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-rw-r--r--libs/ardour/ardour/dsp_filter.h72
-rw-r--r--libs/ardour/dsp_filter.cc229
-rw-r--r--libs/ardour/wscript1
3 files changed, 302 insertions, 0 deletions
diff --git a/libs/ardour/ardour/dsp_filter.h b/libs/ardour/ardour/dsp_filter.h
new file mode 100644
index 0000000000..ae2928de93
--- /dev/null
+++ b/libs/ardour/ardour/dsp_filter.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2016 Robin Gareus <robin@gareus.org>
+ *
+ * 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ */
+#ifndef _dsp_filter_h_
+#define _dsp_filter_h_
+
+#include <stdint.h>
+#include "ardour/libardour_visibility.h"
+
+namespace ARDOUR { namespace DSP {
+
+ void memset (float *data, const float val, const uint32_t n_samples);
+ void mmult (float *data, float *mult, const uint32_t n_samples);
+
+
+ class LIBARDOUR_API LowPass {
+ public:
+ LowPass (double samplerate, float freq);
+ void proc (float *data, const uint32_t n_samples);
+ void ctrl (float *data, const float val, const uint32_t n_samples);
+ void set_cutoff (float freq);
+ void reset () { _z = 0.f; }
+ private:
+ float _rate;
+ float _z;
+ float _a;
+ };
+
+ class LIBARDOUR_API BiQuad {
+ public:
+ enum Type {
+ LowPass,
+ HighPass,
+ BandPassSkirt,
+ BandPass0dB,
+ Notch,
+ AllPass,
+ Peaking,
+ LowShelf,
+ HighShelf
+ };
+
+ BiQuad (double samplerate);
+ BiQuad (const BiQuad &other);
+
+ void run (float *data, const uint32_t n_samples);
+ void compute (Type, double freq, double Q, double gain);
+ void reset () { _z1 = _z2 = 0.0; }
+ private:
+ double _rate;
+ float _z1, _z2;
+ double _a1, _a2;
+ double _b0, _b1, _b2;
+ };
+
+} } /* namespace */
+#endif
diff --git a/libs/ardour/dsp_filter.cc b/libs/ardour/dsp_filter.cc
new file mode 100644
index 0000000000..3026666b79
--- /dev/null
+++ b/libs/ardour/dsp_filter.cc
@@ -0,0 +1,229 @@
+/*
+ * Copyright (C) 2016 Robin Gareus <robin@gareus.org>
+ *
+ * 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
+ *
+ */
+
+#include <stdlib.h>
+#include <math.h>
+#include "ardour/dsp_filter.h"
+
+#ifndef M_PI
+#define M_PI 3.14159265358979323846
+#endif
+
+using namespace ARDOUR::DSP;
+
+void
+ARDOUR::DSP::memset (float *data, const float val, const uint32_t n_samples) {
+ for (uint32_t i = 0; i < n_samples; ++i) {
+ data[i] = val;
+ }
+}
+
+void
+ARDOUR::DSP::mmult (float *data, float *mult, const uint32_t n_samples) {
+ for (uint32_t i = 0; i < n_samples; ++i) {
+ data[i] *= mult[i];
+ }
+}
+
+
+LowPass::LowPass (double samplerate, float freq)
+ : _rate (samplerate)
+ , _z (0)
+{
+ set_cutoff (freq);
+}
+
+void
+LowPass::set_cutoff (float freq)
+{
+ _a = 1.f - expf (-2.f * M_PI * freq / _rate);
+}
+
+void
+LowPass::proc (float *data, const uint32_t n_samples)
+{
+ // localize variables
+ const float a = _a;
+ float z = _z;
+ for (uint32_t i = 0; i < n_samples; ++i) {
+ data[i] += a * (data[i] - z);
+ z = data[i];
+ }
+ _z = z;
+}
+
+void
+LowPass::ctrl (float *data, const float val, const uint32_t n_samples)
+{
+ // localize variables
+ const float a = _a;
+ float z = _z;
+ for (uint32_t i = 0; i < n_samples; ++i) {
+ data[i] += a * (val - z);
+ z = data[i];
+ }
+ _z = z;
+}
+
+///////////////////////////////////////////////////////////////////////////////
+
+BiQuad::BiQuad (double samplerate)
+ : _rate (samplerate)
+ , _z1 (0.0)
+ , _z2 (0.0)
+ , _a1 (0.0)
+ , _a2 (0.0)
+ , _b0 (1.0)
+ , _b1 (0.0)
+ , _b2 (0.0)
+{
+}
+
+BiQuad::BiQuad (const BiQuad &other)
+ : _rate (other._rate)
+ , _z1 (0.0)
+ , _z2 (0.0)
+ , _a1 (other._a1)
+ , _a2 (other._a2)
+ , _b0 (other._b0)
+ , _b1 (other._b1)
+ , _b2 (other._b2)
+{
+}
+
+void
+BiQuad::run (float *data, const uint32_t n_samples)
+{
+ for (uint32_t i = 0; i < n_samples; ++i) {
+ const float xn = data[i];
+ const float z = _b0 * xn + _z1;
+ _z1 = _b1 * xn - _a1 * z + _z2;
+ _z2 = _b2 * xn - _a2 * z;
+ data[i] = z;
+ }
+}
+
+void
+BiQuad::compute (Type type, double freq, double Q, double gain)
+{
+ /* Compute biquad filter settings.
+ * Based on 'Cookbook formulae for audio EQ biquad filter coefficents'
+ * by Robert Bristow-Johnson
+ */
+ const double A = pow (10.0, (gain / 40.0));
+ const double W0 = (2.0 * M_PI * freq) / _rate;
+ const double sinW0 = sin (W0);
+ const double cosW0 = cos (W0);
+ const double alpha = sinW0 / (2.0 * Q);
+ const double beta = sqrt (A) / Q;
+
+ double _a0;
+
+ switch (type) {
+ case LowPass:
+ _b0 = (1.0 - cosW0) / 2.0;
+ _b1 = 1.0 - cosW0;
+ _b2 = (1.0 - cosW0) / 2.0;
+ _a0 = 1.0 + alpha;
+ _a1 = -2.0 * cosW0;
+ _a2 = 1.0 - alpha;
+ break;
+
+ case HighPass:
+ _b0 = (1.0 + cosW0) / 2.0;
+ _b1 = -(1.0 + cosW0);
+ _b2 = (1.0 + cosW0) / 2.0;
+ _a0 = 1.0 + alpha;
+ _a1 = -2.0 * cosW0;
+ _a2 = 1.0 - alpha;
+ break;
+
+ case BandPassSkirt: /* Constant skirt gain, peak gain = Q */
+ _b0 = sinW0 / 2.0;
+ _b1 = 0.0;
+ _b2 = -sinW0 / 2.0;
+ _a0 = 1.0 + alpha;
+ _a1 = -2.0 * cosW0;
+ _a2 = 1.0 - alpha;
+ break;
+
+ case BandPass0dB: /* Constant 0 dB peak gain */
+ _b0 = alpha;
+ _b1 = 0.0;
+ _b2 = -alpha;
+ _a0 = 1.0 + alpha;
+ _a1 = -2.0 * cosW0;
+ _a2 = 1.0 - alpha;
+ break;
+
+ case Notch:
+ _b0 = 1.0;
+ _b1 = -2.0 * cosW0;
+ _b2 = 1.0;
+ _a0 = 1.0 + alpha;
+ _a1 = -2.0 * cosW0;
+ _a2 = 1.0 - alpha;
+ break;
+
+ case AllPass:
+ _b0 = 1.0 - alpha;
+ _b1 = -2.0 * cosW0;
+ _b2 = 1.0 + alpha;
+ _a0 = 1.0 + alpha;
+ _a1 = -2.0 * cosW0;
+ _a2 = 1.0 - alpha;
+ break;
+
+ case Peaking:
+ _b0 = 1.0 + (alpha * A);
+ _b1 = -2.0 * cosW0;
+ _b2 = 1.0 - (alpha * A);
+ _a0 = 1.0 + (alpha / A);
+ _a1 = -2.0 * cosW0;
+ _a2 = 1.0 - (alpha / A);
+ break;
+
+ case LowShelf:
+ _b0 = A * ((A + 1) - ((A - 1) * cosW0) + (beta * sinW0));
+ _b1 = (2.0 * A) * ((A - 1) - ((A + 1) * cosW0));
+ _b2 = A * ((A + 1) - ((A - 1) * cosW0) - (beta * sinW0));
+ _a0 = (A + 1) + ((A - 1) * cosW0) + (beta * sinW0);
+ _a1 = -2.0 * ((A - 1) + ((A + 1) * cosW0));
+ _a2 = (A + 1) + ((A - 1) * cosW0) - (beta * sinW0);
+ break;
+
+ case HighShelf:
+ _b0 = A * ((A + 1) + ((A - 1) * cosW0) + (beta * sinW0));
+ _b1 = -(2.0 * A) * ((A - 1) + ((A + 1) * cosW0));
+ _b2 = A * ((A + 1) + ((A - 1) * cosW0) - (beta * sinW0));
+ _a0 = (A + 1) - ((A - 1) * cosW0) + (beta * sinW0);
+ _a1 = 2.0 * ((A - 1) - ((A + 1) * cosW0));
+ _a2 = (A + 1) - ((A - 1) * cosW0) - (beta * sinW0);
+ break;
+ default:
+ abort(); /*NOTREACHED*/
+ break;
+ }
+
+ _b0 /= _a0;
+ _b1 /= _a0;
+ _b2 /= _a0;
+ _a1 /= _a0;
+ _a2 /= _a0;
+}
diff --git a/libs/ardour/wscript b/libs/ardour/wscript
index 22aeb21dfa..844df2ed21 100644
--- a/libs/ardour/wscript
+++ b/libs/ardour/wscript
@@ -65,6 +65,7 @@ libardour_sources = [
'delivery.cc',
'directory_names.cc',
'diskstream.cc',
+ 'dsp_filter.cc',
'ebur128_analysis.cc',
'element_import_handler.cc',
'element_importer.cc',