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CabIV.cc
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CabIV.cc
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/*
CabIV.cc
Copyright 2002-18 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
Loudspeaker cabinet emulation through IIR/FIR filtering.
*/
/*
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.
*/
#include "basics.h"
#include "dsp/RBJ.h"
#include "Cabinet.h"
#include "Descriptor.h"
#include "CabIV_64_128.h"
void
CabinetIV::init()
{
model = 0;
/* choose oversampling ratio and 44.1 vs 48 k model */
int f = (int) (fs/1000 + .5);
over = 1;
while (f > 48)
over <<= 1,
f >>= 1;
if (over >= 4) over4.init(.75);
else if (over == 2) over2.init(.75);
}
void
CabinetIV::switch_model (int m)
{
model = m;
if (model < 0) return;
gain = CabIVModels[m].gain;
bank.set_a (1, CabIVModels[m].a1);
bank.set_a (2, CabIVModels[m].a2);
bank.set_b (1, CabIVModels[m].b1);
bank.set_b (2, CabIVModels[m].b2);
bank.reset();
fir.set_kernel (CabIVModels[m].fir);
fir.reset();
}
void
CabinetIV::activate()
{
switch_model ((int) getport(0));
remain = 0;
}
void
CabinetIV::cycle (uint frames)
{
static DSP::NoOversampler over1;
if (over == 1) subcycle<DSP::NoOversampler,1>(frames,over1);
else if (over == 2) subcycle<DSP::Oversampler<2,32>,2>(frames,over2);
else if (over == 4) subcycle<DSP::Oversampler<4,64>,4>(frames,over4);
}
template <class O, int Ratio>
void
CabinetIV::subcycle (uint frames, O & Over)
{
int m = (int) getport (0);
if (m != model) switch_model (m);
double g = gain * db2lin (getport(1));
sample_t * s = ports[2];
sample_t * d = ports[3];
uint n;
/* now we have to jump through some hoops to support odd block sizes
* while maintaining the processing rhythm of the undersampling ratio
* continuing over block borders.
*/
int rshift = Ratio == 4 ? 2 : (Ratio == 2 ? 1 : 0);
if (Ratio == 1)
n = frames;
else /* undersampling */
{
n = min (remain,frames);
for (uint i = 0; i < n; ++i)
{
Over.downstore(s[i]);
sample_t x = Over.uppad(i+Ratio-remain);
d[i] = x;
}
remain -= n;
frames -= n;
s += n;
d += n;
if (!frames) return;
n = (frames >> rshift) << rshift;
remain = frames - n;
}
/* at 1:1 sample ratios, the following loop is all that's needed */
uint i;
for (i = 0; i < n; ++i)
{
sample_t x = s[i];
x = Over.downsample (x);
/* process */
x = g*x + normal;
v4f_t a = (v4f_t) {x,x,x,x};
a = bank.process_no_a0 (a);
x = v4f_sum(a + fir.process(x));
x = Over.upsample (x);
d[i] = x;
for (int o=1; o < Ratio; ++o)
{
Over.downstore(s[++i]);
x = Over.uppad(o);
d[i] = x;
}
}
if (Ratio==1 || !remain) return;
/* at least 1 sample remaining in block, process it */
remain = Ratio-1;
frames -= n+1;
sample_t x = s[i];
x = Over.downsample (x);
x = g*x + normal;
v4f_t a = (v4f_t) {x,x,x,x};
a = bank.process_no_a0 (a);
x = v4f_sum(a + fir.process(x));
x = Over.upsample (x);
d[i] = x;
n = 1+min(frames,remain);
/* downsample the remainder of the block */
for (uint o=1; o < n; ++o, --remain)
{
Over.downstore(s[++i]);
x = Over.uppad(o);
d[i] = x;
}
}
/* //////////////////////////////////////////////////////////////////////// */
PortInfo
CabinetIV::port_info [] =
{
{ "model", CTRL_IN, {INTEGER | DEFAULT_MID, 0, NCabIVModels-1}, CabIVModelDict },
{ "gain (dB)", CTRL_IN | GROUP, {DEFAULT_0, -24, 24} },
{ "in", INPUT | AUDIO },
{ "out", OUTPUT | AUDIO }
};
template <> void
Descriptor<CabinetIV>::setup()
{
Label = "CabinetIV";
Name = CAPS "CabinetIV - Idealised loudspeaker cabinet";
autogen();
}