Add basic windowed-sinc audio filters.

3 files changed, 214 insertions, 1 deletions

diff --git a/src/lib/audio_filter.cc b/src/lib/audio_filter.cc
new file mode 100644
index 000000000..0cd2b18fb
--- /dev/null
+++ b/src/lib/audio_filter.cc
@@ -0,0 +1,133 @@
+/*
+    Copyright (C) 2014 Carl Hetherington <cth@carlh.net>
+
+    This program 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,
+    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.
+
+*/
+
+#include <cmath>
+#include "audio_filter.h"
+#include "audio_buffers.h"
+
+using std::vector;
+using std::min;
+using boost::shared_ptr;
+
+vector<float>
+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;
+		} else {
+			/* sinc */
+			ir[i] = sin (2 * M_PI * cutoff * (i - _M / 2)) / (i - _M / 2);
+			/* Blackman window */
+			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;
+}
+
+shared_ptr<AudioBuffers>
+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;
+			for (int k = 0; k <= _M; ++k) {
+				if ((j - k) < 0) {
+					s += _tail->data(i)[j - k + _M + 1] * _ir[k];
+				} else {
+					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;
+}
+
+LowPassAudioFilter::LowPassAudioFilter (float transition_bandwidth, float cutoff)
+	: AudioFilter (transition_bandwidth)
+{
+	_ir = sinc_blackman (cutoff, false);
+}
+
+
+HighPassAudioFilter::HighPassAudioFilter (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)
+{
+	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;
+}
diff --git a/src/lib/audio_filter.h b/src/lib/audio_filter.h
new file mode 100644
index 000000000..9dfcec58b
--- /dev/null
+++ b/src/lib/audio_filter.h
@@ -0,0 +1,80 @@
+/*
+    Copyright (C) 2014 Carl Hetherington <cth@carlh.net>
+
+    This program 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,
+    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.
+
+*/
+
+#include <vector>
+#include <boost/shared_ptr.hpp>
+
+class AudioBuffers;
+class audio_filter_impulse_kernel_test;
+class audio_filter_impulse_input_test;
+
+class AudioFilter
+{
+public:
+	AudioFilter (float transition_bandwidth)
+	{
+		_M = 4 / transition_bandwidth;
+		if (_M % 2) {
+			++_M;
+		}
+	}
+
+	boost::shared_ptr<AudioBuffers> run (boost::shared_ptr<AudioBuffers> in);
+
+protected:
+	friend class audio_filter_impulse_kernel_test;
+	friend class audio_filter_impulse_input_test;
+
+	std::vector<float> sinc_blackman (float cutoff, bool invert) const;
+
+	std::vector<float> _ir;
+	int _M;
+	boost::shared_ptr<AudioBuffers> _tail;
+};
+
+class LowPassAudioFilter : public AudioFilter
+{
+public:
+	/** Construct a windowed-sinc low-pass filter using the Blackman window.
+	 *  @param transition_bandwidth Transition bandwidth as a fraction of the sampling rate.
+	 *  @param cutoff Cutoff frequency as a fraction of the sampling rate.
+	 */
+	LowPassAudioFilter (float transition_bandwidth, float cutoff);
+};
+
+class HighPassAudioFilter : public AudioFilter
+{
+public:
+	/** Construct a windowed-sinc high-pass filter using the Blackman window.
+	 *  @param transition_bandwidth Transition bandwidth as a fraction of the sampling rate.
+	 *  @param cutoff Cutoff frequency as a fraction of the sampling rate.
+	 */
+	HighPassAudioFilter (float transition_bandwidth, float cutoff);
+};
+
+class BandPassAudioFilter : public AudioFilter
+{
+public:
+	/** Construct a windowed-sinc band-pass filter using the Blackman window.
+	 *  @param transition_bandwidth Transition bandwidth as a fraction of the sampling rate.
+	 *  @param lower Lower cutoff frequency as a fraction of the sampling rate.
+	 *  @param higher Higher cutoff frequency as a fraction of the sampling rate.
+	 */
+	BandPassAudioFilter (float transition_bandwidth, float lower, float higher);
+};
diff --git a/src/lib/wscript b/src/lib/wscript
index 9c929a671..ee8a59c0b 100644
--- a/src/lib/wscript
+++ b/src/lib/wscript
@@ -7,6 +7,7 @@ sources = """
           audio_buffers.cc
           audio_content.cc
           audio_decoder.cc
+          audio_filter.cc
           audio_mapping.cc
           audio_processor.cc
           cinema.cc
@@ -73,7 +74,6 @@ sources = """
           single_stream_audio_content.cc
           sndfile_content.cc
           sndfile_decoder.cc
-          sox_audio_processor.cc
           subrip.cc
           subrip_content.cc
           subrip_decoder.cc