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caps.diff
diff --git a/plugins/LadspaEffect/caps/Amp.cc b/plugins/LadspaEffect/caps/Amp.cc
index 10a5f1389..44037479a 100644
--- a/plugins/LadspaEffect/caps/Amp.cc
+++ b/plugins/LadspaEffect/caps/Amp.cc
@@ -1,19 +1,17 @@
/*
Amp.cc
- Copyright 2003-7
- Tim Goetze <tim@quitte.de>
- David Yeh <dtyeh@ccrma.stanford.edu> (Tone Stack in TS models)
+ Copyright 2003-14 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- Tube amplifier models
+ Idealised guitar amplification.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -28,610 +26,195 @@
*/
#include "basics.h"
-#include <stdio.h>
#include "Amp.h"
#include "Descriptor.h"
+#include "dsp/polynomials.h"
void
-AmpStub::init (bool adjust_downsampler)
-{
- dc_blocker.set_f (10. / fs);
-
- /* going a bit lower than nominal with fc */
- double f = .7 * M_PI / OVERSAMPLE;
-
- /* construct the upsampler filter kernel */
- DSP::sinc (f, up.c, FIR_SIZE);
- DSP::kaiser<DSP::apply_window> (up.c, FIR_SIZE, 6.4);
-
- /* copy upsampler filter kernel for downsampler, make sum */
- double s = 0;
- for (int i = 0; i < up.n; ++i)
- down.c[i] = up.c[i],
- s += up.c[i];
-
- s = 1 / s;
-
- /* scale downsampler kernel for unity gain + correction for transfer */
- double t = adjust_downsampler ?
- s / max (fabs (tube.clip[0].value), fabs (tube.clip[1].value)) : s;
-
- for (int i = 0; i < down.n; ++i)
- down.c[i] *= t;
-
- /* scale upsampler kernel for unity gain */
- s *= OVERSAMPLE;
- for (int i = 0; i < up.n; ++i)
- up.c[i] *= s;
-}
-
-/* //////////////////////////////////////////////////////////////////////// */
-
-void
-AmpIII::init()
+AmpVTS::init()
{
- this->AmpStub::init (false);
-
- /* need to filter out dc before the power amp stage, which is running at
- * the oversampled rate */
- dc_blocker.set_f (10. / (fs * OVERSAMPLE));
+ tonestack.init (fs);
- DSP::RBJ::LoShelve (200 / fs, .2, -3, filter.a, filter.b);
+ dc2.set_f (25*over_fs);
+ DSP::RBJ::LP (1*over_fs, .7, biaslp);
+ /* compress is initialised in activate() */
}
-template <sample_func_t F, int OVERSAMPLE>
void
-AmpIII::one_cycle (int frames)
+AmpVTS::activate()
{
- sample_t * s = ports[0];
-
- sample_t gain = getport(1);
- sample_t temp = getport(2) * tube.scale;
-
- drive = getport(3) * .5;
- i_drive = 1 / (1 - drive);
-
- sample_t * d = ports[4];
-
- *ports[5] = OVERSAMPLE;
+ hp1.reset();
+ lp.reset();
- double g = current.g;
+ remain = 0;
+ compress.init (fs);
+ compress.set_threshold(0);
+ compress.set_release(.0);
- current.g = max (gain < 1 ? gain : exp2 (gain - 1), .000001);
- current.g *= tube.scale / fabs (tube.transfer (temp));
+ /* dc1 is reset in setratio() */
+ dc2.reset();
- /* recursive fade to prevent zipper noise from the 'gain' knob */
- if (g == 0) g = current.g;
-
- double one_over_n = frames > 0 ? 1. / frames : 1;
- double gf = pow (current.g / g, one_over_n);
-
- for (int i = 0; i < frames; ++i)
- {
- register sample_t a = s[i];
-
- a = g * tube.transfer (a * temp);
- a = filter.process (a + normal);
-
- a = tube.transfer_clip (up.upsample (a));
- a = power_transfer (dc_blocker.process (a));
-
- a = down.process (a);
-
- for (int o = 1; o < OVERSAMPLE; ++o)
- down.store (
- power_transfer (
- dc_blocker.process (
- normal + tube.transfer_clip (up.pad (o)))));
-
- F (d, i, a, adding_gain);
-
- g *= gf;
- }
-
- current.g = g;
+ model = -1;
+ ratio = -1;
}
-/* //////////////////////////////////////////////////////////////////////// */
-
-PortInfo
-AmpIII::port_info [] =
-{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "gain",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_1, 0, 10}
- }, {
- "temperature",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0.005, 1}
- }, {
- "drive",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MAX, 0.0001, 1} /* ^2 gives the nice drive */
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }, {
- "latency",
- OUTPUT | CONTROL,
- {0}
- }
-};
-
-template <> void
-Descriptor<AmpIII>::setup()
+/* set oversampling ratio */
+void
+AmpVTS::setratio (int r)
{
- UniqueID = 1786;
- Label = "AmpIII";
- Properties = HARD_RT;
+ if (r == ratio)
+ return;
- Name = CAPS "AmpIII - Tube amp";
- Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2002-7";
+ ratio = r;
+ dc1.set_f (72./(ratio*fs));
+ dc1.reset();
- /* fill port info and vtable */
- autogen();
+ over2.reset();
+ over4.reset();
+ over8.reset();
}
-/* //////////////////////////////////////////////////////////////////////// */
-
void
-AmpIV::init()
+AmpVTS::cycle (uint frames)
{
- this->AmpStub::init (false);
-
- /* need to filter out dc before the power amp stage, which is running at
- * the oversampled rate */
- dc_blocker.set_f (10. / (fs * OVERSAMPLE));
-
- tone.init (fs);
+ int r = 2 << (int) getport(0);
+ setratio(r);
+ if (r == 8) subcycle (frames, over8);
+ else if (r == 4) subcycle (frames, over4);
+ else subcycle (frames, over2);
}
-template <sample_func_t F, int OVERSAMPLE>
-void
-AmpIV::one_cycle (int frames)
-{
- double one_over_n = frames > 0 ? 1. / frames : 1;
-
- sample_t * s = ports[0];
-
- sample_t gain = getport(1);
- sample_t temp = getport(2) * tube.scale;
-
- tone.start_cycle (ports + 3, one_over_n);
-
- drive = getport(7) * .5;
- i_drive = 1 / (1 - drive);
-
- sample_t * d = ports[8];
-
- *ports[9] = OVERSAMPLE;
-
- double g = current.g;
-
- current.g = max (gain < 1 ? gain : exp2 (gain - 1), .000001);
- current.g *= tube.scale / fabs (tube.transfer (temp));
-
- /* recursive fade to prevent zipper noise from the 'gain' knob */
- if (g == 0) g = current.g;
-
- double gf = pow (current.g / g, one_over_n);
-
- for (int i = 0; i < frames; ++i)
- {
- register sample_t a = s[i] + normal;
-
- a = g * tube.transfer (a * temp);
- a = tone.process (a);
-
- a = tube.transfer_clip (up.upsample (a));
- a = power_transfer (dc_blocker.process (a));
-
- a = down.process (a);
-
- for (int o = 1; o < OVERSAMPLE; ++o)
- down.store (
- power_transfer (
- dc_blocker.process (
- normal + tube.transfer_clip (up.pad (o)))));
-
- F (d, i, a, adding_gain);
-
- g *= gf;
- }
-
- current.g = g;
-}
-
-/* //////////////////////////////////////////////////////////////////////// */
-
-PortInfo
-AmpIV::port_info [] =
-{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "gain",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_1, 0, 10}
- }, {
- "temperature",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0.005, 1}
- }, {
- "bass",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -20, 20}
- }, {
- "mid",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -20, 20}
- }, {
- "treble",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -20, 20}
- }, {
- "hi",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -20, 20}
- }, {
- "drive",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MAX, 0.0001, 1} /* ^2 gives the nice drive */
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }, {
- "latency",
- OUTPUT | CONTROL,
- {0}
- }
-};
-
-template <> void
-Descriptor<AmpIV>::setup()
-{
- UniqueID = 1794;
- Label = "AmpIV";
- Properties = HARD_RT;
-
- Name = CAPS "AmpIV - Tube amp + tone controls";
- Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2002-7";
-
- /* fill port info and vtable */
- autogen();
-}
+/* saturating function selection */
+#define preamp DSP::Polynomial::one5
+#define poweramp DSP::Polynomial::atan
-/* //////////////////////////////////////////////////////////////////////// */
+/* rough correction for tonestack model gain differences */
+static float tsgain[] = {.639, 1.290, .534, 1.008, .542, .936, .605, 1.146, .211};
+template <class Over>
void
-AmpV::init()
+AmpVTS::subcycle (uint frames, Over & over)
{
- this->AmpStub::init (false);
-
- /* need to filter out dc before the power amp stage, which is running at
- * the oversampled rate */
- dc_blocker.set_f (10. / (fs * OVERSAMPLE));
-
- DSP::RBJ::LoShelve (210. / fs, .2, -1, filter[0].a, filter[0].b);
- DSP::RBJ::LoShelve (4200. / fs, 1.2, +6, filter[1].a, filter[1].b);
- DSP::RBJ::LoShelve (420. / fs, .2, +2, filter[2].a, filter[2].b);
+ int m = getport(4);
+ if (m != model) tonestack.setmodel (model = m);
+ tonestack.updatecoefs (getport(5),getport(6),getport(7));
- /* power supply cap */
- for (int i = 0; i < 2; ++i)
- DSP::RBJ::LP (10. / fs, .3, power_cap[i].a, power_cap[i].b);
-}
+ float x=getport(1), y=getport(3); /* = gain,powa :: shorthand for gain calc */
+ float bright = getport(2)*(1-x*.5);
+ DSP::RBJ::LP ((500+6500*bright*bright)/(over.Ratio*fs), .7, lp);
-template <sample_func_t F, int OVERSAMPLE>
-void
-AmpV::one_cycle (int frames)
-{
- sample_t * s = ports[0];
-
- sample_t gain = getport(1);
+ float gain = x*x;
+ float powa = .2*gain + (1-.2*gain)*y; /* ramp up powa with gain */
- if (*ports[2] != cut)
- {
- cut = getport(2);
- DSP::RBJ::LoShelve (210. / fs, .2, cut, filter[0].a, filter[0].b);
- }
- if (*ports[3] != tone)
- {
- tone = getport(3);
- double f = tone * tone * 8400 + 420;
- double q = tone * .4 + .2;
- double db = tone * 2 + 2;
- DSP::RBJ::LoShelve (f / fs, q, db, filter[2].a, filter[2].b);
- }
+ compress.set_attack (.6*(1-.5*x)*getport(8));
+ float squash = .0 + .8*getport(9)*(1 - .2*powa*gain);
- drive = getport(4) * .5;
- i_drive = 1 / (1 - drive);
-
- #define MAX_WATTS port_info[5].range.UpperBound
- sample_t sag = (MAX_WATTS - getport(5)) / MAX_WATTS;
- sag = .6 * sag * sag;
-
- sample_t * d = ports[6];
-
- *ports[7] = OVERSAMPLE;
+ float bias = .62*powa;
- double g = current.g;
+ /* roughly correcting for loudness increase TODO: revise */
+ float makeup = (.086-.06*y)/(11.6+exp((12.1-5*y)*(.81-.08*y-x)))+0.00032+.0026*y;
+ makeup = 0 ? 1 : .0006/makeup; /* debug switch for makeup */
- current.g = max (gain < 1 ? gain : pow (20, gain - 1), .000001);
- #if 0
- if ( (++_turn & 127) == 0)
- fprintf (stderr, "supply = %.3f sag = %.3f\n", supply, sag);
- #endif
+ float lowcut = .1 + 392*getport(10);
+ hp1.set_f (1.5*lowcut*over_fs);
- if (g == 0) g = current.g;
+ gain = pow (200, x) * -tsgain[model];
+ powa = pow (125, powa);
- /* recursive fade to prevent zipper noise from the 'gain' knob */
- double one_over_n = frames > 0 ? 1. / frames : 1;
- double gf = pow (current.g / g, one_over_n);
+ sample_t * s = ports[11];
+ sample_t * d = ports[12];
- for (int i = 0; i < frames; ++i)
+ while (frames)
{
- register sample_t a = s[i];
- register sample_t v = 3 - supply;
- /* alternative curve: v = v * v * .1 + .1; */
- v = v * v * .06 + .46;
-
- a = filter[0].process (a + normal);
- if (0)
- a = filter[2].process (a);
-
- a = g * (a + supply * .001);
-
- a = v * tube.transfer_clip (up.upsample (a));
- a = power_transfer (dc_blocker.process (a));
-
- a = down.process (a);
-
- a = filter[1].process (a - normal);
- if (1)
- a = filter[2].process (a + normal);
-
+ if (remain == 0)
{
- for (int o = 1; o < OVERSAMPLE; ++o)
- down.store (
- power_transfer (
- dc_blocker.process (
- normal + tube.transfer_clip (
- up.pad (o)))));
+ remain = compress.blocksize;
+ compress.start_block (squash);
}
- F (d, i, a, adding_gain);
-
- /* integrate for an approximation of cumulative output power */
- supply += sag * fabs (a) + normal;
- /* filter integrated power consumption */
- for (int j = 0; j < 2; ++j)
- supply = 0.9 * (power_cap[j].process (supply));
+ uint n = min (frames, remain);
+ for (uint i = 0; i < n; ++i)
+ {
+ sample_t a = s[i];
+ sample_t b = biaslp.process (bias*compress.power.current - .00002);
+
+ a = hp1.process(a);
+ a *= gain*compress.get();
+ a = tonestack.process(a + normal);
+ a += .5*b;
+
+ a = over.upsample(a);
+ a = preamp(a);
+ a = dc1.process(a);
+ a = lp.process(a);
+ a = poweramp(powa*a - b);
+ a = over.downsample(a);
+
+ for (int o=1; o < over.Ratio; ++o)
+ {
+ sample_t a = over.uppad(o);
+ a = preamp(a);
+ a = dc1.process(a);
+ a = lp.process(a);
+ a = poweramp(powa*a - b);
+ over.downstore(a);
+ }
+
+ a = dc2.process(a+normal);
+ compress.store(a);
+
+ a *= makeup;
+
+ d[i] = a;
+ }
- g *= gf;
- normal = -normal;
+ s+=n;d+=n;
+ frames-=n;
+ remain-=n;
}
-
- current.g = g;
}
/* //////////////////////////////////////////////////////////////////////// */
PortInfo
-AmpV::port_info [] =
-{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "gain",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_1, 0, 3}
- }, {
- "bass",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -9, 9}
- }, {
- "tone",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MIN, 0, 1}
- }, {
- "drive",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_HIGH, 0.0001, 1} /* ^2 gives the nice drive */
- }, {
- "watts",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 5, 150}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }, {
- "latency",
- OUTPUT | CONTROL,
- {0}
- }
-};
-
-template <> void
-Descriptor<AmpV>::setup()
-{
- UniqueID = 2587;
- Label = "AmpV";
- Properties = HARD_RT;
-
- Name = CAPS "AmpV - Tube amp";
- Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2002-7";
-
- /* fill port info and vtable */
- autogen();
-}
-
-/* //////////////////////////////////////////////////////////////////////// */
-
-void
-AmpVTS::init()
-{
- this->AmpStub::init (false);
-
- /* need to filter out dc before the power amp stage, which is running at
- * the oversampled rate */
- dc_blocker.set_f (10. / (fs * OVERSAMPLE));
-
- /* power supply capacitance */
- for (int i = 0; i < 2; ++i)
- DSP::RBJ::LP (10. / fs, .3, power_cap[i].a, power_cap[i].b);
-
- tonestack.init (fs);
-}
-
-template <sample_func_t F, int OVERSAMPLE>
-void
-AmpVTS::one_cycle (int frames)
+AmpVTS::port_info [] =
{
- sample_t * s = ports[0];
+ { "over", CTRL_IN, {DEFAULT_1 | INTEGER, 0, 2}, "{0:'2x',1:'4x',2:'8x'}"},
- tonestack.start_cycle (ports + 1, 2);
- sample_t gain = getport(2);
-
- drive = getport(6) * .5;
- i_drive = 1 / (1 - drive);
-
- sample_t sag = 1 - max (0.0001, min (1, getport(7)));
- sag = .6 * sag * sag; /* map to log space makes slider better */
-
- sample_t * d = ports[8];
+ /* 1 */
+ { "gain", CTRL_IN | GROUP, {DEFAULT_LOW, 0, 1} },
+ { "bright", CTRL_IN, {DEFAULT_HIGH, 0, 1} },
+ { "power", CTRL_IN, {DEFAULT_MID, 0, 1} },
- *ports[9] = OVERSAMPLE;
+ /* 4 */
+ { "tonestack", CTRL_IN | GROUP, {DEFAULT_1 | INTEGER, 0, 8}, DSP::ToneStack::presetdict},
- double g = current.g;
+ { "bass", CTRL_IN | GROUP, {DEFAULT_LOW, 0, 1} },
+ { "mid", CTRL_IN, {DEFAULT_1, 0, 1} },
+ { "treble", CTRL_IN, {DEFAULT_HIGH, 0, 1} },
- if (gain < 1)
- current.g = max (gain, .001);
- else
- {
- gain -= 1;
- gain *= gain;
- current.g = pow (10, gain);
- }
+ /* 8 */
+ { "attack", CTRL_IN | GROUP, {DEFAULT_HIGH, 0, 1} },
+ { "squash", CTRL_IN, {DEFAULT_LOW, 0, 1} },
- /* recursive fade to prevent zipper noise from the 'gain' knob */
- double one_over_n = frames > 0 ? 1. / frames : 1;
- double gf = pow (current.g / g, one_over_n);
-
- for (int i = 0; i < frames; ++i)
- {
- register double a = s[i];
- register double v = 3 - supply;
- v = v * v * .06 + .46;
+ /* 10 */
+ { "lowcut", CTRL_IN | GROUP, {DEFAULT_MID, 0, 1} },
- a = tube.transfer (a);
- a = tonestack.process (a + normal);
-
- a = g * (a + supply * .001);
-
- a = v * tube.transfer_clip (up.upsample (a));
- a = power_transfer (dc_blocker.process (a));
-
- a = down.process (a);
-
- {
- for (int o = 1; o < OVERSAMPLE; ++o)
- down.store (
- power_transfer (
- dc_blocker.process (
- normal + tube.transfer_clip (
- up.pad (o)))));
- }
-
- F (d, i, a, adding_gain);
-
- /* integrate for an approximation of cumulative output power */
- supply += sag * fabs (a) + normal;
- /* filter integrated power consumption */
- for (int j = 0; j < 2; ++j)
- supply = 0.9 * (power_cap[j].process (supply + normal));
-
- g *= gf;
- normal = -normal;
- }
-
- current.g = g;
-}
-
-/* //////////////////////////////////////////////////////////////////////// */
-
-PortInfo
-AmpVTS::port_info [] =
-{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "model",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0 | INTEGER, 0, 5} /* no way to set dyn at compile t */
- }, {
- "gain",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_HIGH, 0, 3}
- }, {
- "bass",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, 1}
- }, {
- "mid",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_1, 0, 1}
- }, {
- "treble",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_HIGH, 0, 1}
- }, {
- "drive",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0.0001, 1}
- }, {
- "watts",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_HIGH, 0.0001, 1}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }, {
- "latency",
- OUTPUT | CONTROL,
- {0}
- }
+ { "in", INPUT | AUDIO },
+ { "out", OUTPUT | AUDIO },
};
template <> void
Descriptor<AmpVTS>::setup()
{
- UniqueID = 2592;
Label = "AmpVTS";
- Properties = HARD_RT;
- Name = CAPS "AmpVTS - Tube amp + Tone stack";
- Maker = "David Yeh <dtyeh@ccrma.stanford.edu> & Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2002-7";
+ Name = CAPS "AmpVTS - Idealised guitar amplification";
+ Maker = "Tim Goetze <tim@quitte.de>, David Yeh <dtyeh@ccrma.stanford.edu>";
+ Copyright = "2002-14";
/* fill port info and vtable */
autogen();
}
+
diff --git a/plugins/LadspaEffect/caps/Amp.h b/plugins/LadspaEffect/caps/Amp.h
index e06f7128f..5d2a9f3d2 100644
--- a/plugins/LadspaEffect/caps/Amp.h
+++ b/plugins/LadspaEffect/caps/Amp.h
@@ -1,17 +1,17 @@
/*
Amp.h
- Copyright 2002-9 Tim Goetze <tim@quitte.de>
+ Copyright 2002-14 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- Oversampled tube amplifier emulation.
+ Guitar amplification.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,353 +25,50 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _AMP_H_
-#define _AMP_H_
+#ifndef AMP_H
+#define AMP_H
#include "dsp/util.h"
-#include "dsp/OnePole.h"
-#include "dsp/BiQuad.h"
-#include "dsp/TwelveAX7.h"
-#include "dsp/Roessler.h"
+#include "dsp/IIR1.h"
+#include "dsp/IIR2.h"
-#include "dsp/FIR.h"
-#include "dsp/sinc.h"
-#include "dsp/windows.h"
+#include "dsp/Oversampler.h"
#include "dsp/RBJ.h"
-#include "dsp/Eq.h"
#include "dsp/ToneStack.h"
+#include "dsp/Compress.h"
-class AmpStub
-: public Plugin
-{
- public:
- DSP::TwelveAX7_3 tube;
-
- sample_t drive, i_drive;
-
- struct {
- /* gain (remember current setting and fade to port setting in run) */
- double g;
- /* should also do this for temperature to remove another potential
- * source of zippering, but that would be overkill, at the cost of
- * at least one pow() per block. */
- } current;
-
- /* input is hipass-filtered first */
- DSP::OnePoleHP dc_blocker;
-
- enum {
- OVERSAMPLE = 8,
- FIR_SIZE = 64,
- };
-
- /* antialias filters */
- DSP::FIRUpsampler up;
- DSP::FIR down;
-
- AmpStub()
- : up (FIR_SIZE, OVERSAMPLE),
- down (FIR_SIZE, up.c)
- { }
-
- void init (bool adjust_downsampler = false);
-
- inline sample_t power_transfer (sample_t a)
- {
- return i_drive * (a - drive * fabs (a) * a);
- }
-};
-
-/* /////////////////////////////////////////////////////////////////////// */
-
-class PreampIII
-: public AmpStub
-{
- public:
- template <sample_func_t F, int OVERSAMPLE>
- void one_cycle (int frames);
-
- DSP::BiQuad filter;
-
- public:
- static PortInfo port_info[];
-
- sample_t adding_gain;
-
- void init();
- void activate()
- {
- current.g = 1;
-
- filter.reset();
- up.reset();
- down.reset();
- dc_blocker.reset();
- }
-
- void run (int n)
- {
- one_cycle<store_func, OVERSAMPLE> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func, OVERSAMPLE> (n);
- }
-};
-
-/* /////////////////////////////////////////////////////////////////////// */
-
-class AmpIII
-: public AmpStub
-{
- public:
- template <sample_func_t F, int OVERSAMPLE>
- void one_cycle (int frames);
-
- DSP::BiQuad filter;
-
- public:
- static PortInfo port_info[];
-
- sample_t adding_gain;
-
- void init();
- void activate()
- {
- current.g = 1;
-
- up.reset();
- down.reset();
- dc_blocker.reset();
- filter.reset();
- }
-
- void run (int n)
- {
- one_cycle<store_func, OVERSAMPLE> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func, OVERSAMPLE> (n);
- }
-};
-
-/* /////////////////////////////////////////////////////////////////////// */
+/* TODO: mega wookie bands 80 240 750 2200 6600 */
-typedef struct
- {float center, Q, adjust;}
-PreampBand;
-
-class ToneControls
+class AmpVTS
+: public Plugin
{
public:
- sample_t eq_gain[4];
- DSP::Eq<4> eq;
- static PreampBand bands[4];
-
- public:
- void init (double _fs);
- void activate (sample_t **);
-
- inline void
- start_cycle (sample_t ** ports, double one_over_n)
- {
- for (int i = 0; i < 4; ++i)
- {
- if (*ports[i] == eq_gain[i])
- {
- eq.gf[i] = 1;
- continue;
- }
-
- eq_gain[i] = *ports [i];
-
- double want = get_band_gain (i, eq_gain[i]);
- eq.gf[i] = pow (want / eq.gain[i], one_over_n);
- }
- }
+ int ratio;
+ DSP::Oversampler<2,32> over2;
+ DSP::Oversampler<4,32> over4;
+ DSP::Oversampler<8,64> over8;
- double get_band_gain (int i, double g);
- void set_band_gain (int i, float g);
+ DSP::IIR2<sample_t> lp, biaslp;
+ DSP::HP1<sample_t> hp1, dc1, dc2; /* dc blockers */
- inline sample_t process (sample_t x)
- {
- return eq.process (x);
- }
-};
+ int model;
+ DSP::ToneStack tonestack;
-/* /////////////////////////////////////////////////////////////////////// */
+ uint remain;
+ DSP::CompressRMS compress;
-class PreampIV
-: public PreampIII
-{
- public:
- ToneControls tone;
+ void setratio (int r);
- template <sample_func_t F, int OVERSAMPLE>
- void one_cycle (int frames);
+ void cycle (uint frames);
+ template <class Over> void subcycle (uint frames, Over & over);
public:
static PortInfo port_info[];
- sample_t adding_gain;
-
void init();
void activate();
-
- void run (int n)
- {
- one_cycle<store_func, OVERSAMPLE> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func, OVERSAMPLE> (n);
- }
-};
-
-/* /////////////////////////////////////////////////////////////////////// */
-
-class AmpIV
-: public AmpStub
-{
- public:
- ToneControls tone;
-
- template <sample_func_t F, int OVERSAMPLE>
- void one_cycle (int frames);
-
- public:
- static PortInfo port_info[];
-
- sample_t adding_gain;
-
- void init();
- void activate()
- {
- current.g = 1;
-
- tone.activate (ports + 3);
-
- up.reset();
- down.reset();
- dc_blocker.reset();
- }
-
- void run (int n)
- {
- one_cycle<store_func, OVERSAMPLE> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func, OVERSAMPLE> (n);
- }
-};
-
-/* /////////////////////////////////////////////////////////////////////// */
-
-class AmpV
-: public AmpStub
-{
- public:
- template <sample_func_t F, int OVERSAMPLE>
- void one_cycle (int frames);
-
- DSP::BiQuad filter[3];
-
- sample_t cut, tone;
-
- /* supply voltage sag */
- sample_t supply;
- DSP::BiQuad power_cap[2];
-
- public:
- static PortInfo port_info[];
-
- sample_t adding_gain;
-
- void init();
- void activate()
- {
- current.g = 1;
-
- for (int i = 0; i < 2; ++i)
- filter[i].reset(),
- power_cap[i].reset();
-
- up.reset();
- down.reset();
- dc_blocker.reset();
-
- cut = 2;
- supply = 0.;
-
- tone = -1; /* causes initialisation of the filter at first cycle */
- }
-
- void run (int n)
- {
- one_cycle<store_func, OVERSAMPLE> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func, OVERSAMPLE> (n);
- }
-};
-
-/* /////////////////////////////////////////////////////////////////////// */
-
-class AmpVTS
-: public AmpStub
-{
- public:
- DSP::ToneStack tonestack;
-
- template <sample_func_t F, int OVERSAMPLE>
- void one_cycle (int frames);
-
- sample_t cut, tone;
-
- /* supply voltage sag */
- sample_t supply;
- DSP::BiQuad power_cap[2];
-
- public:
- static PortInfo port_info[];
-
- sample_t adding_gain;
-
- void init();
- void activate()
- {
- current.g = 1;
-
- for (int i = 0; i < 2; ++i)
- power_cap[i].reset();
-
- up.reset();
- down.reset();
- dc_blocker.reset();
-
- cut = 2;
- supply = 0.;
- }
-
- void run (int n)
- {
- one_cycle<store_func, OVERSAMPLE> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func, OVERSAMPLE> (n);
- }
};
-#endif /* _AMP_H_ */
+#endif /* AMP_H */
diff --git a/plugins/LadspaEffect/caps/CHANGES b/plugins/LadspaEffect/caps/CHANGES
index 19c915d70..6948990ec 100644
--- a/plugins/LadspaEffect/caps/CHANGES
+++ b/plugins/LadspaEffect/caps/CHANGES
@@ -1,10 +1,299 @@
+0.9.24
+ * documentation updates including switch to 48k for all spectra
+ * "twin" and "stanford" tonestack models swap places
+ * ChorusI feedback changed to interpolation, softens zipper noise on t changes
+ * use vestigial FPTruncateMode only on pre-SSE2 hardware
+ * 4x128 oversampling option removed from Compress plugins
+ * state port displaying current gain reduction on Compress plugins
+ * _latency port for parallel Eq
+ * Saturate fade-in bug eliminated
+ * Noisegate hysteresis interval lengthened
+ * Click sounds "stick" brightened and "beep" shortened
+ * run_adding() removed, reducing binary size by 30 % and compile time similarly
+ * JVRev tail modified, low-passed
+ * all ports reordered to consistent ctrl i/o audio i/o order
+ * Eq4p crossfade reimplemented as equal-power squared cosine, smoother now
+ * EqFA4p added
+ * non-SSE v4f_shuffle surrogate fixed
+ (fixes Eq4p on ARM, reported by Jens Dreske)
+ * dropped 44.1 kHz ToneStackLT
+ * OnePole header renamed IIR1, filters renamed LP1 and HP1
+ * BiQuad header and filter renamed IIR2, SIMD *4f filters renamed *v4
+ * CabinetIII (less demanding loudspeaker emulation resurrected again)
+ * ChorusI port ranges and defaults adjusted
+ * Compress parameter mappings and defaults adjusted
+
+0.9.23
+ * ChorusI hp filter denormal protection added
+
+0.9.22
+ * ChorusI blend behaviour changed
+ * ChorusI parameters adjusted
+ * ChorusI hp filter before delay added
+ * AutoFilter parameters adjusted
+ * AutoFilter 'range' port becomes 'depth'
+
+0.9.21
+ * Eq10 Q increased
+ * AmpVTS bright/compression/gain mappings adjusted
+ * AutoFilter reverted to 0.9.1 version with adjusted parameters
+
+0.9.20
+ * ToneStack ports rearranged (following "control before audio" convention)
+ * ToneStack LT model selection fixed
+
+0.9.19
+ * Noisegate partially rewritten, defaults adjusted
+ * Noisegate bug causing spurious gain fluctuations in closed state eliminated
+ * AmpVTS compression controls adjusted
+ * influence of total gain on compression in AmpVTS reduced
+ * documentation polish
+
+0.9.18
+ * restored ToneStack presets to present the full list
+ * AmpVTS 'bright' filter given slight resonance
+ * AmpVTS compression control range widened
+
+0.9.17
+ * multiple inclusion header guards changed to comply with ISO/IEC 9899:2011
+ (at the insistence of Markus Elfring)
+ * PlateX2 fixed to read correct port for damping parameter
+ (reported by James Morris)
+ * uninstall removes rdf
+ * DSP::FIR4f::set_kernel size overflow in default parameter fixed
+ (reported and isolated by Ricardo Crudo)
+ * AmpVTS "lowcut" filter changed from 2- to 1-pole
+ * "HK 20" tonestack renamed to "AK 20"
+ * AmpVTS lowcut mapping changed, default lowered
+ * AmpVTS highpass 1 cutoff raised
+ * lowpass filter added to Noisegate's opening peak detector
+ * Noisegate not NoiseGate
+ * Click normal addition limited to idle loop
+ * NoiseGate logic loop bug fixed, hysteresis added
+ * documentation polish
+
+0.9.16
+ * NoiseGate simplified, sped up, gain lowpassed, click bug eliminated
+ * Sin smoothened, defaults updated
+ * PhaserII defaults updated
+ * Fractal defaults updated
+ * documentation polish
+
+0.9.15
+ * no-vector-arithmetic compilation fixed
+
+0.9.14
+ * AutoFilter x/y parameter eliminated
+ * Fractal hp changed to be configurable, attractors recalibrated
+ * DSP::Lorenz instantly initialises to useful state
+ * documentation polish
+
+0.9.13
+ * Eq4p softens drastic parameter changes by crossfading static filters
+ * Eq4p employing single v4f filter, slightly quicker
+ * Eq4p 'off' filter mode
+ * documentation and interface polish
+
+0.9.12
+ * non-SSE compilation fixed
+
+0.9.11
+ * DSP::Roessler instantly initialises to useful state
+ * Logarithmic port bounds and values fixed to reflect actual intent,
+ affecting Spice, AutoFilter and ChorusI
+ * clicking automatic gain fade on activate() eliminated in Saturate
+ * Dirac merged into Click
+ * Lorenz and Roessler merged into Fractal
+ * JVRev removed
+ * running plugins for zero frames returns without touching state
+ * CabinetII and III removed
+ * PhaserII sine LFO and LFO choice port added
+ * PhaserII fractal modulation extensively revised
+ * ChorusI rate lower bound slightly increased
+ * ChorusI delay line length adjusted to accommodate full t+width range
+ * all Chorus and Phaser plugins removed save for ChorusI and PhaserII
+ * major documentation revision
+ * Eq4p parametric equaliser plugin
+ * RBJ shelving filter prototype implementation fixed
+ * "2x2" stereo plugin label postfix changed to "X2"
+ * Eq renamed Eq10
+ * SIMD implementation updates
+ * DSP::RBJ coefficient calculation rearranged
+ * Click bpm range changed, default lowered substantially
+
+0.9.10
+ * DSP::RMS protected against negative sum of squares
+ * documentation updates
+
+0.9.9
+ * OSX compilation configuration refinements
+
+0.9.8
+ * Spice enabled by default, Spice2x2 added
+ * AmpVTS default oversampling ratio changed to 4x
+ * JVRev refinements
+ * Cabinet gain upper limit reinstated to 24 dB
+ * AutoFilter modulation slightly adjusted
+ * more *ChorusII modulation smoothness
+ * 1st-order IIR lp set_f fix part 2
+ * 1st-order IIR lp set_f fix part 1
+ * 1st order low-pass for AutoFilter modulation LFO
+ * DelayTapA removed
+ * README updated
+ * Click 'stick' sound brightened
+ * moved processor_specific_denormal_measures() to activate(), speedup
+
+0.9.7
+ * ChorusI is back
+ * AutoFilter modulation range and default parameter updates
+
+0.9.6
+ * Click fixed to compile by older g++ versions
+
+0.9.5
+ * AmpVTS model port moved, renamed to the more apt 'tonestack'
+ * documentation refinements
+ * appreciable AmpVTS speedup by going back to sample_t for accumulator
+ * NoiseRatio attack time control
+ * sample rate dependencies in NoiseGate removed, time constants revised
+ * Tonestacks reordered
+ * Plugin::over_fs
+ * AmpVTS model setting fixed to follow the correct input
+ * CabinetIII resurrected
+ * AmpVTS 'ratio' port renamed to 'over'
+ * more filters for AutoFilter, ladder stage gain control
+ * AutoFilter oversampling
+ * AmpVTS bias in poweramp, dcblock before output
+ * AutoFilter envelope calculation fixed for fs > 153.6 kHz
+ * peak/rms selection for Compress
+ * AmpVTS tuning
+
+0.9.4
+ * AmpVTS bright mapping revised
+ * Compress hi-pass removed
+ * Compress saturation changed from tanh to atan
+ * beep wave for Click
+ * selectable oversampling ratio for Compress in saturating mode
+ * fixed Compress to apply gain before saturation
+ * replaced sample_func_t with the more apt yield_func_t
+ * minor cosmetics
+ * caps-doc is a separate package
+ * selectable wave for Click, parfiltmodel added
+ * further smoothened ChorusII modulation
+ * selectable oversampling ratio for Amp
+ * moved some compress setup calls from Amp's run() to activate()
+ * rdf target prerequisite for dist
+ * Eq Q fixed, flat response at 0 gain
+
+0.9.3
+ * Smoother modulation for ChorusII
+ * AmpVTS filters retuned
+ * correction for ToneStack gain in AmpVTS
+ * ToneStack models reordered, 'wookie', 'porky', 'HK 20', 'nihon ace' models added
+ * Amp 'bias' now a filtered function of output power, control removed
+ * division operation in 1st-order hipass setup replaced by multiplication
+ * Saturate gain = 0 dB in 'bypass' mode
+ * Saturate 'rectify' mode
+ * AmpVTS, Plate control mapping adjusted
+
+0.9.2
+ * included dsp/polynomials.cc in the tarball
+
+0.9.1
+ * Lorenz/Roessler rate linked to fs
+ * AutoFilter replaces SweepVFI
+ * CabinetIV
+ * fixed erroneous float promotion of upsampler z padding parameter
+ * lots of parameter tuning
+ * many cleanuppies here and there
+ * SweepVFI/AutoFilter saturating SVFII ladder (instead of oversampled SVFI)
+ * UniqueIDs moved into interface.cc
+ * SweepVFI modulation comes as range + x/z now
+ * GPL v3
+ * more Phaser notches
+ * LR filters for Spice
+ * Saturate inverse gain for approximately constant loudness
+ * Scape extensively reworked
+ * further refinements to Chorus and Phaser units for smoother modulation
+ * SweepVFII removed
+ * LADSPA_PORT_GROUP
+ * Compress* input gain removed
+ * ToneStackLT rolled into ToneStack and eliminated
+ * fixed w miscalculation in DSP::Sine::set_f with fs given
+ * AmpPlus/AmpVTS total rewrite
+ * TubeCompress merged into Compress, for great template goodness
+ * *Chorus* modulation rate decoupled from samplerate
+ * tanh approximation, available in the Saturate plugin
+ * *Chorus* units reworked for smoother modulation and sound
+ * StereoChorusII2x2 plugin
+ * JVRev bandwidth filter/control added
+ * VCO* removed
+ * HRTF removed
+ * ChorusI, StereoChorusI, PhaserI removed
+ * CabinetI removed
+ * No more AmpStub silliness
+ * All Amp and Preamp plugins removed, save for AmpVTS
+ * Incite rechristened Spice
+ * JVRev blend range normalised to [0..1]
+ * Plate input value ranges adjusted to uniform [0..1]
+ * Phaser* spread value range upper bound adjusted to 1 from pi multiples
+
+0.9.0
+ * White bleached (was slightly pinkish before)
+ * up+down oversampling functionality rolled into one class
+ * Incite plugin
+ * total redesign of the Clip plugin, renamed to Saturate
+ * NoiseGate plugin
+ * RBJ prototypes changed to accept biquad instead of coeff. arrays
+ * various default value changes
+ * all FIRs changed to use templated kernel size, appreciable
+ speedup for all oversampled plugins
+ * reduced the number of arithmetic operations in the tonestack
+ coefficient update function, speedup for ToneStack and AmpVTS
+ * TubeCompress2x2 plugin
+ * TubeCompress plugin
+ * no more borrowed kernels!
+ * CabinetII gain adjusted for Mesa and Pro Jr models (much lower now and
+ more in line with the other models)
+ * Amp final drive gain reduced slightly to prevent foldover distortion
+ * SweepVF fractal rate control mapping adjusted
+ * Pan completely redesigned with Orban-like stereo image synthesis
+ * Narrower mode switch and mid/side operation added
+ * Plate parameter ranges modified for a brighter and less reverberant
+ default sound
+ * VCOd sync option had to be dropped due to compiler incompatibility
+ * Lorenz, Roessler fitted with a dc-blocking filter
+ * all int sample loop variables changed to unsigned, slightest speedup
+ * Compress2x2 plugin
+ * Compress plugin replaced with a completely new design
+ * Phaser feedback port ranges sanitised
+ * StereoPhaser plugins
+ * PhaserII gain reduced by 6dB
+ * "latency" outputs of all plugins removed
+ * Descriptor array handling changed to disable individual plugins with
+ relative ease
+ * fixed instances of g++ -O3 and above compiled code segfaulting,
+ performance gained everywhere
+ * documentation reworked extensively
+ * CEO wave is resampled at runtime from fs = 8kHz source
+ * CEO mantra changed (now an flite recording)
+ * Click changed to use modal synthesis to generate the click wavetable,
+ for improved sound and independency of sample rate
+ * Plate changed to do delay summation using doubles
+ * 1st and 2nd order filters changed to use templated types for coefficient
+ and state storage
+ * 1st order input hipass added to AmpVTS plugin
+ * filter 'mode' port added to AutoWah, and renamed to AutoFilter
+ * fixed improper 'model' labels meant to be 'mode' in the rdf generator
+
0.4.5
* Narrower plugin added
* fixed 'configure.py' to work with python3
* fixed Sin, Roessler and Lorenz gain smoothing on activation
0.4.4
-
+ * had a good time
+
0.4.3
* basics.h cleanup / comments
* minor Makefile cleanup
diff --git a/plugins/LadspaEffect/caps/Cabinet.h b/plugins/LadspaEffect/caps/Cabinet.h
index 4b6f1957d..4a1775928 100644
--- a/plugins/LadspaEffect/caps/Cabinet.h
+++ b/plugins/LadspaEffect/caps/Cabinet.h
@@ -1,21 +1,23 @@
/*
Cabinet.h
- Copyright 2002-5 Tim Goetze <tim@quitte.de>
+ Copyright 2002-14 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- CabinetI - 16th order IIR filters modeled after various impulse responses
- from Steve Harris' 'imp' plugin. Limited to 44.1 kHz sample rate.
+ CabinetIV - IIR/FIR combination filters for cabinet emulation, following
+ Bank's parfilt approach (warped Prony), see
+
+ Balázs Bank,
+ "Direct Design of Parallel Second-Order Filters for Instrument Body Modeling",
+ Proc. of the Int. Computer Music Conference (ICMC07)
+ Copenhagen, Denmark, Aug. 2007, vol. I., pp. 458-465.
- CabinetII - 32nd order IIR filters modeled after the same impulse responses
- using a different algorithm. Versions for 44.1 / 48 / 88.2 / 96 kHz sample
- rates, switched at runtime.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -29,107 +31,87 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _CABINET_H_
-#define _CABINET_H_
+#ifndef CABINET_H
+#define CABINET_H
#include "dsp/util.h"
+#include "dsp/Oversampler.h"
+#include "dsp/v4f.h"
+#include "dsp/v4f_FIR.h"
+#include "dsp/v4f_IIR2.h"
-/* cabinet_float sets the data type used for the IIR history and thus the
- * computing precision. doubles tend to make the sound more vivid and lively.
- * You can squeeze out a few extra cycles by making this 'float' if needed.
- * Be warned though that CabinetII has not been tested with 32-bit floats and
- * might become unstable due to the lower computing precision. */
typedef double cabinet_float;
typedef struct {
- int n;
- cabinet_float a[16], b[16];
- float gain;
-} Model16;
-
-typedef struct {
- int n;
+ sample_t gain;
cabinet_float a[32], b[32];
- float gain;
} Model32;
-class CabinetI
+class CabinetIII
: public Plugin
{
public:
sample_t gain;
- static Model16 models [];
+ static Model32 allmodels[];
+
+ Model32 * models;
int model;
void switch_model (int m);
- int n, h;
+ int h;
cabinet_float * a, * b;
- cabinet_float x[16], y[16];
+ cabinet_float x[32], y[32];
- template <sample_func_t F>
- void one_cycle (int frames);
+ void cycle (uint frames);
public:
static PortInfo port_info [];
void init();
-
void activate();
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
};
-/* Second version with 32nd order filters precalculated for
- * 44.1 / 48 / 88.2 / 96 kHz sample rates */
+/* /////////////////////////////////////////////////////////////////////// */
+
+template <uint N, uint FIR>
+class ParModel {
+ public:
+ float gain;
+ float a1[4*N], a2[4*N], b1[4*N], b2[4*N];
+ float fir[FIR];
+};
-class CabinetII
+class CabinetIV
: public Plugin
{
public:
- sample_t gain;
+ uint remain; /* when oversampling, an oversampling cycle may be incomplete */
- static Model32 models44100 [];
- static Model32 models48000 [];
- static Model32 models88200 [];
- static Model32 models96000 [];
+ int over; /* oversampling ratio */
+ DSP::Oversampler<2,32> over2;
+ DSP::Oversampler<4,64> over4;
- Model32 * models;
int model;
void switch_model (int m);
- int n, h;
- cabinet_float * a, * b;
- cabinet_float x[32], y[32];
-
- template <sample_func_t F>
- void one_cycle (int frames);
+ enum {
+ N = 64/4, /* number of bands/4 */
+ FIR = 128 /* FIR filter taps */
+ };
+ DSP::IIR2v4Bank<N> bank;
+ DSP::FIRv4<FIR> fir;
+
+ double gain;
+
+ void cycle (uint frames);
+ template <class O, int Ratio> void subcycle (uint frames, O & Over);
public:
static PortInfo port_info [];
- sample_t adding_gain;
-
void init();
void activate();
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
};
-#endif /* _CABINET_H_ */
+#endif /* CABINET_H */
diff --git a/plugins/LadspaEffect/caps/Chorus.cc b/plugins/LadspaEffect/caps/Chorus.cc
index cf3afe4a8..d096827af 100644
--- a/plugins/LadspaEffect/caps/Chorus.cc
+++ b/plugins/LadspaEffect/caps/Chorus.cc
@@ -1,17 +1,17 @@
/*
Chorus.cc
- Copyright 2004-7 Tim Goetze <tim@quitte.de>
+ Copyright 2004-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- mono and mono-to-stereo chorus units.
+ Chorus units.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -30,298 +30,65 @@
#include "Chorus.h"
#include "Descriptor.h"
-template <sample_func_t F>
void
-ChorusI::one_cycle (int frames)
+ChorusI::activate()
{
- sample_t * s = ports[0];
-
- double one_over_n = 1 / (double) frames;
- double ms = .001 * fs;
-
- double t = time;
- time = getport(1) * ms;
- double dt = (time - t) * one_over_n;
-
- double w = width;
- width = getport(2) * ms;
- /* clamp, or we need future samples from the delay line */
- if (width >= t - 3) width = t - 3;
- double dw = (width - w) * one_over_n;
-
- if (rate != *ports[3])
- lfo.set_f (max (rate = getport(3), .000001), fs, lfo.get_phase());
-
- double blend = getport(4);
- double ff = getport(5);
- double fb = getport(6);
-
- sample_t * d = ports[7];
-
- DSP::FPTruncateMode truncate;
-
- for (int i = 0; i < frames; ++i)
- {
- sample_t x = s[i];
-
- /* truncate the feedback tap to integer, better quality for less
- * cycles (just a bit of zipper when changing 't', but it does sound
- * interesting) */
- int ti;
- fistp (t, ti);
- x -= fb * delay[ti];
-
- delay.put (x + normal);
-
-# if 0
- /* allpass delay sounds a little cleaner for a chorus
- * but sucks big time when flanging. */
- x = blend * x + ff * tap.get (delay, t + w * lfo.get());
-# elif 0
- /* linear interpolation */
- x = blend * x + ff * delay.get_at (t + w * lfo.get());
-# else
- /* cubic interpolation */
- x = blend * x + ff * delay.get_cubic (t + w * lfo.get());
-# endif
-
- F (d, i, x, adding_gain);
-
- t += dt;
- w += dw;
- }
+ setrate (getport(0));
+ time = 0;
+ width = 0;
+ delay.reset();
+ hp.reset();
+ hp.set_f(250*over_fs);
}
-/* //////////////////////////////////////////////////////////////////////// */
-
-PortInfo
-ChorusI::port_info [] =
-{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "t (ms)",
- INPUT | CONTROL,
- {BOUNDED | LOG | DEFAULT_LOW, 2.5, 40}
- }, {
- "width (ms)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_1, .5, 10}
- }, {
- "rate (Hz)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 5}
- }, {
- "blend",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_1, 0, 1}
- }, {
- "feedforward",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 1}
- }, {
- "feedback",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, 0, 1}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }
-};
-
-template <> void
-Descriptor<ChorusI>::setup()
-{
- UniqueID = 1767;
- Label = "ChorusI";
- Properties = HARD_RT;
-
- Name = CAPS "ChorusI - Mono chorus/flanger";
- Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
-
- /* fill port info and vtable */
- autogen();
-}
-
-/* //////////////////////////////////////////////////////////////////////// */
-
-template <sample_func_t F>
void
-StereoChorusI::one_cycle (int frames)
-{
- sample_t * s = ports[0];
-
- double one_over_n = 1 / (double) frames;
- double ms = .001 * fs;
-
- double t = time;
- time = getport(1) * ms;
- double dt = (time - t) * one_over_n;
-
- double w = width;
- width = getport(2) * ms;
- /* clamp, or we need future samples from the delay line */
- if (width >= t - 1) width = t - 1;
- double dw = (width - w) * one_over_n;
-
- if (rate != *ports[3] && phase != *ports[4])
- {
- rate = getport(3);
- phase = getport(4);
- double phi = left.lfo.get_phase();
- left.lfo.set_f (max (rate, .000001), fs, phi);
- right.lfo.set_f (max (rate, .000001), fs, phi + phase * M_PI);
- }
-
- double blend = getport(5);
- double ff = getport(6);
- double fb = getport(7);
-
- sample_t * dl = ports[8];
- sample_t * dr = ports[9];
-
- /* to go sure (on i386) that the fistp instruction does the right thing
- * when looking up fractional sample indices */
- DSP::FPTruncateMode truncate;
-
- for (int i = 0; i < frames; ++i)
- {
- sample_t x = s[i];
-
- /* truncate the feedback tap to integer, better quality for less
- * cycles (just a bit of zipper when changing 't', but it does sound
- * interesting) */
- int ti;
- fistp (t, ti);
- x -= fb * delay[ti];
-
- delay.put (x + normal);
-
- sample_t l = blend * x + ff * delay.get_cubic (t + w * left.lfo.get());
- sample_t r = blend * x + ff * delay.get_cubic (t + w * right.lfo.get());
-
- F (dl, i, l, adding_gain);
- F (dr, i, r, adding_gain);
-
- t += dt;
- w += dw;
- }
-}
-
-/* //////////////////////////////////////////////////////////////////////// */
-
-PortInfo
-StereoChorusI::port_info [] =
+ChorusI::setrate (float r)
{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "t (ms)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MIN, 2.5, 40}
- }, {
- "width (ms)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_1, .5, 10}
- }, {
- "rate (Hz)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 5}
- }, {
- "phase",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MAX, 0, 1}
- }, {
- "blend",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_1, 0, 1}
- }, {
- "feedforward",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 1}
- }, {
- "feedback",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, 0, 1}
- }, {
- "out:l",
- OUTPUT | AUDIO,
- {0}
- }, {
- "out:r",
- OUTPUT | AUDIO,
- {0}
- }
-};
-
-template <> void
-Descriptor<StereoChorusI>::setup()
-{
- UniqueID = 1768;
- Label = "StereoChorusI";
- Properties = HARD_RT;
-
- Name = CAPS "StereoChorusI - Stereo chorus/flanger";
- Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
-
- /* fill port info and vtable */
- autogen();
+ if (r == rate) return;
+ rate = r;
+ lfo.sine.set_f (rate, fs, lfo.sine.get_phase());
}
-/* //////////////////////////////////////////////////////////////////////// */
-
-template <sample_func_t F>
void
-ChorusII::one_cycle (int frames)
+ChorusI::cycle (uint frames)
{
- sample_t * s = ports[0];
-
- double one_over_n = 1 / (double) frames;
- double ms = .001 * fs;
+ float one_over_n = 1/(float)frames;
+ float ms = fs*.001;
- double t = time;
- time = getport(1) * ms;
- double dt = (time - t) * one_over_n;
+ float t = time;
+ time = getport(0)*ms;
+ float dt = (time - t)*one_over_n;
- double w = width;
- width = getport(2) * ms;
+ float w = width;
+ width = getport(1)*ms;
/* clamp, or we need future samples from the delay line */
if (width >= t - 3) width = t - 3;
- double dw = (width - w) * one_over_n;
+ float dw = (width - w) * one_over_n;
- if (rate != *ports[3])
- set_rate (*ports[3]);
-
- double blend = getport(4);
- double ff = getport(5);
- double fb = getport(6);
+ setrate (getport(2));
+
+ float blend = getport(3);
+ float ff = getport(4);
+ float fb = getport(5);
+ sample_t * s = ports[6];
sample_t * d = ports[7];
DSP::FPTruncateMode truncate;
- for (int i = 0; i < frames; ++i)
+ for (uint i = 0; i < frames; ++i)
{
- sample_t x = s[i];
+ sample_t x=s[i], y=x;
- x -= fb * delay.get_cubic (t);
+ x = hp.process(x+normal);
- delay.put (filter.process (x + normal));
+ y -= fb*delay.get_linear(t);
- double a = 0;
- for (int j = 0; j < Taps; ++j)
- a += taps[j].get (delay, t, w);
+ delay.put (y + normal);
- x = blend * x + ff * a;
+ y += blend*x + ff*delay.get_cubic (t + w*lfo.sine.get());
- F (d, i, x, adding_gain);
+ d[i] = y;
t += dt;
w += dw;
@@ -331,173 +98,29 @@ ChorusII::one_cycle (int frames)
/* //////////////////////////////////////////////////////////////////////// */
PortInfo
-ChorusII::port_info [] =
-{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "t (ms)",
- INPUT | CONTROL,
- {BOUNDED | LOG | DEFAULT_LOW, 2.5, 40}
- }, {
- "width (ms)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_1, .5, 10}
- }, {
- "rate",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 1}
- }, {
- "blend",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_1, 0, 1}
- }, {
- "feedforward",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 1}
- }, {
- "feedback",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, 0, 1}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }
-};
-
-template <> void
-Descriptor<ChorusII>::setup()
-{
- UniqueID = 2583;
- Label = "ChorusII";
- Properties = HARD_RT;
-
- Name = CAPS "ChorusII - Mono chorus/flanger modulated by a fractal";
- Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
-
- /* fill port info and vtable */
- autogen();
-}
-
-/* //////////////////////////////////////////////////////////////////////// */
-
-template <sample_func_t F>
-void
-StereoChorusII::one_cycle (int frames)
+ChorusI::port_info [] =
{
- sample_t * s = ports[0];
-
- double one_over_n = 1 / (double) frames;
- double ms = .001 * fs;
-
- double t = time;
- time = getport(1) * ms;
- double dt = (time - t) * one_over_n;
-
- double w = width;
- width = getport(2) * ms;
- /* clamp, or we need future samples from the delay line */
- if (width >= t - 1) width = t - 1;
- double dw = (width - w) * one_over_n;
-
- set_rate (*ports[3]);
-
- double blend = getport(4);
- double ff = getport(5);
- double fb = getport(6);
-
- sample_t * dl = ports[7];
- sample_t * dr = ports[8];
-
- /* to go sure (on i386) that the fistp instruction does the right thing
- * when looking up fractional sample indices */
- DSP::FPTruncateMode truncate;
-
- for (int i = 0; i < frames; ++i)
- {
- sample_t x = s[i];
+ { "t (ms)", CTRL_IN, {LOG | DEFAULT_MID, 2.5, 40} },
+ { "width (ms)", CTRL_IN, {DEFAULT_LOW, .5, 10} },
- /* truncate the feedback tap to integer, better quality for less
- * cycles (just a bit of zipper when changing 't', but it does sound
- * interesting) */
- int ti;
- fistp (t, ti);
- x -= fb * delay[ti];
+ { "rate (Hz)", CTRL_IN | GROUP, {LOG | DEFAULT_LOW, 0.02, 5} },
- delay.put (x + normal);
+ { "blend", CTRL_IN | GROUP, {DEFAULT_LOW, 0, 1} },
+ { "feedforward", CTRL_IN, {DEFAULT_LOW, 0, 1} },
+ { "feedback", CTRL_IN, {DEFAULT_LOW, 0, 1} },
- double m;
- m = left.lfo_lp.process (left.fractal.get());
- sample_t l = blend * x + ff * delay.get_cubic (t + w * m);
- m = right.lfo_lp.process (right.fractal.get());
- sample_t r = blend * x + ff * delay.get_cubic (t + w * m);
-
- F (dl, i, l, adding_gain);
- F (dr, i, r, adding_gain);
-
- t += dt;
- w += dw;
- }
-}
-
-/* //////////////////////////////////////////////////////////////////////// */
-
-PortInfo
-StereoChorusII::port_info [] =
-{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "t (ms)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 2.5, 40}
- }, {
- "width (ms)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, .5, 10}
- }, {
- "rate",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 1}
- }, {
- "blend",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_1, 0, 1}
- }, {
- "feedforward",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, 1}
- }, {
- "feedback",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, 0, 1}
- }, {
- "out:l",
- OUTPUT | AUDIO,
- {0}
- }, {
- "out:r",
- OUTPUT | AUDIO,
- {0}
- }
+ { "in", INPUT | AUDIO },
+ { "out", OUTPUT | AUDIO }
};
template <> void
-Descriptor<StereoChorusII>::setup()
+Descriptor<ChorusI>::setup()
{
- UniqueID = 2584;
- Label = "StereoChorusII";
- Properties = HARD_RT;
+ Label = "ChorusI";
- Name = CAPS "StereoChorusII - Stereo chorus/flanger modulated by a fractal";
+ Name = CAPS "ChorusI - Mono chorus/flanger";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ Copyright = "GPL, 2004-13";
/* fill port info and vtable */
autogen();
diff --git a/plugins/LadspaEffect/caps/Chorus.h b/plugins/LadspaEffect/caps/Chorus.h
index 03ed256ec..83db371ee 100644
--- a/plugins/LadspaEffect/caps/Chorus.h
+++ b/plugins/LadspaEffect/caps/Chorus.h
@@ -1,18 +1,15 @@
/*
Chorus.h
- Copyright 2004-5 Tim Goetze <tim@quitte.de>
+ Copyright 2004-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- mono and stereo chorus/flanger units, traditional designs and some
- differentiated a bit further.
-
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -26,276 +23,44 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _CHORUS_H_
-#define _CHORUS_H_
+#ifndef CHORUS_H
+#define CHORUS_H
#include "dsp/Sine.h"
#include "dsp/Roessler.h"
#include "dsp/Lorenz.h"
#include "dsp/Delay.h"
-#include "dsp/OnePole.h"
-#include "dsp/BiQuad.h"
+#include "dsp/IIR1.h"
+#include "dsp/IIR2.h"
#include "dsp/RBJ.h"
-class ChorusStub
-: public Plugin
-{
- public:
- sample_t time, width, rate;
-};
-
class ChorusI
-: public ChorusStub
-{
- public:
- DSP::Sine lfo;
- DSP::Delay delay;
- DSP::DelayTapA tap;
-
- template <sample_func_t>
- void one_cycle (int frames);
-
- public:
- static PortInfo port_info [];
-
- void init()
- {
- rate = .15;
- delay.init ((int) (.040 * fs));
- }
-
- void activate()
- {
- time = 0;
- width = 0;
-
- rate = *ports[3];
-
- delay.reset();
- tap.reset();
-
- lfo.set_f (rate, fs, 0);
- }
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
-};
-
-class StereoChorusI
-: public ChorusStub
-{
- public:
- sample_t rate;
- sample_t phase;
-
- DSP::Delay delay;
-
- struct {
- DSP::Sine lfo;
- DSP::DelayTapA tap;
- } left, right;
-
- template <sample_func_t>
- void one_cycle (int frames);
-
- public:
- static PortInfo port_info [];
-
- void init()
- {
- rate = .15;
- phase = .5; /* pi */
-
- delay.init ((int) (.040 * fs));
- }
-
- void activate()
- {
- time = 0;
- width = 0;
-
- delay.reset();
-
- left.tap.reset();
- right.tap.reset();
-
- left.lfo.set_f (rate, fs, 0);
- right.lfo.set_f (rate, fs, phase * M_PI);
- }
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
-};
-
-/* //////////////////////////////////////////////////////////////////////// */
-
-#define FRACTAL_RATE 0.02
-
-/* fractally modulated Chorus units */
-
-class FracTap
-{
- public:
- DSP::Lorenz f1;
- DSP::Roessler f2;
- DSP::OnePoleLP lp;
-
- void init (double fs)
- {
- lp.set_f (30. / fs);
- f1.init (.001, frandom());
- f2.init (.001, frandom());
- }
-
- void set_rate (sample_t r)
- {
- f1.set_rate (r * FRACTAL_RATE);
- f2.set_rate (3.3 * r * FRACTAL_RATE);
- }
-
- /* t = time, w = width, should inline nicely */
- sample_t get (DSP::Delay & d, double t, double w)
- {
- double m = lp.process (f1.get() + .3 * f2.get());
- return d.get_cubic (t + w * m);
- }
-};
-
-class ChorusII
-: public ChorusStub
+: public Plugin
{
public:
- enum {
- Taps = 1
- };
-
- FracTap taps[Taps];
- DSP::BiQuad filter;
- DSP::Delay delay;
-
- template <sample_func_t>
- void one_cycle (int frames);
-
- void set_rate (sample_t r)
- {
- rate = r;
- for (int i = 0; i < Taps; ++i)
- {
- taps[i].set_rate (rate * (i * FRACTAL_RATE) / Taps);
- // fprintf (stderr, "[%d] %.3f\n", i, (rate * (i * FRACTAL_RATE) / Taps));
- }
- }
-
- public:
- static PortInfo port_info [];
-
- void init()
- {
- delay.init ((int) (.040 * fs));
- for (int i = 0; i < Taps; ++i)
- taps[i].init (fs);
- DSP::RBJ::HiShelve (1000. / fs, 1., 6, filter.a, filter.b);
- }
+ DSP::HP1<sample_t> hp;
+ float time, width, rate;
- void activate()
- {
- time = 0;
- width = 0;
-
- set_rate (*ports[3]);
-
- delay.reset();
- filter.reset();
- }
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
-};
-
-class StereoChorusII
-: public ChorusStub
-{
- public:
- sample_t rate;
- sample_t phase;
+ struct { DSP::Sine sine; } lfo;
DSP::Delay delay;
- struct {
- DSP::Roessler fractal;
- DSP::OnePoleLP lfo_lp;
- DSP::DelayTapA tap;
- } left, right;
-
- template <sample_func_t>
- void one_cycle (int frames);
+ void cycle(uint frames);
- void set_rate (sample_t r)
- {
- rate = r;
- left.fractal.set_rate (rate * FRACTAL_RATE);
- right.fractal.set_rate (rate * FRACTAL_RATE);
- left.lfo_lp.set_f (3. / fs);
- right.lfo_lp.set_f (3. / fs);
- }
+ void setrate(float r);
public:
static PortInfo port_info [];
- sample_t adding_gain;
void init()
{
- phase = .5; /* pi */
-
- delay.init ((int) (.040 * fs));
-
- left.fractal.init (.001, frandom());
- right.fractal.init (.001, frandom());
- }
-
- void activate()
- {
- time = 0;
- width = 0;
-
- delay.reset();
-
- left.tap.reset();
- right.tap.reset();
-
- set_rate (*ports[3]);
+ rate = .15;
+ lfo.sine.set_f (rate, fs, 0);
+ delay.init ((int) (.050 * fs));
}
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
+ void activate();
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
};
-#endif /* _CHORUS_H_ */
+#endif /* CHORUS_H */
diff --git a/plugins/LadspaEffect/caps/Click.cc b/plugins/LadspaEffect/caps/Click.cc
index d862c55f4..2a159a28e 100644
--- a/plugins/LadspaEffect/caps/Click.cc
+++ b/plugins/LadspaEffect/caps/Click.cc
@@ -1,7 +1,7 @@
/*
Click.cc
- Copyright 2002-7 Tim Goetze <tim@quitte.de>
+ Copyright 2002-14 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -11,7 +11,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -26,30 +26,33 @@
*/
#include "basics.h"
-
-#include "waves/click.h"
-#include "waves/money.h"
+#include "dsp/RBJ.h"
+#include "dsp/White.h"
#include "Click.h"
#include "Descriptor.h"
+template <int Waves>
void
-ClickStub::init (float * _wave, int _N)
+ClickStub<Waves>::initwave (int i, int16 * _wave, uint _N)
{
- wave = _wave;
- N = _N;
- bpm = -1;
+ wave[i].data = _wave;
+ wave[i].N = _N;
}
-template <sample_func_t F>
+template <int Waves>
void
-ClickStub::one_cycle (int frames)
+ClickStub<Waves>::cycle (uint frames)
{
- bpm = getport(0);
- sample_t gain = getport(1) * *ports[1];
- lp.set (1 - *ports[2]);
+ static double scale16 = 1./32768;
+ int p = Waves > 1 ? 1 : 0; /* port */
+ int w = p ? getport(0) : 0; /* wave */
+ bpm = getport(p);
+ sample_t gain = getport(p+1);
+ gain *= scale16 * gain;
+ lp.set (1 - getport (p+2));
- sample_t * d = ports[3];
+ sample_t * d = ports[p+3];
while (frames)
{
@@ -59,28 +62,25 @@ ClickStub::one_cycle (int frames)
played = 0;
}
- int n = min (frames, period);
+ uint n = min (frames, period);
- if (played < N)
+ if (played < wave[w].N)
{
- n = min (n, N - played);
+ n = min (n, wave[w].N - played);
- for (int i = 0; i < n; ++i)
+ for (uint i = 0; i < n; ++i)
{
- sample_t x = gain * wave [played + i] + normal;
- F (d, i, lp.process (x), adding_gain);
- normal = -normal;
+ double x = gain * wave[w].data[played+i];
+ x = lp.process (x);
+ d[i] = x;
}
played += n;
}
else
{
- for (int i = 0; i < n; ++i)
- {
- F (d, i, lp.process (normal), adding_gain);
- normal = -normal;
- }
+ for (uint i = 0; i < n; ++i)
+ d[i] = lp.process(normal);
}
period -= n;
@@ -91,93 +91,155 @@ ClickStub::one_cycle (int frames)
/* //////////////////////////////////////////////////////////////////////// */
-PortInfo
-ClickStub::port_info [] =
+/* initialising the wavetable for the click sound from a few oscillation modes */
+void
+Click::initsimple()
{
+ enum { Peaks = 7 };
+ /* {f, gain} pairs */
+ static float _peaks[Peaks][2] = {
+ {897, 1.0},
+ {1423, 0.20},
+ {2054, 0.92},
+ {2353, 0.244},
+ {2843, 0.719},
+ {3567, 0.191},
+ {5117, 0.091},
+ };
+
+ DSP::LP1<sample_t> lp1;
+ lp1.set_f (800*over_fs);
+ DSP::IIR2<sample_t> lp;
+ DSP::RBJ::LP (8000*over_fs, .2, lp);
+
+ DSP::IIR2<sample_t> peaks[Peaks];
+ for (int i = 0; i < Peaks; ++i)
{
- "bpm",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 4, 244}
- }, {
- "volume",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, 1}
- }, {
- "damping",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_HIGH, 0, 1}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
+ /* tune to g' = 784 Hz */
+ float f = .8740245*_peaks[i][0]*over_fs;
+ float g = _peaks[i][1];
+ DSP::RBJ::BP (f, 22*g, peaks[i]);
}
-};
-/* //////////////////////////////////////////////////////////////////////// */
+ DSP::IIR2<sample_t> bp;
+ DSP::RBJ::BP (150*over_fs, 3.8, bp);
-#define LENGTH(W) ((int) (sizeof (W) / sizeof (float)))
+ DSP::IIR2<sample_t> post;
+ DSP::RBJ::PeakingEQ (1000*over_fs, 1.8, 24, post);
-void
-Click::init()
-{
- this->ClickStub::init (click, LENGTH (click));
+ int n = (int) (fs * 2800. / 44100.);
+ int16 * click = new int16 [n];
+
+ DSP::White white;
+ int m = 8;
+ sample_t mi = 1./m;
+ sample_t x;
+ for (int i = 0; i < n; ++i)
+ {
+ if (i < m) /* simplistic noise excitation signal */
+ x = .5 * white.get() * (m-i)*mi;
+ x = lp.process (x);
+ double a = x;
+ for (int j = 0; j < Peaks; ++j)
+ a += peaks[j].process_bp (x);
+ a = post.process (a);
+ /* add some ring-modulated noisz */
+ a += a*bp.process(white.get());
+ click[i] = (int16) (a * 32767.);
+ x = 0;
+ }
+
+ initwave (0, click, n);
}
-template <> void
-Descriptor<Click>::setup()
+/* using parfilt models to generate the click */
+#include "Cabinet.h"
+#include "waves/click.h"
+
+void
+Click::initparfilt()
{
- UniqueID = 1769;
- Label = "Click";
- Properties = HARD_RT;
+ DSP::IIR2v4Bank<128> bank;
+
+ ParModel<128,1> * model;
+ if (fs > 120000) model = &waves_click_wav_176000;
+ else if (fs > 60000) model = &waves_click_wav_88200;
+ else model = &waves_click_wav_44100;
+
+ bank.set_a (1, model->a1);
+ bank.set_a (2, model->a2);
+ bank.set_b (1, model->b1);
+ bank.set_b (2, model->b2);
+ bank.reset();
+
+ int n = (int) (fs*2800/44100);
+ int16 * click = new int16 [n];
- Name = CAPS "Click - Metronome";
- Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ DSP::IIR2<sample_t> hp;
+ DSP::RBJ::HP (1520*over_fs, .7, hp);
- /* fill port info and vtable */
- autogen();
-}
+ DSP::White white;
+ int m = 3;
+ sample_t mi = 1./m;
+ sample_t x;
+ for (int i = 0; i < n; ++i)
+ {
+ if (i < m) /* simplistic noise excitation signal */
+ x = .5 * white.get() * (m-i)*mi;
+ x = v4f_sum (bank.process_bp(v4f(x)));
+ x = hp.process(x);
+ click[i] = (int16) (x * 32767.);
+ x = 0;
+ }
-/* //////////////////////////////////////////////////////////////////////// */
+ initwave (1, click, n);
+}
-PortInfo
-CEO::port_info [] =
+void
+Click::initsine()
{
+ float f = 2*784;
+ DSP::Sine sin (2*M_PI*f*over_fs);
+
+ int n = (int) (12*fs/f);
+ int m = 6*n/4;
+ int16 * click = new int16 [m];
+
+ DSP::IIR2<sample_t> lp;
+ DSP::RBJ::BP (f*over_fs,2.5,lp);
+
+ float a = .4 * 32767;
+ for (int i = 0; i < n; ++i)
{
- "mpm",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_HIGH, 4, 244}
- }, {
- "volume",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, 1}
- }, {
- "damping",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MIN, 0, 1}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
+ sample_t x = a*sin.get();
+ x = lp.process(x);
+ click[i] = (int16) (x);
}
-};
+ for (int i = n; i < m; ++i)
+ {
+ sample_t x = lp.process(NOISE_FLOOR);
+ click[i] = (int16) (x);
+ }
+
+ initwave (2, click, m);
+}
void
-CEO::init()
+Click::initdirac()
{
- this->ClickStub::init (money, LENGTH (money));
+ int16 * dirac = new int16[1];
+ *dirac = 32767;
+ initwave (3, dirac, 1);
}
template <> void
-Descriptor<CEO>::setup()
+Descriptor<Click>::setup()
{
- UniqueID = 1770;
- Label = "CEO";
- Properties = HARD_RT;
+ Label = "Click";
- Name = CAPS "CEO - Chief Executive Oscillator";
+ Name = CAPS "Click - Metronome";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ Copyright = "2004-14";
/* fill port info and vtable */
autogen();
@@ -185,46 +247,90 @@ Descriptor<CEO>::setup()
/* //////////////////////////////////////////////////////////////////////// */
-float dirac [] = { 1, };
+PortInfo
+Click::port_info [] =
+{
+ { "model", CTRL_IN, {INTEGER | DEFAULT_1, 0, 3},
+ "{0:'box',1:'stick',2:'beep',3:'dirac'}" },
+ { "bpm", CTRL_IN | GROUP, {DEFAULT_LOW, 4, 240} },
+ { "volume", CTRL_IN | GROUP, {DEFAULT_HIGH, 0, 1} },
+ { "damping", CTRL_IN, {DEFAULT_HIGH, 0, 1} },
+
+ { "out", OUTPUT | AUDIO}
+};
+
+/* //////////////////////////////////////////////////////////////////////// */
PortInfo
-Dirac::port_info [] =
+CEO::port_info [] =
{
- {
- "ppm",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MIN, 30, 60}
- }, {
- "volume",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, 0, 1}
- }, {
- "damping",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, 0, 1}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }
+ { "ppm", CTRL_IN, {DEFAULT_LOW, 30, 232} },
+ { "volume", CTRL_IN | GROUP, {DEFAULT_HIGH, 0, 1}},
+ { "damping", CTRL_IN, {DEFAULT_0, 0, 1} },
+
+ { "out", OUTPUT | AUDIO, {0} }
};
+#include "waves/profit.h"
+
void
-Dirac::init()
+CEO::init()
{
- this->ClickStub::init (dirac, LENGTH (dirac));
+ int m = sizeof(profit)/sizeof(*profit);
+
+ float s = fs/8000., dx = 1/s;
+
+ int n = (int) (s*m);
+ int16 * wave = new int16 [n];
+
+ DSP::IIR2<sample_t> lp;
+ /* suppress aliasing with an additional lowpass; also slight gain at 3 kHz */
+ DSP::RBJ::LP (3000*over_fs,1.5,lp);
+ #if 1 /* linear */
+ float x = 0;
+ for (int i = 0; i < n-1; ++i)
+ {
+ int j = (int) x;
+ float a = x-j;
+ a = (1-a)*profit[j] + a*profit[j+1];
+ a = lp.process(a);
+ wave[i] = (int16) a;
+ x += dx;
+ }
+ #else /* cubic, unneeded */
+ float x = 0;
+ for (int i=0; i < n; ++i, x+=dx)
+ {
+ int j = (int) x;
+ float f = x-j;
+ sample_t x_1 = profit[j-1];
+ sample_t x0 = profit[j];
+ sample_t x1 = profit[j+1];
+ sample_t x2 = profit[j+2];
+
+ sample_t a =
+ .5*(3*(x0 - x1) - x_1 + x2);
+ sample_t b =
+ 2*x1 + x_1 - .5*(5*x0 + x2);
+ sample_t c =
+ .5*(x1 - x_1);
+
+ a = x0 + (((a * f) + b) * f + c) * f;
+ wave[i] = (int16) a;
+ }
+ #endif
+
+ initwave (0, wave, n-1);
}
template <> void
-Descriptor<Dirac>::setup()
+Descriptor<CEO>::setup()
{
- UniqueID = 2585;
- Label = "Dirac";
- Properties = HARD_RT;
+ Label = "CEO";
- Name = CAPS "Dirac - One-sample impulse generator";
+ Name = CAPS "CEO - Chief Executive Oscillator";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ Copyright = "2004-12";
/* fill port info and vtable */
autogen();
diff --git a/plugins/LadspaEffect/caps/Click.h b/plugins/LadspaEffect/caps/Click.h
index feb5bc7a5..7d6ced9a1 100644
--- a/plugins/LadspaEffect/caps/Click.h
+++ b/plugins/LadspaEffect/caps/Click.h
@@ -1,17 +1,17 @@
/*
Click.h
- Copyright 2004-5 Tim Goetze <tim@quitte.de>
+ Copyright 2004-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- units perpetually repeating a recorded sample.
+ Units repeatedly playing back a wavetable.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,69 +25,71 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _CLICK_H_
-#define _CLICK_H_
+#ifndef CLICK_H
+#define CLICK_H
-#include "dsp/OnePole.h"
+#include "dsp/IIR1.h"
+#include "dsp/IIR2.h"
#include "dsp/util.h"
+template <int Waves>
class ClickStub
: public Plugin
{
public:
sample_t bpm;
- float * wave;
- int N; /* number of samples in wave */
+ struct {
+ int16 * data;
+ uint N; /* number of samples in wave */
+ } wave[Waves];
- DSP::OnePoleLP lp;
+ DSP::LP1<sample_t> lp;
- int period; /* frames remaining in period */
- int played; /* frames played from sample */
+ uint period; /* frames remaining in period */
+ uint played; /* frames played from sample */
- template <sample_func_t F>
- void one_cycle (int frames);
+ void cycle (uint frames);
- public:
- static PortInfo port_info [];
+ ClickStub()
+ {
+ for (int i=0; i < Waves; ++i)
+ wave[i].data = 0;
+ }
+ ~ClickStub()
+ {
+ for (int i=0; i < Waves; ++i)
+ if (wave[i].data) delete [] wave[i].data;
+ }
- void init (float * _wave, int _N);
+ public:
+ void initwave (int i, int16 * wave, uint N);
void activate()
{
played = 0;
period = 0;
- }
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
+ bpm = -1;
}
};
class Click
-: public ClickStub
+: public ClickStub<4>
{
public:
- void init();
-};
+ void initsimple();
+ void initparfilt();
+ void initsine();
+ void initdirac();
-class CEO
-: public ClickStub
-{
- public:
- void init();
+ void init()
+ { initsimple(); initparfilt(); initsine(); initdirac(); }
static PortInfo port_info [];
};
-class Dirac
-: public ClickStub
+class CEO
+: public ClickStub<1>
{
public:
void init();
@@ -95,4 +97,4 @@ class Dirac
static PortInfo port_info [];
};
-#endif /* _CLICK_H_ */
+#endif /* CLICK_H */
diff --git a/plugins/LadspaEffect/caps/Compress.cc b/plugins/LadspaEffect/caps/Compress.cc
index 3d9725a64..582a05c08 100644
--- a/plugins/LadspaEffect/caps/Compress.cc
+++ b/plugins/LadspaEffect/caps/Compress.cc
@@ -1,18 +1,15 @@
/*
Compress.cc
- Copyright 2004-7 Tim Goetze <tim@quitte.de>
+ Copyright 2011-14 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
-
- mono compressor suitable for solo instruments. adaptation of Steve
- Harris' 'sc1' unit, with minor tweaks: table lookup for attack and
- release time to frames mapping replaced with exp. port order changed.
+
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -31,69 +28,105 @@
#include "Compress.h"
#include "Descriptor.h"
-template <sample_func_t F>
+template <int Channels>
void
-Compress::one_cycle (int frames)
+CompressStub<Channels>::activate()
{
- sample_t * s = ports[0];
-
- sample_t range = DSP::db2lin (getport(1));
- sample_t ratio = (*ports[2] - 1) / getport(2);
-
- /* sc1 has lookup tables here, and they're only 40 % used (400 ms/1 s).
- * thus, sc1's attack/release controls are a bit coarse due to truncation
- * error. calling exp() once per cycle doesn't seem too expensive.
- *
- * besides, i have a suspicion that the attack and release parameters
- * don't work like they should. if they, as the code suggests, control
- * an exponential envelope fade, pow (.5, n_frames) or something like
- * that seems more appropriate. so ...
- *
- * TODO: check whether these parameters work like they should, try pow()
- */
- double ga = exp (-1 / (fs * getport(3)));
- double gr = exp (-1 / (fs * getport(4)));
-
- sample_t threshold = getport(5);
- sample_t knee = getport(6);
-
- sample_t * d = ports[7];
-
- sample_t knee0 = DSP::db2lin (threshold - knee);
- sample_t knee1 = DSP::db2lin (threshold + knee);
-
- sample_t ef_a = ga * .25;
- sample_t ef_ai = 1 - ef_a;
-
- for (int i = 0; i < frames; ++i)
- {
- sum += s[i] * s[i];
+ compress.peak.init (fs);
+ compress.rms.init (fs);
+ remain = 0;
+}
+
+template <int Channels>
+void
+CompressStub<Channels>::cycle (uint frames)
+{
+ int c = getport(0);
+ if (c == 0) subcycle<DSP::CompressPeak> (frames, compress.peak);
+ else subcycle<DSP::CompressRMS> (frames, compress.rms);
+}
+
+/* do-nothing 'saturator' as template for linear instead of saturating mode */
+struct NoSat { sample_t process(sample_t x) { return x; } };
+
+template <int Channels>
+template <class Comp>
+void
+CompressStub<Channels>::subcycle (uint frames, Comp & comp)
+{
+ static NoSat none;
+
+ int s = getport(1);
+ if (s == 1) subsubcycle<Comp,CompSat2>
+ (frames, comp, saturate[0].two, saturate[1].two);
+ else if (s == 2) subsubcycle<Comp,CompSat4>
+ (frames, comp, saturate[0].four, saturate[1].four);
+ #if 0
+ else if (s == 3) subsubcycle<Comp,CompSat8>
+ (frames, comp, saturate[0].eight, saturate[1].eight);
+ #endif
+ else subsubcycle<Comp, NoSat>
+ (frames, comp, none, none);
+}
+
+template <int Channels>
+template <class Comp, class Sat>
+void
+CompressStub<Channels>::subsubcycle (uint frames, Comp & comp, Sat & satl, Sat & satr)
+{
+ comp.set_threshold (pow(getport(2), 1.6));
+ sample_t strength = pow(getport(3), 1.4); /* more resolution in lower range */
+ comp.set_attack (getport(4));
+ comp.set_release (getport(5));
+ sample_t gain_out = db2lin (getport (6));
- if (amp > env)
- env = env * ga + amp * (1 - ga);
- else
- env = env * gr + amp * (1 - gr);
+ sample_t * sl = ports[Stereo ? 8 : 8]; /* ;) */
+ sample_t * sr = ports[Stereo ? 9 : 8];
- if ((count++ & 3) == 3)
+ sample_t * dl = ports[Stereo ? 10 : 9];
+ sample_t * dr = ports[Stereo ? 11 : 9];
+
+ sample_t state = 1;
+
+ while (frames)
+ {
+ if (remain == 0)
{
- amp = rms.process (sum * .25);
- sum = 0;
-
- if (env < knee0)
- gain_t = 1;
- else if (env < knee1)
- {
- float x = -(threshold - knee - DSP::lin2db (env)) / knee;
- gain_t = DSP::db2lin (-knee * ratio * x * x * 0.25f);
- }
- else
- gain_t = DSP::db2lin ((threshold - DSP::lin2db (env)) * ratio);
+ remain = comp.blocksize;
+ comp.start_block(strength);
+ state = min(state,comp.gain.state);
}
- gain = gain * ef_a + gain_t * ef_ai;
+ uint n = min (frames, remain);
+
+ for (uint i=0; i<n; ++i)
+ {
+ sample_t xl = sl[i], xr = sr[i];
+
+ if (Stereo)
+ comp.store(xl, xr);
+ else
+ comp.store(xl);
+
+ sample_t gain = gain_out*comp.get();
+
+ xl = satl.process(xl*gain);
+ if (Stereo)
+ xr = satr.process(xr*gain);
+
+ dl[i] = xl;
+ if (Stereo)
+ dr[i] = xr;
+ }
- F (d, i, s[i] * gain * range, adding_gain);
+ sl += n;
+ sr += n;
+ dl += n;
+ dr += n;
+ frames -= n;
+ remain -= n;
}
+ *ports[7] = lin2db(state);
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -101,51 +134,64 @@ Compress::one_cycle (int frames)
PortInfo
Compress::port_info [] =
{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "gain (dB)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 24}
- }, {
- "ratio (1:n)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MIN, 1, 10}
- }, {
- "attack (s)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MIN, .001, 1}
- }, {
- "release (s)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, .001, 1}
- }, {
- "threshold (dB)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -30, 400}
- }, {
- "knee radius (dB)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 1, 10}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }
+ { "measure", CTRL_IN, {INTEGER | DEFAULT_0, 0, 1}, "{0:'peak',1:'rms'}" },
+ { "mode", CTRL_IN | GROUP, {INTEGER | DEFAULT_1, 0, 2},
+ "{0:'no limiting',1:'saturating 2x',2:'saturating 4x',3:'saturating 4x128'}" },
+ /* 2 */
+ { "threshold", CTRL_IN | GROUP, {DEFAULT_MID, 0, 1} },
+ { "strength", CTRL_IN, {DEFAULT_LOW, 0, 1} },
+ { "attack", CTRL_IN | GROUP, {DEFAULT_HIGH, 0, 1} },
+ { "release", CTRL_IN, {DEFAULT_MID, 0, 1} },
+ /* 6 */
+ { "gain (dB)", CTRL_IN | GROUP, {DEFAULT_MID, -12, 18} },
+ { "state (dB)", CONTROL|OUTPUT| GROUP, {DEFAULT_0,-144,0} },
+
+ { "in", INPUT | AUDIO, {BOUNDED, -1, 1} },
+ { "out", OUTPUT | AUDIO, {0} }
};
template <> void
Descriptor<Compress>::setup()
{
- UniqueID = 1772;
Label = "Compress";
- Properties = HARD_RT;
- Name = CAPS "Compress - Mono compressor";
- Maker = "Tim Goetze <tim@quitte.de>, Steve Harris <steve@plugin.org.uk>";
- Copyright = "GPL, 2004-7";
+ Name = CAPS "Compress - Compressor and saturating limiter";
+ Maker = "Tim Goetze <tim@quitte.de>";
+ Copyright = "2011-14";
+
+ /* fill port info and vtable */
+ autogen();
+}
+
+/* //////////////////////////////////////////////////////////////////////// */
+
+PortInfo
+CompressX2::port_info [] =
+{
+ { "measure", CTRL_IN, {INTEGER | DEFAULT_0, 0, 1},
+ "{0:'peak',1:'rms'}" },
+ { "mode", CTRL_IN | GROUP, {INTEGER | DEFAULT_1, 0, 2},
+ "{0:'linear',1:'saturating 2x',2:'saturating 4x',3:'saturating 4x128'}" },
+ { "threshold", CTRL_IN | GROUP, {DEFAULT_MID, 0, 1} },
+ { "strength", CTRL_IN, {DEFAULT_LOW, 0, 1} },
+ { "attack", CTRL_IN | GROUP, {DEFAULT_HIGH, 0, 1} },
+ { "release", CTRL_IN, {DEFAULT_MID, 0, 1} },
+ { "gain (dB)", CTRL_IN | GROUP, {DEFAULT_MID, -12, 18} },
+ { "state (dB)", CONTROL|OUTPUT| GROUP, {DEFAULT_0,-144,0} },
+ { "in.l", INPUT | AUDIO },
+ { "in.r", INPUT | AUDIO },
+ { "out.l", OUTPUT | AUDIO },
+ { "out.r", OUTPUT | AUDIO },
+};
+
+template <> void
+Descriptor<CompressX2>::setup()
+{
+ Label = "CompressX2";
+
+ Name = CAPS "CompressX2 - Stereo compressor and saturating limiter";
+ Maker = "Tim Goetze <tim@quitte.de>";
+ Copyright = "2011-14";
/* fill port info and vtable */
autogen();
diff --git a/plugins/LadspaEffect/caps/Compress.h b/plugins/LadspaEffect/caps/Compress.h
index 2bd722c7e..074dffb4b 100644
--- a/plugins/LadspaEffect/caps/Compress.h
+++ b/plugins/LadspaEffect/caps/Compress.h
@@ -1,17 +1,17 @@
/*
Compress.h
- Copyright 2004-5 Tim Goetze <tim@quitte.de>, Steve Harris
+ Copyright 2013 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- mono compressor.
+ Dynamic range processing.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,54 +25,118 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _COMPRESS_H_
-#define _COMPRESS_H_
+#ifndef COMPRESS_H
+#define COMPRESS_H
-#include "dsp/RMS.h"
#include "dsp/util.h"
-
-class Compress
-: public Plugin
+#include "dsp/Compress.h"
+#include "dsp/polynomials.h"
+#include "dsp/FIR.h"
+#include "dsp/sinc.h"
+#include "dsp/windows.h"
+
+template <int Over, int FIRSize>
+class CompSaturate
{
public:
- double fs;
- sample_t f;
+ /* antialias filters */
+ DSP::FIRUpsampler<FIRSize, Over> up;
+ DSP::FIRn<FIRSize> down;
- DSP::RMS rms;
- sample_t sum, amp, env, gain, gain_t;
+ public:
+ void init (double fs)
+ {
+ /* going a bit lower than nominal with fc */
+ double f = .7 * M_PI/Over;
+
+ DSP::sinc (f, up.c, FIRSize);
+ DSP::kaiser<DSP::apply_window> (up.c, FIRSize, 6.4);
+
+ /* copy upsampler filter kernel for downsampler, make sum */
+ double s = 0;
+ for (uint i = 0; i < FIRSize; ++i)
+ down.c[i] = up.c[i],
+ s += up.c[i];
+
+ s = 1/s;
- int count;
+ /* scale kernels for unity gain */
+ for (uint i=0; i<FIRSize; ++i)
+ down.c[i] *= s;
- template <sample_func_t F>
- void one_cycle (int frames);
+ s *= Over;
+ for (uint i=0; i<FIRSize; ++i)
+ up.c[i] *= s;
+ }
- public:
- static PortInfo port_info [];
- void init() {}
- void activate()
- {
- rms.reset();
+ sample_t process (sample_t x)
+ {
+ x = up.upsample (x);
+ x = DSP::Polynomial::tanh(x);
+ x = down.process (x);
- sum = 0;
- count = 0;
-
- amp = 0;
- env = 0;
+ for (int o = 1; o < Over; ++o)
+ down.store (DSP::Polynomial::atan1 (up.pad (o)));
- gain = 0;
- gain_t = 0;
+ return x;
}
+};
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
+typedef CompSaturate<2,32> CompSat2;
+typedef CompSaturate<4,64> CompSat4;
+typedef CompSaturate<4,128> CompSat8;
+
+template <int Channels>
+class CompressStub
+: public Plugin
+{
+ public:
+ uint remain;
+ struct {
+ DSP::CompressPeak peak;
+ DSP::CompressRMS rms;
+ } compress;
+
+ enum { Stereo = (Channels == 2) };
+
+ struct {
+ CompSat2 two;
+ CompSat4 four;
+ } saturate [Channels];
+
+ void cycle (uint frames);
+ template <class Comp>
+ void subcycle (uint frames, Comp & comp);
+ template <class Comp, class Sat>
+ void subsubcycle (uint frames, Comp & comp, Sat & satl, Sat & satr);
+
+ public:
+ static PortInfo port_info [];
+
+ void init()
{
- one_cycle<adding_func> (n);
+ for (int i=0; i < Channels; ++i)
+ {
+ saturate[i].two.init(fs);
+ saturate[i].four.init(fs);
+ }
}
+ void activate();
+};
+
+class Compress
+: public CompressStub<1>
+{
+ public:
+ static PortInfo port_info [];
+};
+
+class CompressX2
+: public CompressStub<2>
+{
+ public:
+ static PortInfo port_info [];
};
-#endif /* _COMPRESS_H_ */
+#endif /* COMPRESS_H */
diff --git a/plugins/LadspaEffect/caps/Descriptor.h b/plugins/LadspaEffect/caps/Descriptor.h
index 12c5d1c88..fb803ab35 100644
--- a/plugins/LadspaEffect/caps/Descriptor.h
+++ b/plugins/LadspaEffect/caps/Descriptor.h
@@ -1,7 +1,7 @@
/*
Descriptor.h
- Copyright 2004-10 Tim Goetze <tim@quitte.de>
+ Copyright 2004-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -17,7 +17,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -31,8 +31,8 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DESCRIPTOR_H_
-#define _DESCRIPTOR_H_
+#ifndef DESCRIPTOR_H
+#define DESCRIPTOR_H
#ifdef __SSE__
#include <xmmintrin.h>
@@ -41,6 +41,21 @@
#include <pmmintrin.h>
#endif
+/* not used */
+inline void
+processor_specific_denormal_measures()
+{
+ #ifdef __SSE3__
+ /* this one works reliably on a 6600 Core2 */
+ _MM_SET_DENORMALS_ZERO_MODE (_MM_DENORMALS_ZERO_ON);
+ #endif
+
+ #ifdef __SSE__
+ /* this one doesn't ... */
+ _MM_SET_FLUSH_ZERO_MODE (_MM_FLUSH_ZERO_ON);
+ #endif
+}
+
/* common stub for Descriptor makes it possible to delete() without special-
* casing for every plugin class.
*/
@@ -48,10 +63,7 @@ class DescriptorStub
: public LADSPA_Descriptor
{
public:
- DescriptorStub()
- {
- PortCount = 0;
- }
+ DescriptorStub() { memset(this,0,sizeof(*this)); }
~DescriptorStub()
{
@@ -64,20 +76,6 @@ class DescriptorStub
}
};
-inline void
-processor_specific_denormal_measures()
-{
- #ifdef __SSE3__
- /* this one works reliably on a 6600 Core2 */
- _MM_SET_DENORMALS_ZERO_MODE (_MM_DENORMALS_ZERO_ON);
- #endif
-
- #ifdef __SSE__
- /* this one doesn't ... */
- _MM_SET_FLUSH_ZERO_MODE (_MM_FLUSH_ZERO_ON);
- #endif
-}
-
template <class T>
class Descriptor
: public DescriptorStub
@@ -86,38 +84,43 @@ class Descriptor
LADSPA_PortRangeHint * ranges;
public:
- Descriptor() { setup(); }
- void setup();
+ Descriptor (uint id) { UniqueID = id; setup(); }
+ void setup();
+ /* setup() is in the plugin's .cc implementation file because it needs
+ * access to the port_info implementation, instantiating the following
+ * function: */
void autogen()
{
+ Properties = HARD_RT;
PortCount = (sizeof (T::port_info) / sizeof (PortInfo));
- /* unroll PortInfo members */
+ ImplementationData = T::port_info;
+
+ /* convert PortInfo members to Descriptor properties */
const char ** names = new const char * [PortCount];
- LADSPA_PortDescriptor * desc = new LADSPA_PortDescriptor [PortCount];
+ PortNames = names;
+
+ LADSPA_PortDescriptor * desc = new LADSPA_PortDescriptor[PortCount];
+ PortDescriptors = desc;
+
ranges = new LADSPA_PortRangeHint [PortCount];
+ PortRangeHints = ranges;
- /* could also assign directly but const_cast is ugly. */
for (int i = 0; i < (int) PortCount; ++i)
{
names[i] = T::port_info[i].name;
desc[i] = T::port_info[i].descriptor;
ranges[i] = T::port_info[i].range;
+ if (desc[i] & INPUT)
+ ranges[i].HintDescriptor |= BOUNDED;
}
-
- PortNames = names;
- PortDescriptors = desc;
- PortRangeHints = ranges;
- /* LADSPA_Descriptor vtable entries */
+ /* Descriptor vtable */
instantiate = _instantiate;
connect_port = _connect_port;
activate = _activate;
run = _run;
- run_adding = _run_adding;
- set_run_adding_gain = _set_run_adding_gain;
- deactivate = 0;
cleanup = _cleanup;
}
@@ -125,18 +128,18 @@ class Descriptor
const struct _LADSPA_Descriptor * d, ulong fs)
{
T * plugin = new T();
- int n = (int) d->PortCount;
LADSPA_PortRangeHint * ranges = ((Descriptor *) d)->ranges;
plugin->ranges = ranges;
+ int n = (int) d->PortCount;
plugin->ports = new sample_t * [n];
-
/* connect to lower bound as a safety measure */
for (int i = 0; i < n; ++i)
plugin->ports[i] = &(ranges[i].LowerBound);
plugin->fs = fs;
+ plugin->over_fs = 1./fs;
plugin->normal = NOISE_FLOOR;
plugin->init();
@@ -150,47 +153,31 @@ class Descriptor
static void _activate (LADSPA_Handle h)
{
- T * plugin = (T *) h;
+ Plugin * plugin = (Plugin *) h;
plugin->first_run = 1;
/* since none of the plugins do any RT-critical work in
- * activate(), it's safe to defer the actual call to the
- * plugin's activate() method for the first run() after
- * the host called in here.
+ * activate(), it's safe to defer the actual call into
+ * the first run() after the host called activate().
*
* It's the simplest way to prevent a parameter smoothing sweep
* in the first audio block after activation.
+ *
+ * While it would be preferable to set up the plugin's internal
+ * state from the current set of parameters, ladspa.h allows hosts to
+ * call activate() without even having connected the inputs, so that
+ * is out of the question.
plugin->activate();
*/
}
static void _run (LADSPA_Handle h, ulong n)
{
- T * plugin = (T *) h;
-
- /* We don't reset the processor flags later, it's true. */
- processor_specific_denormal_measures();
-
- /* If this is the first audio block after activation,
- * initialize the plugin from the current set of parameters. */
- if (plugin->first_run)
- {
- plugin->activate();
- plugin->first_run = 0;
- }
+ if (!n) return;
- plugin->run (n);
- plugin->normal = -plugin->normal;
- }
-
- static void _run_adding (LADSPA_Handle h, ulong n)
- {
T * plugin = (T *) h;
- /* We don't reset the processor flags later, it's true. */
- processor_specific_denormal_measures();
-
/* If this is the first audio block after activation,
* initialize the plugin from the current set of parameters. */
if (plugin->first_run)
@@ -199,24 +186,17 @@ class Descriptor
plugin->first_run = 0;
}
- plugin->run_adding (n);
+ plugin->cycle (n);
plugin->normal = -plugin->normal;
}
- static void _set_run_adding_gain (LADSPA_Handle h, LADSPA_Data g)
- {
- T * plugin = (T *) h;
-
- plugin->adding_gain = g;
- }
-
static void _cleanup (LADSPA_Handle h)
{
- T * plugin = (T *) h;
+ Plugin * plugin = (Plugin *) h;
delete [] plugin->ports;
delete plugin;
}
};
-#endif /* _DESCRIPTOR_H_ */
+#endif /* DESCRIPTOR_H */
diff --git a/plugins/LadspaEffect/caps/Eq.cc b/plugins/LadspaEffect/caps/Eq.cc
index dabd1b183..8d9ec729d 100644
--- a/plugins/LadspaEffect/caps/Eq.cc
+++ b/plugins/LadspaEffect/caps/Eq.cc
@@ -1,17 +1,17 @@
/*
Eq.cc
- Copyright 2002-7 Tim Goetze <tim@quitte.de>
+ Copyright 2002-14 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- 10-band octave-spread equalizer.
+ Various equaliser plugins.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,8 +25,10 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
+/* todo: Eq5p -- 20-400, 60-1k, 150-2.5k, 500-8k, 1k-20k */
+
#include "basics.h"
-#include <stdio.h>
+#include "dsp/Sine.h"
#include "Eq.h"
#include "Descriptor.h"
@@ -47,38 +49,35 @@ adjust_gain (int i, double g)
return g * adjust[i];
}
-#define Q 1.414
+#define Eq10Q 1.2
void
-Eq::init()
+Eq10::init()
{
- eq.init (fs, Q);
+ eq.init (fs, Eq10Q);
}
void
-Eq::activate()
+Eq10::activate()
{
for (int i = 0; i < 10; ++i)
{
- gain[i] = getport (1 + i);
- eq.gain[i] = adjust_gain (i, DSP::db2lin (gain[i]));
+ gain[i] = getport(i);
+ eq.gain[i] = adjust_gain (i, db2lin (gain[i]));
eq.gf[i] = 1;
}
}
-template <sample_func_t F>
void
-Eq::one_cycle (int frames)
+Eq10::cycle (uint frames)
{
- sample_t * s = ports[0];
-
/* evaluate band gain changes and compute recursion factor to prevent
* zipper noise */
double one_over_n = frames > 0 ? 1. / frames : 1;
for (int i = 0; i < 10; ++i)
{
- sample_t g = getport (1 + i);
+ sample_t g = getport(i);
if (g == gain[i])
{
/* no gain factoring */
@@ -87,17 +86,18 @@ Eq::one_cycle (int frames)
}
gain[i] = g;
- double want = adjust_gain (i, DSP::db2lin (g));
+ double want = adjust_gain (i, db2lin (g));
eq.gf[i] = pow (want / eq.gain[i], one_over_n);
}
+ sample_t * s = ports[10];
sample_t * d = ports[11];
- for (int i = 0; i < frames; ++i)
+ for (uint i = 0; i < frames; ++i)
{
sample_t x = s[i];
x = eq.process (x);
- F (d, i, x, adding_gain);
+ d[i] = x;
}
eq.normal = -normal;
@@ -107,69 +107,31 @@ Eq::one_cycle (int frames)
/* //////////////////////////////////////////////////////////////////////// */
PortInfo
-Eq::port_info [] =
+Eq10::port_info [] =
{
- {
- "in",
- INPUT | AUDIO,
- {0, -1, 1}
- }, {
- "31 Hz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "63 Hz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "125 Hz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "250 Hz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "500 Hz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "1 kHz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "2 kHz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "4 kHz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "8 kHz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "16 kHz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }
+ {"31 Hz", CTRL_IN, {DEFAULT_0, -48, 24}},
+ {"63 Hz", CTRL_IN, {DEFAULT_0, -48, 24}},
+ {"125 Hz", CTRL_IN | GROUP, {DEFAULT_0, -48, 24}},
+ {"250 Hz", CTRL_IN, {DEFAULT_0, -48, 24}},
+ {"500 Hz", CTRL_IN, {DEFAULT_0, -48, 24}},
+ {"1 kHz", CTRL_IN | GROUP, {DEFAULT_0, -48, 24}},
+ {"2 kHz", CTRL_IN, {DEFAULT_0, -48, 24}},
+ {"4 kHz", CTRL_IN, {DEFAULT_0, -48, 24}},
+ {"8 kHz", CTRL_IN | GROUP, {DEFAULT_0, -48, 24}},
+ {"16 kHz", CTRL_IN, {DEFAULT_0, -48, 24}},
+
+ {"in", INPUT | AUDIO},
+ {"out", OUTPUT | AUDIO}
};
template <> void
-Descriptor<Eq>::setup()
+Descriptor<Eq10>::setup()
{
- UniqueID = 1773;
- Label = "Eq";
- Properties = HARD_RT;
+ Label = "Eq10";
- Name = CAPS "Eq - 10-band equalizer";
+ Name = CAPS "Eq10 - 10-band equaliser";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ Copyright = "2004-13";
/* fill port info and vtable */
autogen();
@@ -178,31 +140,30 @@ Descriptor<Eq>::setup()
/* //////////////////////////////////////////////////////////////////////// */
void
-Eq2x2::init()
+Eq10X2::init()
{
for (int c = 0; c < 2; ++c)
- eq[c].init (fs, Q);
+ eq[c].init (fs, Eq10Q);
}
void
-Eq2x2::activate()
+Eq10X2::activate()
{
/* Fetch current parameter settings so we won't sweep band gains in the
* first block to process.
*/
for (int i = 0; i < 10; ++i)
{
- gain[i] = getport (2 + i);
- double a = adjust_gain (i, DSP::db2lin (gain[i]));
+ gain[i] = getport(i);
+ double a = adjust_gain (i, db2lin (gain[i]));
for (int c = 0; c < 2; ++c)
eq[c].gf[i] = 1,
eq[c].gain[i] = a;
}
}
-template <sample_func_t F>
void
-Eq2x2::one_cycle (int frames)
+Eq10X2::cycle (uint frames)
{
/* evaluate band gain changes and compute recursion factor to prevent
* zipper noise */
@@ -212,15 +173,15 @@ Eq2x2::one_cycle (int frames)
{
double a;
- if (*ports [2 + i] == gain[i])
+ if (*ports[i] == gain[i])
/* still same value, no gain fade */
a = 1;
else
{
- gain[i] = getport (2 + i);
+ gain[i] = getport(i);
/* prepare factor for logarithmic gain fade */
- a = adjust_gain (i, DSP::db2lin (gain[i]));
+ a = adjust_gain (i, db2lin (gain[i]));
a = pow (a / eq[0].gain[i], one_over_n);
}
@@ -231,14 +192,14 @@ Eq2x2::one_cycle (int frames)
for (int c = 0; c < 2; ++c)
{
sample_t
- * s = ports[c],
+ * s = ports[10 + c],
* d = ports[12 + c];
- for (int i = 0; i < frames; ++i)
+ for (uint i = 0; i < frames; ++i)
{
sample_t x = s[i];
x = eq[c].process (x);
- F (d, i, x, adding_gain);
+ d[i] = x;
}
}
@@ -251,77 +212,189 @@ Eq2x2::one_cycle (int frames)
}
PortInfo
-Eq2x2::port_info [] =
+Eq10X2::port_info [] =
+{
+ {"31 Hz", CTRL_IN, {DEFAULT_0, -48, 24}},
+ {"63 Hz", CTRL_IN, {DEFAULT_0, -48, 24}},
+ {"125 Hz", CTRL_IN | GROUP, {DEFAULT_0, -48, 24}},
+ {"250 Hz", CTRL_IN, {DEFAULT_0, -48, 24}},
+ {"500 Hz", CTRL_IN, {DEFAULT_0, -48, 24}},
+ {"1 kHz", CTRL_IN | GROUP, {DEFAULT_0, -48, 24}},
+ {"2 kHz", CTRL_IN, {DEFAULT_0, -48, 24}},
+ {"4 kHz", CTRL_IN, {DEFAULT_0, -48, 24}},
+ {"8 kHz", CTRL_IN | GROUP, {DEFAULT_0, -48, 24}},
+ {"16 kHz", CTRL_IN, {DEFAULT_0, -48, 24}},
+
+ {"in.l", INPUT | AUDIO},
+ {"in.r", INPUT | AUDIO},
+ {"out.l", OUTPUT | AUDIO},
+ {"out.r", OUTPUT | AUDIO}
+};
+
+template <> void
+Descriptor<Eq10X2>::setup()
+{
+ Label = "Eq10X2";
+
+ Name = CAPS "Eq10X2 - Stereo 10-band equaliser";
+ Maker = "Tim Goetze <tim@quitte.de>";
+ Copyright = "2004-13";
+
+ /* fill port info and vtable */
+ autogen();
+}
+
+/* //////////////////////////////////////////////////////////////////////// */
+
+void
+Eq4p::init()
+{
+ /* limit filter frequency to slightly under Nyquist to be on the safe side */
+ float limit = .48*fs;
+ for (int i = 0; i < 4; ++i)
+ {
+ state[i].f = -1; /* ensure all coefficients updated */
+ ranges[4*i + 1].UpperBound = min(ranges[4*i + 1].UpperBound, limit);
+ }
+}
+
+void
+Eq4p::activate()
+{
+ filter[0].reset();
+ filter[1].reset();
+
+ updatestate();
+ filter[0] = filter[1];
+ xfade = false;
+}
+
+typedef struct {sample_t a[3], b[3];} IIR2_ab;
+
+void
+Eq4p::updatestate()
{
+ for (int i=0; i<4; ++i)
{
- "in:l",
- INPUT | AUDIO,
- {0, -1, 1}
- }, {
- "in:r",
- INPUT | AUDIO,
- {0, -1, 1}
- }, {
- "31 Hz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "63 Hz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "125 Hz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "250 Hz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "500 Hz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "1 kHz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "2 kHz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "4 kHz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "8 kHz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "16 kHz",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -48, 24}
- }, {
- "out:l",
- OUTPUT | AUDIO,
- {0}
- }, {
- "out:r",
- OUTPUT | AUDIO,
- {0}
+ sample_t mode = getport(i*4);
+ sample_t f = getport(i*4 + 1);
+ sample_t Q = getport(i*4 + 2);
+ sample_t gain = getport(i*4 + 3);
+
+ if (mode==state[i].mode && gain==state[i].gain && f==state[i].f && Q==state[i].Q)
+ continue;
+
+ xfade = true;
+
+ state[i].mode = mode;
+ state[i].Q = Q;
+ state[i].f = f;
+ state[i].gain = gain;
+
+ IIR2_ab c;
+
+ f *= over_fs;
+ /* Zoelzer shelve: H(s) = (A*s^2 + s*(sqrt(A)/Q) + 1) / (s^2 + s/Q + 1) */
+ /* maxima: solve([a/(1-b*0)=.5,a/(1-b)=50,a/(1-b*x)=.707],[a,b,x]); */
+ Q = .5/(1-.99*Q);
+ if (mode < 0)
+ c.a[0]=1,c.a[1]=0,c.a[2]=0,c.b[1]=0,c.b[2]=0; /* off = identity filter */
+ else if (mode < 0.5)
+ DSP::RBJ::LoShelve (f,Q,gain,c);
+ else if (mode < 1.5)
+ DSP::RBJ::PeakingEQ (f,Q,gain,c);
+ else /* if (mode < 2.5) */
+ DSP::RBJ::HiShelve (f,Q,gain,c);
+
+ filter[1].set_ab (i, c.a, c.b);
}
+}
+
+void
+Eq4p::cycle (uint frames)
+{
+ *ports[16] = 3; /* _latency */
+
+ sample_t * s = ports[17];
+ sample_t * d = ports[18];
+
+ updatestate();
+
+ if (!xfade)
+ {
+ for (uint i = 0; i < frames; ++i)
+ {
+ sample_t x = s[i] + normal;
+ x = filter[0].seriesprocess(x);
+ d[i] = x;
+ }
+ }
+ else
+ {
+ float over_n = frames ? 1./frames : 1;
+ DSP::Sine gf0 (.5*M_PI*over_n,.5*M_PI);
+ DSP::Sine gf1 (.5*M_PI*over_n,0);
+ for (uint i = 0; i < frames; ++i)
+ {
+ sample_t x = s[i];
+ sample_t g0 = gf0.get();
+ sample_t g1 = gf1.get();
+ x = g0*g0*filter[0].seriesprocess(x) + g1*g1*filter[1].seriesprocess(x);
+ d[i] = x;
+ }
+
+ filter[0] = filter[1];
+ filter[1].reset();
+ xfade = false;
+ }
+}
+
+/* //////////////////////////////////////////////////////////////////////// */
+
+static const char * Eq4pBandModes = "{-1:'off',0:'lowshelve',1:'band',2:'hishelve'}";
+
+PortInfo
+Eq4p::port_info [] =
+{
+ {"a.mode", CTRL_IN, {DEFAULT_0 | INTEGER, -1,2}, Eq4pBandModes},
+ {"a.f (Hz)", CTRL_IN, {DEFAULT_LOW | LOG, 20, 14000}},
+ {"a.Q", CTRL_IN, {DEFAULT_LOW, 0, 1}},
+ {"a.gain (dB)", CTRL_IN, {DEFAULT_0, -48, 24}},
+
+ /* 4 */
+ {"b.mode", CTRL_IN | GROUP, {DEFAULT_1 | INTEGER, -1,2}, Eq4pBandModes},
+ {"b.f (Hz)", CTRL_IN, {DEFAULT_MID | LOG, 20, 14000}},
+ {"b.Q", CTRL_IN, {DEFAULT_MID, 0, 1}},
+ {"b.gain (dB)", CTRL_IN, {DEFAULT_0, -48, 24}},
+
+ /* 8 */
+ {"c.mode", CTRL_IN | GROUP, {DEFAULT_1 | INTEGER, -1,2}, Eq4pBandModes},
+ {"c.f (Hz)", CTRL_IN, {DEFAULT_MID | LOG, 20, 14000}},
+ {"c.Q", CTRL_IN, {DEFAULT_LOW, 0, 1}},
+ {"c.gain (dB)", CTRL_IN, {DEFAULT_0, -48, 24}},
+
+ /* 12 */
+ {"d.mode", CTRL_IN | GROUP, {DEFAULT_MAX | INTEGER, -1,2}, Eq4pBandModes},
+ {"d.f (Hz)", CTRL_IN, {DEFAULT_HIGH | LOG, 20, 14000}},
+ {"d.Q", CTRL_IN, {DEFAULT_LOW, 0, 1}},
+ {"d.gain (dB)", CTRL_IN, {DEFAULT_0, -48, 24}},
+
+ /* 16 */
+ {"_latency", OUTPUT|CONTROL|GROUP, {INTEGER|DEFAULT_MAX,3,3}, "{3:'3 samples'}"},
+
+ /* 17 */
+ {"in", INPUT | AUDIO, {0, -1, 1}},
+ {"out", OUTPUT | AUDIO, {0}}
};
template <> void
-Descriptor<Eq2x2>::setup()
+Descriptor<Eq4p>::setup()
{
- UniqueID = 2594;
- Label = "Eq2x2";
- Properties = HARD_RT;
+ Label = "Eq4p";
- Name = CAPS "Eq2x2 - stereo 10-band equalizer";
+ Name = CAPS "Eq4p - 4-band parametric shelving equaliser";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ Copyright = "2013-14";
/* fill port info and vtable */
autogen();
@@ -329,7 +402,154 @@ Descriptor<Eq2x2>::setup()
/* //////////////////////////////////////////////////////////////////////// */
-/*
- todo: parametric -- 20-400, 60-1k, 150-2.5k, 500-8k, 1k-20k
- bandwidth 0-2 octaves
- */
+void
+EqFA4p::init()
+{
+ /* limit filter frequency to slightly under Nyquist to be on the safe side */
+ float limit = .48*fs;
+ for (int i = 0; i < 4; ++i)
+ {
+ state[i].f = -1; /* ensure all coefficients updated */
+ ranges[4*i + 1].UpperBound = min(ranges[4*i + 1].UpperBound, limit);
+ }
+}
+
+void
+EqFA4p::activate()
+{
+ filter[0].reset();
+ filter[1].reset();
+
+ updatestate();
+ filter[0] = filter[1];
+ xfade = false;
+ gain = db2lin(getport(16));
+}
+
+void
+EqFA4p::updatestate()
+{
+ for (int i=0; i<4; ++i)
+ {
+ sample_t mode = getport(i*4 + 0);
+ sample_t f = getport(i*4 + 1);
+ sample_t bw = getport(i*4 + 2);
+ sample_t gain = getport(i*4 + 3);
+
+ if (mode==state[i].mode && gain==state[i].gain && f==state[i].f && bw==state[i].bw)
+ continue;
+
+ xfade = true;
+
+ state[i].mode = mode;
+ state[i].bw = bw;
+ state[i].f = f;
+ state[i].gain = gain;
+
+ if (!mode)
+ filter[1].unity(i);
+ else
+ filter[1].set(i, f*over_fs, bw, db2lin(gain));
+ }
+}
+
+void
+EqFA4p::cycle (uint frames)
+{
+ updatestate();
+
+ sample_t g = db2lin(getport(16));
+ sample_t over_n = frames ? 1./frames : 1;
+ sample_t gf = pow(g/gain, over_n);
+
+ *ports[17] = 3; /* _latency */
+ sample_t * s = ports[18];
+ sample_t * d = ports[19];
+
+ if (!xfade)
+ {
+ for (uint i = 0; i < frames; ++i)
+ {
+ sample_t x = s[i];
+ x = filter[0].seriesprocess(x);
+ x = gain*x;
+ gain *= gf;
+ d[i] = x;
+ }
+ }
+ else
+ {
+ DSP::Sine gf0 (.5*M_PI*over_n,.5*M_PI);
+ DSP::Sine gf1 (.5*M_PI*over_n,0);
+ for (uint i = 0; i < frames; ++i)
+ {
+ sample_t x = s[i];
+ sample_t g0 = gf0.get();
+ sample_t g1 = gf1.get();
+ x = g0*g0*filter[0].seriesprocess(x) + g1*g1*filter[1].seriesprocess(x);
+ x = gain*x;
+ gain *= gf;
+ d[i] = x;
+ }
+
+ filter[0] = filter[1];
+ filter[1].reset();
+ xfade = false;
+ }
+}
+
+/* //////////////////////////////////////////////////////////////////////// */
+
+static const char * EqFA4pBandModes = "{0:'off',1:'on'}";
+
+PortInfo
+EqFA4p::port_info [] =
+{
+ {"a.act", CTRL_IN | GROUP, {DEFAULT_0 | INTEGER, 0,1}, EqFA4pBandModes},
+ {"a.f (Hz)", CTRL_IN, {DEFAULT_LOW | LOG, 20, 14000}},
+ {"a.bw", CTRL_IN, {DEFAULT_1, .125, 8}},
+ {"a.gain (dB)", CTRL_IN, {DEFAULT_0, -24, 24}},
+
+ /* 4 */
+ {"b.act", CTRL_IN | GROUP, {DEFAULT_0 | INTEGER, 0,1}, EqFA4pBandModes},
+ {"b.f (Hz)", CTRL_IN, {DEFAULT_MID | LOG, 20, 14000}},
+ {"b.bw", CTRL_IN, {DEFAULT_1, .125, 8}},
+ {"b.gain (dB)", CTRL_IN, {DEFAULT_0, -24, 24}},
+
+ /* 8 */
+ {"c.act", CTRL_IN | GROUP, {DEFAULT_0 | INTEGER, 0,1}, EqFA4pBandModes},
+ {"c.f (Hz)", CTRL_IN, {DEFAULT_MID | LOG, 20, 14000}},
+ {"c.bw", CTRL_IN, {DEFAULT_1, .125, 8}},
+ {"c.gain (dB)", CTRL_IN, {DEFAULT_0, -24, 24}},
+
+ /* 12 */
+ {"d.act", CTRL_IN | GROUP, {DEFAULT_0 | INTEGER, 0,1}, EqFA4pBandModes},
+ {"d.f (Hz)", CTRL_IN, {DEFAULT_HIGH | LOG, 20, 14000}},
+ {"d.bw", CTRL_IN, {DEFAULT_1, .125, 8}},
+ {"d.gain (dB)", CTRL_IN, {DEFAULT_0, -24, 24}},
+
+ /* 16 */
+ {"gain", CTRL_IN|GROUP, {DEFAULT_0, -24, 24}},
+
+ /* 17 */
+ {"_latency", OUTPUT|CONTROL|GROUP, {INTEGER|DEFAULT_MAX,3,3}, "{3:'3 samples'}"},
+
+ /* 18 */
+ {"in", INPUT | AUDIO},
+ {"out", OUTPUT | AUDIO}
+};
+
+template <> void
+Descriptor<EqFA4p>::setup()
+{
+ Label = "EqFA4p";
+
+ Name = CAPS "EqFA4p - 4-band parametric eq";
+ Maker = "Tim Goetze <tim@quitte.de>";
+ Copyright = "2014";
+
+ /* fill port info and vtable */
+ autogen();
+}
+
+
diff --git a/plugins/LadspaEffect/caps/Eq.h b/plugins/LadspaEffect/caps/Eq.h
index 3532b958f..57666c548 100644
--- a/plugins/LadspaEffect/caps/Eq.h
+++ b/plugins/LadspaEffect/caps/Eq.h
@@ -1,17 +1,17 @@
/*
Eq.h
- Copyright 2004-5 Tim Goetze <tim@quitte.de>
+ Copyright 2004-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- equalizer circuit using recursive filtering.
+ IIR equalisation filters.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,15 +25,18 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _EQ_H_
-#define _EQ_H_
+#ifndef EQ_H
+#define EQ_H
#include "dsp/util.h"
#include "dsp/Eq.h"
-#include "dsp/BiQuad.h"
+#include "dsp/IIR2.h"
#include "dsp/RBJ.h"
+#include "dsp/v4f.h"
+#include "dsp/v4f_IIR2.h"
-class Eq
+/* octave-band variants, mono and stereo */
+class Eq10
: public Plugin
{
public:
@@ -43,51 +46,72 @@ class Eq
int block;
enum { BlockSize = 64 };
- template <sample_func_t F>
- void one_cycle (int frames);
+ void cycle (uint frames);
public:
static PortInfo port_info [];
void init();
void activate();
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
};
-class Eq2x2
+class Eq10X2
: public Plugin
{
public:
sample_t gain[10];
DSP::Eq<10> eq[2];
- template <sample_func_t F>
- void one_cycle (int frames);
+ void cycle (uint frames);
+
+ public:
+ static PortInfo port_info [];
+
+ void init();
+ void activate();
+};
+
+/* 4-way parametric, parallel implementation */
+class Eq4p
+: public Plugin
+{
+ public:
+ struct {sample_t mode,gain,f,Q;} state[4]; /* parameters */
+
+ DSP::IIR2v4 filter[2];
+
+ bool xfade;
+ void updatestate();
+
+ void cycle (uint frames);
public:
static PortInfo port_info [];
void init();
void activate();
+};
+
+/* 4-way parametric, parallel implementation */
+class EqFA4p
+: public Plugin
+{
+ public:
+ struct {sample_t mode,gain,f,bw;} state[4]; /* parameters */
+
+ DSP::MREqv4 filter[2];
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
+ bool xfade;
+ void updatestate();
+ sample_t gain;
+
+ void cycle (uint frames);
+
+ public:
+ static PortInfo port_info [];
+
+ void init();
+ void activate();
};
-#endif /* _EQ_H_ */
+#endif /* EQ_H */
diff --git a/plugins/LadspaEffect/caps/Pan.cc b/plugins/LadspaEffect/caps/Pan.cc
index 0449f616b..b83cf1372 100644
--- a/plugins/LadspaEffect/caps/Pan.cc
+++ b/plugins/LadspaEffect/caps/Pan.cc
@@ -5,14 +5,14 @@
http://quitte.de/dsp/
- panorama with width control,
- stereo image width reduction
+ Stereo image synthesis (inspired by the Orban 245F unit)
+ and manipulation.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -27,134 +27,92 @@
*/
#include "basics.h"
+#include "dsp/RBJ.h"
#include "Pan.h"
#include "Descriptor.h"
void
-Pan::init()
+Wider::init()
{
- delay.init ((int) (.040 * fs));
+ pan = FLT_MAX; /* invalid setting to make sure gain is updated */
}
void
-Pan::activate()
+Wider::activate()
{
- delay.reset();
- tap.reset (400 / fs);
- set_pan (getport (1));
+ set_pan (getport(1));
+ float fc[3] = { 150, 900, 5000 };
+ for (int i = 0; i < 3; ++i)
+ DSP::RBJ::AllPass (fc[i]*over_fs, .707, ap[i]);
}
inline void
-Pan::set_pan (sample_t p)
+Wider::set_pan (sample_t p)
{
- pan = p;
-
- double phi = (pan + 1) * M_PI * .25;
+ if (pan == p) return;
- gain_l = cos (phi);
- gain_r = sin (phi);
+ pan = p;
+ double phi = (pan + 1)*M_PI*.25;
+ gain_l = cos(phi);
+ gain_r = sin(phi);
}
-template <sample_func_t F>
void
-Pan::one_cycle (int frames)
+Wider::cycle (uint frames)
{
- sample_t * s = ports[0];
-
- if (pan != *ports[1])
- set_pan (getport(1));
-
- sample_t g = getport(2);
- sample_t
- width_l = g * gain_r,
- width_r = g * gain_l;
-
- tap.t = (int) (getport(3) * fs * .001);
-
- bool mono = getport(4);
-
- sample_t * dl = ports[5];
- sample_t * dr = ports[6];
-
- sample_t x, xt;
+ sample_t p = getport(0);
+ if (p != pan) set_pan (p);
- if (mono) for (int i = 0; i < frames; ++i)
- {
- x = s[i];
-
- xt = tap.get (delay);
+ sample_t width = getport(1);
- delay.put (x + normal);
+ /* need to limit width as pan increases in order to prevent
+ * excessive phase cancellation */
+ width *= 1 - fabs (p);
+ width *= width;
- x = (gain_l * x + gain_r * x + width_l * xt + width_r * xt) * .5;
+ sample_t * src = ports[2];
+ sample_t * dl = ports[3];
+ sample_t * dr = ports[4];
- F (dl, i, x, adding_gain);
- F (dr, i, x, adding_gain);
-
- normal = -normal;
- }
- else /* stereo */ for (int i = 0; i < frames; ++i)
+ for (uint i = 0; i < frames; ++i)
{
- x = s[i];
-
- xt = tap.get (delay);
-
- delay.put (x + normal);
-
- F (dl, i, gain_l * x + width_l * xt, adding_gain);
- F (dr, i, gain_r * x + width_r * xt, adding_gain);
-
- normal = -normal;
+ sample_t m = .707*src[i] + normal;
+ sample_t s = m;
+ s = ap[0].process(s);
+ s = ap[1].process(s);
+ s = ap[2].process(s);
+
+ s *= width;
+
+ sample_t l = m - s;
+ sample_t r = m + s;
+ dl[i] = gain_l*l;
+ dr[i] = gain_r*r;
}
}
/* //////////////////////////////////////////////////////////////////////// */
PortInfo
-Pan::port_info [] =
+Wider::port_info [] =
{
- {
- "in",
- INPUT | AUDIO,
- {0}
- }, {
- "pan",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, -1, 1}
- }, {
- "width",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0, 0, 1}
- }, {
- "t",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0.1, 40}
- }, {
- "mono",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0 | INTEGER | TOGGLE, 0, 1}
- }, {
- "out:l",
- OUTPUT | AUDIO,
- {0}
- }, {
- "out:r",
- OUTPUT | AUDIO,
- {0}
- }
+ { "pan", INPUT | CONTROL, {DEFAULT_0, -1, 1} },
+ { "width", INPUT | CONTROL, {DEFAULT_1, 0, 1} },
+
+ { "in", INPUT | AUDIO },
+ { "out.l", OUTPUT | AUDIO },
+ { "out.r", OUTPUT | AUDIO }
};
template <> void
-Descriptor<Pan>::setup()
+Descriptor<Wider>::setup()
{
- UniqueID = 1788;
- Label = "Pan";
- Properties = HARD_RT;
+ Label = "Wider";
- Name = CAPS "Pan - Pan and width";
+ Name = CAPS "Wider - Stereo image synthesis";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ Copyright = "2011-13";
/* fill port info and vtable */
autogen();
@@ -163,46 +121,55 @@ Descriptor<Pan>::setup()
/* //////////////////////////////////////////////////////////////////////// */
void
-Narrower::init()
-{
-}
-
-void
-Narrower::activate()
-{
-}
-
-template <sample_func_t F>
-void
-Narrower::one_cycle (int frames)
+Narrower::cycle (uint frames)
{
- sample_t * sl = ports[0];
- sample_t * sr = ports[1];
-
- if (strength != *ports[2])
- strength = *ports[2];
-
- sample_t * dl = ports[3];
- sample_t * dr = ports[4];
+ float mode = getport(0);
+ strength = getport(1);
- double xl, xr, m;
- double dry = 1 - strength, wet = strength;
+ sample_t * sl = ports[2];
+ sample_t * sr = ports[3];
+ sample_t * dl = ports[4];
+ sample_t * dr = ports[5];
- for (int i = 0; i < frames; ++i)
+ if (mode)
{
- xl = sl[i];
- xr = sr[i];
-
- m = wet * (xl + xr) * .5;
-
- xl *= dry;
- xr *= dry;
-
- xl += m;
- xr += m;
-
- F (dl, i, xl, adding_gain);
- F (dr, i, xr, adding_gain);
+ for (uint i = 0; i < frames; ++i)
+ {
+ sample_t xl = sl[i];
+ sample_t xr = sr[i];
+
+ sample_t m = xl + xr, s = xl - xr;
+
+ m += strength * s;
+ s *= (1 - strength);
+
+ xl = .5 * (m + s);
+ xr = .5 * (m - s);
+
+ dl[i] = xl;
+ dr[i] = xr;
+ }
+ }
+ else
+ {
+ sample_t dry = 1 - strength, wet = strength;
+
+ for (uint i = 0; i < frames; ++i)
+ {
+ sample_t xl = sl[i];
+ sample_t xr = sr[i];
+
+ sample_t m = wet * (xl + xr) * .5;
+
+ xl *= dry;
+ xr *= dry;
+
+ xl += m;
+ xr += m;
+
+ dl[i] = xl;
+ dr[i] = xr;
+ }
}
}
@@ -211,39 +178,24 @@ Narrower::one_cycle (int frames)
PortInfo
Narrower::port_info [] =
{
- {
- "in:l",
- INPUT | AUDIO,
- {0}
- }, {
- "in:r",
- INPUT | AUDIO,
- {0}
- }, {
- "strength",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 1}
- }, {
- "out:l",
- OUTPUT | AUDIO,
- {0}
- }, {
- "out:r",
- OUTPUT | AUDIO,
- {0}
- }
+ { "mode", INPUT | CONTROL, {INTEGER | DEFAULT_0, 0, 1},
+ "{0:'crossfeed mixing',1:'mid/side processing',}" },
+ { "strength", INPUT | CONTROL | GROUP, {DEFAULT_LOW, 0, 1} },
+
+ { "in.l", INPUT | AUDIO },
+ { "in.r", INPUT | AUDIO },
+ { "out.l", OUTPUT | AUDIO },
+ { "out.r", OUTPUT | AUDIO }
};
template <> void
Descriptor<Narrower>::setup()
{
- UniqueID = 2595;
Label = "Narrower";
- Properties = HARD_RT;
Name = CAPS "Narrower - Stereo image width reduction";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2011";
+ Copyright = "2011-12";
/* fill port info and vtable */
autogen();
diff --git a/plugins/LadspaEffect/caps/Pan.h b/plugins/LadspaEffect/caps/Pan.h
index 469ccb192..d2191b4f7 100644
--- a/plugins/LadspaEffect/caps/Pan.h
+++ b/plugins/LadspaEffect/caps/Pan.h
@@ -5,14 +5,14 @@
http://quitte.de/dsp/
- panorama with width control,
+ Panorama with width synthesis,
stereo image width reduction
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -26,31 +26,13 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _PAN_H_
-#define _PAN_H_
+#ifndef PAN_H
+#define PAN_H
#include "dsp/Delay.h"
-#include "dsp/OnePole.h"
+#include "dsp/IIR2.h"
-class PanTap
-{
- public:
- int t;
- DSP::OnePoleLP damper;
-
- sample_t get (DSP::Delay & delay)
- {
- return damper.process (delay[t]);
- }
-
- void reset (double c)
- {
- damper.set_f (c);
- damper.reset();
- }
-};
-
-class Pan
+class Wider
: public Plugin
{
public:
@@ -58,11 +40,9 @@ class Pan
sample_t gain_l, gain_r;
- DSP::Delay delay;
- PanTap tap;
+ DSP::IIR2<sample_t> ap[3];
- template <sample_func_t F>
- void one_cycle (int frames);
+ void cycle (uint frames);
inline void set_pan (sample_t);
@@ -71,16 +51,6 @@ class Pan
void init();
void activate();
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
};
/* stereo width reduction */
@@ -90,25 +60,13 @@ class Narrower
public:
sample_t strength;
- template <sample_func_t F>
- void one_cycle (int frames);
+ void cycle (uint frames);
public:
static PortInfo port_info [];
- void init();
- void activate();
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
-
+ void init() {}
+ void activate() {}
};
-#endif /* _PAN_H_ */
+#endif /* PAN_H */
diff --git a/plugins/LadspaEffect/caps/Phaser.cc b/plugins/LadspaEffect/caps/Phaser.cc
index b0116b448..d50c2b079 100644
--- a/plugins/LadspaEffect/caps/Phaser.cc
+++ b/plugins/LadspaEffect/caps/Phaser.cc
@@ -1,20 +1,17 @@
/*
Phaser.cc
- Copyright 2002-7 Tim Goetze <tim@quitte.de>
+ Copyright 2002-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- One simple mono phaser, 6 all-pass lines, the usual controls.
-
- Another unit in the same vein with the filter modulation controlled by
- a Lorenz fractal.
+ Phaser effect
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -33,146 +30,56 @@
#include "Phaser.h"
#include "Descriptor.h"
-template <sample_func_t F>
-void
-PhaserI::one_cycle (int frames)
-{
- sample_t * s = ports[0];
-
- if (rate != *ports[1])
- {
- rate = getport(1);
- lfo.set_f (max (.001, rate * (double) blocksize), fs, lfo.get_phase());
- }
-
- double depth = getport(2);
- double spread = 1 + getport(3);
- double fb = getport(4);
-
- sample_t * dst = ports[5];
-
- while (frames)
- {
- if (remain == 0) remain = 32;
-
- int n = min (remain, frames);
-
- double d = delay.bottom + delay.range * (1. - fabs (lfo.get()));
-
- for (int j = 5; j >= 0; --j)
- {
- ap[j].set (d);
- d *= spread;
- }
-
- for (int i = 0; i < n; ++i)
- {
- sample_t x = s[i];
- sample_t y = x + y0 * fb + normal;
-
- for (int j = 5; j >= 0; --j)
- y = ap[j].process (y);
-
- y0 = y;
-
- F (dst, i, x + y * depth, adding_gain);
- }
-
- s += n;
- dst += n;
- frames -= n;
- remain -= n;
- }
-}
-
-/* //////////////////////////////////////////////////////////////////////// */
+#define FBSCALE .9 /* feedback scale from 0..1 parameter range */
-PortInfo
-PhaserI::port_info [] =
-{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "rate (Hz)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_1, 0, 10}
- }, {
- "depth",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_HIGH, 0, 1}
- }, {
- "spread",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, M_PI}
- }, {
- "feedback",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_HIGH, 0, .999}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }
-};
-
-template <> void
-Descriptor<PhaserI>::setup()
-{
- UniqueID = 1775;
- Label = "PhaserI";
- Properties = HARD_RT;
-
- Name = CAPS "PhaserI - Mono phaser";
- Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2002-7";
-
- /* fill port info and vtable */
- autogen();
-}
-
-/* //////////////////////////////////////////////////////////////////////// */
-
-template <sample_func_t F>
void
-PhaserII::one_cycle (int frames)
+PhaserII::cycle (uint frames)
{
- sample_t * s = ports[0];
+ sample_t * s = ports[5];
+ sample_t * dst = ports[6];
- lorenz.set_rate (getport(1) * .08);
+ rate = getport(0);
+ lfo.sine.set_f (max(.001,rate*blocksize), fs, lfo.sine.get_phase());
+ lfo.lp.set_f (5*(1+rate)*over_fs);
+ lfo.roessler.set_rate(.05*rate);
- double depth = getport(2);
- double spread = 1 + getport(3);
- double fb = getport(4);
+ float mode = getport(1);
- sample_t * dst = ports[5];
+ sample_t depth = getport(2);
+ sample_t spread = 1 + .5*M_PI*getport(3);
+ sample_t fb = FBSCALE * getport(4);
while (frames)
{
- if (remain == 0) remain = 32;
+ if (remain == 0) remain = blocksize;
int n = min (remain, frames);
- double d = delay.bottom + delay.range * (.3 * lorenz.get());
+ float d;
+ if (mode < .5)
+ d = fabs (lfo.sine.get()),
+ d *= d;
+ else
+ d = min (.99,fabs (lfo.lp.process(4.3*lfo.roessler.get())));
+ d = delay.bottom + d*delay.range;
- for (int j = 5; j >= 0; --j)
+ for (int j = 0; j < Notches; ++j)
{
- ap[j].set (d);
+ ap[j].set(d);
d *= spread;
}
for (int i = 0; i < n; ++i)
{
- sample_t x = s[i];
+ sample_t x = .5 * s[i];
sample_t y = x + y0 * fb + normal;
- for (int j = 5; j >= 0; --j)
+ for (int j = 0; j < Notches; ++j)
y = ap[j].process (y);
y0 = y;
- F (dst, i, x + y * depth, adding_gain);
+ dst[i] = x + y*depth;
}
s += n;
@@ -187,43 +94,25 @@ PhaserII::one_cycle (int frames)
PortInfo
PhaserII::port_info [] =
{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "rate",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 1}
- }, {
- "depth",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_HIGH, 0, 1}
- }, {
- "spread",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, M_PI * .5}
- }, {
- "feedback",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_HIGH, 0, .999}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }
+ { "rate", CTRL_IN, {DEFAULT_LOW, 0, 1} },
+ { "lfo", CTRL_IN, {DEFAULT_0 | INTEGER, 0, 1}, "{0:'sine',1:'fractal'}"},
+
+ { "depth", CTRL_IN | GROUP, {DEFAULT_HIGH, 0, 1} },
+ { "spread", CTRL_IN, {DEFAULT_HIGH, 0, 1} },
+ { "resonance", CTRL_IN, {DEFAULT_LOW, 0, 1} },
+
+ { "in", INPUT | AUDIO },
+ { "out", OUTPUT | AUDIO }
};
template <> void
Descriptor<PhaserII>::setup()
{
- UniqueID = 2586;
Label = "PhaserII";
- Properties = HARD_RT;
- Name = CAPS "PhaserII - Mono phaser modulated by a Lorenz fractal";
+ Name = CAPS "PhaserII - Mono phaser";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2002-7";
+ Copyright = "2002-13";
/* fill port info and vtable */
autogen();
diff --git a/plugins/LadspaEffect/caps/Phaser.h b/plugins/LadspaEffect/caps/Phaser.h
index c79b93053..52a2ec3a1 100644
--- a/plugins/LadspaEffect/caps/Phaser.h
+++ b/plugins/LadspaEffect/caps/Phaser.h
@@ -1,17 +1,17 @@
/*
Phaser.h
- Copyright 2002-5 Tim Goetze <tim@quitte.de>
+ Copyright 2002-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- Standard and fractal-modulated phaser units.
+ Phaser effect plugin.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,97 +25,43 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _PHASER_H_
-#define _PHASER_H_
+#ifndef PHASER_H
+#define PHASER_H
#include "dsp/Sine.h"
-#include "dsp/Lorenz.h"
+#include "dsp/Roessler.h"
#include "dsp/Delay.h"
+#include "dsp/IIR1.h"
-/* all-pass as used by the phaser. */
class PhaserAP
{
public:
sample_t a, m;
- PhaserAP()
- {
- a = m = 0.;
- }
+ PhaserAP() { a=m=0.; }
- void set (double delay)
- {
- a = (1 - delay) / (1 + delay);
- }
+ void set (sample_t delay)
+ { a = (1-delay) / (1+delay); }
sample_t process (sample_t x)
- {
- register sample_t y = -a * x + m;
- m = a * y + x;
-
- return y;
- }
-};
-
-class PhaserI
-: public Plugin
-{
- public:
- PhaserAP ap[6];
- DSP::Sine lfo;
-
- sample_t rate;
- sample_t y0;
-
- struct {
- double bottom, range;
- } delay;
-
- template <sample_func_t>
- void one_cycle (int frames);
-
- int blocksize, remain;
-
- public:
- static PortInfo port_info [];
-
- void init()
- {
- blocksize = 32;
- }
-
- void activate()
- {
- y0 = 0.;
- remain = 0;
-
- delay.bottom = 400. / fs;
- delay.range = 2200. / fs;
-
- rate = -1; /* force lfo reset in one_cycle() */
- }
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
{
- one_cycle<adding_func> (n);
+ sample_t y = -a*x + m;
+ m = a*y + x;
+ return y;
}
};
-/* same as above, but filter sweep is controlled by a Lorenz fractal */
-
class PhaserII
: public Plugin
{
public:
- double fs;
-
- PhaserAP ap[6];
- DSP::Lorenz lorenz;
+ enum {Notches = 12};
+ PhaserAP ap[Notches];
+ struct {
+ DSP::Sine sine;
+ DSP::Roessler roessler;
+ DSP::LP1<sample_t> lp;
+ } lfo;
sample_t rate;
sample_t y0;
@@ -124,18 +70,22 @@ class PhaserII
double bottom, range;
} delay;
- template <sample_func_t>
- void one_cycle (int frames);
+ void cycle (uint frames);
- int blocksize, remain;
+ uint blocksize, remain;
public:
static PortInfo port_info [];
void init()
{
- blocksize = 32;
- lorenz.init();
+ blocksize = 16;
+ if (fs > 32000) blocksize *= 2;
+ if (fs > 64000) blocksize *= 2;
+ if (fs > 128000) blocksize *= 2;
+
+ lfo.roessler.init();
+ lfo.sine.set_f (300*over_fs,0);
}
void activate()
@@ -143,22 +93,9 @@ class PhaserII
y0 = 0.;
remain = 0;
- delay.bottom = 400. / fs;
- delay.range = 2200. / fs;
-
- rate = -1; /* force lfo reset in one_cycle() */
- }
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
+ delay.bottom = 400*over_fs;
+ delay.range = 2200*over_fs;
}
};
-
-#endif /* _PHASER_H_ */
+#endif /* PHASER_H */
diff --git a/plugins/LadspaEffect/caps/README b/plugins/LadspaEffect/caps/README
index 58fc72ec8..78a2d6af0 100644
--- a/plugins/LadspaEffect/caps/README
+++ b/plugins/LadspaEffect/caps/README
@@ -1,7 +1,10 @@
-This is CAPS, the C Audio Plugin Suite
-======================================
+This is CAPS, the C* Audio Plugin Suite
+=======================================
-For all questions, please be referred to the HTML documentation of
-this software package in the file 'caps.html', also available from
+For all questions, please be referred to the documentation at
-http://quitte.de/dsp/caps.html
+ http://quitte.de/dsp/caps.html
+
+If that doesn't help, you can contact
+
+ tim@quitte.de
diff --git a/plugins/LadspaEffect/caps/Reverb.cc b/plugins/LadspaEffect/caps/Reverb.cc
index 9061daaf5..2a57275d8 100644
--- a/plugins/LadspaEffect/caps/Reverb.cc
+++ b/plugins/LadspaEffect/caps/Reverb.cc
@@ -1,15 +1,15 @@
/*
Reverb.cc
- Copyright 2002-7 Tim Goetze <tim@quitte.de>
+ Copyright 2002-14 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- three reverb units: JVRev, Plate and Plate2x2.
+ Three reverb units: JVRev, Plate and PlateX2.
- the former is a rewrite of STK's JVRev, a traditional design.
+ The former is a rewrite of STK's JVRev, a traditional design.
- original comment:
+ Original comment:
This is based on some of the famous
Stanford CCRMA reverbs (NRev, KipRev)
@@ -18,17 +18,18 @@
networks of simple allpass and comb
delay filters.
- the algorithm is mostly unchanged in this implementation; the delay
+ The algorithm is mostly unchanged in this implementation; the delay
line lengths have been fiddled with to make the stereo field more
- evenly weighted, and denormal protection has been added.
+ evenly weighted, denormal protection and a bandwidth control have been
+ added as well.
- the latter two are based on the circuit discussed in Jon Dattorro's
- september 1997 JAES paper on effect design (part 1: reverb & filters).
+ The latter two are based on the circuit discussed in Jon Dattorro's
+ September 1997 JAES paper on effect design (part 1: reverb & filters).
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -47,32 +48,20 @@
#include "Reverb.h"
#include "Descriptor.h"
-int
-JVRev::default_length[9] = {
-#if 1 /* slightly modified, tg */
- 1777, 1847, 1993, 2137, 389, 127, 43, 211, 209
-#else
- 4799, 4999, 5399, 5801, 1051, 337, 113, 573, 487
-#endif
-};
+int JVRev_length[9] = { 4199, 4999, 5399, 5801, 1051, 337, 113, 573, 487 };
void
JVRev::init()
{
- memcpy (length, default_length, sizeof (length));
+ double s = fs/44100.;
- if (fs != 44100)
+ for (int i = 0; i < 9; ++i)
{
- double s = fs / 44100.;
-
- for (int i = 0; i < 9; ++i)
- {
- int v = (int) (s * length[i]);
- v |= 1;
- while (!DSP::isprime (v))
- v += 2;
- length[i] = v;
- }
+ int v = (int) (s * JVRev_length[i]);
+ v |= 1;
+ while (!DSP::isprime(v))
+ v += 2;
+ length[i] = v;
}
for (int i = 0; i < 4; ++i)
@@ -84,7 +73,7 @@ JVRev::init()
left.init (length[7]);
right.init (length[8]);
- /* such a simple number, but i couldn't find a better one. */
+ /* such a simple number, yet I couldn't find a better one. */
apc = .7;
}
@@ -93,61 +82,73 @@ JVRev::set_t60 (sample_t t)
{
t60 = t;
- t = max (.00001, t);
+ t = max(.00001, t);
+ t = -3/(t*fs);
- for (int i = 0; i < 4; ++i)
- comb[i].c = pow (10, (-3 * length[i] / (t * fs)));
+ for (int i=0; i<4; ++i)
+ comb[i].c = pow (10, t*length[i]);
}
void
JVRev::activate()
{
- for (int i = 0; i < 3; ++i)
+ bandwidth.reset();
+ tone.reset();
+
+ for (int i=0; i<3; ++i)
allpass[i].reset();
- for (int i = 0; i < 4; ++i)
+ for (int i=0; i<4; ++i)
comb[i].reset();
left.reset();
right.reset();
- set_t60 (getport(1));
+ set_t60(getport(1));
+ tone.set_f(1800*over_fs);
}
-template <sample_func_t F>
void
-JVRev::one_cycle (int frames)
+JVRev::cycle (uint frames)
{
- sample_t * s = ports[0];
+ sample_t bw = .005 + .994*getport(0);
+ bandwidth.set(exp(-M_PI*(1. - bw)));
if (t60 != *ports[1])
- set_t60 (getport(1));
+ set_t60(getport(1));
- double wet = getport(2), dry = 1 - wet;
+ double wet = getport(2);
+ wet = .38*wet*wet;
+ double dry = 1 - wet;
- sample_t * dl = ports[3];
- sample_t * dr = ports[4];
+ sample_t * s = ports[3];
- for (int i = 0; i < frames; ++i)
+ sample_t * dl = ports[4];
+ sample_t * dr = ports[5];
+
+ for (uint i = 0; i < frames; ++i)
{
sample_t x = s[i], a = x + normal;
+ a = bandwidth.process(a);
x *= dry;
/* diffusors */
- a = allpass[0].process (a, -apc);
- a = allpass[1].process (a, -apc);
- a = allpass[2].process (a, -apc);
+ a = allpass[0].process(a,-apc);
+ a = allpass[1].process(a,-apc);
+ a = allpass[2].process(a,-apc);
/* tank */
sample_t t = 0;
a -= normal;
- for (int j = 0; j < 4; ++j)
- t += comb[j].process (a);
+ for (int j=0; j<4; ++j)
+ t += comb[j].process(a);
+
+ t = tone.process(t);
- F (dl, i, x + wet * left.putget (t), adding_gain);
- F (dr, i, x + wet * right.putget (t), adding_gain);
+ dl[i] = x + wet*left.putget(t);
+ dr[i] = x + wet*right.putget(t);
}
}
@@ -156,39 +157,23 @@ JVRev::one_cycle (int frames)
PortInfo
JVRev::port_info [] =
{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "t60 (s)",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, 4.6}
- }, {
- "blend",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, .28}
- }, {
- "out:l",
- OUTPUT | AUDIO,
- {0}
- }, {
- "out:r",
- OUTPUT | AUDIO,
- {0}
- }
+ { "bandwidth", INPUT | CONTROL, {DEFAULT_MID, 0, 1} },
+ { "t60 (s)", INPUT | CONTROL | GROUP, {DEFAULT_MID, 0, 5.6} },
+ { "blend", INPUT | CONTROL, {DEFAULT_LOW, 0, 1} },
+
+ { "in", INPUT | AUDIO },
+ { "out.l", OUTPUT | AUDIO },
+ { "out.r", OUTPUT | AUDIO }
};
template <> void
Descriptor<JVRev>::setup()
{
- UniqueID = 1778;
Label = "JVRev";
- Properties = HARD_RT;
Name = CAPS "JVRev - Stanford-style reverb from STK";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ Copyright = "2004-12";
/* fill port info and vtable */
autogen();
@@ -219,8 +204,8 @@ PlateStub::init()
input.lattice[3].init (L(3));
/* modulated, width about 12 samples @ 44.1 */
- tank.mlattice[0].init (L(4), (int) (0.00040322707570310132 * fs));
- tank.mlattice[1].init (L(5), (int) (0.00040322707570310132 * fs));
+ tank.mlattice[0].init (L(4), (int) (0.000403221 * fs));
+ tank.mlattice[1].init (L(5), (int) (0.000403221 * fs));
/* lh */
tank.delay[0].init (L(6));
@@ -267,8 +252,8 @@ PlateStub::process (sample_t x, sample_t decay, sample_t * _xl, sample_t * _xr)
x = input.lattice[3].process (x, indiff2);
/* summation point */
- register sample_t xl = x + decay * tank.delay[3].get();
- register sample_t xr = x + decay * tank.delay[1].get();
+ register double xl = x + decay * tank.delay[3].get();
+ register double xr = x + decay * tank.delay[1].get();
/* lh */
xl = tank.mlattice[0].process (xl, dediff1);
@@ -307,29 +292,31 @@ PlateStub::process (sample_t x, sample_t decay, sample_t * _xl, sample_t * _xr)
/* //////////////////////////////////////////////////////////////////////// */
-template <sample_func_t F>
void
-Plate::one_cycle (int frames)
+Plate::cycle (uint frames)
{
- sample_t * s = ports[0];
+ sample_t bw = .005 + .994*getport(0);
+ input.bandwidth.set (exp (-M_PI * (1. - bw)));
- input.bandwidth.set (exp (-M_PI * (1. - getport(1))));
+ sample_t decay = .749*getport(1);
- sample_t decay = getport(2);
-
- double damp = exp (-M_PI * getport(3));
+ double damp = exp (-M_PI * (.0005+.9995*getport(2)));
tank.damping[0].set (damp);
tank.damping[1].set (damp);
- sample_t blend = getport(4), dry = 1 - blend;
+ sample_t blend = getport(3);
+ blend = pow (blend, 1.6); /* linear is not a good choice for this pot */
+ sample_t dry = 1 - blend;
+
+ sample_t * s = ports[4];
sample_t * dl = ports[5];
sample_t * dr = ports[6];
- /* the modulated lattices interpolate, which needs truncated float */
+ /* modulated lattice interpolation needs float truncation */
DSP::FPTruncateMode _truncate;
- for (int i = 0; i < frames; ++i)
+ for (uint i = 0; i < frames; ++i)
{
normal = -normal;
sample_t x = s[i] + normal;
@@ -340,8 +327,8 @@ Plate::one_cycle (int frames)
x = dry * s[i];
- F (dl, i, x + blend * xl, adding_gain);
- F (dr, i, x + blend * xr, adding_gain);
+ dl[i] = x + blend*xl;
+ dr[i] = x + blend*xr;
}
}
@@ -350,47 +337,24 @@ Plate::one_cycle (int frames)
PortInfo
Plate::port_info [] =
{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "bandwidth",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0.005, .999} /* .9995 */
- }, {
- "tail",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, .749} /* .5 */
- }, {
- "damping",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, .0005, 1} /* .0005 */
- }, {
- "blend",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 1}
- }, {
- "out:l",
- OUTPUT | AUDIO,
- {0}
- }, {
- "out:r",
- OUTPUT | AUDIO,
- {0}
- }
+ {"bandwidth", INPUT | CONTROL, {DEFAULT_HIGH, 0, 1} /* .9995 */ },
+ {"tail", INPUT | CONTROL | GROUP, {DEFAULT_MID, 0, 1} /* .5 */ },
+ {"damping", INPUT | CONTROL, {DEFAULT_LOW, 0, 1} /* .0005 */ },
+ {"blend", INPUT | CONTROL | GROUP, {DEFAULT_LOW, 0, 1} },
+
+ {"in", INPUT | AUDIO},
+ {"out.l", OUTPUT | AUDIO},
+ {"out.r", OUTPUT | AUDIO}
};
template <> void
Descriptor<Plate>::setup()
{
- UniqueID = 1779;
Label = "Plate";
- Properties = HARD_RT;
Name = CAPS "Plate - Versatile plate reverb";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ Copyright = "2004-11";
/* fill port info and vtable */
autogen();
@@ -398,30 +362,31 @@ Descriptor<Plate>::setup()
/* //////////////////////////////////////////////////////////////////////// */
-template <sample_func_t F>
void
-Plate2x2::one_cycle (int frames)
+PlateX2::cycle (uint frames)
{
- sample_t * sl = ports[0];
- sample_t * sr = ports[1];
+ sample_t bw = .005 + .994*getport(0);
+ input.bandwidth.set (exp (-M_PI * (1. - bw)));
- input.bandwidth.set (exp (-M_PI * (1. - getport(2))));
+ sample_t decay = .749*getport(1);
- sample_t decay = getport(3);
-
- double damp = exp (-M_PI * getport(4));
+ double damp = exp (-M_PI * (.0005+.9995*getport(2)));
tank.damping[0].set (damp);
tank.damping[1].set (damp);
- sample_t blend = getport(5), dry = 1 - blend;
+ sample_t blend = getport(3);
+ blend = pow (blend, 1.53);
+ sample_t dry = 1 - blend;
+ sample_t * sl = ports[4];
+ sample_t * sr = ports[5];
sample_t * dl = ports[6];
sample_t * dr = ports[7];
/* the modulated lattices interpolate, which needs truncated float */
DSP::FPTruncateMode _truncate;
- for (int i = 0; i < frames; ++i)
+ for (uint i = 0; i < frames; ++i)
{
normal = -normal;
sample_t x = (sl[i] + sr[i] + normal) * .5;
@@ -429,64 +394,35 @@ Plate2x2::one_cycle (int frames)
sample_t xl, xr;
PlateStub::process (x, decay, &xl, &xr);
- xl = blend * xl + dry * sl[i];
- xr = blend * xr + dry * sr[i];
-
- F (dl, i, xl, adding_gain);
- F (dr, i, xr, adding_gain);
+ dl[i] = blend*xl + dry*sl[i];
+ dr[i] = blend*xr + dry*sr[i];
}
}
/* //////////////////////////////////////////////////////////////////////// */
PortInfo
-Plate2x2::port_info [] =
+PlateX2::port_info [] =
{
- {
- "in:l",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "in:r",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "bandwidth",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0.005, .999} /* .9995 */
- }, {
- "tail",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, .749} /* .5 */
- }, {
- "damping",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, .0005, 1} /* .0005 */
- }, {
- "blend",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 1}
- }, {
- "out:l",
- OUTPUT | AUDIO,
- {0}
- }, {
- "out:r",
- OUTPUT | AUDIO,
- {0}
- }
+ {"bandwidth", INPUT | CONTROL, {DEFAULT_HIGH, 0, 1} /* .9995 */ },
+ {"tail", INPUT | CONTROL | GROUP, {DEFAULT_MID, 0, 1} /* .5 */ },
+ {"damping", INPUT | CONTROL, {DEFAULT_LOW, 0, 1} /* .0005 */ },
+ {"blend", INPUT | CONTROL | GROUP, {DEFAULT_LOW, 0, 1} },
+
+ {"in.l", INPUT | AUDIO},
+ {"in.r", INPUT | AUDIO},
+ {"out.l", OUTPUT | AUDIO},
+ {"out.r", OUTPUT | AUDIO}
};
template <> void
-Descriptor<Plate2x2>::setup()
+Descriptor<PlateX2>::setup()
{
- UniqueID = 1795;
- Label = "Plate2x2";
- Properties = HARD_RT;
+ Label = "PlateX2";
- Name = CAPS "Plate2x2 - Versatile plate reverb, stereo inputs";
+ Name = CAPS "PlateX2 - Versatile plate reverb, stereo inputs";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ Copyright = "2004-11";
/* fill port info and vtable */
autogen();
diff --git a/plugins/LadspaEffect/caps/Reverb.h b/plugins/LadspaEffect/caps/Reverb.h
index 568f1c558..dbdfcdac8 100644
--- a/plugins/LadspaEffect/caps/Reverb.h
+++ b/plugins/LadspaEffect/caps/Reverb.h
@@ -1,7 +1,7 @@
/*
Reverb.h
- Copyright 2002-5 Tim Goetze <tim@quitte.de>
+ Copyright 2002-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -30,7 +30,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -44,13 +44,13 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _REVERB_H_
-#define _REVERB_H_
+#ifndef REVERB_H
+#define REVERB_H
#include <stdio.h>
#include "dsp/Delay.h"
-#include "dsp/OnePole.h"
+#include "dsp/IIR1.h"
#include "dsp/Sine.h"
#include "dsp/util.h"
@@ -59,13 +59,12 @@ class Lattice
: public DSP::Delay
{
public:
- inline sample_t
- process (sample_t x, double d)
+ sample_t process (sample_t x, double d)
{
sample_t y = get();
- x -= d * y;
- put (x);
- return d * x + y;
+ x -= d*y;
+ put(x);
+ return d*x + y;
}
};
@@ -76,11 +75,10 @@ class JVComb
public:
float c;
- inline sample_t
- process (sample_t x)
+ sample_t process (sample_t x)
{
- x += c * get();
- put (x);
+ x += c*get();
+ put(x);
return x;
}
};
@@ -89,20 +87,20 @@ class JVRev
: public Plugin
{
public:
- static int default_length[9];
+ DSP::LP1<sample_t> bandwidth, tone;
+
sample_t t60;
- Lattice allpass [3];
+ int length[9];
+
+ Lattice allpass[3];
JVComb comb[4];
DSP::Delay left, right;
double apc;
- template <sample_func_t F>
- void one_cycle (int frames);
-
- int length [9];
+ void cycle (uint frames);
void set_t60 (sample_t t);
@@ -111,16 +109,6 @@ class JVRev
void init();
void activate();
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
};
/* /////////////////////////////////////////////////////////////////////// */
@@ -132,7 +120,6 @@ class ModLattice
DSP::Delay delay;
DSP::Sine lfo;
- DSP::DelayTapA tap;
void init (int n, int w)
{
@@ -144,14 +131,12 @@ class ModLattice
void reset()
{
delay.reset();
- tap.reset();
}
inline sample_t
process (sample_t x, double d)
{
- /* TODO: try all-pass interpolation */
- sample_t y = delay.get_at (n0 + width * lfo.get());
+ sample_t y = delay.get_linear (n0 + width * lfo.get());
x += d * y;
delay.put (x);
return y - d * x; /* note sign */
@@ -167,7 +152,7 @@ class PlateStub
sample_t indiff1, indiff2, dediff1, dediff2;
struct {
- DSP::OnePoleLP bandwidth;
+ DSP::LP1<sample_t> bandwidth;
Lattice lattice[4];
} input;
@@ -175,7 +160,7 @@ class PlateStub
ModLattice mlattice[2];
Lattice lattice[2];
DSP::Delay delay[4];
- DSP::OnePoleLP damping[2];
+ DSP::LP1<sample_t> damping[2];
int taps[12];
} tank;
@@ -212,44 +197,22 @@ class Plate
: public PlateStub
{
public:
- template <sample_func_t F>
- void one_cycle (int frames);
+ void cycle (uint frames);
public:
static PortInfo port_info [];
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
};
/* /////////////////////////////////////////////////////////////////////// */
-class Plate2x2
+class PlateX2
: public PlateStub
{
public:
- template <sample_func_t F>
- void one_cycle (int frames);
+ void cycle (uint frames);
public:
static PortInfo port_info [];
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
};
-#endif /* _REVERB_H_ */
+#endif /* REVERB_H */
diff --git a/plugins/LadspaEffect/caps/Scape.cc b/plugins/LadspaEffect/caps/Scape.cc
index be9e5a879..952ae0d88 100644
--- a/plugins/LadspaEffect/caps/Scape.cc
+++ b/plugins/LadspaEffect/caps/Scape.cc
@@ -1,7 +1,7 @@
/*
Scape.cc
- Copyright 2004-7 Tim Goetze <tim@quitte.de>
+ Copyright 2004-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -9,7 +9,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -28,6 +28,31 @@
#include "Scape.h"
#include "Descriptor.h"
+static double
+dividers [] = {
+ 1 /* not used, 0 sentinel */,
+ 1, 0.5, 0.66666666666666666667, 0.75
+};
+
+static float
+frandom2()
+{
+ float f = frandom();
+ return f*f*f;
+}
+
+void
+Scape::init()
+{
+ delay.init ((int) (2.01 * fs)); /* two seconds = 30 bpm + */
+ for (int i = 0; i < 2; ++i)
+ {
+ lfo[i].lorenz.init(),
+ lfo[i].lorenz.set_rate (.00000001*fs);
+ lfo[i].lp.set_f (3*over_fs);
+ }
+}
+
void
Scape::activate()
{
@@ -36,93 +61,71 @@ Scape::activate()
for (int i = 0; i < 4; ++i)
svf[i].reset(),
- svf[i].set_out (SVF::Band),
- hipass[i].set_f (250. / fs);
- svf[3].set_out (SVF::Low),
+ svf[i].set_out (1), /* band pass */
+ hipass[i].set_f (250*over_fs);
+ svf[3].set_out (0); /* low pass */
delay.reset();
period = 0;
}
-static double
-dividers [] = {
- 1 /* 0 sentinel */,
- 1, 0.5, 0.66666666666666666667, 0.75
-};
-
-float
-frandom2()
+inline double
+pick_f (float range, float tune)
{
- float f = frandom();
- return f * f * f;
+ static double over_12 = 1./12;
+ int n = 48 + (int) (4*12*range*frandom());
+ return tune * pow (2, (n - 69) * over_12);
}
-template <sample_func_t F>
+
void
-Scape::one_cycle (int frames)
+Scape::cycle (uint frames)
{
- sample_t * s = ports[0];
-
- // double one_over_n = 1 / (double) frames;
-
/* delay times */
- double t1 = fs * 60. / getport(1);
- int div = (int) getport(2);
+ double t1 = fs * 60. / getport(0);
+ int div = (int) getport(1);
double t2 = t1 * dividers[div];
- fb = getport(3);
+ fb = .94*getport(2);
- double dry = getport(4);
+ double dry = getport(3);
dry = dry * dry;
- double blend = getport(5);
+ double blend = getport(4);
+ float tune = getport(5);
- sample_t * dl = ports[6];
- sample_t * dr = ports[7];
+ sample_t * s = ports[6];
+ sample_t * dl = ports[7];
+ sample_t * dr = ports[8];
DSP::FPTruncateMode truncate;
while (frames)
{
- /* flip 'renormal' addition constant */
- normal = -normal;
-
/* retune filters */
if (period <= 1)
{
period = t2 * .5;
- float f, q;
+ float f;
f = frandom2();
- svf[0].set_f_Q (300 + 300 * f / fs, .3);
- svf[3].set_f_Q (300 + 600 * 2 * f / fs, .6);
+ svf[0].set_f_Q (pick_f(.3,tune)*over_fs, .3);
+ svf[3].set_f_Q (pick_f(.5,tune)*over_fs, .6); /* LP */
f = frandom2();
- q = f;
- svf[1].set_f_Q (400 + 2400 * f / fs, q);
- q = 1 - f;
- svf[2].set_f_Q (400 + 2400 * f / fs, q);
+ svf[1].set_f_Q (pick_f(.8,tune)*over_fs, .9*f);
+ svf[2].set_f_Q (pick_f(.9,tune)*over_fs, .5*f);
}
- int n = min ((int) period, frames);
- if (n < 1)
- {
- /* not reached */
- #ifdef DEBUG
- fprintf (stderr, "Scape: %d - %d/%d frames, t2 = %.3f?!?\n", (int) period, n, frames, t2);
- #endif
- return;
- }
-
- /* sample loop */
- for (int i = 0; i < n; ++i)
+ uint n = min((uint) period, frames);
+ for (uint i=0; i < n; ++i)
{
sample_t x = s[i] + normal;
- sample_t x1 = delay.get_at (t1);
- sample_t x2 = delay.get_at (t2);
+ sample_t x1 = delay.get_linear (t1);
+ sample_t x2 = delay.get_linear (t2);
- delay.put (x + fb * x1 + normal);
- x = dry * x + .2 * svf[0].process (x) + .6 * svf[3].process(x);
+ delay.put (x + fb*x1);
+ x = dry*x + .2*svf[0].process (x) + .6*svf[3].process(x);
x1 = svf[1].process (x1 - normal);
x2 = svf[2].process (x2 - normal);
@@ -131,13 +134,13 @@ Scape::one_cycle (int frames)
x2 = hipass[2].process (x2);
sample_t x1l, x1r, x2l, x2r;
- x1l = fabs (lfo[0].get());
+ x1l = fabs (lfo[0].lp.process(lfo[0].lorenz.get()));
x1r = 1 - x1l;
- x2r = fabs (lfo[1].get());
+ x2r = fabs (lfo[1].lp.process(lfo[1].lorenz.get()));
x2l = 1 - x2r;
- F (dl, i, x + blend * (x1 * x1l + x2 * x2l), adding_gain);
- F (dr, i, x + blend * (x2 * x2r + x1 * x1r), adding_gain);
+ dl[i] = x + blend*(x1*x1l + x2*x2l);
+ dr[i] = x + blend*(x1*x1r + x2*x2r);
}
frames -= n;
@@ -153,51 +156,27 @@ Scape::one_cycle (int frames)
PortInfo
Scape::port_info [] =
{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "bpm",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 30, 240}
- }, {
- "divider",
- INPUT | CONTROL,
- {BOUNDED | INTEGER | DEFAULT_MIN, 2, 4}
- }, {
- "feedback",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 1}
- }, {
- "dry",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_LOW, 0, 1}
- }, {
- "blend",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_HIGH, 0, 1}
- }, {
- "out:l",
- OUTPUT | AUDIO,
- {0}
- }, {
- "out:r",
- OUTPUT | AUDIO,
- {0}
- }
+ { "bpm", CTRL_IN, {DEFAULT_MID, 30, 164} },
+ { "divider", CTRL_IN, {INTEGER | DEFAULT_MID, 2, 4},
+ "{2:'eighths',3:'triplets',4:'sixteenths'}" },
+ { "feedback", CTRL_IN | GROUP, {DEFAULT_HIGH, 0, 1} },
+ { "dry", CTRL_IN | GROUP, {DEFAULT_MID, 0, 1} },
+ { "blend", CTRL_IN, {DEFAULT_1, 0, 1} },
+ { "tune (Hz)", CTRL_IN | GROUP, {DEFAULT_440, 415, 467} },
+
+ { "in", AUDIO_IN },
+ { "out.l", AUDIO_OUT },
+ { "out.r", AUDIO_OUT }
};
template <> void
Descriptor<Scape>::setup()
{
- UniqueID = 2588;
Label = "Scape";
- Properties = HARD_RT;
- Name = CAPS "Scape - Stereo delay + Filters";
+ Name = CAPS "Scape - Stereo delay with chromatic resonances";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ Copyright = "2004-12";
/* fill port info and vtable */
autogen();
diff --git a/plugins/LadspaEffect/caps/Scape.h b/plugins/LadspaEffect/caps/Scape.h
index 25cca9a1e..e1b02caa5 100644
--- a/plugins/LadspaEffect/caps/Scape.h
+++ b/plugins/LadspaEffect/caps/Scape.h
@@ -9,7 +9,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -23,20 +23,18 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _SCAPE_H_
-#define _SCAPE_H_
+#ifndef SCAPE_H
+#define SCAPE_H
#include "dsp/Sine.h"
#include "dsp/Roessler.h"
#include "dsp/Lorenz.h"
#include "dsp/Delay.h"
-#include "dsp/OnePole.h"
-#include "dsp/BiQuad.h"
+#include "dsp/IIR1.h"
+#include "dsp/IIR2.h"
#include "dsp/RBJ.h"
#include "dsp/SVF.h"
-typedef DSP::SVF<1> SVF;
-
class Scape
: public Plugin
{
@@ -44,36 +42,22 @@ class Scape
sample_t time, fb;
double period;
- DSP::Lorenz lfo[2];
+ struct {
+ DSP::Lorenz lorenz;
+ DSP::LP1<sample_t> lp;
+ } lfo[2];
+
DSP::Delay delay;
- SVF svf[4];
- DSP::OnePoleHP hipass[4];
+ DSP::SVFI<1> svf[4];
+ DSP::HP1<float> hipass[4];
- template <sample_func_t>
- void one_cycle (int frames);
+ void cycle (uint frames);
public:
static PortInfo port_info [];
- void init()
- {
- delay.init ((int) (2.01 * fs)); /* two seconds = 30 bpm + */
- for (int i = 0; i < 2; ++i)
- lfo[i].init(),
- lfo[i].set_rate (.00000001 * fs);
- }
-
+ void init();
void activate();
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
};
-#endif /* _SCAPE_H_ */
+#endif /* SCAPE_H */
diff --git a/plugins/LadspaEffect/caps/Sin.cc b/plugins/LadspaEffect/caps/Sin.cc
index 150ddf88b..c97da2917 100644
--- a/plugins/LadspaEffect/caps/Sin.cc
+++ b/plugins/LadspaEffect/caps/Sin.cc
@@ -1,7 +1,7 @@
/*
Sin.cc
- Copyright 2002-7 Tim Goetze <tim@quitte.de>
+ Copyright 2002-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -11,7 +11,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -31,28 +31,41 @@
#include "Descriptor.h"
void
-Sin::init()
-{
- sin.set_f (f = .005, fs, 0);
- gain = 0;
+Sin::activate()
+{
+ gain = getport(1);
+ f = getport(0);
+ sin.set_f (f, fs, 0);
}
-template <sample_func_t F>
void
-Sin::one_cycle (int frames)
+Sin::cycle (uint frames)
{
- if (f != *ports[0])
- sin.set_f (f = getport(0), fs, sin.get_phase());
-
- double g = (gain == *ports[1]) ?
- 1 : pow (getport(1) / gain, 1. / (double) frames);
-
sample_t * d = ports[2];
+ double g = getport(1);
+ g = (g == gain ? 1 : pow (g/gain, 1./(double) frames));
- for (int i = 0; i < frames; ++i)
+ float ff = getport(0);
+ if (ff == f)
+ {
+ for (uint i = 0; i < frames; ++i)
+ {
+ d[i] = gain*sin.get();
+ gain *= g;
+ }
+ }
+ else /* crossfade old and new frequency */
{
- F (d, i, gain * sin.get(), adding_gain);
- gain *= g;
+ sample_t g0=1, g1=0, dg=1./frames;
+ DSP::Sine sin0 = sin;
+ sin.set_f (f = ff, fs, sin.get_phase());
+ for (uint i = 0; i < frames; ++i)
+ {
+ sample_t x = g0*sin0.get() + g1*sin.get();
+ g0 -= dg, g1 += dg;
+ d[i] = gain*x;
+ gain *= g;
+ }
}
gain = getport(1);
@@ -63,31 +76,20 @@ Sin::one_cycle (int frames)
PortInfo
Sin::port_info [] =
{
- {
- "f",
- INPUT | CONTROL,
- {BOUNDED | LOG | DEFAULT_100, 0.0001, 20000}
- }, {
- "volume",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, MIN_GAIN, 1}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }
+ { "f (Hz)", CTRL_IN, {LOG | DEFAULT_440, 0.0001, 20000} },
+ { "volume", CTRL_IN, {DEFAULT_MID, MIN_GAIN, 1} },
+
+ { "out", OUTPUT | AUDIO, {0} }
};
template <> void
Descriptor<Sin>::setup()
{
- UniqueID = 1781;
Label = "Sin";
- Properties = HARD_RT;
Name = CAPS "Sin - Sine wave generator";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ Copyright = "2004-13";
/* fill port info and vtable */
autogen();
diff --git a/plugins/LadspaEffect/caps/Sin.h b/plugins/LadspaEffect/caps/Sin.h
index fd540aa4d..819c79c3c 100644
--- a/plugins/LadspaEffect/caps/Sin.h
+++ b/plugins/LadspaEffect/caps/Sin.h
@@ -1,7 +1,7 @@
/*
Sin.h
- Copyright 2004-11 Tim Goetze <tim@quitte.de>
+ Copyright 2004-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -11,7 +11,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,8 +25,8 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _SIN_H_
-#define _SIN_H_
+#ifndef SIN_H
+#define SIN_H
#include "dsp/Sine.h"
@@ -38,25 +38,13 @@ class Sin
DSP::Sine sin;
- template <sample_func_t F>
- void one_cycle (int frames);
+ void cycle (uint frames);
public:
static PortInfo port_info [];
- void init();
- void activate()
- { gain = getport(1); }
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
+ void init() {}
+ void activate();
};
-#endif /* _SIN_H_ */
+#endif /* SIN_H */
diff --git a/plugins/LadspaEffect/caps/ToneStack.cc b/plugins/LadspaEffect/caps/ToneStack.cc
index 4005f1851..c80ee04cf 100644
--- a/plugins/LadspaEffect/caps/ToneStack.cc
+++ b/plugins/LadspaEffect/caps/ToneStack.cc
@@ -3,7 +3,9 @@
Copyright 2006-7
David Yeh <dtyeh@ccrma.stanford.edu>
- Tim Goetze <tim@quitte.de> (cosmetics)
+
+ 2007-2013
+ Tim Goetze <tim@quitte.de> (cosmetics)
Tone Stack emulation.
*
@@ -11,7 +13,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -26,65 +28,79 @@
*/
#include "basics.h"
+#include "dsp/util.h"
+#include "dsp/Sine.h"
#include "ToneStack.h"
#include "Descriptor.h"
-#include "dsp/tonestack/ks_tab.h"
-#include "dsp/tonestack/vs_tab.h"
+const char *
+DSP::ToneStack::presetdict =
+ "{0:'basswoman', 1:'stanford', 2:'wookie', 3:'DC 30', 4:'juice 800',"
+ "5:'twin', 6:'AK 20', 7:'nihon ace', 8:'porky',}";
DSP::TSParameters
DSP::ToneStack::presets[] = {
- /* for convenience, temporarily define k and MOhms as well as nF and pF */
- #define k * 1000
- #define M * 1000000
- #define nF * 1e-9
- #define pF * 1e-12
+ /* for convenience, */
+ #define k *1e3
+ #define M *1e6
+ #define nF *1e-9
+ #define pF *1e-12
/* parameter order is R1 - R4, C1 - C3 */
+ /* R1=treble R2=Bass R3=Mid, C1-3 related caps, R4 = parallel resistor */
/* { 250000, 1000000, 25000, 56000, 0.25e-9, 20e-9, 20e-9 }, DY */
- /* Fender */
- {250 k, 1 M, 25 k, 56 k, 250 pF, 20 nF, 20 nF}, /* 59 Bassman 5F6-A */
- {250 k, 250 k, 10 k, 100 k, 120 pF, 100 nF, 47 nF}, /* 69 Twin Reverb AA270 */
+ {250 k, 1 M, 25 k, 56 k, 250 pF, 20 nF, 20 nF}, /* 59 Bassman 5F6-A */
{250 k, 250 k, 4.8 k, 100 k, 250 pF, 100 nF, 47 nF}, /* 64 Princeton AA1164 */
- /* Marshall */
+ {250 k, 1 M, 25 k, 47 k, 600 pF, 20 nF, 20 nF}, /* Mesa Dual Rect. 'Orange' */
+ /* Vox -- R3 is fixed (circuit differs anyway) */
+ {1 M, 1 M, 20 k, 100 k, 50 pF, 22 nF, 22 nF}, /* 59/86 Vox AC-30 */
+
{220 k, 1 M, 22 k, 33 k, 470 pF, 22 nF, 22 nF}, /* 59/81 JCM-800 Lead 100 2203 */
- /* R4 is a 10 k fixed + 100 k pot in series actually */
- {250 k, 1 M, 25 k, 56 k, 500 pF, 22 nF, 22 nF}, /* 81 2000 Lead */
+ {250 k, 250 k, 10 k, 100 k, 120 pF, 100 nF, 47 nF}, /* 69 Twin Reverb AA270 */
+ {500 k, 1 M, 25 k, 47 k, 150 pF, 22 nF, 22 nF}, /* Hughes & Kettner Tube 20 */
+ {250 k, 250 k, 10 k, 100 k, 150 pF, 82 nF, 47 nF}, /* Roland Jazz Chorus */
+ {250 k, 1 M, 50 k, 33 k, 100 pF, 22 nF, 22 nF}, /* Pignose G40V */
#if 0
- {220 k, 1 M, 22 k, 33 k, 470 pF, 22 nF, 22 nF}, /* 90 JCM-900 Master 2100 (same as JCM-800) */
+ /* R4 is a 10 k fixed + 100 k pot in series actually */
{250 k, 1 M, 25 k, 33 k, 500 pF, 22 nF, 22 nF}, /* 67 Major Lead 200 */
+
+ {250 k, 1 M, 25 k, 56 k, 500 pF, 22 nF, 22 nF}, /* 81 2000 Lead */
{250 k, 250 k, 25 k, 56 k, 250 pF, 47 nF, 47 nF}, /* undated M2199 30W solid state */
#endif
- /* Vox -- R3 is fixed (circuit differs anyway) */
- {1 M, 1 M, 10 k, 100 k, 50 pF, 22 nF, 22 nF}, /* 59/86 AC-30 */
#undef k
#undef M
#undef nF
#undef pF
};
-int DSP::ToneStack::n_presets = TS_N_PRESETS;
-
void
ToneStack::activate()
{
- tonestack.activate (ports + 2);
+ model = -1;
}
-template <sample_func_t F>
void
-ToneStack::one_cycle (int frames)
+ToneStack::cycle (uint frames)
{
- sample_t * s = ports[0];
- tonestack.start_cycle (ports + 1);
+ int m = getport(0);
+
+ if (m != model)
+ {
+ model = m;
+ tonestack.setmodel (model);
+ }
+
+ sample_t * s = ports[4];
sample_t * d = ports[5];
- for (int i = 0; i < frames; ++i)
+ float bass=getport(1), mid=getport(2), treble=getport(3);
+
+ tonestack.updatecoefs (bass, mid, treble);
+ for (uint i = 0; i < frames; ++i)
{
- register sample_t a = s[i];
- a = tonestack.process (a + normal);
- F (d, i, a, adding_gain);
+ sample_t a = s[i];
+ d[i] = tonestack.process(a + normal);
}
}
@@ -92,104 +108,26 @@ ToneStack::one_cycle (int frames)
PortInfo
ToneStack::port_info [] =
{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "model",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_0 | INTEGER, 0, TS_N_PRESETS - 1}
- }, {
- "bass",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, 1}
- }, {
- "mid",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, 1}
- }, {
- "treble",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, 1}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }
+ { "model", CTRL_IN, {DEFAULT_0 | INTEGER, 0, 8}, DSP::ToneStack::presetdict },
+ { "bass", CTRL_IN | GROUP, {DEFAULT_MID, 0, 1} },
+ { "mid", CTRL_IN, {DEFAULT_MID, 0, 1} },
+ { "treble", CTRL_IN, {DEFAULT_MID, 0, 1} },
+
+ { "in", INPUT | AUDIO, {BOUNDED, -1, 1} },
+ { "out", OUTPUT | AUDIO }
};
template <> void
Descriptor<ToneStack>::setup()
{
- UniqueID = 2589;
Label = "ToneStack";
- Properties = HARD_RT;
- Name = CAPS "ToneStack - Tone stack emulation";
- Maker = "David Yeh <dtyeh@ccrma.stanford.edu>";
- Copyright = "GPL, 2006-7";
+ Name = CAPS "ToneStack - Classic amplifier tone stack emulation";
+ Maker = "David T. Yeh <dtyeh@ccrma.stanford.edu>";
+ Copyright = "2006-12";
/* fill port info and vtable */
autogen();
}
-/* //////////////////////////////////////////////////////////////////////// */
-
-template <sample_func_t F>
-void
-ToneStackLT::one_cycle (int frames)
-{
- sample_t * s = ports[0];
- tonestack.updatecoefs (ports + 1);
- sample_t * d = ports[4];
-
- for (int i = 0; i < frames; ++i)
- {
- register sample_t a = s[i];
- a = tonestack.process (a + normal);
- F (d, i, a, adding_gain);
- }
-}
-
-PortInfo
-ToneStackLT::port_info [] =
-{
- {
- "in",
- INPUT | AUDIO,
- {BOUNDED, -1, 1}
- }, {
- "bass",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, 1}
- }, {
- "mid",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, 1}
- }, {
- "treble",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, 0, 1}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }
-};
-
-template <> void
-Descriptor<ToneStackLT>::setup()
-{
- UniqueID = 2590;
- Label = "ToneStackLT";
- Properties = HARD_RT;
-
- Name = CAPS "ToneStackLT - Tone stack emulation, lattice filter 44.1";
- Maker = "David Yeh <dtyeh@ccrma.stanford.edu>";
- Copyright = "GPL, 2006-7";
-
- /* fill port info and vtable */
- autogen();
-}
diff --git a/plugins/LadspaEffect/caps/ToneStack.h b/plugins/LadspaEffect/caps/ToneStack.h
index f98a0a0d6..c88d7b559 100644
--- a/plugins/LadspaEffect/caps/ToneStack.h
+++ b/plugins/LadspaEffect/caps/ToneStack.h
@@ -3,6 +3,7 @@
Copyright 2006-7
David Yeh <dtyeh@ccrma.stanford.edu>
+ 2006-14
Tim Goetze <tim@quitte.de> (cosmetics)
Tone Stack emulation.
@@ -11,7 +12,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,8 +26,8 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _TONESTACK_H_
-#define _TONESTACK_H_
+#ifndef TONESTACK_H
+#define TONESTACK_H
#include "dsp/util.h"
#include "dsp/windows.h"
@@ -35,64 +36,18 @@
class ToneStack
: public Plugin
{
- private:
- DSP::ToneStack tonestack;
-
- template <sample_func_t F>
- void one_cycle (int frames);
-
public:
- static PortInfo port_info [];
-
- void init()
- {
- tonestack.init (fs);
- }
-
- void activate();
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
-};
-
-/* /////////////////////////////////////////////////////////////////////// */
+ int model;
-class ToneStackLT
-: public Plugin
-{
- private:
- DSP::ToneStackLT tonestack;
+ DSP::ToneStack tonestack;
- template <sample_func_t F>
- void one_cycle (int frames);
+ void cycle (uint frames);
public:
static PortInfo port_info [];
- void init()
- {
- tonestack.init (fs);
- }
-
- void activate()
- { tonestack.activate (ports + 1); }
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
+ void init() { tonestack.init(fs); }
+ void activate();
};
-#endif /* _TONESTACK_H_ */
+#endif /* TONESTACK_H */
diff --git a/plugins/LadspaEffect/caps/White.cc b/plugins/LadspaEffect/caps/White.cc
index 28b2c3643..381d73b05 100644
--- a/plugins/LadspaEffect/caps/White.cc
+++ b/plugins/LadspaEffect/caps/White.cc
@@ -1,17 +1,17 @@
/*
White.cc
- Copyright 2004-7 Tim Goetze <tim@quitte.de>
+ Copyright 2004-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- white noise generation.
+ Approximating white pseudonoise generation.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -30,18 +30,28 @@
#include "White.h"
#include "Descriptor.h"
-template <sample_func_t F>
void
-White::one_cycle (int frames)
+White::activate()
+{
+ gain = getport(0);
+ white.init (frandom());
+ cream.init (frandom());
+ hp.set_f (.48);
+}
+
+void
+White::cycle (uint frames)
{
double g = (gain == *ports[0]) ?
1 : pow (getport(0) / gain, 1. / (double) frames);
sample_t * d = ports[1];
- for (int i = 0; i < frames; ++i)
+ for (uint i = 0; i < frames; ++i)
{
- F (d, i, gain * white.get(), adding_gain);
+ sample_t x = .4*white.get();
+ x += hp.process(cream.get());
+ d[i] = gain*x;
gain *= g;
}
@@ -53,27 +63,18 @@ White::one_cycle (int frames)
PortInfo
White::port_info [] =
{
- {
- "volume",
- INPUT | CONTROL,
- {BOUNDED | DEFAULT_MID, MIN_GAIN, 1}
- }, {
- "out",
- OUTPUT | AUDIO,
- {0}
- }
+ { "volume", INPUT | CONTROL, {DEFAULT_HIGH, MIN_GAIN, 1} },
+ { "out", OUTPUT | AUDIO}
};
template <> void
Descriptor<White>::setup()
{
- UniqueID = 1785;
Label = "White";
- Properties = HARD_RT;
- Name = CAPS "White - White noise generator";
+ Name = CAPS "White - Noise generator";
Maker = "Tim Goetze <tim@quitte.de>";
- Copyright = "GPL, 2004-7";
+ Copyright = "2004-13";
/* fill port info and vtable */
autogen();
diff --git a/plugins/LadspaEffect/caps/White.h b/plugins/LadspaEffect/caps/White.h
index 5633599e8..65bde1930 100644
--- a/plugins/LadspaEffect/caps/White.h
+++ b/plugins/LadspaEffect/caps/White.h
@@ -1,17 +1,17 @@
/*
White.h
- Copyright 2004-5 Tim Goetze <tim@quitte.de>
+ Copyright 2004-12 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- white noise generator.
+ Noise generator.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,10 +25,11 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _WHITE_H_
-#define _WHITE_H_
+#ifndef WHITE_H
+#define WHITE_H
#include "dsp/White.h"
+#include "dsp/IIR1.h"
class White
: public Plugin
@@ -36,27 +37,16 @@ class White
public:
sample_t gain;
- DSP::White white;
+ DSP::White white, cream;
+ DSP::HP1<sample_t> hp;
- template <sample_func_t F>
- void one_cycle (int frames);
+ void cycle (uint frames);
public:
static PortInfo port_info [];
void init() {}
- void activate()
- { gain = getport(0); }
-
- void run (int n)
- {
- one_cycle<store_func> (n);
- }
-
- void run_adding (int n)
- {
- one_cycle<adding_func> (n);
- }
+ void activate();
};
-#endif /* _WHITE_H_ */
+#endif /* WHITE_H */
diff --git a/plugins/LadspaEffect/caps/basics.h b/plugins/LadspaEffect/caps/basics.h
index df24e8c05..643d96e2d 100644
--- a/plugins/LadspaEffect/caps/basics.h
+++ b/plugins/LadspaEffect/caps/basics.h
@@ -1,18 +1,18 @@
/*
basics.h
- Copyright 2004-9 Tim Goetze <tim@quitte.de>
+ Copyright 2004-12 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- common constants, typedefs, utility functions
+ Common constants, typedefs, utility functions
and simplified LADSPA #defines.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -26,8 +26,8 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _BASICS_H_
-#define _BASICS_H_
+#ifndef BASICS_H
+#define BASICS_H
#define _GNU_SOURCE 1
#define _USE_GNU 1
@@ -42,90 +42,78 @@
#include <string.h>
#include <math.h>
+#include <float.h>
#include <assert.h>
#include <stdio.h>
-#include <stdint.h>
-
-#include <ladspa.h>
-
-/* reducing LADSPA_DEFINES_WITH_LOTS_OF_CHARACTERS_REALLY verbosity */
-#define BOUNDED (LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_BOUNDED_ABOVE)
-#define INTEGER LADSPA_HINT_INTEGER
-#define LOG LADSPA_HINT_LOGARITHMIC
-#define TOGGLE LADSPA_HINT_TOGGLED
-
-#define DEFAULT_0 LADSPA_HINT_DEFAULT_0
-#define DEFAULT_1 LADSPA_HINT_DEFAULT_1
-#define DEFAULT_100 LADSPA_HINT_DEFAULT_100
-#define DEFAULT_440 LADSPA_HINT_DEFAULT_440
-#define DEFAULT_MIN LADSPA_HINT_DEFAULT_MINIMUM
-#define DEFAULT_LOW LADSPA_HINT_DEFAULT_LOW
-#define DEFAULT_MID LADSPA_HINT_DEFAULT_MIDDLE
-#define DEFAULT_HIGH LADSPA_HINT_DEFAULT_HIGH
-#define DEFAULT_MAX LADSPA_HINT_DEFAULT_MAXIMUM
-
-#define INPUT LADSPA_PORT_INPUT
-#define OUTPUT LADSPA_PORT_OUTPUT
-#define AUDIO LADSPA_PORT_AUDIO
-#define CONTROL LADSPA_PORT_CONTROL
-#define HARD_RT LADSPA_PROPERTY_HARD_RT_CAPABLE
+#include "ladspa.h"
+
+typedef __int8_t int8;
+typedef __uint8_t uint8;
+typedef __int16_t int16;
+typedef __uint16_t uint16;
+typedef __int32_t int32;
+typedef __uint32_t uint32;
+typedef __int64_t int64;
+typedef __uint64_t uint64;
-#define TEN_TO_THE_SIXTH 1000000
+#define MIN_GAIN 1e-6 /* -120 dB */
+/* smallest non-denormal 32 bit IEEE float is 1.18e-38 */
+#define NOISE_FLOOR 1e-20 /* -400 dB */
-#define MIN_GAIN .000001 /* -120 dB */
+#define HARD_RT LADSPA_PROPERTY_HARD_RT_CAPABLE
-/* smallest non-denormal 32 bit IEEE float is 1.18�10-38 */
-#define NOISE_FLOOR .00000000000005 /* -266 dB */
+/* some LADSPA_DEFINES_THAT_COME_WITH_LOTS_OF_CHARACTERS */
+#define INPUT LADSPA_PORT_INPUT
+#define OUTPUT LADSPA_PORT_OUTPUT
+#define CONTROL LADSPA_PORT_CONTROL
+#define AUDIO LADSPA_PORT_AUDIO
+
+#define AUDIO_IN AUDIO|INPUT
+#define AUDIO_OUT AUDIO|OUTPUT
+#define CTRL_IN CONTROL|INPUT
+#define CTRL_OUT CONTROL|OUTPUT
+
+/* extending LADSPA_PORT_* */
+#define LADSPA_PORT_GROUP (AUDIO<<1) /* 16 */
+#define GROUP LADSPA_PORT_GROUP
+
+/* more LADSPA_DEFINES_THAT_REALLY_COME_WITH_LOTS_OF_CHARACTERS */
+#define BOUNDED (LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_BOUNDED_ABOVE)
+#define INTEGER LADSPA_HINT_INTEGER
+#define LOG LADSPA_HINT_LOGARITHMIC
+#define TOGGLE LADSPA_HINT_TOGGLED
+
+#define DEFAULT_0 LADSPA_HINT_DEFAULT_0
+#define DEFAULT_1 LADSPA_HINT_DEFAULT_1
+#define DEFAULT_100 LADSPA_HINT_DEFAULT_100
+#define DEFAULT_440 LADSPA_HINT_DEFAULT_440
+#define DEFAULT_MIN LADSPA_HINT_DEFAULT_MINIMUM
+#define DEFAULT_LOW LADSPA_HINT_DEFAULT_LOW
+#define DEFAULT_MID LADSPA_HINT_DEFAULT_MIDDLE
+#define DEFAULT_HIGH LADSPA_HINT_DEFAULT_HIGH
+#define DEFAULT_MAX LADSPA_HINT_DEFAULT_MAXIMUM
-typedef int8_t int8;
-typedef uint8_t uint8;
-typedef int16_t int16;
-typedef uint16_t uint16;
-typedef int32_t int32;
-typedef uint32_t uint32;
-typedef int64_t int64;
-typedef uint64_t uint64;
+/* //////////////////////////////////////////////////////////////////////// */
typedef struct {
const char * name;
LADSPA_PortDescriptor descriptor;
LADSPA_PortRangeHint range;
+ const char * meta;
} PortInfo;
typedef LADSPA_Data sample_t;
+typedef unsigned int uint;
typedef unsigned long ulong;
-/* flavours for sample store functions run() and run_adding() */
-typedef void (*sample_func_t) (sample_t *, int, sample_t, sample_t);
-
-inline void
-store_func (sample_t * s, int i, sample_t x, sample_t gain)
-{
- s[i] = x;
-}
-
-inline void
-adding_func (sample_t * s, int i, sample_t x, sample_t gain)
-{
- s[i] += gain * x;
-}
+/* prototype that takes a sample and yields a sample */
+typedef sample_t (*clip_func_t) (sample_t);
#ifndef max
-
-template <class X, class Y>
-X min (X x, Y y)
-{
- return x < y ? x : (X) y;
-}
-
-template <class X, class Y>
-X max (X x, Y y)
-{
- return x > y ? x : (X) y;
-}
-
+template <class X, class Y> X min (X x, Y y) { return x < (X)y ? x : (X)y; }
+template <class X, class Y> X max (X x, Y y) { return x > (X)y ? x : (X)y; }
#endif /* ! max */
template <class T>
@@ -136,11 +124,7 @@ T clamp (T value, T lower, T upper)
return value;
}
-static inline float
-frandom()
-{
- return (float) rand() / (float) RAND_MAX;
-}
+static inline float frandom() { return (float) random() / (float) RAND_MAX; }
/* NB: also true if 0 */
inline bool
@@ -150,7 +134,7 @@ is_denormal (float & f)
return ((i & 0x7f800000) == 0);
}
-/* todo: not sure if this double version is correct, actually ... */
+/* not used, check validity before using */
inline bool
is_denormal (double & f)
{
@@ -158,7 +142,28 @@ is_denormal (double & f)
return ((i & 0x7fe0000000000000ll) == 0);
}
-#ifdef __i386__
+/* lovely algorithm from
+ http://graphics.stanford.edu/~seander/bithacks.html#RoundUpPowerOf2Float
+*/
+inline uint
+next_power_of_2 (uint n)
+{
+ assert (n <= 0x40000000);
+
+ --n;
+ n |= n >> 1;
+ n |= n >> 2;
+ n |= n >> 4;
+ n |= n >> 8;
+ n |= n >> 16;
+
+ return ++n;
+}
+
+inline double db2lin (double db) { return pow(10, .05*db); }
+inline double lin2db (double lin) { return 20*log10(lin); }
+
+#if defined(__i386__) || defined(__amd64__)
#define TRAP asm ("int $3;")
#else
#define TRAP
@@ -168,10 +173,11 @@ is_denormal (double & f)
#define CAPS "C* "
-class Plugin {
+class Plugin
+{
public:
- double fs; /* sample rate */
- double adding_gain; /* for run_adding() */
+ float fs, over_fs; /* sample rate and 1/fs */
+ float adding_gain; /* for run_adding() */
int first_run; /* 1st block after activate(), do no parameter smoothing */
sample_t normal; /* renormal constant */
@@ -196,4 +202,4 @@ class Plugin {
}
};
-#endif /* _BASICS_H_ */
+#endif /* BASICS_H */
diff --git a/plugins/LadspaEffect/caps/caps.rdf b/plugins/LadspaEffect/caps/caps.rdf
index dbcfcc566..64c927a97 100644
--- a/plugins/LadspaEffect/caps/caps.rdf
+++ b/plugins/LadspaEffect/caps/caps.rdf
@@ -1,447 +1,998 @@
-<?xml version='1.0' encoding='ISO-8859-1'?>
-<!--
+<?xml version='1.0' encoding='utf-8'?><!--
caps.rdf
- Metadata for the CAPS Audio Plugin Suite
+ Metadata for the CAPS Audio Plugin Suite
+ Automatically generated based on work done by
- Automatically generated based on work done by
- Paul Winkler,
- Pete Leigh and
- Tim Goetze.
+ Paul Winkler, Pete Leigh, and Tim Goetze.
- Welcome to the Dept. of Redundancy Dept. Dept. Dept.
+ Welcome to the Dept. of Redundancy Dept.
- -->
-<!DOCTYPE rdf:RDF
- [
+-->
+<!DOCTYPE rdf:RDF [
<!ENTITY rdf 'http://www.w3.org/1999/02/22-rdf-syntax-ns#'>
<!ENTITY rdfs 'http://www.w3.org/2000/01/rdf-schema#'>
<!ENTITY dc 'http://purl.org/dc/elements/1.1/'>
- <!ENTITY ladspa 'http://ladspa.org/ontology#'>
- ]
->
+ <!ENTITY caps 'http://quitte.de/dsp/caps.html#'>
+ <!ENTITY ladspa 'http://ladspa.org/ontology#'> ]>
<rdf:RDF
xmlns:rdf="&rdf;"
xmlns:rdfs="&rdfs;"
xmlns:dc="&dc;"
- xmlns:ladspa="&ladspa;"
->
+ xmlns:caps="&caps;"
+ xmlns:ladspa="&ladspa;">
- <!-- CabinetI -->
+ <!-- Noisegate -->
+ <ladspa:FilterPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Noisegate'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2602.3"
+ ladspa:hasLabel="mains (Hz)">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="off" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="50"
+ ladspa:hasLabel="global" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="60"
+ ladspa:hasLabel="imperial" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ </ladspa:FilterPlugin>
+
+ <!-- Compress -->
+ <ladspa:CompressorPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Compress'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;1772.0"
+ ladspa:hasLabel="measure">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="peak" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="rms" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;1772.1"
+ ladspa:hasLabel="mode">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="no limiting" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="saturating 2x" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="saturating 4x" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="3"
+ ladspa:hasLabel="saturating 4x128" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ </ladspa:CompressorPlugin>
+
+ <!-- CompressX2 -->
+ <ladspa:CompressorPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#CompressX2'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2598.0"
+ ladspa:hasLabel="measure">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="peak" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="rms" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2598.1"
+ ladspa:hasLabel="mode">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="linear" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="saturating 2x" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="saturating 4x" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="3"
+ ladspa:hasLabel="saturating 4x128" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ </ladspa:CompressorPlugin>
+
+ <!-- ToneStack -->
<ladspa:SimulatorPlugin
- rdf:about='http://ladspa.org/ontology#1766'>
- <ladspa:hasPort>
- <ladspa:InputControlPort
- rdf:about="&ladspa;1766.1"
+ rdf:about='http://quitte.de/dsp/caps.html#ToneStack'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2589.0"
ladspa:hasLabel="model">
- <ladspa:hasScale>
- <ladspa:Scale>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
rdf:value="0"
- ladspa:hasLabel="none" />
+ ladspa:hasLabel="basswoman" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="1"
- ladspa:hasLabel="Unmatched off-axis" />
+ ladspa:hasLabel="stanford" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="2"
- ladspa:hasLabel="Unmatched on-axis" />
+ ladspa:hasLabel="wookie" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="3"
- ladspa:hasLabel="Supertramp" />
+ ladspa:hasLabel="DC 30" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="4"
- ladspa:hasLabel="Little Wing 68" />
+ ladspa:hasLabel="juice 800" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="5"
- ladspa:hasLabel="Martial" />
+ ladspa:hasLabel="twin" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="6"
- ladspa:hasLabel="Mesa" />
+ ladspa:hasLabel="AK 20" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="7"
- ladspa:hasLabel="Pro Jr" />
+ ladspa:hasLabel="nihon ace" />
</ladspa:hasPoint>
- </ladspa:Scale>
- </ladspa:hasScale>
- </ladspa:InputControlPort>
- </ladspa:hasPort>
- </ladspa:SimulatorPlugin>
-
- <!-- ChorusI -->
- <ladspa:ChorusPlugin
- rdf:about='http://ladspa.org/ontology#1767'>
- </ladspa:ChorusPlugin>
-
- <!-- StereoChorusI -->
- <ladspa:ChorusPlugin
- rdf:about='http://ladspa.org/ontology#1768'>
- </ladspa:ChorusPlugin>
-
- <!-- Click -->
- <ladspa:TimePlugin
- rdf:about='http://ladspa.org/ontology#1769'>
- </ladspa:TimePlugin>
-
- <!-- CEO -->
- <ladspa:OscillatorPlugin
- rdf:about='http://ladspa.org/ontology#1770'>
- </ladspa:OscillatorPlugin>
-
- <!-- Clip -->
- <ladspa:DistortionPlugin
- rdf:about='http://ladspa.org/ontology#1771'>
- </ladspa:DistortionPlugin>
-
- <!-- Compress -->
- <ladspa:CompressorPlugin
- rdf:about='http://ladspa.org/ontology#1772'>
- </ladspa:CompressorPlugin>
-
- <!-- Eq -->
- <ladspa:EQPlugin
- rdf:about='http://ladspa.org/ontology#1773'>
- </ladspa:EQPlugin>
-
- <!-- Lorenz -->
- <ladspa:GeneratorPlugin
- rdf:about='http://ladspa.org/ontology#1774'>
- </ladspa:GeneratorPlugin>
-
- <!-- PhaserI -->
- <ladspa:PhaserPlugin
- rdf:about='http://ladspa.org/ontology#1775'>
- </ladspa:PhaserPlugin>
-
- <!-- PreampIII -->
- <ladspa:SimulatorPlugin
- rdf:about='http://ladspa.org/ontology#1776'>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="8"
+ ladspa:hasLabel="porky" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
</ladspa:SimulatorPlugin>
- <!-- PreampIV -->
+ <!-- AmpVTS -->
<ladspa:SimulatorPlugin
- rdf:about='http://ladspa.org/ontology#1777'>
+ rdf:about='http://quitte.de/dsp/caps.html#AmpVTS'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2592.0"
+ ladspa:hasLabel="over">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="2x" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="4x" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="8x" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2592.4"
+ ladspa:hasLabel="tonestack">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="basswoman" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="stanford" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="wookie" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="3"
+ ladspa:hasLabel="DC 30" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="4"
+ ladspa:hasLabel="juice 800" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="5"
+ ladspa:hasLabel="twin" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="6"
+ ladspa:hasLabel="AK 20" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="7"
+ ladspa:hasLabel="nihon ace" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="8"
+ ladspa:hasLabel="porky" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
</ladspa:SimulatorPlugin>
- <!-- JVRev -->
- <ladspa:ReverbPlugin
- rdf:about='http://ladspa.org/ontology#1778'>
- </ladspa:ReverbPlugin>
-
- <!-- Plate -->
- <ladspa:ReverbPlugin
- rdf:about='http://ladspa.org/ontology#1779'>
- </ladspa:ReverbPlugin>
-
- <!-- Roessler -->
+ <!-- AmpVI -->
<ladspa:GeneratorPlugin
- rdf:about='http://ladspa.org/ontology#1780'>
- </ladspa:GeneratorPlugin>
-
- <!-- Sin -->
- <ladspa:OscillatorPlugin
- rdf:about='http://ladspa.org/ontology#1781'>
- </ladspa:OscillatorPlugin>
-
- <!-- SweepVFI -->
- <ladspa:FilterPlugin
- rdf:about='http://ladspa.org/ontology#1782'>
- <ladspa:hasPort>
- <ladspa:InputControlPort
- rdf:about="&ladspa;1782.3"
- ladspa:hasLabel="model">
- <ladspa:hasScale>
- <ladspa:Scale>
- <ladspa:hasPoint>
- <ladspa:Point
+ rdf:about='http://quitte.de/dsp/caps.html#AmpVI'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;1789.0"
+ ladspa:hasLabel="over">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="4x" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="8x" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;1789.5"
+ ladspa:hasLabel="tonestack">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
rdf:value="0"
- ladspa:hasLabel="low pass" />
+ ladspa:hasLabel="basswoman" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="1"
- ladspa:hasLabel="band pass" />
+ ladspa:hasLabel="stanford" />
</ladspa:hasPoint>
- </ladspa:Scale>
- </ladspa:hasScale>
- </ladspa:InputControlPort>
- </ladspa:hasPort>
- </ladspa:FilterPlugin>
-
- <!-- VCOs -->
- <ladspa:OscillatorPlugin
- rdf:about='http://ladspa.org/ontology#1783'>
- </ladspa:OscillatorPlugin>
-
- <!-- VCOd -->
- <ladspa:OscillatorPlugin
- rdf:about='http://ladspa.org/ontology#1784'>
- </ladspa:OscillatorPlugin>
-
- <!-- White -->
- <ladspa:GeneratorPlugin
- rdf:about='http://ladspa.org/ontology#1785'>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="wookie" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="3"
+ ladspa:hasLabel="DC 30" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="4"
+ ladspa:hasLabel="juice 800" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="5"
+ ladspa:hasLabel="twin" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="6"
+ ladspa:hasLabel="AK 20" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="7"
+ ladspa:hasLabel="nihon ace" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="8"
+ ladspa:hasLabel="porky" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;1789.14"
+ ladspa:hasLabel="cab">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="twin" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="supro" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="chief" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="3"
+ ladspa:hasLabel="wookie" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="4"
+ ladspa:hasLabel="tweedie" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="5"
+ ladspa:hasLabel="vox" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
</ladspa:GeneratorPlugin>
- <!-- AmpIII -->
+ <!-- CabinetIII -->
<ladspa:SimulatorPlugin
- rdf:about='http://ladspa.org/ontology#1786'>
+ rdf:about='http://quitte.de/dsp/caps.html#CabinetIII'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2601.0"
+ ladspa:hasLabel="model">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="off" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="twin" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="wookie" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="3"
+ ladspa:hasLabel="blue" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="4"
+ ladspa:hasLabel="indigo" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="5"
+ ladspa:hasLabel="supra" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="6"
+ ladspa:hasLabel="tweedie" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="7"
+ ladspa:hasLabel="basswoman" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="8"
+ ladspa:hasLabel="martial" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="9"
+ ladspa:hasLabel="poser" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="10"
+ ladspa:hasLabel="cabby" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="11"
+ ladspa:hasLabel="angel" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="12"
+ ladspa:hasLabel="matched" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="13"
+ ladspa:hasLabel="yellow" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="14"
+ ladspa:hasLabel="piquey" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="15"
+ ladspa:hasLabel="paladin" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="16"
+ ladspa:hasLabel="turtle" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2601.1"
+ ladspa:hasLabel="alt">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="off" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="on" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
</ladspa:SimulatorPlugin>
- <!-- HRTF -->
+ <!-- CabinetIV -->
<ladspa:SimulatorPlugin
- rdf:about='http://ladspa.org/ontology#1787'>
+ rdf:about='http://quitte.de/dsp/caps.html#CabinetIV'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2606.0"
+ ladspa:hasLabel="model">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="mega wookie 800" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="mega wookie 812" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="mega wookie 828" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="3"
+ ladspa:hasLabel="mega wookie 868" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="4"
+ ladspa:hasLabel="mega wookie 908" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="5"
+ ladspa:hasLabel="mega wookie 912" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="6"
+ ladspa:hasLabel="mega wookie 936" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="7"
+ ladspa:hasLabel="mega wookie 968" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="8"
+ ladspa:hasLabel="mega wookie 992" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="9"
+ ladspa:hasLabel="unmatched" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="10"
+ ladspa:hasLabel="twin A" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="11"
+ ladspa:hasLabel="twin B" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="12"
+ ladspa:hasLabel="twin C" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="13"
+ ladspa:hasLabel="blue A" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="14"
+ ladspa:hasLabel="blue B" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="15"
+ ladspa:hasLabel="tweedie A" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="16"
+ ladspa:hasLabel="tweedie B" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="17"
+ ladspa:hasLabel="mini wookie A" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="18"
+ ladspa:hasLabel="mini wookie B" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="19"
+ ladspa:hasLabel="rosie A" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="20"
+ ladspa:hasLabel="rosie B" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="21"
+ ladspa:hasLabel="indigo" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="22"
+ ladspa:hasLabel="angel" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="23"
+ ladspa:hasLabel="sixty-one" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="24"
+ ladspa:hasLabel="sixty-two" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
</ladspa:SimulatorPlugin>
- <!-- Pan -->
- <ladspa:DelayPlugin
- rdf:about='http://ladspa.org/ontology#1788'>
- </ladspa:DelayPlugin>
-
- <!-- AmpIV -->
- <ladspa:SimulatorPlugin
- rdf:about='http://ladspa.org/ontology#1794'>
- </ladspa:SimulatorPlugin>
+ <!-- Plate -->
+ <ladspa:ReverbPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Plate'>
+ </ladspa:ReverbPlugin>
- <!-- Plate2x2 -->
+ <!-- PlateX2 -->
<ladspa:ReverbPlugin
- rdf:about='http://ladspa.org/ontology#1795'>
+ rdf:about='http://quitte.de/dsp/caps.html#PlateX2'>
</ladspa:ReverbPlugin>
- <!-- CabinetII -->
- <ladspa:SimulatorPlugin
- rdf:about='http://ladspa.org/ontology#2581'>
- <ladspa:hasPort>
- <ladspa:InputControlPort
- rdf:about="&ladspa;2581.1"
- ladspa:hasLabel="model">
- <ladspa:hasScale>
- <ladspa:Scale>
- <ladspa:hasPoint>
- <ladspa:Point
+ <!-- Saturate -->
+ <ladspa:DistortionPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Saturate'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;1771.0"
+ ladspa:hasLabel="mode">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
rdf:value="0"
- ladspa:hasLabel="none" />
+ ladspa:hasLabel="bypass" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="1"
- ladspa:hasLabel="Unmatched off-axis" />
+ ladspa:hasLabel="atan" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="2"
- ladspa:hasLabel="Unmatched on-axis" />
+ ladspa:hasLabel="atan15" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="3"
- ladspa:hasLabel="Supertramp" />
+ ladspa:hasLabel="clip" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="4"
- ladspa:hasLabel="Little Wing 68" />
+ ladspa:hasLabel="one5" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="5"
- ladspa:hasLabel="Martial" />
+ ladspa:hasLabel="one53" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="6"
- ladspa:hasLabel="Mesa" />
+ ladspa:hasLabel="clip3" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="7"
- ladspa:hasLabel="Pro Jr" />
+ ladspa:hasLabel="clip9" />
</ladspa:hasPoint>
- </ladspa:Scale>
- </ladspa:hasScale>
- </ladspa:InputControlPort>
- </ladspa:hasPort>
- </ladspa:SimulatorPlugin>
-
- <!-- SweepVFII -->
- <ladspa:FilterPlugin
- rdf:about='http://ladspa.org/ontology#2582'>
- <ladspa:hasPort>
- <ladspa:InputControlPort
- rdf:about="&ladspa;2582.3"
- ladspa:hasLabel="model">
- <ladspa:hasScale>
- <ladspa:Scale>
- <ladspa:hasPoint>
- <ladspa:Point
- rdf:value="0"
- ladspa:hasLabel="low pass" />
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="8"
+ ladspa:hasLabel="sin1" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
- rdf:value="1"
- ladspa:hasLabel="band pass" />
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="9"
+ ladspa:hasLabel="pow7" />
</ladspa:hasPoint>
- </ladspa:Scale>
- </ladspa:hasScale>
- </ladspa:InputControlPort>
- </ladspa:hasPort>
- </ladspa:FilterPlugin>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="10"
+ ladspa:hasLabel="tanh" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="11"
+ ladspa:hasLabel="rectify" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ </ladspa:DistortionPlugin>
- <!-- ChorusII -->
- <ladspa:ChorusPlugin
- rdf:about='http://ladspa.org/ontology#2583'>
- </ladspa:ChorusPlugin>
+ <!-- Spice -->
+ <ladspa:DistortionPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Spice'>
+ </ladspa:DistortionPlugin>
+
+ <!-- SpiceX2 -->
+ <ladspa:DistortionPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#SpiceX2'>
+ </ladspa:DistortionPlugin>
- <!-- StereoChorusII -->
+ <!-- ChorusI -->
<ladspa:ChorusPlugin
- rdf:about='http://ladspa.org/ontology#2584'>
+ rdf:about='http://quitte.de/dsp/caps.html#ChorusI'>
</ladspa:ChorusPlugin>
- <!-- Dirac -->
- <ladspa:GeneratorPlugin
- rdf:about='http://ladspa.org/ontology#2585'>
- </ladspa:GeneratorPlugin>
-
<!-- PhaserII -->
<ladspa:PhaserPlugin
- rdf:about='http://ladspa.org/ontology#2586'>
+ rdf:about='http://quitte.de/dsp/caps.html#PhaserII'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2586.1"
+ ladspa:hasLabel="lfo">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="sine" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="fractal" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
</ladspa:PhaserPlugin>
- <!-- AmpV -->
- <ladspa:SimulatorPlugin
- rdf:about='http://ladspa.org/ontology#2587'>
- </ladspa:SimulatorPlugin>
-
- <!-- Scape -->
- <ladspa:DelayPlugin
- rdf:about='http://ladspa.org/ontology#2588'>
- </ladspa:DelayPlugin>
-
- <!-- ToneStack -->
- <ladspa:SimulatorPlugin
- rdf:about='http://ladspa.org/ontology#2589'>
- <ladspa:hasPort>
- <ladspa:InputControlPort
- rdf:about="&ladspa;2589.1"
- ladspa:hasLabel="model">
- <ladspa:hasScale>
- <ladspa:Scale>
- <ladspa:hasPoint>
- <ladspa:Point
+ <!-- AutoFilter -->
+ <ladspa:FilterPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#AutoFilter'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2593.0"
+ ladspa:hasLabel="mode">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
rdf:value="0"
- ladspa:hasLabel="'59 Bassman" />
+ ladspa:hasLabel="low pass" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="1"
- ladspa:hasLabel="'69 Twin Reverb" />
+ ladspa:hasLabel="band pass" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2593.1"
+ ladspa:hasLabel="filter">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="breathy" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="fat" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ </ladspa:FilterPlugin>
+
+ <!-- Scape -->
+ <ladspa:DelayPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Scape'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2588.1"
+ ladspa:hasLabel="divider">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
rdf:value="2"
- ladspa:hasLabel="'64 Princeton" />
+ ladspa:hasLabel="eighths" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="3"
- ladspa:hasLabel="'59/'81 JCM 800" />
+ ladspa:hasLabel="triplets" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="4"
- ladspa:hasLabel="'78 Club &amp; Country" />
- </ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
- rdf:value="5"
- ladspa:hasLabel="'59/'86 AC-30 of sorts" />
+ ladspa:hasLabel="sixteenths" />
</ladspa:hasPoint>
- </ladspa:Scale>
- </ladspa:hasScale>
- </ladspa:InputControlPort>
- </ladspa:hasPort>
- </ladspa:SimulatorPlugin>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ </ladspa:DelayPlugin>
- <!-- ToneStackLT -->
- <ladspa:SimulatorPlugin
- rdf:about='http://ladspa.org/ontology#2590'>
- </ladspa:SimulatorPlugin>
+ <!-- Eq10 -->
+ <ladspa:EQPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Eq10'>
+ </ladspa:EQPlugin>
- <!-- AmpVTS -->
- <ladspa:SimulatorPlugin
- rdf:about='http://ladspa.org/ontology#2592'>
- <ladspa:hasPort>
- <ladspa:InputControlPort
- rdf:about="&ladspa;2592.1"
- ladspa:hasLabel="model">
- <ladspa:hasScale>
- <ladspa:Scale>
- <ladspa:hasPoint>
- <ladspa:Point
+ <!-- Eq10X2 -->
+ <ladspa:EQPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Eq10X2'>
+ </ladspa:EQPlugin>
+
+ <!-- Eq4p -->
+ <ladspa:EQPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Eq4p'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2608.0"
+ ladspa:hasLabel="a.mode">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
rdf:value="0"
- ladspa:hasLabel="'59 Bassman" />
+ ladspa:hasLabel="lowshelve" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="1"
- ladspa:hasLabel="'69 Twin Reverb" />
+ ladspa:hasLabel="band" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
+ <ladspa:hasPoint><ladspa:Point
rdf:value="2"
- ladspa:hasLabel="'64 Princeton" />
+ ladspa:hasLabel="hishelve" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="-1"
+ ladspa:hasLabel="off" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2608.4"
+ ladspa:hasLabel="b.mode">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="lowshelve" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
- rdf:value="3"
- ladspa:hasLabel="'59/'81 JCM 800" />
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="band" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
- rdf:value="4"
- ladspa:hasLabel="'78 Club &amp; Country" />
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="hishelve" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="-1"
+ ladspa:hasLabel="off" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2608.8"
+ ladspa:hasLabel="c.mode">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="lowshelve" />
</ladspa:hasPoint>
- <ladspa:hasPoint>
- <ladspa:Point
- rdf:value="5"
- ladspa:hasLabel="'59/'86 AC-30 of sorts" />
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="band" />
</ladspa:hasPoint>
- </ladspa:Scale>
- </ladspa:hasScale>
- </ladspa:InputControlPort>
- </ladspa:hasPort>
- </ladspa:SimulatorPlugin>
-
- <!-- AutoWah -->
- <ladspa:FilterPlugin
- rdf:about='http://ladspa.org/ontology#2593'>
- </ladspa:FilterPlugin>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="hishelve" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="-1"
+ ladspa:hasLabel="off" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2608.12"
+ ladspa:hasLabel="d.mode">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="lowshelve" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="band" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="hishelve" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="-1"
+ ladspa:hasLabel="off" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2608.16"
+ ladspa:hasLabel="_latency">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="3"
+ ladspa:hasLabel="3 samples" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ </ladspa:EQPlugin>
- <!-- Eq2x2 -->
+ <!-- EqFA4p -->
<ladspa:EQPlugin
- rdf:about='http://ladspa.org/ontology#2594'>
+ rdf:about='http://quitte.de/dsp/caps.html#EqFA4p'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2609.0"
+ ladspa:hasLabel="a.act">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="off" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="on" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2609.4"
+ ladspa:hasLabel="b.act">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="off" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="on" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2609.8"
+ ladspa:hasLabel="c.act">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="off" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="on" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2609.12"
+ ladspa:hasLabel="d.act">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="off" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="on" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2609.17"
+ ladspa:hasLabel="_latency">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="3"
+ ladspa:hasLabel="3 samples" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
</ladspa:EQPlugin>
+ <!-- Wider -->
+ <ladspa:DelayPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Wider'>
+ </ladspa:DelayPlugin>
+
<!-- Narrower -->
<ladspa:FilterPlugin
- rdf:about='http://ladspa.org/ontology#2595'>
+ rdf:about='http://quitte.de/dsp/caps.html#Narrower'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;2595.0"
+ ladspa:hasLabel="mode">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="crossfeed mixing" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="mid/side processing" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
</ladspa:FilterPlugin>
- <!-- Goodbye, Dept. of Redundancy Dept. Dept. Dept. -->
+ <!-- Sin -->
+ <ladspa:OscillatorPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Sin'>
+ </ladspa:OscillatorPlugin>
+
+ <!-- White -->
+ <ladspa:GeneratorPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#White'>
+ </ladspa:GeneratorPlugin>
+
+ <!-- Fractal -->
+ <ladspa:GeneratorPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Fractal'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;1774.1"
+ ladspa:hasLabel="mode">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="lorenz" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="roessler" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ </ladspa:GeneratorPlugin>
+
+ <!-- Click -->
+ <ladspa:TimePlugin
+ rdf:about='http://quitte.de/dsp/caps.html#Click'>
+ <ladspa:hasPort>
+ <ladspa:InputControlPort rdf:about="&ladspa;1769.0"
+ ladspa:hasLabel="model">
+ <ladspa:hasScale><ladspa:Scale>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="0"
+ ladspa:hasLabel="box" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="1"
+ ladspa:hasLabel="stick" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="2"
+ ladspa:hasLabel="beep" />
+ </ladspa:hasPoint>
+ <ladspa:hasPoint><ladspa:Point
+ rdf:value="3"
+ ladspa:hasLabel="dirac" />
+ </ladspa:hasPoint>
+ </ladspa:Scale></ladspa:hasScale>
+ </ladspa:InputControlPort>
+ </ladspa:hasPort>
+ </ladspa:TimePlugin>
+
+ <!-- CEO -->
+ <ladspa:OscillatorPlugin
+ rdf:about='http://quitte.de/dsp/caps.html#CEO'>
+ </ladspa:OscillatorPlugin>
+ <!-- Goodbye, Dept. of Redundancy Dept. -->
</rdf:RDF>
diff --git a/plugins/LadspaEffect/caps/configure.py b/plugins/LadspaEffect/caps/configure.py
index 227a1328f..82cfc3537 100755
--- a/plugins/LadspaEffect/caps/configure.py
+++ b/plugins/LadspaEffect/caps/configure.py
@@ -2,29 +2,43 @@
import os
CFLAGS = []
-OSX_LDFLAGS = "-bundle -undefined suppress -flat_namespace"
+ARCH = []
+
+def dude_we_think_so_different():
+ try: return 'Darwin' == os.popen ('uname -s').read().strip()
+ except: return 0
+OSX = dude_we_think_so_different()
+if OSX:
+ OSX_LDFLAGS = "-bundle -undefined suppress -flat_namespace"
+ ARCH += ("-arch","i386","-arch","x86_64")
+ import ctypes as C
+ libc = C.CDLL("libc.dylib")
+ def osx_query(k):
+ size = C.c_uint(0)
+ libc.sysctlbyname(k,None,C.byref(size),None,0)
+ buf = C.create_string_buffer(size.value)
+ libc.sysctlbyname(k,buf,C.byref(size),None,0)
+ return buf.value
def we_have_sse():
+ if OSX: return osx_query("hw.optional.sse2")
try: return 'sse' in open ('/proc/cpuinfo').read().split()
except: return 0
def we_have_ssse3():
+ if OSX: return osx_query("hw.optional.sse3")
try: return 'ssse3' in open ('/proc/cpuinfo').read().split()
except: return 0
-
-def we_think_so_different_dude():
- try: return 'Darwin' == os.popen ('uname -s').read().strip()
- except: return 0
-
+
def store():
f = open ('defines.make', 'w')
f.write ("_CFLAGS=" + ' '.join (CFLAGS) + "\n")
- if we_think_so_different_dude():
+ f.write ("ARCH=" + ' '.join (ARCH) + "\n")
+ if OSX:
f.write ("_LDFLAGS=" + OSX_LDFLAGS + "\n")
f.write ("STRIP = echo\n")
-
+ os.system ("make clean")
+
if __name__ == '__main__':
- if we_have_sse():
- CFLAGS += ('-msse', '-mfpmath=sse')
- if we_have_ssse3():
- CFLAGS += ('-msse3',)
+ if we_have_sse(): CFLAGS += ('-msse', '-mfpmath=sse')
+ if we_have_ssse3(): CFLAGS += ('-msse3',)
store()
diff --git a/plugins/LadspaEffect/caps/dsp/Delay.h b/plugins/LadspaEffect/caps/dsp/Delay.h
index a9aa517ec..3be9972a5 100644
--- a/plugins/LadspaEffect/caps/dsp/Delay.h
+++ b/plugins/LadspaEffect/caps/dsp/Delay.h
@@ -1,7 +1,7 @@
/*
dsp/Delay.h
- Copyright 2003-4, 2010 Tim Goetze <tim@quitte.de>
+ Copyright 2003-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -9,13 +9,13 @@
and an allpass interpolating tap (which needs more work).
delay line storage is aligned to powers of two for simplified wrapping
- checks (no conditional or modulo, binary and suffices instead).
+ checks (no conditional or modulo, binary 'and' suffices instead).
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -29,8 +29,8 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DSP_DELAY_H_
-#define _DSP_DELAY_H_
+#ifndef DSP_DELAY_H
+#define DSP_DELAY_H
#include "util.h"
#include "FPTruncateMode.h"
@@ -40,26 +40,20 @@ namespace DSP {
class Delay
{
public:
- int size;
+ uint size;
sample_t * data;
- int read, write;
+ uint read, write;
- Delay()
- {
- read = write = 0;
- data = 0;
- }
+ Delay() { read = write = 0; data = 0; }
- ~Delay()
- {
- if (data) free (data);
- }
+ ~Delay() { free (data); }
- void init (int n)
+ void init (uint n)
{
size = next_power_of_2 (n);
+ assert (size <= (1 << 20));
data = (sample_t *) calloc (sizeof (sample_t), size);
- size -= 1;
+ --size; /* used as mask for confining access */
write = n;
}
@@ -68,37 +62,25 @@ class Delay
memset (data, 0, (size + 1) * sizeof (sample_t));
}
- sample_t &
- operator [] (int i)
- {
- return data [(write - i) & size];
- }
+ sample_t & operator [] (int i) { return data [(write - i) & size]; }
- inline void
- put (sample_t x)
+ inline void put (sample_t x)
{
data [write] = x;
write = (write + 1) & size;
}
- inline sample_t
- get()
+ inline sample_t get()
{
sample_t x = data [read];
read = (read + 1) & size;
return x;
}
-
- inline sample_t
- putget (sample_t x)
- {
- put (x);
- return get();
- }
+ inline sample_t peek() { return data [read]; }
+ inline sample_t putget (sample_t x) {put(x); return get();}
/* fractional lookup, linear interpolation */
- inline sample_t
- get_at (float f)
+ inline sample_t get_linear (float f)
{
int n;
fistp (f, n); /* read: i = (int) f; relies on FPTruncateMode */
@@ -108,8 +90,7 @@ class Delay
}
/* fractional lookup, cubic interpolation */
- inline sample_t
- get_cubic (float f)
+ inline sample_t get_cubic (float f)
{
int n;
fistp (f, n); /* see FPTruncateMode */
@@ -132,40 +113,38 @@ class Delay
}
};
-/* allpass variant */
-
-class DelayTapA
+class MovingAverage
+: public Delay
{
public:
- sample_t x1, y1;
+ sample_t state, over_n;
- DelayTapA()
+ void init (uint n)
{
- reset();
+ this->Delay::init (n);
+ over_n = 1. / n;
+ /* adjust write pointer so we have a full history of zeros */
+ write = (write + size + 1) & size;
+ state = 0;
}
void reset()
{
- x1 = y1 = 0;
+ this->Delay::reset();
+ state = 0;
}
- sample_t get (Delay & d, float f)
- {
- int n;
- fistp (f, n); /* read: n = (int) f; relies on FPTruncateMode */
- f -= n;
- if (0 && f < .5)
- f += 1,
- n -= 1;
-
- sample_t x = d[n];
- f = (1 - f) / (1 + f);
- y1 = x1 + f * x - f * y1;
- x1 = x;
- return y1;
+ void process (sample_t x)
+ {
+ x *= over_n;
+ state -= this->Delay::get();
+ state += x;
+ this->Delay::put (x);
}
+
+ sample_t get() { return state; }
};
}; /* namespace DSP */
-#endif /* _DSP_DELAY_H_ */
+#endif /* DSP_DELAY_H */
diff --git a/plugins/LadspaEffect/caps/dsp/Eq.h b/plugins/LadspaEffect/caps/dsp/Eq.h
index 92639e8a1..feac91443 100644
--- a/plugins/LadspaEffect/caps/dsp/Eq.h
+++ b/plugins/LadspaEffect/caps/dsp/Eq.h
@@ -1,18 +1,17 @@
/*
Eq.h
- Copyright 2004-7 Tim Goetze <tim@quitte.de>
+ Copyright 2004-14 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- Equalizer circuit using recursive filtering.
- Based on a motorola paper implementing a similar circuit on a DSP56001.
+ Filter prototypes from a motorola paper implementing a similar circuit on a DSP56k.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -26,47 +25,21 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DSP_EQ_H_
-#define _DSP_EQ_H_
+#ifndef DSP_EQ_H
+#define DSP_EQ_H
namespace DSP {
-/* A single bandpass as used by the Eq, expressed as a biquad. Like all
- * band-pass filters I know of, the filter works with a FIR coefficient of 0
- * for x[-1], so a generic biquad isn't the optimum implementation.
- *
- * This routine isn't used anywhere, just here for testing purposes.
- */
-template <class T>
-void
-_BP (double fc, double Q, T * ca, T * cb)
-{
- double theta = 2 * fc * M_PI;
-
- double
- b = (Q - theta * .5) / (2 * Q + theta),
- a = (.5 - b) / 2,
- c = (.5 + b) * cos (theta);
-
- ca[0] = 2 * a;
- ca[1] = 0;
- ca[2] = -2 * a;
-
- cb[0] = 0;
- cb[1] = 2 * c;
- cb[2] = -2 * b;
-}
-
template <int Bands, class eq_sample = float>
class Eq
{
public:
/* recursion coefficients, 3 per band */
- eq_sample __attribute__ ((aligned)) a[Bands], b[Bands], c[Bands];
+ eq_sample a[Bands], b[Bands], c[Bands];
/* past outputs, 2 per band */
- eq_sample __attribute__ ((aligned)) y[2][Bands];
+ eq_sample y[2][Bands];
/* current gain and recursion factor, each 1 per band = 2 */
- eq_sample __attribute__ ((aligned)) gain[Bands], gf[Bands];
+ eq_sample gain[Bands], gf[Bands];
/* input history */
eq_sample x[2];
/* history index */
@@ -83,10 +56,11 @@ class Eq
void reset()
{
for (int z = 0; z < 2; ++z)
- {
- memset( y[z], 0, Bands*sizeof( eq_sample ) );
- x[z] = 0;
- }
+ for (int i = 0; i < Bands; ++i)
+ y[z][i] = 0;
+
+ for (int i = 0; i < 2; ++i)
+ x[i] = 0;
}
void init (double fs, double Q)
@@ -94,8 +68,8 @@ class Eq
double f = 31.25;
int i = 0;
- for (i = 0; i < Bands && f < fs / 2; ++i, f *= 2)
- init_band (i, 2 * f * M_PI / fs, Q);
+ for (i = 0; i < Bands && f < .48*fs; ++i, f *= 2)
+ init_band (i, 2*f*M_PI/fs, Q);
/* just in case, zero the remaining coefficients */
for ( ; i < Bands; ++i)
zero_band (i);
@@ -105,21 +79,18 @@ class Eq
void init_band (int i, double theta, double Q)
{
- b[i] = (Q - theta * .5) / (2 * Q + theta);
- a[i] = (.5 - b[i]) / 2;
- c[i] = (.5 + b[i]) * cos (theta);
+ b[i] = (Q - .5*theta)/(2*Q + theta);
+ a[i] = (.5 - b[i])*.5;
+ c[i] = (.5 + b[i])*cos(theta);
/* fprintf (stderr, "%02d %f %f %f\n", i, a[i], b[i], c[i]); */
gain[i] = 1;
gf[i] = 1;
}
- void zero_band (int i)
- {
- a[i] = b[i] = c[i] = 0;
- }
+ void zero_band (int i) { a[i] = b[i] = c[i] = 0; }
/* per-band recursion:
- * y = 2 * (a * (x - x[-2]) + c * y[-1] - b * y[-2])
+ * y = 2*(a*(x - x[-2]) + c*y[-1] - b*y[-2])
*/
eq_sample process (eq_sample s)
{
@@ -133,7 +104,7 @@ class Eq
for (int i = 0; i < Bands; ++i)
{
- y2[i] = normal + 2 * (a[i] * x_x2 + c[i] * y1[i] - b[i] * y2[i]);
+ y2[i] = normal + 2*(a[i]*x_x2 + c[i]*y1[i] - b[i]*y2[i]);
r += gain[i] * y2[i];
gain[i] *= gf[i];
}
@@ -155,4 +126,4 @@ class Eq
} /* namespace DSP */
-#endif /* _DSP_EQ_H_ */
+#endif /* DSP_EQ_H */
diff --git a/plugins/LadspaEffect/caps/dsp/FIR.h b/plugins/LadspaEffect/caps/dsp/FIR.h
index 8cb69ef0f..20c5fe318 100644
--- a/plugins/LadspaEffect/caps/dsp/FIR.h
+++ b/plugins/LadspaEffect/caps/dsp/FIR.h
@@ -1,17 +1,25 @@
/*
dsp/FIR.h
- Copyright 2003-10 Tim Goetze <tim@quitte.de>
+ Copyright 2003-12 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
finite impulse response filters, with options for up- and down-sampling.
*/
+/*
+ Somewhat surprisingly, exploiting the symmetry of linear-phase filter
+ kernels by only storing one half of the kernel does not improve
+ performance on this Core2 chip, on the contrary. apparently,
+ memory bandwidth (the full kernel likely being read into the CPU
+ cache as soon as it is accessed) is not an issue, but additional index
+ arithmetic becomes one.
+*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,59 +33,32 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _FIR_H_
-#define _FIR_H_
+#ifndef FIR_H
+#define FIR_H
#include "util.h"
namespace DSP {
-/* brute-force FIR filter with downsampling method (decimating).
- *
- * CAVEAT: constructing it from another FIR makes the filter share the other's
- * kernel data set. IOW, the other FIR must be valid throughout the lifetime
- * of this instance.
- */
+/*
+ Brute-force FIR filter with downsampling method (decimating).
+*/
class FIR
{
public:
/* kernel length, history length - 1 */
- int n, m;
+ uint n, m;
/* coefficients, history */
sample_t * c, * x;
- bool borrowed_kernel;
/* history index */
int h;
- FIR (int N)
- {
- c = 0;
- init (N);
- }
-
- FIR (FIR & fir)
- {
- c = fir.c;
- init (fir.n);
- }
-
- FIR (int n, sample_t * kernel)
- {
- c = 0;
- init (n);
- memcpy (c, kernel, n * sizeof (*c));
- }
-
- ~FIR()
- {
- if (!borrowed_kernel)
- free (c);
- free (x);
- }
+ FIR() { c = x = 0; }
+ ~FIR() { free(c); free(x); }
- void init (int N)
+ void init (uint N)
{
n = N;
@@ -85,12 +66,7 @@ class FIR
* history pointer by & instead of %, saving a few cpu cycles. */
m = next_power_of_2 (n);
- if (c)
- borrowed_kernel = true;
- else
- borrowed_kernel = false,
- c = (sample_t *) malloc (n * sizeof (sample_t));
-
+ c = (sample_t *) malloc (n * sizeof (sample_t));
x = (sample_t *) malloc (m * sizeof (sample_t));
m -= 1;
@@ -104,14 +80,13 @@ class FIR
memset (x, 0, n * sizeof (sample_t));
}
- /* TODO: write an SSE-enabled version */
inline sample_t process (sample_t s)
{
x[h] = s;
s *= c[0];
- for (int Z = 1, z = h - 1; Z < n; --z, ++Z)
+ for (uint Z = 1, z = h - 1; Z < n; --z, ++Z)
s += c[Z] * x[z & m];
h = (h + 1) & m;
@@ -123,7 +98,7 @@ class FIR
* OVER is the oversampling factor. just here for documentation, use
* a FIRUpsampler instead.
*/
- template <int Z, int OVER>
+ template <uint Z, uint OVER>
inline sample_t upsample (sample_t s)
{
x[h] = s;
@@ -133,7 +108,7 @@ class FIR
/* for the interpolation, iterate over the history in z ^ -OVER
* steps -- all the samples between are 0.
*/
- for (int j = Z, z = h - Z; j < n; --z, j += OVER)
+ for (uint j = Z, z = h - Z; j < n; --z, j += OVER)
s += c[j] * x[z & m];
h = (h + 1) & m;
@@ -149,40 +124,20 @@ class FIR
}
};
-/* close relative of FIR, but distinct enough to not justify inheritance.
- *
- * the difference to the FIR is the shorter history length. don't need
- * to clutter the d-cache with interleaved 0s.
- *
- * however, an initial test shows this to be a fraction *slower* than a
- * complete FIR for N = 32, OVER = 4.
- */
+/* FIR upsampler, optimised not to store the 0 samples */
+template <int N, int Oversample>
class FIRUpsampler
{
public:
- /* kernel length, history length - 1 */
- int n, m;
+ uint m; /* history length - 1 */
+ int h; /* history index */
- /* oversampling ratio */
- int over;
-
- /* coefficients, history */
- sample_t * c, * x;
+ sample_t * c, * x; /* coefficients, history */
- /* history index */
- int h;
-
- FIRUpsampler (int _n, int _over)
+ FIRUpsampler()
{
c = x = 0;
- init (_n, _over);
- }
-
- FIRUpsampler (FIR & fir, int _over)
- {
- c = x = 0;
- init (fir.n, _over);
- memcpy (c, fir.c, n * sizeof (sample_t));
+ init();
}
~FIRUpsampler()
@@ -191,20 +146,16 @@ class FIRUpsampler
if (x) free (x);
}
- void init (int _n, int _over)
+ void init()
{
- /* oversampling ratio must be multiple of FIR kernel length */
- // assert (_n % _over == 0);
-
- n = _n;
- over = _over;
+ /* FIR kernel length must be a multiple of the oversampling ratio */
+ assert (N % Oversample == 0);
/* like FIR, keep the history buffer a power of 2; additionally,
- * compress and don't store the 0 samples inbetween.
- */
- m = next_power_of_2 ((n + over - 1) / over);
+ * don't store the 0 samples inbetween. */
+ m = next_power_of_2 ((N + Oversample - 1) / Oversample);
- c = (sample_t *) malloc (n * sizeof (sample_t));
+ c = (sample_t *) malloc (N * sizeof (sample_t));
x = (sample_t *) malloc (m * sizeof (sample_t));
m -= 1;
@@ -225,7 +176,7 @@ class FIRUpsampler
s = 0;
- for (int Z = 0, z = h; Z < n; --z, Z += over)
+ for (uint Z = 0, z = h; Z < N; --z, Z += Oversample)
s += c[Z] * x[z & m];
h = (h + 1) & m;
@@ -235,18 +186,70 @@ class FIRUpsampler
/* upsample a zero sample (interleaving), Z being the time, in samples,
* since the last non-0 sample. */
- inline sample_t pad (int Z)
+ inline sample_t pad (uint Z)
{
sample_t s = 0;
- for (int z = h - 1; Z < n; --z, Z += over)
+ for (uint z = h-1; Z < N; --z, Z += Oversample)
+ s += c[Z] * x[z & m];
+
+ return s;
+ }
+
+};
+
+/* templating for kernel length allows g++ to optimise aggressively
+ * resulting in appreciable performance gains. */
+template <int N>
+class FIRn
+{
+ public:
+ /* history length - 1 */
+ uint m;
+
+ /* coefficients, history */
+ sample_t c[N], x[N];
+
+ /* history index */
+ int h;
+
+ FIRn()
+ {
+ /* keeping history size a power of 2 makes it possible to wrap the
+ * history pointer by & instead of %, saving a few cpu cycles. */
+ m = next_power_of_2 (N) - 1;
+
+ reset();
+ }
+
+ void reset()
+ {
+ h = 0;
+ memset (x, 0, N * sizeof (sample_t));
+ }
+
+ inline sample_t process (sample_t s)
+ {
+ x[h] = s;
+
+ s *= c[0];
+
+ for (uint Z = 1, z = h - 1; Z < N; --z, ++Z)
s += c[Z] * x[z & m];
+ h = (h + 1) & m;
+
return s;
}
+ /* used in downsampling */
+ inline void store (sample_t s)
+ {
+ x[h] = s;
+ h = (h + 1) & m;
+ }
};
-}; /* namespace DSP */
+} /* namespace DSP */
-#endif /* _FIR_H_ */
+#endif /* FIR_H */
diff --git a/plugins/LadspaEffect/caps/dsp/FPTruncateMode.h b/plugins/LadspaEffect/caps/dsp/FPTruncateMode.h
index 354d062be..8900c39de 100644
--- a/plugins/LadspaEffect/caps/dsp/FPTruncateMode.h
+++ b/plugins/LadspaEffect/caps/dsp/FPTruncateMode.h
@@ -1,19 +1,42 @@
-/* Copyright 2001-4 tim goetze <tim@quitte.de> -- see 'COPYING'. */
-
-/* Sets the FP rounding mode to 'truncate' in the constructor
- * and loads the previous FP conrol word in the destructor.
- *
- * By directly using the machine instruction to convert float to int
- * we avoid the performance hit that loading the control word twice for
- * every (int) cast causes on i386.
- *
- * On other architectures this is a no-op.
- */
-
-#ifndef _DSP_FP_TRUNCATE_MODE_H_
-#define _DSP_FP_TRUNCATE_MODE_H_
-
-#ifdef __i386__
+/*
+ FPTruncateMode.h
+
+ Copyright 2001-11 Tim Goetze <tim@quitte.de>
+
+ http://quitte.de/dsp/
+
+ Sets the FP rounding mode to 'truncate' in the constructor
+ and loads the previous FP conrol word in the destructor.
+
+ By directly using the machine instruction to convert float to int
+ we avoid the performance hit that loading the control word twice for
+ every (int) cast causes on i386.
+
+ On other architectures this is a no-op.
+
+*/
+/*
+ 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 3
+ 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., 59 Temple Place - Suite 330, Boston, MA
+ 02111-1307, USA or point your web browser to http://www.gnu.org.
+*/
+
+
+#ifndef DSP_FP_TRUNCATE_MODE_H
+#define DSP_FP_TRUNCATE_MODE_H
+
+#if defined(__i386__) && !defined(__SSE2__)
#define fstcw(i) \
__asm__ __volatile__ ("fstcw %0" : "=m" (i))
@@ -32,6 +55,14 @@
#endif
namespace DSP {
+
+static inline int
+fast_trunc (float f)
+{
+ int i;
+ fistp (f, i);
+ return i;
+}
class FPTruncateMode
{
@@ -53,4 +84,4 @@ class FPTruncateMode
} /* namespace DSP */
-#endif /* _DSP_FP_TRUNCATE_MODE_H_ */
+#endif /* DSP_FP_TRUNCATE_MODE_H */
diff --git a/plugins/LadspaEffect/caps/dsp/Lorenz.h b/plugins/LadspaEffect/caps/dsp/Lorenz.h
index bb57a7cc3..5247c43a1 100644
--- a/plugins/LadspaEffect/caps/dsp/Lorenz.h
+++ b/plugins/LadspaEffect/caps/dsp/Lorenz.h
@@ -1,7 +1,7 @@
/*
dsp/Lorenz.h
- Copyright 2001-4 Tim Goetze <tim@quitte.de>
+ Copyright 2001-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -11,7 +11,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,8 +25,8 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DSP_LORENZ_H_
-#define _DSP_LORENZ_H_
+#ifndef DSP_LORENZ_H
+#define DSP_LORENZ_H
namespace DSP {
@@ -56,16 +56,9 @@ class Lorenz
{
I = 0;
- x[0] = .1 + seed - frandom() * .1;
- y[0] = 0;
- z[0] = 0;
-
- /* progress quickly to get a 'stable' system */
- h = .001;
-
- int n = 10000 + min ((int) (10000 * seed), 10000);
- for (int i = 0; i < n; ++i)
- step();
+ x[0] = -2.884960 + seed;
+ y[0] = -5.549104;
+ z[0] = 7.801511;
h = _h;
}
@@ -80,29 +73,23 @@ class Lorenz
{
int J = I ^ 1;
- x[J] = x[I] + h * a * (y[I] - x[I]);
- y[J] = y[I] + h * (x[I] * (b - z[I]) - y[I]);
- z[J] = z[I] + h * (x[I] * y[I] - c * z[I]);
+ x[J] = x[I] + h*a*(y[I] - x[I]);
+ y[J] = y[I] + h*(x[I] * (b - z[I]) - y[I]);
+ z[J] = z[I] + h*(x[I] * y[I] - c * z[I]);
I = J;
}
-
- double get_x()
- {
- return .024 * (x[I] - .172);
- }
-
- double get_y()
- {
- return .018 * (y[I] - .172);
- }
-
- double get_z()
- {
- return .019 * (z[I] - 25.43);
- }
+#if 0
+ double get_x() {return x[I];}
+ double get_y() {return y[I];}
+ double get_z() {return z[I];}
+#else
+ double get_x() {return (x[I]+0.01661)*-.04;}
+ double get_y() {return (y[I]-0.02379)*-.03;}
+ double get_z() {return (z[I]-24.1559)*.03;}
+#endif
};
} /* namespace DSP */
-#endif /* _DSP_LORENZ_H_ */
+#endif /*DSP_LORENZ_H */
diff --git a/plugins/LadspaEffect/caps/dsp/RBJ.h b/plugins/LadspaEffect/caps/dsp/RBJ.h
index 0a1db6b39..7f0acb977 100644
--- a/plugins/LadspaEffect/caps/dsp/RBJ.h
+++ b/plugins/LadspaEffect/caps/dsp/RBJ.h
@@ -1,17 +1,15 @@
/*
dsp/RBJ.h
- Copyright
- 1998 Robert Bristow-Johnson
- 2004-10 Tim Goetze <tim@quitte.de>
+ Copyright 2004-13 Tim Goetze <tim@quitte.de>
+
+ Filter prototypes from Robert Bristow-Johnson's lovely eq cookbook.
- biquad prototypes according to the eq cookbook. easy-to-use, nice,
- predictable filters. thanks rbj!
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,10 +23,10 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DSP_RBJ_H_
-#define _DSP_RBJ_H_
+#ifndef DSP_RBJ_H
+#define DSP_RBJ_H
-#include "BiQuad.h"
+#include "IIR2.h"
namespace DSP {
namespace RBJ {
@@ -65,11 +63,11 @@ class RBJ
ca[2] = b[2] * a0i;
/* our bi-quad implementation /adds/ b[i] * y[i] so we need to
- * toggle the sign for the b[] coefficients.
- */
- cb[0] = 0;
- cb[1] = -a[1] * a0i;
- cb[2] = -a[2] * a0i;
+ * toggle the sign for the b[] coefficients. */
+ a0i = -a0i;
+
+ cb[1] = a[1] * a0i;
+ cb[2] = a[2] * a0i;
}
};
@@ -80,12 +78,9 @@ class LP
: public RBJ
{
public:
- LP (double f, double Q, BiQuad & bq) : RBJ (f, Q)
- { ab (bq.a, bq.b); }
-
template <class T>
- LP (double f, double Q, T * ca, T * cb) : RBJ (f, Q)
- { ab (ca, cb); }
+ LP (double f, double Q, T& filter) : RBJ (f, Q)
+ { ab (filter.a, filter.b); }
template <class T>
void ab (T * ca, T * cb)
@@ -106,12 +101,9 @@ class BP
: public RBJ
{
public:
- BP (double f, double Q, BiQuad & bq) : RBJ (f, Q)
- { ab (bq.a, bq.b); }
-
template <class T>
- BP (double f, double Q, T * ca, T * cb) : RBJ (f, Q)
- { ab (ca, cb); }
+ BP (double f, double Q, T& filter) : RBJ (f, Q)
+ { ab (filter.a, filter.b); }
template <class T>
void ab (T * ca, T * cb)
@@ -132,12 +124,9 @@ class HP
: public RBJ
{
public:
- HP (double f, double Q, BiQuad & bq) : RBJ (f, Q)
- { ab (bq.a, bq.b); }
-
template <class T>
- HP (double f, double Q, T * ca, T * cb) : RBJ (f, Q)
- { ab (ca, cb); }
+ HP (double f, double Q, T& filter) : RBJ (f, Q)
+ { ab (filter.a, filter.b); }
template <class T>
void ab (T * ca, T * cb)
@@ -158,12 +147,9 @@ class Notch
: public RBJ
{
public:
- Notch (double f, double Q, BiQuad & bq) : RBJ (f, Q)
- { ab (bq.a, bq.b); }
-
template <class T>
- Notch (double f, double Q, T * ca, T * cb) : RBJ (f, Q)
- { ab (ca, cb); }
+ Notch (double f, double Q, T& filter) : RBJ (f, Q)
+ { ab (filter.a, filter.b); }
template <class T>
void ab (T * ca, T * cb)
@@ -180,6 +166,30 @@ class Notch
}
};
+class AllPass
+: public RBJ
+{
+ public:
+ template <class T>
+ AllPass (double f, double Q, T& filter) : RBJ (f, Q)
+ { ab (filter.a, filter.b); }
+
+ template <class T>
+ void ab (T * ca, T * cb)
+ {
+ b[0] = 1 - alpha;
+ b[1] = -2 * cos;
+ b[2] = 1 + alpha;
+
+ a[0] = 1 + alpha;
+ a[1] = -2 * cos;
+ a[2] = 1 - alpha;
+
+ make_direct_I (ca, cb);
+ }
+};
+
+
/* shelving and peaking dept. ////////////////////////////////////////////// */
class PeakShelve
@@ -194,7 +204,7 @@ class PeakShelve
{
A = pow (10, dB * .025);
double S = Q; /* slope */
- beta = sqrt ((A * A + 1) / S - (A - 1) * (A - 1));
+ beta = sqrt ((A*A + 1)/S - (A-1)*(A-1));
}
};
@@ -202,28 +212,24 @@ class LoShelve
: public PeakShelve
{
public:
- LoShelve (double f, double Q, double dB, BiQuad & bq)
- : PeakShelve (f, Q, dB)
- { ab (bq.a, bq.b); }
-
template <class T>
- LoShelve (double f, double Q, double dB, T * ca, T * cb)
+ LoShelve (double f, double Q, double dB, T& filter)
: PeakShelve (f, Q, dB)
- { ab (ca, cb); }
+ { ab (filter.a, filter.b); }
template <class T>
void ab (T * ca, T * cb)
{
- double Ap1 = A + 1, Am1 = A - 1;
- double beta_sin = beta * sin;
+ double Ap1 = A+1, Am1 = A-1;
+ double beta = 2*sqrt(A)*alpha;
- b[0] = A * (Ap1 - Am1 * cos + beta_sin);
- b[1] = 2 * A * (Am1 - Ap1 * cos);
- b[2] = A * (Ap1 - Am1 * cos - beta_sin);
+ b[0] = A*(Ap1 - Am1*cos + beta);
+ b[1] = 2*A*(Am1 - Ap1*cos);
+ b[2] = A*(Ap1 - Am1*cos - beta);
- a[0] = Ap1 + Am1 * cos + beta_sin;
- a[1] = -2 * (Am1 + Ap1 * cos);
- a[2] = Ap1 + Am1 * cos - beta_sin;
+ a[0] = Ap1 + Am1*cos + beta;
+ a[1] = -2 * (Am1 + Ap1*cos);
+ a[2] = Ap1 + Am1*cos - beta;
make_direct_I (ca, cb);
}
@@ -233,14 +239,10 @@ class PeakingEQ
: public PeakShelve
{
public:
- PeakingEQ (double f, double Q, double dB, BiQuad & bq)
- : PeakShelve (f, Q, dB)
- { ab (bq.a, bq.b); }
-
template <class T>
- PeakingEQ (double f, double Q, double dB, T * ca, T * cb)
+ PeakingEQ (double f, double Q, double dB, T& filter)
: PeakShelve (f, Q, dB)
- { ab (ca, cb); }
+ { ab (filter.a, filter.b); }
template <class T>
void ab (T * ca, T * cb)
@@ -261,28 +263,24 @@ class HiShelve
: public PeakShelve
{
public:
- HiShelve (double f, double Q, double dB, BiQuad & bq)
- : PeakShelve (f, Q, dB)
- { ab (bq.a, bq.b); }
-
template <class T>
- HiShelve (double f, double Q, double dB, T * ca, T * cb)
+ HiShelve (double f, double Q, double dB, T& filter)
: PeakShelve (f, Q, dB)
- { ab (ca, cb); }
+ { ab (filter.a, filter.b); }
template <class T>
void ab (T * ca, T * cb)
{
double Ap1 = A + 1, Am1 = A - 1;
- double beta_sin = beta * sin;
+ double beta = 2*sqrt(A)*alpha;
- b[0] = A * (Ap1 + Am1 * cos + beta_sin);
+ b[0] = A * (Ap1 + Am1 * cos + beta);
b[1] = -2 * A * (Am1 + Ap1 * cos);
- b[2] = A * (Ap1 + Am1 * cos - beta_sin);
+ b[2] = A * (Ap1 + Am1 * cos - beta);
- a[0] = Ap1 - Am1 * cos + beta_sin;
+ a[0] = Ap1 - Am1 * cos + beta;
a[1] = 2 * (Am1 - Ap1 * cos);
- a[2] = Ap1 - Am1 * cos - beta_sin;
+ a[2] = Ap1 - Am1 * cos - beta;
make_direct_I (ca, cb);
}
@@ -291,4 +289,4 @@ class HiShelve
} /* ~namespace RBJ */
} /* ~namespace DSP */
-#endif /* _DSP_RBJ_H_ */
+#endif /* DSP_RBJ_H */
diff --git a/plugins/LadspaEffect/caps/dsp/RMS.h b/plugins/LadspaEffect/caps/dsp/RMS.h
index 3b7806729..4b21c0384 100644
--- a/plugins/LadspaEffect/caps/dsp/RMS.h
+++ b/plugins/LadspaEffect/caps/dsp/RMS.h
@@ -1,17 +1,17 @@
/*
dsp/RMS.h
- Copyright 2004 Tim Goetze <tim@quitte.de>
+ Copyright 2004-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- root-mean-square accumulator.
+ Sliding window, running sum root-mean-square.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,22 +25,26 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DSP_RMS_H_
-#define _DSP_RMS_H_
+#ifndef DSP_RMS_H
+#define DSP_RMS_H
namespace DSP {
+template <int N>
class RMS
{
protected:
- sample_t buffer[64];
+ sample_t buffer[N];
int write;
public:
- double sum;
+ double sum, over_N;
RMS()
{
+ /* g++ doesn't reliably optimise the division in get() below into
+ * a multiplication with 1/N */
+ over_N = 1./N;
write = 0;
reset();
}
@@ -51,27 +55,21 @@ class RMS
memset (buffer, 0, sizeof (buffer));
}
- /* caution: pass in the *squared* sample value */
+ /* caution: pass in the SQUARED! sample value */
void store (sample_t x)
{
sum -= buffer[write];
sum += (buffer[write] = x);
- write = (write + 1) & 63;
+ write = (write+1) & (N-1);
}
- sample_t process (sample_t x)
+ sample_t get()
{
- store (x);
- return rms();
- }
-
- sample_t rms()
- {
- /* fabs it before sqrt, just in case ... */
- return sqrt (fabs (sum) / 64);
+ /* lack of running sum accuracy necessitates fabs() */
+ return sqrt(fabs(sum * over_N));
}
};
} /* namespace DSP */
-#endif /* _DSP_RMS_H_ */
+#endif /* DSP_RMS_H */
diff --git a/plugins/LadspaEffect/caps/dsp/Roessler.h b/plugins/LadspaEffect/caps/dsp/Roessler.h
index e4ec86303..f45d97b77 100644
--- a/plugins/LadspaEffect/caps/dsp/Roessler.h
+++ b/plugins/LadspaEffect/caps/dsp/Roessler.h
@@ -1,17 +1,17 @@
/*
dsp/Roessler.h
- Copyright 2003-4 Tim Goetze <tim@quitte.de>
+ Copyright 2003-13 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- Roessler fractal.
+ Roessler attractor.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,8 +25,8 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DSP_ROESSLER_H_
-#define _DSP_ROESSLER_H_
+#ifndef DSP_ROESSLER_H
+#define DSP_ROESSLER_H
namespace DSP {
@@ -58,43 +58,39 @@ class Roessler
I = 0;
- x[0] = .0001 + .0001 * seed;
- y[0] = .0001;
- z[0] = .0001;
-
- for (int i = 0; i < 5000; ++i)
- get();
+ x[0] = -0.327732 + .0001 * seed;
+ y[0] = 2.569375;
+ z[0] = 0.036099;
}
- sample_t get()
+ void step()
{
int J = I ^ 1;
- x[J] = x[I] + h * (- y[I] - z[I]);
- y[J] = y[I] + h * (x[I] + a * y[I]);
- z[J] = z[I] + h * (b + z[I] * (x[I] - c));
+ x[J] = x[I] + h*(-y[I] - z[I]);
+ y[J] = y[I] + h*(x[I] + a*y[I]);
+ z[J] = z[I] + h*(b + z[I]*(x[I] - c));
I = J;
-
- return x[I] * .01725 + z[I] * .015;
}
- double get_x()
- {
- return x[I];
- }
-
- double get_y()
+ sample_t get()
{
- return y[I];
+ step();
+ return x[I]*.01725 + z[I]*.015;
}
- double get_z()
- {
- return z[I];
- }
+#if 0
+ double get_x() { return x[I]; }
+ double get_y() { return y[I]; }
+ double get_z() { return z[I]; }
+#else
+ double get_x() {return (x[I]-0.22784)*-.08;}
+ double get_y() {return (y[I]+1.13942)*-.09;}
+ double get_z() {return (z[I]-1.13929)*.055;}
+#endif
};
} /* namespace DSP */
-#endif /* _DSP_ROESSLER_H_ */
+#endif /* DSP_ROESSLER_H */
diff --git a/plugins/LadspaEffect/caps/dsp/SVF.h b/plugins/LadspaEffect/caps/dsp/SVF.h
index ccd5734ab..7d387bf86 100644
--- a/plugins/LadspaEffect/caps/dsp/SVF.h
+++ b/plugins/LadspaEffect/caps/dsp/SVF.h
@@ -1,18 +1,20 @@
/*
dsp/SVF.h
- Copyright 2002-4 Tim Goetze <tim@quitte.de>
+ Copyright 2002-12 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- ladder filter in Chamberlin topology. supports largely independent
- f and Q adjustments and sweeps.
+ SVF*: State-variable filter in Chamberlin topology,
+ supports f and Q sweeps.
+
+ StackedSVF: building ladders from the above.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -26,7 +28,7 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
/*
- inspired by this music-dsp entry:
+ Based on this music-dsp entry:
State Variable Filter (Double Sampled, Stable)
Type : 2 Pole Low, High, Band, Notch and Peaking
@@ -69,14 +71,14 @@
output[i] = out;
}
*/
-
-#ifndef _DSP_SVF_H_
-#define _DSP_SVF_H_
+#ifndef DSP_SVF_H
+#define DSP_SVF_H
namespace DSP {
-template <int OVERSAMPLE>
-class SVF
+/* oversampling only makes sense for LP */
+template <int Oversample>
+class SVFI
{
protected:
/* loop parameters */
@@ -94,7 +96,7 @@ class SVF
High = 2
};
- SVF()
+ SVFI()
{
set_out (Low);
set_f_Q (.1, .1);
@@ -108,7 +110,7 @@ class SVF
void set_f_Q (double fc, double Q)
{
/* this is a very tight limit */
- f = min (.25, 2 * sin (M_PI * fc / OVERSAMPLE));
+ f = min (.25, 2 * sin (M_PI * fc / Oversample));
q = 2 * cos (pow (Q, .1) * M_PI * .5);
q = min (q, min (2., 2 / f - f * .5));
@@ -135,56 +137,179 @@ class SVF
{
x = qnorm * x;
- for (int pass = 0; pass < OVERSAMPLE; ++pass)
+ for (int pass = 0; pass < Oversample; ++pass)
{
hi = x - lo - q * band;
band += f * hi;
lo += f * band;
- /* zero-padding, not 0th order holding. */
+ /* zero padding */
x = 0;
}
- /* peak and notch outputs don't belong in the loop, put them
- * here (best in a template) if needed. */
-
return *out;
}
};
-template <int STACKED, int OVERSAMPLE>
+/* //////////////////////////////////////////////////////////////////////// */
+
+/*
+ Alternative SVF implementation based on (this is the corrected version):
+
+ From andy@cytomic.com Mon May 23 21:47:40 2011
+ Subject: Re: [music-dsp] Trapezoidal and other integration methods applied tomusical resonant filters
+ From: Andrew Simper <andy@cytomic.com>
+ To: A discussion list for music-related DSP <music-dsp@music.columbia.edu>
+ Reply-To: A discussion list for music-related DSP <music-dsp@music.columbia.edu>
+ Date: Tue, 17 May 2011 17:48:27 +1000
+
+ Hi Vadim,
+
+ I want to thank you for your excellent paper "Generation of
+ bandlimited sync transitions for sine waveforms", which I feel is the
+ most thorough coverage in a paper of the method I use in Synth Squad,
+ which I call "corrective grains". You also outline in that paper
+ another couple of methods which I didn't even know existed, so I thank
+ you for showing me some interesting new ways to approach the problem
+ of band limiting transitions in waveforms.
+
+ I'm not sure what a "non-zero impedance approach" is, but standard
+ circuit simulation stuff handles any number of arbitrary impedance
+ devices connected in any topology, so I'm happy to not try
+ re-inventing anything since I can apply these techniques in a brain
+ dead crank the handle type way and get great results for both the
+ linear and non-linear cases.
+
+ Thanks for letting me know about all the papers you know of, it sounds
+ like I should get on and publish something as it isn't really covered
+ in detail anywhere yet. As a start below I have included the
+ difference equation for the linear trapezoidal svf:
+
+ init:
+ v1 = v2 = 0;
+ v0z = v1z = v2z = 0;
+
+ process:
+ g = tan (pi * cutoff / samplerate);
+ k = damping factor (typically in the range 2 to 0);
+ v1z = v1;
+ v2z = v2;
+ v0 = input;
+ v1 = v1z + g * (v0 + v0z - 2*(g + k)*v1z - 2*v2z) / (1 + g*(g + k));
+ v2 = v2z + g * (v1 + v1z);
+ v0z = v0;
+
+ outputs (the same as the analog circuit):
+ band = v1;
+ low = v2;
+ high = v0 - k*v1 - v2;
+ notch = high + low;
+ peak = high - low;
+
+
+ Andy
+ --
+ cytomic - sound music software
+ skype: andrewsimper
+*/
+
+class SVFII
+{
+ public:
+ sample_t v[3];
+ sample_t k, g, c1, c2;
+ int out;
+
+ public:
+ SVFII() { out = 0; reset(); }
+
+ void set_out (int o) { out = 2-(o&1); }
+
+ void reset()
+ {
+ for (uint i=0; i < 3; ++i)
+ v[i] = 0;
+ }
+
+ void set_f_Q (sample_t fc, sample_t q)
+ {
+ k = (1-.99*q);
+ g = tan (fc*M_PI);
+ c1 = 2*(g+k);
+ c2 = g/(1 + g*(g+k));
+ }
+
+ sample_t process (sample_t x)
+ { _process (x); return v[out]; }
+
+ void _process (sample_t x)
+ {
+ sample_t v1 = v[1] + c2*(x + v[0] - c1*v[1] - 2*v[2]);
+ sample_t v2 = v[2] + g*(v1 + v[1]);
+ v[0] = x;
+ v[1] = v1;
+ v[2] = v2;
+ }
+
+ /*
+ outputs (the same as the analog circuit):
+ band = v1;
+ low = v2;
+ high = v0 - k*v1 - v2;
+ notch = high + low;
+ peak = high - low;
+ */
+
+ sample_t process_lp (sample_t x)
+ { _process (x); return v[2]; }
+ sample_t process_bp (sample_t x)
+ { _process (x); return v[1]; }
+ sample_t process_hp (sample_t x)
+ { _process (x); return v[0] - k*v[1] - v[2]; }
+};
+
+/* //////////////////////////////////////////////////////////////////////// */
+
+template <int N, class SVF>
class StackedSVF
{
public:
- SVF<OVERSAMPLE> svf [STACKED];
+ SVF svf[N];
void reset()
{
- for (int i = 0; i < STACKED; ++i)
+ for (int i = 0; i < N; ++i)
svf[i].reset();
}
void set_out (int out)
{
- for (int i = 0; i < STACKED; ++i)
+ for (int i = 0; i < N; ++i)
svf[i].set_out (out);
}
void set_f_Q (double f, double Q)
{
- for (int i = 0; i < STACKED; ++i)
+ for (int i = 0; i < N; ++i)
svf[i].set_f_Q (f, Q);
}
sample_t process (sample_t x)
{
- for (int i = 0; i < STACKED; ++i)
+ for (int i = 0; i < N; ++i)
x = svf[i].process (x);
+ return x;
+ }
+ template<clip_func_t clip>
+ sample_t process (sample_t x, sample_t g)
+ {
+ for (int i = 0; i < N; ++i)
+ x = clip (svf[i].process (g*x));
return x;
}
};
} /* namespace DSP */
-#endif /* _DSP_SVF_H_ */
+#endif /* DSP_SVF_H */
diff --git a/plugins/LadspaEffect/caps/dsp/Sine.h b/plugins/LadspaEffect/caps/dsp/Sine.h
index a54c66161..a3c0a2737 100644
--- a/plugins/LadspaEffect/caps/dsp/Sine.h
+++ b/plugins/LadspaEffect/caps/dsp/Sine.h
@@ -1,17 +1,18 @@
/*
dsp/Sine.h
- Copyright 2003-4 Tim Goetze <tim@quitte.de>
+ Copyright 2003-14 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- direct form I recursive sin() generator.
+ Direct form I recursive sin() generator. Utilising doubles
+ for stability.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,14 +26,14 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DSP_SINE_H_
-#define _DSP_SINE_H_
+#ifndef DSP_SINE_H
+#define DSP_SINE_H
namespace DSP {
class Sine
{
- protected:
+ public:
int z;
double y[2];
double b;
@@ -50,28 +51,28 @@ class Sine
set_f (f, fs, phase);
}
- Sine (double omega, double phase = 0.)
+ Sine (double w, double phase = 0.)
{
- set_f (omega, phase);
+ set_f (w, phase);
}
inline void set_f (double f, double fs, double phase)
{
- set_f (f * M_PI / fs, phase);
+ set_f (f*2*M_PI/fs, phase);
}
inline void set_f (double w, double phase)
{
- b = 2 * cos (w);
+ b = 2*cos(w);
y[0] = sin (phase - w);
- y[1] = sin (phase - w * 2);
+ y[1] = sin (phase - 2*w);
z = 0;
}
/* advance and return 1 sample */
inline double get()
{
- register double s = b * y[z];
+ register double s = b*y[z];
z ^= 1;
s -= y[z];
return y[z] = s;
@@ -79,17 +80,34 @@ class Sine
double get_phase()
{
- double x0 = y[z], x1 = b * y[z] - y[z^1];
- double phi = asin (x0);
+ double x0 = y[z], x1 = b*y[z] - y[z^1];
+ double phi = asin(x0);
- /* slope is falling, we're into the 2nd half. */
- if (x1 < x0)
- return M_PI - phi;
+ /* slope is falling: into the 2nd half. */
+ return x1 < x0 ? M_PI - phi : phi;
+ }
+};
+
+/* same as above but including a damping coefficient d */
+class DampedSine
+: public Sine
+{
+ public:
+ double d;
- return phi;
+ public:
+ DampedSine()
+ { d = 1; }
+
+ inline double get()
+ {
+ register double s = b * y[z];
+ z ^= 1;
+ s -= d * y[z];
+ return y[z] = d * s;
}
};
} /* namespace DSP */
-#endif /* _DSP_SINE_H_ */
+#endif /* DSP_SINE_H */
diff --git a/plugins/LadspaEffect/caps/dsp/TDFII.h b/plugins/LadspaEffect/caps/dsp/TDFII.h
index 0252bc47a..eb2028bb0 100644
--- a/plugins/LadspaEffect/caps/dsp/TDFII.h
+++ b/plugins/LadspaEffect/caps/dsp/TDFII.h
@@ -3,6 +3,7 @@
Copyright 2006-7
David Yeh <dtyeh@ccrma.stanford.edu> (implementation)
+ 2006-14
Tim Goetze <tim@quitte.de> (cosmetics)
transposed Direct Form II digital filter.
@@ -13,7 +14,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -27,23 +28,23 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DSP_TDFII_H_
-#define _DSP_TDFII_H_
+#ifndef DSP_TDFII_H
+#define DSP_TDFII_H
namespace DSP {
-// ORDER is the highest power of s in the transfer function
-template <int Order>
+/* N is the highest power of s in the transfer function */
+template <int N>
class TDFII
{
public:
- double a[Order + 1];
- double b[Order + 1];
- double h[Order + 1];
+ double a[N+1];
+ double b[N+1];
+ double h[N+1];
void reset()
{
- for (int i = 0; i <= Order; ++i)
+ for (int i = 0; i <= N; ++i)
h[i] = 0; // zero state
}
@@ -55,8 +56,9 @@ class TDFII
void clear()
{
- for (int i=0; i<= Order; i++)
- a[i] = b[i] = 1;
+ for (int i=0; i<= N; i++)
+ a[i] = b[i] = 0;
+ b[0] = 1;
}
/* per-band recursion:
@@ -64,12 +66,12 @@ class TDFII
*/
sample_t process (sample_t s)
{
- double y = h[0] + b[0] * s;
+ double y = h[0] + b[0]*s;
- for (int i = 1; i < Order; ++i)
- h[i - 1] = h[i] + b[i] * s - a[i] * y;
+ for (int i=1; i<N; ++i)
+ h[i-1] = h[i] + b[i]*s - a[i]*y;
- h[Order - 1] = b[Order] * s - a[Order] * y;
+ h[N-1] = b[N]*s - a[N]*y;
return (sample_t) y;
}
@@ -77,4 +79,4 @@ class TDFII
} /* namespace DSP */
-#endif /* _DSP_TDFII_H_ */
+#endif /* DSP_TDFII_H */
diff --git a/plugins/LadspaEffect/caps/dsp/ToneStack.h b/plugins/LadspaEffect/caps/dsp/ToneStack.h
index 4d0600856..82e300241 100644
--- a/plugins/LadspaEffect/caps/dsp/ToneStack.h
+++ b/plugins/LadspaEffect/caps/dsp/ToneStack.h
@@ -1,8 +1,10 @@
/*
ToneStack.h
- Copyright 2006-9
+ Copyright 2006
David Yeh <dtyeh@ccrma.stanford.edu> (implementation)
+
+ Copyright 2006-14
Tim Goetze <tim@quitte.de> (cosmetics)
Tone Stack emulation.
@@ -11,7 +13,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,14 +27,12 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DSP_TONESTACK_H_
-#define _DSP_TONESTACK_H_
+#ifndef DSP_TONESTACK_H
+#define DSP_TONESTACK_H
#include "util.h"
#include "windows.h"
#include "TDFII.h"
-#include "LatFilt.h"
-#include "tonestack/tables.h"
namespace DSP {
@@ -41,9 +41,6 @@ typedef struct {
double C1, C2, C3;
} TSParameters;
-#define TS_N_PRESETS (sizeof (DSP::ToneStack::presets) / \
- sizeof (DSP::TSParameters))
-
class ToneStack
{
private:
@@ -57,31 +54,17 @@ class ToneStack
a0, a1d, a1m, a1l, a2m, a2lm, a2m2, a2l, a2d,
a3lm, a3m2, a3m, a3l, a3d; // intermediate calculations
- struct {
- double b1;
- double b2;
- double b3;
- double a1;
- double a2;
- double a3;
- } acoef; // analog coefficients
-
- // digital coefficients
- double dcoef_a[Order + 1];
- double dcoef_b[Order + 1];
-
+ double fs;
TDFII<Order> filter;
public:
- int model;
-
- static TSParameters presets[]; /* in ../ToneStack.cc */
- static int n_presets;
+ /* in ../ToneStack.cc */
+ static const char * presetdict;
+ static TSParameters presets[];
ToneStack()
{
- model = -1;
- setparams (presets[0]);
+ setmodel(0);
}
void init (double _fs)
@@ -89,24 +72,12 @@ class ToneStack
c = 2 * _fs;
}
- void activate (sample_t ** ports)
- {
- filter.reset();
- }
-
- /* pass in pointer to ports and relative index of first eq band control */
- void start_cycle (sample_t ** ports, int bassindex = 1)
- {
- int m = clamp<int> ((int) *ports[0], 0, n_presets - 1);
- if (m != model)
- {
- model = m;
- setparams (presets[model]);
+ void reset() { filter.reset(); }
+ void setmodel (int model)
+ {
+ setparams (presets[model]);
filter.reset();
}
- updatecoefs (ports + bassindex);
- }
-
void setparams (TSParameters & p)
{
@@ -148,12 +119,17 @@ class ToneStack
filter.reset();
}
- inline void updatecoefs (sample_t ** ports)
+ /* extensively reworked TG 5/2011 */
+ inline void updatecoefs (double b, double m, double t)
{
- /* range checks on input */
- double b = clamp<double> (*ports[0], 0, 1);
- double m = clamp<double> (*ports[1], 0, 1);
- double t = clamp<double> (*ports[2], 0, 1);
+ struct {
+ double a1, a2, a3;
+ double b1, b2, b3;
+ } acoef; // analog coefficients
+
+ // digital coefficients
+ double dcoef_a[Order + 1];
+ double dcoef_b[Order + 1];
m = (m - 1) * 3.5;
m = pow (10, m);
@@ -161,93 +137,36 @@ class ToneStack
acoef.a1 = a1d + m*a1m + b*a1l;
acoef.a2 = m*a2m + b*m*a2lm + m*m*a2m2 + b*a2l + a2d;
acoef.a3 = b*m*a3lm + m*m*a3m2 + m*a3m + b*a3l + a3d;
- dcoef_a[0] = -1 - acoef.a1*c - acoef.a2*c*c - acoef.a3*c*c*c; // sets scale
- dcoef_a[1] = -3 - acoef.a1*c + acoef.a2*c*c + 3*acoef.a3*c*c*c;
- dcoef_a[2] = -3 + acoef.a1*c + acoef.a2*c*c - 3*acoef.a3*c*c*c;
- dcoef_a[3] = -1 + acoef.a1*c - acoef.a2*c*c + acoef.a3*c*c*c;
+
+ double c2 = c*c, c3 = c2*c;
+
+ acoef.a1 *= c, acoef.a2 *= c2, acoef.a3 *= c3;
+
+ dcoef_a[0] = -1 - acoef.a1 - acoef.a2 - acoef.a3; // sets scale
+ dcoef_a[1] = -3 - acoef.a1 + acoef.a2 + 3*acoef.a3;
+ dcoef_a[2] = -3 + acoef.a1 + acoef.a2 - 3*acoef.a3;
+ dcoef_a[3] = -1 + acoef.a1 - acoef.a2 + acoef.a3;
acoef.b1 = t*b1t + m*b1m + b*b1l + b1d;
acoef.b2 = t*b2t + m*m*b2m2 + m*b2m + b*b2l + b*m*b2lm + b2d;
acoef.b3 = b*m*b3lm + m*m*b3m2 + m*b3m + t*b3t + t*m*b3tm + t*b*b3tl;
- dcoef_b[0] = - acoef.b1*c - acoef.b2*c*c - acoef.b3*c*c*c;
- dcoef_b[1] = - acoef.b1*c + acoef.b2*c*c + 3*acoef.b3*c*c*c;
- dcoef_b[2] = acoef.b1*c + acoef.b2*c*c - 3*acoef.b3*c*c*c;
- dcoef_b[3] = acoef.b1*c - acoef.b2*c*c + acoef.b3*c*c*c;
+
+ acoef.b1 *= c, acoef.b2 *= c2, acoef.b3 *= c3;
+
+ dcoef_b[0] = - acoef.b1 - acoef.b2 - acoef.b3;
+ dcoef_b[1] = - acoef.b1 + acoef.b2 + 3*acoef.b3;
+ dcoef_b[2] = acoef.b1 + acoef.b2 - 3*acoef.b3;
+ dcoef_b[3] = acoef.b1 - acoef.b2 + acoef.b3;
+ double a0i = 1. / dcoef_a[0];
for (int i=1; i<=3; ++i)
- filter.a[i] = dcoef_a[i] / dcoef_a[0];
+ filter.a[i] = dcoef_a[i] * a0i;
for (int i=0; i<=3; ++i)
- filter.b[i] = dcoef_b[i] / dcoef_a[0];
+ filter.b[i] = dcoef_b[i] * a0i;
}
- // actualy do the DFII filtering, one sample at a time
- inline sample_t process (sample_t x)
- {
- return filter.process (x);
- }
-};
-
-/* /////////////////////////////////////////////////////////////////////// */
-
-/*
- hardcode this, known size memory blocks
- extern double* KS; // 25 x 25 x 3
- extern double* VS; // 25 x 25 x 25 x 4
- extern double KS[NSTEPS][NSTEPS][TSORDER]; //[bass][mid][coefs]
- extern double VS[NSTEPS][NSTEPS][NSTEPS][TSORDER+1]; //[bass][mid][treb][coefs]
- */
-
-class ToneStackLT
-{
- private:
- enum { Order = 3, Steps = 25 };
-
- // digital coefficients
- double *kcoef;
- double *vcoef;
- LatFilt<Order> filter;
-
- public:
- ToneStackLT()
- {
- setparams (250000, 1000000, 25000, 56000, 0.25e-9, 20e-9, 20e-9);
- }
-
- void init (double _fs)
- { }
-
- void activate (sample_t ** ports)
- {
- filter.reset();
- }
-
- void setparams
- (double R1, double R2, double R3, double R4,
- double C1, double C2, double C3) {
- }
-
- void updatecoefs (sample_t ** ports)
- {
- double b = min (Steps - 1, max (*ports[0] * (Steps - 1), 0));
- double m = min (Steps - 1, max (*ports[1] * (Steps - 1), 0));
- double t = min (Steps - 1, max (*ports[2] * (Steps - 1), 0));
-
- int bi = (int) b;
- int mi = (int) m;
- int ti = (int) t;
-
- kcoef = DSP::ToneStackKS + (mi * Steps + bi) * Order;
- vcoef = DSP::ToneStackVS + ((mi * Steps + bi) * Steps + ti) * (Order + 1);
-
- for (int i = 0; i < Order; ++i)
- filter.set_ki (kcoef[i], i);
-
- for (int i = 0; i < Order + 1; ++i)
- filter.set_vi (vcoef[i], i);
- }
-
- // actualy do the DFII filtering, one sample at a time
+ /* actually do the DFII filtering, one sample at a time */
inline sample_t process (sample_t x)
{
return filter.process (x);
@@ -256,4 +175,4 @@ class ToneStackLT
} /* namespace DSP */
-#endif /* _DSP_TONESTACK_H_ */
+#endif /* DSP_TONESTACK_H */
diff --git a/plugins/LadspaEffect/caps/dsp/White.h b/plugins/LadspaEffect/caps/dsp/White.h
index 1a9fe6ea7..bc5506187 100644
--- a/plugins/LadspaEffect/caps/dsp/White.h
+++ b/plugins/LadspaEffect/caps/dsp/White.h
@@ -1,7 +1,7 @@
/*
dsp/White.h
- Copyright 2004 Tim Goetze <tim@quitte.de>
+ Copyright 2004-13 Tim Goetze <tim@quitte.de>
simple white noise generator, based on Jon Dattorro's 3/2002 JAES
paper. quite an elegant design; consumes next to no CPU on a processor
@@ -11,7 +11,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,8 +25,8 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DSP_WHITE_H_
-#define _DSP_WHITE_H_
+#ifndef DSP_WHITE_H
+#define DSP_WHITE_H
namespace DSP {
@@ -56,27 +56,23 @@ class White
/* 32-bit version */
sample_t get()
{
-# define BIT(y) ((b << (31 - y)) & 0x80000000)
-
+ #define BIT(y) ((b << (31 - y)) & 0x80000000)
b = ((BIT (28) ^ BIT (27) ^ BIT (1) ^ BIT (0))) | (b >> 1);
return (4.6566128730773926e-10 * (sample_t) b) - 1;
-
-# undef BIT
+ #undef BIT
}
/* 31-bit version, at least 6 instructions less / sample. probably only
* pays off on a processor not providing a decent binary shift. */
sample_t get_31()
{
-# define BIT(y) ((b << (30 - y)) & 0x40000000)
-
+ #define BIT(y) ((b << (30 - y)) & 0x40000000)
b = ((BIT (3) ^ BIT (0))) | (b >> 1);
return (9.3132257461547852e-10 * (sample_t) b) - 1;
-
-# undef BIT
+ #undef BIT
}
};
} /* namespace DSP */
-#endif /* _DSP_WHITE_H_ */
+#endif /* DSP_WHITE_H */
diff --git a/plugins/LadspaEffect/caps/dsp/sinc.h b/plugins/LadspaEffect/caps/dsp/sinc.h
index b002f234d..2c5231304 100644
--- a/plugins/LadspaEffect/caps/dsp/sinc.h
+++ b/plugins/LadspaEffect/caps/dsp/sinc.h
@@ -11,7 +11,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,8 +25,8 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _SINC_H_
-#define _SINC_H_
+#ifndef DSP_SINC_H
+#define DSP_SINC_H
#include "Sine.h"
@@ -53,6 +53,6 @@ sinc (double omega, sample_t * s, int n)
}
}
-}; /* namespace DSP */
+} /* namespace DSP */
-#endif /* _SINC_H_ */
+#endif /* DSP_SINC_H */
diff --git a/plugins/LadspaEffect/caps/dsp/util.h b/plugins/LadspaEffect/caps/dsp/util.h
index 09416b5a5..50c6cb652 100644
--- a/plugins/LadspaEffect/caps/dsp/util.h
+++ b/plugins/LadspaEffect/caps/dsp/util.h
@@ -1,7 +1,7 @@
/*
dsp/util.h
- Copyright 2002-4 Tim Goetze <tim@quitte.de>
+ Copyright 2002-12 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -11,7 +11,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,21 +25,27 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DSP_UTIL_H_
-#define _DSP_UTIL_H_
+#ifndef DSP_UTIL_H
+#define DSP_UTIL_H
namespace DSP {
-inline int next_power_of_2 (int n)
-{
- assert (n <= 0x40000000);
-
- int m = 1;
-
- while (m < n)
- m <<= 1;
+inline float pow2 (float x) { return x * x; }
+inline float pow3 (float x) { return x * pow2(x); }
+inline float pow4 (float x) { return pow2 (pow2(x)); }
+inline float pow5 (float x) { return x * pow4(x); }
+inline float pow6 (float x) { return pow3 (pow2(x)); }
+inline float pow7 (float x) { return x * (pow6 (x)); }
+inline float pow8 (float x) { return pow2 (pow4 (x)); }
- return m;
+inline float
+sgn (float x)
+{
+ union { float f; uint32 i; } u;
+ u.f = x;
+ u.i &= 0x80000000;
+ u.i |= 0x3F800000;
+ return u.f;
}
inline bool
@@ -58,18 +64,6 @@ isprime (int v)
return true;
}
-inline double
-db2lin (double db)
-{
- return pow (10., db * .05);
-}
-
-inline double
-lin2db (double lin)
-{
- return 20. * log10 (lin);
-}
-
} /* namespace DSP */
-#endif /* _DSP_UTIL_H_ */
+#endif /* DSP_UTIL_H */
diff --git a/plugins/LadspaEffect/caps/dsp/windows.h b/plugins/LadspaEffect/caps/dsp/windows.h
index 949a95818..b0b61b873 100644
--- a/plugins/LadspaEffect/caps/dsp/windows.h
+++ b/plugins/LadspaEffect/caps/dsp/windows.h
@@ -1,17 +1,17 @@
/*
dsp/windows.h
- Copyright 2004-11 Tim Goetze <tim@quitte.de>
+ Copyright 2004-12 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
- select few common windowing algorithms.
+ Window functions.
*/
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -25,8 +25,8 @@
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
-#ifndef _DSP_WINDOWS_H_
-#define _DSP_WINDOWS_H_
+#ifndef DSP_WINDOWS_H
+#define DSP_WINDOWS_H
namespace DSP {
@@ -48,29 +48,43 @@ apply_window (sample_t &d, sample_t s)
template <window_sample_func_t F>
void
-hanning (sample_t * s, int n)
+hann (sample_t * s, int n, double step = 1)
{
- /* could speed up by using DSP::Sine but we rarely use this window */
+ step = step / n;
+
for (int i = 0; i < n; ++i)
{
- register double f = (double) i / n - 1;
+ register double f = i * step;
F (s[i], .5 - .5 * cos (2 * M_PI * f));
}
}
+template <window_sample_func_t F>
+void
+hamming (sample_t * s, int n)
+{
+ register float in = 1. / n;
+
+ for (int i = 0; i < n; ++i)
+ {
+ register double f = i * in;
+ F (s[i], .54 - .46 * cos (2 * M_PI * f));
+ }
+}
+
template <window_sample_func_t F>
void
blackman (sample_t * s, int n)
{
- register float w = n;
+ register float in = 1. / n;
for (int i = 0; i < n; ++i)
{
register float f = (float) i;
register double b = .42f -
- .5f * cos (2.f * f * M_PI / w) +
- .08 * cos (4.f * f * M_PI / w);
+ .5f * cos (2.f * f * M_PI * in) +
+ .08 * cos (4.f * f * M_PI * in);
F (s[i], b);
}
@@ -93,8 +107,6 @@ blackman_harris (sample_t * s, int n)
.14128f * cos (w2 * f) -
.01168f * cos (w3 * f);
- bh *= .761f;
-
F (s[i], bh);
}
}
@@ -135,28 +147,39 @@ besseli (double x)
return ans;
}
+/* step = .5 : window [-n to 0] */
template <window_sample_func_t F>
void
-kaiser (sample_t * s, int n, double beta)
+kaiser (sample_t * s, int n, double beta, double step = 1)
{
double bb = besseli (beta);
int si = 0;
- for (double i = -n / 2 + .1; si < n; ++si, ++i)
+ for (double i = -n / 2 + .1; si < n; ++si, i += step)
{
double k = besseli ((beta * sqrt (1 - pow ((2 * i / (n - 1)), 2)))) / bb;
/* can you spell hack */
- if (!isfinite (k) || isnan(k))
+ if (!finite (k) || isnan(k))
k = 0;
F (s[si], k);
}
- /* asymmetrical hack: sort out first value!
- win[0] = win[len-1];
- */
}
-}; /* namespace DSP */
+template <window_sample_func_t F>
+void
+xfade (sample_t * s, int n, int dir) /* dir [-1,1] */
+{
+ DSP::Sine cos(.5*M_PI/n, 0);
+ if (dir > 0)
+ for (int i=0; i<n; ++i)
+ s[i] *= cos.get();
+ else
+ for (int i=n; --i>=0; )
+ s[i] *= cos.get();
+}
+
+} /* namespace DSP */
-#endif /* _DSP_WINDOWS_H_ */
+#endif /* DSP_WINDOWS_H */
diff --git a/plugins/LadspaEffect/caps/interface.cc b/plugins/LadspaEffect/caps/interface.cc
index 96e3d9806..82a4dc257 100644
--- a/plugins/LadspaEffect/caps/interface.cc
+++ b/plugins/LadspaEffect/caps/interface.cc
@@ -1,7 +1,7 @@
/*
interface.cc
- Copyright 2004-11 Tim Goetze <tim@quitte.de>
+ Copyright 2004-14 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -11,7 +11,7 @@
/*
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
+ as published by the Free Software Foundation; either version 3
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
@@ -29,117 +29,99 @@
(2541 - 2580 donated to artemio@kdemail.net)
*/
-#include <sys/time.h>
-
+#include "version.h"
#include "basics.h"
#include "Cabinet.h"
#include "Chorus.h"
#include "Phaser.h"
#include "Sin.h"
-#include "Lorenz.h"
-#include "Roessler.h"
+#include "Fractals.h"
#include "Reverb.h"
#include "Compress.h"
#include "Click.h"
#include "Eq.h"
-#include "Clip.h"
+#include "Saturate.h"
#include "White.h"
-#include "SweepVF.h"
-#include "VCO.h"
+#include "AutoFilter.h"
#include "Amp.h"
-#include "HRTF.h"
#include "Pan.h"
#include "Scape.h"
#include "ToneStack.h"
+#include "Noisegate.h"
+#ifdef SUMMER
+#include "AmpVI.h"
+#endif
#include "Descriptor.h"
-#define N 39
-static DescriptorStub * descriptors [N];
-
-/*static inline void
-seed()
-{
- static struct timeval tv;
- gettimeofday (&tv, 0);
+#define N 35
- srand (tv.tv_sec ^ tv.tv_usec);
-}*/
+static DescriptorStub * descriptors[N+1];
extern "C" {
+const LADSPA_Descriptor *
+ladspa_descriptor (unsigned long i) { return i < N ? descriptors[i] : 0; }
+
__attribute__ ((constructor))
void caps_so_init()
{
DescriptorStub ** d = descriptors;
-
- *d++ = new Descriptor<Eq>();
- *d++ = new Descriptor<Eq2x2>();
- *d++ = new Descriptor<Compress>();
- *d++ = new Descriptor<Pan>();
- *d++ = new Descriptor<Narrower>();
-
- *d++ = new Descriptor<PreampIII>();
- *d++ = new Descriptor<PreampIV>();
- *d++ = new Descriptor<ToneStack>();
- *d++ = new Descriptor<ToneStackLT>();
- *d++ = new Descriptor<AmpIII>();
- *d++ = new Descriptor<AmpIV>();
- *d++ = new Descriptor<AmpV>();
- *d++ = new Descriptor<AmpVTS>();
- *d++ = new Descriptor<CabinetI>();
- *d++ = new Descriptor<CabinetII>();
- *d++ = new Descriptor<Clip>();
-
- *d++ = new Descriptor<ChorusI>();
- *d++ = new Descriptor<StereoChorusI>();
- *d++ = new Descriptor<ChorusII>();
- *d++ = new Descriptor<StereoChorusII>();
- *d++ = new Descriptor<PhaserI>();
- *d++ = new Descriptor<PhaserII>();
- *d++ = new Descriptor<SweepVFI>();
- *d++ = new Descriptor<SweepVFII>();
- *d++ = new Descriptor<AutoWah>();
- *d++ = new Descriptor<Scape>();
-
- *d++ = new Descriptor<VCOs>();
- *d++ = new Descriptor<VCOd>();
- *d++ = new Descriptor<CEO>();
- *d++ = new Descriptor<Sin>();
- *d++ = new Descriptor<White>();
- *d++ = new Descriptor<Lorenz>();
- *d++ = new Descriptor<Roessler>();
-
- *d++ = new Descriptor<JVRev>();
- *d++ = new Descriptor<Plate>();
- *d++ = new Descriptor<Plate2x2>();
-
- *d++ = new Descriptor<Click>();
- *d++ = new Descriptor<Dirac>();
- *d++ = new Descriptor<HRTF>();
-
- /* make sure N is correct */
- assert (d - descriptors == N);
-
- //seed();
+ /* make sure uninitialised array members are safe to pass to the host */
+ memset (d, 0, sizeof (descriptors));
+
+ *d++ = new Descriptor<Noisegate>(2602);
+ *d++ = new Descriptor<Compress>(1772);
+ *d++ = new Descriptor<CompressX2>(2598);
+
+ *d++ = new Descriptor<ToneStack>(2589);
+ *d++ = new Descriptor<AmpVTS>(2592);
+ #ifdef SUMMER
+ *d++ = new Descriptor<AmpVI>(1789);
+ #endif
+ *d++ = new Descriptor<CabinetIII>(2601);
+ *d++ = new Descriptor<CabinetIV>(2606);
+
+ #ifdef WITH_JVREV
+ *d++ = new Descriptor<JVRev>(1778);
+ #endif
+ *d++ = new Descriptor<Plate>(1779);
+ *d++ = new Descriptor<PlateX2>(1795);
+
+ *d++ = new Descriptor<Saturate>(1771);
+ *d++ = new Descriptor<Spice>(2603);
+ *d++ = new Descriptor<SpiceX2>(2607);
+
+ *d++ = new Descriptor<ChorusI>(1767);
+ *d++ = new Descriptor<PhaserII>(2586);
+
+ *d++ = new Descriptor<AutoFilter>(2593);
+ *d++ = new Descriptor<Scape>(2588);
+
+ *d++ = new Descriptor<Eq10>(1773);
+ *d++ = new Descriptor<Eq10X2>(2594);
+ *d++ = new Descriptor<Eq4p>(2608);
+ *d++ = new Descriptor<EqFA4p>(2609);
+
+ *d++ = new Descriptor<Wider>(1788);
+ *d++ = new Descriptor<Narrower>(2595);
+
+ *d++ = new Descriptor<Sin>(1781);
+ *d++ = new Descriptor<White>(1785);
+ *d++ = new Descriptor<Fractal>(1774);
+
+ *d++ = new Descriptor<Click>(1769);
+ *d++ = new Descriptor<CEO>(1770);
+
+ assert (d - descriptors <= N);
}
__attribute__ ((destructor))
void caps_so_fini()
{
- for (ulong i = 0; i < N; ++i)
- delete descriptors[i];
-}
-
-/* /////////////////////////////////////////////////////////////////////// */
-
-const LADSPA_Descriptor *
-ladspa_descriptor (unsigned long i)
-{
- if (i < N)
- return descriptors[i];
- return 0;
+ DescriptorStub ** d = descriptors;
+ while (*d) delete *d++;
}
}; /* extern "C" */
diff --git a/plugins/LadspaEffect/caps/waves/click.h b/plugins/LadspaEffect/caps/waves/click.h
index 35950d618..7b70c33c4 100644
--- a/plugins/LadspaEffect/caps/waves/click.h
+++ b/plugins/LadspaEffect/caps/waves/click.h
@@ -1,520 +1,811 @@
-float click [] = {
- -0.013062, -0.013062, -0.012817, -0.011078, -0.031097, -0.266479,
- -0.610718, -0.255005, 0.698975, 0.999969, 0.505188, -0.724579,
- -1.000000, -0.493225, 0.532135, 0.789459, 0.131744, -0.270599,
- -0.080780, -0.065796, 0.584381, 0.718567, -0.355316, -0.760559,
- -0.649994, -0.039032, 0.442017, 0.700043, 0.434845, -0.556427,
- -1.000000, -0.400391, 0.590729, 0.999969, 0.999969, 0.592896,
- -0.248932, -0.662750, -0.074615, 0.424774, 0.274933, -0.416504,
- -0.772736, -0.359833, -0.218140, -0.254456, -0.379547, -0.768280,
- -0.707428, 0.108124, 0.755127, 0.680847, 0.429352, 0.126129,
- -0.424316, -0.451813, 0.197510, 0.862762, 0.964508, 0.263824,
- -0.383057, -0.753571, -0.848297, -0.491425, -0.042114, -0.222504,
- -0.664490, -0.690155, -0.227905, 0.286682, 0.742981, 0.812225,
- 0.326691, 0.182770, 0.381195, 0.536194, 0.446930, 0.085754,
- -0.650604, -1.000000, -1.000000, -0.774658, -0.152191, -0.112122,
- -0.416290, -0.491547, -0.253235, 0.241760, 0.749847, 0.618042,
- 0.338440, 0.431335, 0.773712, 0.936890, 0.752563, 0.291595,
- -0.206970, -0.552460, -0.447449, -0.152069, -0.315735, -0.891846,
- -1.000000, -1.000000, -0.682281, 0.066925, 0.477264, 0.447144,
- 0.209167, 0.149994, 0.374176, 0.621185, 0.597046, 0.315826,
- 0.034210, -0.004089, 0.084534, 0.127075, 0.060608, -0.134949,
- -0.275421, -0.347870, -0.505371, -0.618835, -0.443878, -0.217682,
- -0.157898, -0.241058, -0.424103, -0.494690, -0.286438, 0.111542,
- 0.487793, 0.654907, 0.571869, 0.484650, 0.413208, 0.254089,
- 0.136414, 0.021118, -0.223724, -0.366364, -0.290771, -0.126251,
- -0.080048, -0.175262, -0.290100, -0.281403, -0.071808, 0.078613,
- 0.216339, 0.275787, 0.223938, 0.178131, 0.162659, 0.110779,
- 0.104553, 0.158661, 0.208252, 0.271484, 0.256531, 0.153046,
- 0.068054, -0.009003, -0.116119, -0.155304, -0.099213, 0.066589,
- 0.293732, 0.392303, 0.313416, 0.196320, 0.123932, 0.145874,
- 0.279053, 0.291016, 0.081818, -0.092133, -0.090363, 0.030029,
- 0.099762, -0.021912, -0.133942, -0.025299, 0.257782, 0.466980,
- 0.406006, 0.150085, -0.007385, -0.027588, 0.048920, 0.128998,
- 0.058868, -0.066589, -0.146759, -0.168427, -0.119751, -0.087677,
- -0.082581, -0.029694, -0.041138, -0.139984, -0.156036, -0.033325,
- 0.164612, 0.331726, 0.320435, 0.205811, 0.209717, 0.275909,
- 0.240479, 0.081238, -0.104828, -0.239624, -0.286652, -0.254120,
- -0.304688, -0.428223, -0.479248, -0.534729, -0.453003, -0.144928,
- -0.001709, 0.059448, 0.016937, -0.127991, -0.186432, -0.122986,
- 0.086090, 0.278351, 0.258026, 0.172882, -0.032043, -0.267242,
- -0.289276, -0.293365, -0.288971, -0.256073, -0.212250, -0.214783,
- -0.182709, -0.149170, -0.132599, -0.221649, -0.233063, 0.002472,
- 0.208008, 0.119659, -0.018250, -0.182861, -0.267731, -0.164001,
- -0.052063, 0.009979, 0.058472, 0.147156, 0.111298, 0.017426,
- -0.067780, -0.049835, 0.002167, -0.028259, 0.003418, 0.023743,
- -0.056702, -0.074066, -0.070984, -0.075439, -0.056915, -0.189575,
- -0.324524, -0.241882, -0.051514, 0.075623, 0.160095, 0.094238,
- 0.004425, 0.010193, 0.032898, 0.040222, 0.163605, 0.273682,
- 0.192261, -0.019989, -0.217682, -0.327728, -0.337372, -0.227386,
- -0.043945, 0.107788, 0.244690, 0.226227, -0.034332, -0.252106,
- -0.364136, -0.377136, -0.246277, -0.021973, 0.194763, 0.300110,
- 0.295532, 0.246246, 0.198425, 0.209991, 0.272064, 0.249451,
- 0.078094, -0.143616, -0.276459, -0.257782, -0.228638, -0.161224,
- -0.006012, 0.169342, 0.267456, 0.246094, 0.232147, 0.290436,
- 0.280670, 0.220886, 0.162811, 0.187286, 0.178223, 0.036804,
- -0.043304, -0.098694, -0.101990, 0.008698, 0.102844, 0.132050,
- 0.142578, 0.152924, 0.058960, -0.074615, -0.079041, 0.026123,
- 0.136322, 0.253510, 0.382782, 0.447357, 0.419800, 0.294952,
- 0.129059, 0.004028, -0.119781, -0.257538, -0.320526, -0.211609,
- -0.054565, 0.012085, -0.016663, 0.006592, 0.144958, 0.193237,
- 0.072174, -0.018951, 0.038483, 0.085175, -0.023407, -0.089661,
- -0.029999, -0.023956, -0.079498, -0.135223, -0.154205, -0.095276,
- -0.031830, -0.105988, -0.252563, -0.284424, -0.153198, 0.017273,
- 0.073456, 0.072571, 0.175751, 0.222260, 0.105316, -0.053345,
- -0.180115, -0.192932, -0.185211, -0.167511, -0.079529, -0.070007,
- -0.151276, -0.215210, -0.206390, -0.133820, -0.099060, -0.082855,
- 0.047852, 0.108429, 0.014099, -0.049377, -0.050751, -0.081207,
- -0.060944, -0.028564, -0.124420, -0.139526, -0.033508, 0.054108,
- 0.042664, -0.004669, -0.069611, -0.105774, -0.090881, -0.128601,
- -0.090027, -0.012634, 0.004578, 0.046387, 0.085144, 0.055511,
- 0.036469, 0.001770, -0.019867, 0.016998, 0.031891, 0.089935,
- 0.066254, -0.069183, -0.020081, 0.046356, 0.005554, -0.023224,
- -0.042847, -0.133057, -0.104004, 0.065674, 0.146484, 0.145782,
- 0.006897, -0.059509, 0.070923, 0.058319, -0.002655, 0.000641,
- -0.036011, 0.049988, 0.076752, 0.016998, 0.063049, 0.105835,
- 0.041870, -0.092072, -0.154358, -0.081879, 0.089905, 0.090820,
- 0.054047, -0.110565, -0.285980, -0.219116, -0.103058, 0.083405,
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