X-Git-Url: https://main.carlh.net/gitweb/?p=dcpomatic.git;a=blobdiff_plain;f=src%2Flib%2Faudio_filter.cc;h=281ac966831c8e80de73609a96a12132a763d67a;hp=59b5684ea62e2cd7e447599b215c6d1b6a44fc0e;hb=2d4e8c5f69cc694625ad95dcee554499605f823b;hpb=fc56dc97103d00437843a31e1ef0c4915900ad1a

diff --git a/src/lib/audio_filter.cc b/src/lib/audio_filter.cc
index 59b5684ea..281ac9668 100644
--- a/src/lib/audio_filter.cc
+++ b/src/lib/audio_filter.cc
@@ -1,37 +1,43 @@
 /*
-    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 <cmath>
+
 #include "audio_filter.h"
 #include "audio_buffers.h"
+#include "util.h"
+#include <cmath>
 
-using std::vector;
+
+using std::make_shared;
 using std::min;
-using boost::shared_ptr;
+using std::shared_ptr;
+
 
-vector<float>
+/** @return array of floats which the caller must destroy with delete[] */
+float *
 AudioFilter::sinc_blackman (float cutoff, bool invert) const
 {
-	vector<float> ir (_M + 1);
-	
+	float* ir = new float[_M + 1];
+
 	/* Impulse response */
-	
+
 	for (int i = 0; i <= _M; ++i) {
 		if (i == (_M / 2)) {
 			ir[i] = 2 * M_PI * cutoff;
@@ -42,70 +48,85 @@ AudioFilter::sinc_blackman (float cutoff, bool invert) const
 			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::~AudioFilter ()
+{
+	delete[] _ir;
+}
+
+
 shared_ptr<AudioBuffers>
-AudioFilter::run (shared_ptr<AudioBuffers> in)
+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_silent ();
 	}
-	
-	for (int i = 0; i < in->channels(); ++i) {
-		for (int j = 0; j < in->frames(); ++j) {
+
+	int const channels = in->channels ();
+	int const frames = in->frames ();
+
+	for (int i = 0; i < channels; ++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) {
 				if ((j - k) < 0) {
-					s += _tail->data(i)[j - k + _M + 1] * _ir[k];
+					s += tail_p[j - k + _M + 1] * _ir[k];
 				} else {
-					s += in->data(i)[j - k] * _ir[k];
+					s += in_p[j - k] * _ir[k];
 				}
 			}
-			
-			out->data(i)[j] = s;
+
+			out_p[j] = s;
 		}
 	}
-	
+
 	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)
 {
@@ -119,21 +140,26 @@ HighPassAudioFilter::HighPassAudioFilter (float transition_bandwidth, float cuto
 	_ir = sinc_blackman (cutoff, true);
 }
 
+
 BandPassAudioFilter::BandPassAudioFilter (float transition_bandwidth, float lower, float higher)
 	: AudioFilter (transition_bandwidth)
 {
-	vector<float> lpf = sinc_blackman (lower, false);
-	vector<float> hpf = sinc_blackman (higher, true);
-	
-	_ir.resize (_M + 1);
+	auto lpf = sinc_blackman (lower, false);
+	auto hpf = sinc_blackman (higher, true);
+
+	delete[] _ir;
+	_ir = new float[_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];
 	}
-	
+
 	_ir[_M / 2] += 1;
 }