auto in_p = in->data (i);
+ auto out_p = out->data (i);
for (int j = 0; j < frames; ++j) {
float s = 0;
for (int k = 0; k <= _M; ++k) {
int const amount = min (in->frames(), _tail->frames());
if (amount < _tail->frames ()) {
- _tail->move (amount, 0, _tail->frames() - amount);
+ _tail->move (_tail->frames() - amount, amount, 0);
}
_tail->copy_from (in.get(), amount, in->frames() - amount, _tail->frames () - amount);
return out;
}
+
void
AudioFilter::flush ()
{
_tail.reset ();
}
+
LowPassAudioFilter::LowPassAudioFilter (float transition_bandwidth, float cutoff)
: AudioFilter (transition_bandwidth)
{
_ir = sinc_blackman (cutoff, true);
}
+
BandPassAudioFilter::BandPassAudioFilter (float transition_bandwidth, float lower, float higher)
: AudioFilter (transition_bandwidth)
{
- float* lpf = sinc_blackman (lower, false);
- float* hpf = sinc_blackman (higher, true);
+ auto lpf = sinc_blackman (lower, false);
+ auto hpf = sinc_blackman (higher, true);
- delete[] _ir;
- _ir = new float[_M + 1];
+ _ir.resize(_M + 1);
for (int i = 0; i <= _M; ++i) {
_ir[i] = lpf[i] + hpf[i];
}
- delete[] lpf;
- delete[] hpf;
-
/* We now have a band-stop, so invert for band-pass */
for (int i = 0; i <= _M; ++i) {
_ir[i] = -_ir[i];