//#################################### vaeffects.lib ########################################
// A library of virtual analog filter effects. Its official prefix is `ve`.
//
// #### References
// * <https://github.com/grame-cncm/faustlibraries/blob/master/vaeffects.lib>
//########################################################################################

ma = library("maths.lib");
si = library("signals.lib");
an = library("analyzers.lib");
fi = library("filters.lib");
ef = library("misceffects.lib");

declare name "Faust Virtual Analog Filter Effect Library";
declare version "1.3.0";

//########################################################################################
/************************************************************************
FAUST library file, jos section

Except where noted otherwise, The Faust functions below in this
section are Copyright (C) 2003-2017 by Julius O. Smith III <jos@ccrma.stanford.edu>
([jos](http://ccrma.stanford.edu/~jos/)), and released under the
(MIT-style) [STK-4.3](#stk-4.3-license) license.

All MarkDown comments in this section are Copyright 2016-2017 by Romain
Michon and Julius O. Smith III, and are released under the
[CCA4I](https://creativecommons.org/licenses/by/4.0/) license (TODO: if/when Romain agrees!)

************************************************************************/

//====================================Moog Filters========================================
//========================================================================================

//-------------------------`(ve.)moog_vcf`---------------------------
// Moog "Voltage Controlled Filter" (VCF) in "analog" form. Moog VCF
// implemented using the same logical block diagram as the classic
// analog circuit.  As such, it neglects the one-sample delay associated
// with the feedback path around the four one-poles.
// This extra delay alters the response, especially at high frequencies
// (see reference [1] for details).
// See `moog_vcf_2b` below for a more accurate implementation.
//
// #### Usage
//
// ```
// _ : moog_vcf(res,fr) : _
// ```
// Where:
//
// * `res`: normalized amount of corner-resonance between 0 and 1 
// (0 is no resonance, 1 is maximum)
// * `fr`: corner-resonance frequency in Hz (less than SR/6.3 or so)
//
// #### References
// * <https://ccrma.stanford.edu/~stilti/papers/moogvcf.pdf>
// * <https://ccrma.stanford.edu/~jos/pasp/vegf.html>
//------------------------------------------------------------
declare moog_vcf author "Julius O. Smith III";
declare moog_vcf copyright "Copyright (C) 2003-2019 by Julius O. Smith III <jos@ccrma.stanford.edu>";
declare moog_vcf license "MIT-style STK-4.3 license";
moog_vcf(res,fr) = (+ : seq(i,4,fi.pole(p)) : *(unitygain(p))) ~ *(mk)
with {
     p = 1.0 - fr * 2.0 * ma.PI / ma.SR; // good approximation for fr << SR
     unitygain(p) = pow(1.0-p,4.0); // one-pole unity-gain scaling
     mk = -4.0*max(0,min(res,0.999999)); // need mk > -4 for stability
};

//-----------------------`(ve.)moog_vcf_2b[n]`---------------------------
// Moog "Voltage Controlled Filter" (VCF) as two biquads. Implementation
// of the ideal Moog VCF transfer function factored into second-order
// sections. As a result, it is more accurate than `moog_vcf` above, but
// its coefficient formulas are more complex when one or both parameters
// are varied.  Here, res is the fourth root of that in `moog_vcf`, so, as
// the sampling rate approaches infinity, `moog_vcf(res,fr)` becomes equivalent
// to `moog_vcf_2b[n](res^4,fr)` (when res and fr are constant).
// `moog_vcf_2b` uses two direct-form biquads (`tf2`).
// `moog_vcf_2bn` uses two protected normalized-ladder biquads (`tf2np`).
//
// #### Usage
//
// ```
// _ : moog_vcf_2b(res,fr) : _
// _ : moog_vcf_2bn(res,fr) : _
// ```
//
// Where:
//
// * `res`: normalized amount of corner-resonance between 0 and 1
// 	(0 is min resonance, 1 is maximum)
// * `fr`: corner-resonance frequency in Hz
//------------------------------------------------------------
declare moog_vcf_2b author "Julius O. Smith III";
declare moog_vcf_2b copyright "Copyright (C) 2003-2019 by Julius O. Smith III <jos@ccrma.stanford.edu>";
declare moog_vcf_2b license "MIT-style STK-4.3 license";
moog_vcf_2b(res,fr) = fi.tf2s(0,0,b0,a11,a01,w1) : fi.tf2s(0,0,b0,a12,a02,w1)
with {
 s = 1; // minus the open-loop location of all four poles
 frl = max(20,min(10000,fr)); // limit fr to reasonable 20-10k Hz range
 w1 = 2*ma.PI*frl; // frequency-scaling parameter for bilinear xform
 // Equivalent: w1 = 1; s = 2*PI*frl;
 kmax = sqrt(2)*0.99999; // 0.99999 gives stability margin (tf2 is unprotected)
 k = min(kmax,sqrt(2)*res); // fourth root of Moog VCF feedback gain
 b0 = s^2;
 s2k = sqrt(2) * k;
 a11 = s * (2 + s2k);
 a12 = s * (2 - s2k);
 a01 = b0 * (1 + s2k + k^2);
 a02 = b0 * (1 - s2k + k^2);
};

declare moog_vcf_2bn author "Julius O. Smith III";
declare moog_vcf_2bn copyright "Copyright (C) 2003-2019 by Julius O. Smith III <jos@ccrma.stanford.edu>";
declare moog_vcf_2bn license "MIT-style STK-4.3 license";
moog_vcf_2bn(res,fr) = fi.tf2snp(0,0,b0,a11,a01,w1) : fi.tf2snp(0,0,b0,a12,a02,w1)
with {
 s = 1; // minus the open-loop location of all four poles
 w1 = 2*ma.PI*max(fr,20); // frequency-scaling parameter for bilinear xform
 k = sqrt(2)*0.99999*res; // fourth root of Moog VCF feedback gain
 b0 = s^2;
 s2k = sqrt(2) * k;
 a11 = s * (2 + s2k);
 a12 = s * (2 - s2k);
 a01 = b0 * (1 + s2k + k^2);
 a02 = b0 * (1 - s2k + k^2);
};


//------------------`(ve.)moogLadder`-----------------
// Virtual analog model of the 4th-order Moog Ladder (without any nonlinearities), which is arguably the 
// most well-known ladder filter in analog synthesizers. Several 
// 1st-order filters are cascaded in series. Feedback is then used, in part, to 
// control the cut-off frequency and the resonance.
//
// #### References
//
// [Zavalishin 2012] (revision 2.1.2, February 2020): 
//
// * <https://www.native-instruments.com/fileadmin/ni_media/downloads/pdf/VAFilterDesign_2.1.2.pdf>
//
// This fix is based on Lorenzo Della Cioppa's correction to Pirkle's implementation; see this post: 
// https://www.kvraudio.com/forum/viewtopic.php?f=33&t=571909
//
// #### Usage
//
// ```
// _ : moogLadder(normFreq,Q) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: quality factor between .707 (0 feedback coefficient) to 25 (feedback = 4, which is the self-oscillating threshold).
//---------------------------------------------------------------------
declare moogLadder author "Dario Sanfilippo";
declare moogLadder license "MIT-style STK-4.3 license";
moogLadder(normFreq, Q, x) = loop ~ si.bus(4) : (!,!,!,!,_)
with {
    loop(s1, s2, s3, s4) =  v1 + lp1 , // define s1
                            v2 + lp2 , // define s2
                            v3 + lp3 , // define s3
                            v4 + lp4 , // define s4
                            lp4 // system output
        with {
            invSqrt2 = 1/sqrt(2);
            T = 1.0 / ma.SR;
            cf = 2*(10^(3*normFreq+1));
            k = 4.0 * (Q - invSqrt2) / (25.0 - invSqrt2);
            omegaWarp = tan(ma.PI * cf * T);
            g = omegaWarp / (1.0 + omegaWarp);
            G = g * g * g * g; // ladder's G in generalised form y = G * xi + S
            S = g * g * g * (s1 * (1 - g)) + g * g * (s2 * (1 - g)) + g * (s3 * (1 - g)) + (s4 * (1 - g)); // ladder's S in generalised form y = G * xi + S
            u = (x - k * S) / (1.0 + k * G); // input to the first LP stage: u = (x - kS) / (1 + kG)
            v1 = g * (u - s1); // v-signals in TPT integrator (Zavalishin, Figure 3.30)
            v2 = g * (lp1 - s2); // second stage
            v3 = g * (lp2 - s3); // third stage
            v4 = g * (lp3 - s4); // fourth stage
            lp1 = v1 + s1; // define outputs
            lp2 = v2 + s2;
            lp3 = v3 + s3;
            lp4 = v4 + s4;
        };
};

//------------------`(ve.)lowpassLadder4`-----------------
// Topology-preserving transform implementation of a four-pole ladder lowpass.
// This is essentially the same filter as the moogLadder above except for 
// the parameters, which will be expressed in Hz, for the cutoff, and as a
// raw feedback coefficient, for the resonance. 
// Also, note that the parameter order has changed.
//
// #### References
//
// [Zavalishin 2012] (revision 2.1.2, February 2020): 
//
// * <https://www.native-instruments.com/fileadmin/ni_media/downloads/pdf/VAFilterDesign_2.1.2.pdf>
//
// #### Usage
//
// ```
// _ : lowpassLadder4(k, CF) : _
// ```
//
// Where:
//
// * `k`: feedback coefficient between 0 and 4, which is the stability threshold.
// * `CF`: the filter's cutoff in Hz.
//
// Notes:
// 
// If you want to express the feedback coefficient as the resonance peak, you can use the formula: 
//
//      k = 4.0 - 1.0 / Q;
//
// where Q, between .25 and infinity, corresponds to the peak of the filter at cutoff. 
// I.e., if you feed the filter with a sine whose frequency is the same as the cutoff, the output 
// peak corresponds exactly to that set via the Q-param.
declare lowpassLadder4 author "Dario Sanfilippo";
declare lowpassLadder4 license "MIT License";
lowpassLadder4(k, CF, x) = loop ~ si.bus(4) : si.block(4) , _
    with {
        loop(s0, s1, s2, s3) = u0 , u1 , u2 , u3 , LP3
            with {
                PIT = ma.PI * ma.T;
                g = tan(CF * PIT);
                G = g / (1.0 + g);
                oneMinusG = 1.0 - G;
                globalG = G * G * G * G; 
                globalS = G * (G * (G * S0 + S1) + S2) + S3; 
                u = (x - k * globalS) / (1.0 + k * globalG); 
                v0 = G * (u - s0);
                v1 = G * (LP0 - s1);
                v2 = G * (LP1 - s2);
                v3 = G * (LP2 - s3);
                S0 = oneMinusG * s0;
                S1 = oneMinusG * s1;
                S2 = oneMinusG * s2;
                S3 = oneMinusG * s3;
                LP0 = v0 + s0; 
                LP1 = v1 + s1;
                LP2 = v2 + s2;
                LP3 = v3 + s3;
                u0 = v0 + LP0;
                u1 = v1 + LP1;
                u2 = v2 + LP2;
                u3 = v3 + LP3;
            };
    };

//------------------`(ve.)moogHalfLadder`-----------------
// Virtual analog model of the 2nd-order Moog Half Ladder (simplified version of
// `(ve.)moogLadder`). Several 1st-order filters are cascaded in series. 
// Feedback is then used, in part, to control the cut-off frequency and the 
// resonance.
//
// This filter was implemented in Faust by Eric Tarr during the 
// [2019 Embedded DSP With Faust Workshop](https://ccrma.stanford.edu/workshops/faust-embedded-19/).
// 
// #### References
//
// * <https://www.willpirkle.com/app-notes/virtual-analog-moog-half-ladder-filter>
// * <http://www.willpirkle.com/Downloads/AN-8MoogHalfLadderFilter.pdf>
//
// #### Usage
//
// ```
// _ : moogHalfLadder(normFreq,Q) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare moogHalfLadder author "Eric Tarr";
declare moogHalfLadder license "MIT-style STK-4.3 license";
moogHalfLadder(normFreq,Q) = _ <: (s1,s2,s3,y) : !,!,!,_ 
letrec{
  's1 = -(s3*B3*k):-(s2*B2*k):-(s1*B1*k):*(alpha0):-(s1):*(alpha*2):+(s1);
  's2 = -(s3*B3*k):-(s2*B2*k):-(s1*B1*k):*(alpha0):-(s1):*(alpha):+(s1):-(s2):*(alpha*2):+(s2);
  's3 = -(s3*B3*k):-(s2*B2*k):-(s1*B1*k):*(alpha0):-(s1):*(alpha):+(s1):-(s2):*(alpha):+(s2):-(s3):*(alpha*2):+(s3);
  'y = -(s3*B3*k):-(s2*B2*k):-(s1*B1*k):*(alpha0):-(s1):*(alpha):+(s1):-(s2):*(alpha):+(s2) <:_*-1,((-(s3):*(alpha):+(s3))*2):>_;
}
with{
  invSqrt2 = 1/sqrt(2);
  freq = 2*(10^(3*normFreq+1));
  k = 2.0*(Q - invSqrt2)/(25.0 - invSqrt2);
  wd = 2*ma.PI*freq;
  T = 1/ma.SR;
  wa = (2/T)*tan(wd*T/2);
  g = wa*T/2; 
  G = g/(1.0 + g);
  alpha = G;
  GA = 2*G-1; // All-pass gain
  B1 = GA*G/(1+g);
  B2 = GA/(1+g);
  B3 = 2/(1+g);
  alpha0 = 1/(1 + k*GA*G*G);
};


//------------------`(ve.)diodeLadder`-----------------
// 4th order virtual analog diode ladder filter. In addition to the individual 
// states used within each independent 1st-order filter, there are also additional 
// feedback paths found in the block diagram. These feedback paths are labeled 
// as connecting states. Rather than separately storing these connecting states 
// in the Faust implementation, they are simply implicitly calculated by 
// tracing back to the other states (`s1`,`s2`,`s3`,`s4`) each recursive step.
//
// This filter was implemented in Faust by Eric Tarr during the 
// [2019 Embedded DSP With Faust Workshop](https://ccrma.stanford.edu/workshops/faust-embedded-19/).
// 
// #### References
//
// * <https://www.willpirkle.com/virtual-analog-diode-ladder-filter/>
// * <http://www.willpirkle.com/Downloads/AN-6DiodeLadderFilter.pdf>
//
// #### Usage
//
// ```
// _ : diodeLadder(normFreq,Q) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare diodeLadder author "Eric Tarr";
declare diodeLadder license "MIT-style STK-4.3 license";
diodeLadder(normFreq,Q) = ef.cubicnl(1,0)*1.5 <:(s1,s2,s3,s4,y)  : !,!,!,!,_ 
letrec{
  's1 = _-(s4*B4*SG4*k) : 
    _-((s4*B4*d3+s3)*B3*SG3*k) : 
    _-(((s4*B4*d3+s3)*B3*d2 + s2)*B2*SG3*k) :
    _-((((s4*B4*d3+s3)*B3*d2 + s2)*B2*d1 + s1)*B1*SG1*k) :
    _*alpha0: _*gam1 : _+((s4*B4*d3+s3)*B3*d2 + s2)*B2 : //_+S2
    _+((((s4*B4*d3+s3)*B3*d2 + s2)*B2)*d1 + s1)*B1*G2 : // _ + (S2 ...
    _*a1 : _-s1 :_*alpha*2 : _+s1;

  's2 = _-(s4*B4*SG4*k) : 
    _-((s4*B4*d3+s3)*B3*SG3*k) : 
    _-(((s4*B4*d3+s3)*B3*d2 + s2)*B2*SG3*k):
    _-((((s4*B4*d3+s3)*B3*d2 + s2)*B2*d1 + s1)*B1*SG1*k) :
    _*alpha0: _*gam1 : _+((s4*B4*d3+s3)*B3*d2 + s2)*B2 : //_+S2
    _+((((s4*B4*d3+s3)*B3*d2 + s2)*B2)*d1 + s1)*B1*G2 : // _ + (S2 ...
    _*a1 : _-s1 :_*alpha : _+s1 : _*gam2 :
    _+(s4*B4*d3 + s3)*B3 : //_+S3 :
    _+(((s4*B4*d3 + s3)*B3)*d2 + s2)*B2*G3 : //_+(S3...)
    _*a2 : _-s2 : _*alpha*2 : _+s2;
		
  's3 = _-(s4*B4*SG4*k) : 
    _-((s4*B4*d3+s3)*B3*SG3*k) : 
    _-(((s4*B4*d3+s3)*B3*d2 + s2)*B2*SG3*k) :
    _-((((s4*B4*d3+s3)*B3*d2 + s2)*B2*d1 + s1)*B1*SG1*k) :
    _*alpha0 : _*gam1 : _+((s4*B4*d3+s3)*B3*d2 + s2)*B2 : //_+S2
    _+((((s4*B4*d3+s3)*B3*d2 + s2)*B2)*d1+s1)*B1*G2 : // _ + (S2 ...
    _*a1 : _-s1 :_*alpha : _+s1 : _*gam2 :
    _+(s4*B4*d3 + s3)*B3 : //_+S3 :
    _+(((s4*B4*d3 + s3)*B3)*d2 + s2)*B2*G3 : //_+(S3...)
    _*a2 : _-s2 : _*alpha : _+s2 : _*gam3:
    _+s4*B4 : // _ + S4
    _+((s4*B4)*d3 + s3)*B3*G4: // _ + S4 ...
    _*a3 : _-s3 : _*alpha*2 : _+s3;
  
  's4 = _-(s4*B4*SG4*k) : 
    _-((s4*B4*d3+s3)*B3*SG3*k) : 
    _-(((s4*B4*d3+s3)*B3*d2 + s2)*B2*SG3*k) :
    _-((((s4*B4*d3+s3)*B3*d2 + s2)*B2*d1 + s1 )*B1*SG1*k) :
    _*alpha0 : _*gam1 : _+((s4*B4*d3+s3)*B3*d2 + s2)*B2 : //_+S2
    _+((((s4*B4*d3+s3)*B3*d2 + s2)*B2)*d1 + s1)*B1*G2 : // _ + (S2 ...
    _*a1 : _-s1 :_*alpha : _+s1 : _*gam2 :
    _+(s4*B4*d3 + s3)*B3 : //_+S3 :
    _+(((s4*B4*d3 + s3)*B3)  *d2+s2)*B2*G3 : //_+(S3...)
    _*a2 : _-s2 : _*alpha : _+s2 : _*gam3 :
    _+s4*B4 : // _ + S4
    _+((s4*B4)*d3 + s3)*B3*G4: // _ + S4 ...
    _*a3 : _-s3 : _*alpha : _+s3 : _*gam4 : _*a4 : _-s4 : _*alpha*2 : _+s4;
  
  // Output signal
  'y = _-(s4*B4*SG4*k) : 
    _-((s4*B4*d3+s3)*B3*SG3*k) : 
    _-(((s4*B4*d3+s3)*B3*d2 + s2)*B2*SG3*k) :
    _-((((s4*B4*d3+s3)*B3*d2 + s2)*B2*d1 + s1 )*B1*SG1*k) :
    _*alpha0: _*gam1 : _+((s4*B4*d3+s3)*B3*d2 + s2)*B2 : //_+S2
    _+((((s4*B4*d3+s3)*B3*d2 + s2)*B2)*d1 + s1)*B1*G2 : // _ + (S2 ...
    _*a1 : _-s1 :_*alpha : _+s1 : _*gam2 :
    _+(s4*B4*d3 + s3)*B3 : //_+S3 :
    _+(((s4*B4*d3 + s3)*B3)*d2 + s2)*B2*G3 : //_+(S3...)
    _*a2 : _-s2 : _*alpha : _+s2 : _*gam3 :
    _+s4*B4 : // _ + S4
    _+((s4*B4)*d3 + s3)*B3*G4: // _ + S4 ...
    _*a3 : _-s3 : _*alpha : _+s3 : _*gam4 : _*a4 : _-s4 : _*alpha : _+s4;
}
with{
  freq = 2*(10^(3*normFreq+1));
  invSqrt2 = 1/sqrt(2);
  k = (17 - (normFreq^10)*9.7)*(Q - invSqrt2)/(25.0 - invSqrt2);
  wd = 2*ma.PI*freq;
  T = 1/ma.SR;
  wa = (2/T)*tan(wd*T/2);
  g = wa*T/2; 
  G4 = 0.5*g/(1 + g);
  G3 = 0.5*g/(1 + g - 0.5*g*G4);
  G2 = 0.5*g/(1 + g - 0.5*g*G3);
  G1 = g/(1.0 + g - g*G2);
  Gamma = G1*G2*G3*G4;
  SG1 = G4*G3*G2; // feedback gain pre-calculated
  SG2 = G4*G3; 
  SG3 = G4; 
  SG4 = 1;
  alpha = g/(1+g);
  alpha0 = 1/(1+k*Gamma);
  gam1 = 1+G1*G2;
  gam2 = 1+G2*G3;
  gam3 = 1+G3*G4; 
  gam4 = 1;
  a1 = 1; // a0 for 1st LPF
  a2 = 0.5; // a0 for 2nd LPF
  a3 = 0.5;
  a4 = 0.5;
  B1 = 1/(1+g-g*G2); // Beta for 1st block
  B2 = 1/(1+g-0.5*g*G3);
  B3 = 1/(1+g-0.5*g*G4);
  B4 = 1/(1+g);
  d1 = g; // delta for 1st block
  d2 = 0.5*g;
  d3 = 0.5*g;
  //d4 = 0;
};


//===================================Korg 35 Filters======================================
// The following filters are virtual analog models of the Korg 35 low-pass 
// filter and high-pass filter found in the MS-10 and MS-20 synthesizers.
// The virtual analog models for the LPF and HPF are different, making these 
// filters more interesting than simply tapping different states of the same 
// circuit. 
//
// These filters were implemented in Faust by Eric Tarr during the 
// [2019 Embedded DSP With Faust Workshop](https://ccrma.stanford.edu/workshops/faust-embedded-19/).
//
// #### Filter history:
//
// <https://secretlifeofsynthesizers.com/the-korg-35-filter/>
//========================================================================================

//------------------`(ve.)korg35LPF`-----------------
// Virtual analog models of the Korg 35 low-pass filter found in the MS-10 and 
// MS-20 synthesizers.
//
// #### Usage
//
// ```
// _ : korg35LPF(normFreq,Q) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare korg35LPF author "Eric Tarr";
declare korg35LPF license "MIT-style STK-4.3 license";
korg35LPF(normFreq,Q) = _ <: (s1,s2,s3,y) : !,!,!,_
letrec{
  's1 = _-s1:_*(alpha*2):_+s1;
  's2 = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2):_*alpha0:_-s3:_*alpha:_+s3:_*K:_-s2:_*(alpha*2):_+s2;
  's3 = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2):_*alpha0:_-s3:_*(alpha*2):_+s3;
  'y = _-s1:_*alpha:_+s1:_+(s3*B3):_+(s2*B2) :_*alpha0:_-s3:_*alpha:_+s3;
}
with{
  invSqrt2 = 1/sqrt(2);
  freq = 2*(10^(3*normFreq+1));
  K = 2.0*(Q - invSqrt2)/(10.0 - invSqrt2);
  wd = 2*ma.PI*freq;
  T = 1/ma.SR;
  wa = (2/T)*tan(wd*T/2);
  g = wa*T/2; 
  G = g/(1.0 + g);
  alpha = G;
  B3 = (K - K*G)/(1 + g);
  B2 = -1/(1 + g);
  alpha0 = 1/(1 - K*G + K*G*G); 
};


//------------------`(ve.)korg35HPF`-----------------
// Virtual analog models of the Korg 35 high-pass filter found in the MS-10 and 
// MS-20 synthesizers.
//
// #### Usage
//
// ```
// _ : korg35HPF(normFreq,Q) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare korg35HPF author "Eric Tarr";
declare korg35HPF license "MIT-style STK-4.3 license";
korg35HPF(normFreq,Q) = _ <: (s1,s2,s3,y) : !,!,!,_
letrec{
  's1 = _-s1:_*(alpha*2):_+s1;
  's2 = _<:(_-s1:_*alpha:_+s1)*-1,_:>_+(s3*B3):_+(s2*B2):_*alpha0:_*K:_-s2:_*alpha*2:_+s2;
  's3 = _<:(_-s1:_*alpha:_+s1)*-1,_:>_+(s3*B3):_+(s2*B2):_*alpha0:_*K:_<:(_-s2:_*alpha:_+s2)*-1,_:>_-s3:_*alpha*2:_+s3;
  'y = _<:(_-s1:_*alpha:_+s1)*-1,_:>_+(s3*B3):_+(s2*B2):_*alpha0;
}
with{
  invSqrt2 = 1/sqrt(2);
  freq = 2*(10^(3*normFreq+1));
  K = 2.0*(Q - invSqrt2)/(10.0 - invSqrt2);
  wd = 2*ma.PI*freq;
  T = 1/ma.SR;
  wa = (2/T)*tan(wd*T/2);
  g = wa*T/2; 
  G = g/(1.0 + g);
  alpha = G;
  B3 = 1.0/(1.0 + g);
  B2 = -1.0*G/(1.0 + g);
  alpha0 = 1/(1 - K*G + K*G*G);
};


//==================================Oberheim Filters======================================
// The following filter (4 types) is an implementation of the virtual analog 
// model described in Section 7.2 of the Will Pirkle book, "Designing Software 
// Synthesizer Plug-ins in C++". It is based on the block diagram in Figure 7.5. 
//
// The Oberheim filter is a state-variable filter with soft-clipping distortion 
// within the circuit. 
//
// In many VA filters, distortion is accomplished using the "tanh" function. 
// For this Faust implementation, that distortion function was replaced with 
// the `(ef.)cubicnl` function.
//========================================================================================

//------------------`(ve.)oberheim`-----------------
// Generic multi-outputs Oberheim filter that produces the BSF, BPF, HPF and LPF outputs (see description above).
//
// #### Usage
//
// ```
// _ : oberheim(normFreq,Q) : _,_,_,_
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare oberheim author "Eric Tarr";
declare oberheim license "MIT-style STK-4.3 license";
oberheim(normFreq,Q) = _<:(s1,s2,ybsf,ybpf,yhpf,ylpf) : !,!,_,_,_,_  
letrec{
  's1 = _-s2:_-(s1*FBs1):_*alpha0:_*g<:_,(_+s1:ef.cubicnl(0.0,0)):>_;
  's2 = _-s2:_-(s1*FBs1):_*alpha0:_*g:_+s1:ef.cubicnl(0.0,0):_*g*2:_+s2;
  // Compute the BSF, BPF, HPF, LPF outputs
  'ybsf = _-s2:_-(s1*FBs1):_*alpha0<:(_*g:_+s1:ef.cubicnl(0.0,0):_*g:_+s2),_:>_;
  'ybpf = _-s2:_-(s1*FBs1):_*alpha0:_*g:_+s1:ef.cubicnl(0.0,0);
  'yhpf = _-s2:_-(s1*FBs1):_*alpha0;
  'ylpf = _-s2:_-(s1*FBs1):_*alpha0:_*g :_+s1:ef.cubicnl(0.0,0):_*g:_+s2;
}
with{
  freq = 2*(10^(3*normFreq+1));
  wd = 2*ma.PI*freq;
  T = 1/ma.SR;
  wa = (2/T)*tan(wd*T/2);
  g = wa*T/2; 
  G = g/(1.0 + g);
  R = 1/(2*Q);
  FBs1 = (2*R+g);
  alpha0 = 1/(1 + 2*R*g + g*g);
};


//------------------`(ve.)oberheimBSF`-----------------
// Band-Stop Oberheim filter (see description above). 
// Specialize the generic implementation: keep the first BSF output, 
// the compiler will only generate the needed code.
//
// #### Usage
//
// ```
// _ : oberheimBSF(normFreq,Q) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare oberheimBSF author "Eric Tarr";
declare oberheimBSF license "MIT-style STK-4.3 license";
oberheimBSF(normFreq,Q) = oberheim(normFreq,Q):_,!,!,!;


//------------------`(ve.)oberheimBPF`-----------------
// Band-Pass Oberheim filter (see description above).
// Specialize the generic implementation: keep the second BPF output, 
// the compiler will only generate the needed code.
//
// #### Usage
//
// ```
// _ : oberheimBPF(normFreq,Q) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare oberheimBPF author "Eric Tarr";
declare oberheimBPF license "MIT-style STK-4.3 license";
oberheimBPF(normFreq,Q) = oberheim(normFreq,Q):!,_,!,!;


//------------------`(ve.)oberheimHPF`-----------------
// High-Pass Oberheim filter (see description above).
// Specialize the generic implementation: keep the third HPF output, 
// the compiler will only generate the needed code.
//
// #### Usage
//
// ```
// _ : oberheimHPF(normFreq,Q) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare oberheimHPF author "Eric Tarr";
declare oberheimHPF license "MIT-style STK-4.3 license";
oberheimHPF(normFreq,Q) = oberheim(normFreq,Q):!,!,_,!;


//------------------`(ve.)oberheimLPF`-----------------
// Low-Pass Oberheim filter (see description above). 
// Specialize the generic implementation: keep the fourth LPF output,
// the compiler will only generate the needed code.
//
// #### Usage
//
// ```
// _ : oberheimLPF(normFreq,Q) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare oberheimLPF author "Eric Tarr";
declare oberheimLPF license "MIT-style STK-4.3 license";
oberheimLPF(normFreq,Q) = oberheim(normFreq,Q):!,!,!,_;


//================================Sallen Key Filters======================================
// The following filters were implemented based on VA models of synthesizer 
// filters.
//
// The modeling approach is based on a Topology Preserving Transform (TPT) to 
// resolve the delay-free feedback loop in the corresponding analog filters.  
//
// The primary processing block used to build other filters (Moog, Korg, etc.) is
// based on a 1st-order Sallen-Key filter. 
//
// The filters included in this script are 1st-order LPF/HPF and 2nd-order 
// state-variable filters capable of LPF, HPF, and BPF.  
//
// #### Resources:
//
// * Vadim Zavalishin (2018) "The Art of VA Filter Design", v2.1.0
// <https://www.native-instruments.com/fileadmin/ni_media/downloads/pdf/VAFilterDesign_2.1.0.pdf>
// * Will Pirkle (2014) "Resolving Delay-Free Loops in Recursive Filters Using 
// the Modified Härmä Method", AES 137 <http://www.aes.org/e-lib/browse.cfm?elib=17517>
// * Description and diagrams of 1st- and 2nd-order TPT filters: 
// <https://www.willpirkle.com/706-2/>
//========================================================================================


//------------------`(ve.)sallenKeyOnePole`-----------------
// Sallen-Key generic One Pole filter that produces the LPF and HPF outputs (see description above).
//
// For the Faust implementation of this filter, recursion (`letrec`) is used 
// for storing filter "states". The output (e.g. `y`) is calculated by using 
// the input signal and the previous states of the filter.
// During the current recursive step, the states of the filter (e.g. `s`) for 
// the next step are also calculated.
// Admittedly, this is not an efficient way to implement a filter because it 
// requires independently calculating the output and each state during each 
// recursive step. However, it works as a way to store and use "states"
// within the constraints of Faust. 

// The simplest example is the 1st-order LPF (shown on the cover of Zavalishin 
// 2018 and Fig 4.3 of <https://www.willpirkle.com/706-2/>). Here, the input 
// signal is split in parallel for the calculation of the output signal, `y`, and 
// the state `s`. The value of the state is only used for feedback to the next 
// step of recursion. It is blocked (!) from also being routed to the output. 
// A trick used for calculating the state `s` is to observe that the input to 
// the delay block is the sum of two signal: what appears to be a feedforward 
// path and a feedback path. In reality, the signals being summed are identical 
// (signal*2) plus the value of the current state.
//
// #### Usage
//
// ```
// _ : sallenKeyOnePole(normFreq) : _,_
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)

declare sallenKeyOnePole author "Eric Tarr";
declare sallenKeyOnePole license "MIT-style STK-4.3 license";
sallenKeyOnePole(normFreq) = _<:(s,ylpf,-(ylpf)) : !,_,_
letrec {  
  's = -(s):*(2*G):+(s);
  'ylpf = -(s):*(G):+(s);
}
with{
  freq = 2*(10^(3*normFreq+1));
  wd = 2*ma.PI*freq;
  T = 1/ma.SR;
  wa = (2/T)*tan(wd*T/2);
  g = wa*T/2; 
  G = g/(1.0 + g);
};


//------------------`(ve.)sallenKeyOnePoleLPF`-----------------
// Sallen-Key One Pole lowpass filter (see description above).
// Specialize the generic implementation: keep the first LPF output, 
// the compiler will only generate the needed code.
//
// #### Usage
//
// ```
// _ : sallenKeyOnePoleLPF(normFreq) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)

declare sallenKeyOnePoleLPF author "Eric Tarr";
declare sallenKeyOnePoleLPF license "MIT-style STK-4.3 license";
sallenKeyOnePoleLPF(normFreq) = sallenKeyOnePole(normFreq) : _,!;


//------------------`(ve.)sallenKeyOnePoleHPF`-----------------
// Sallen-Key One Pole Highpass filter (see description above). The dry input 
// signal is routed in parallel to the output. The LPF'd signal is subtracted 
// from the input so that the HPF remains.
// Specialize the generic implementation: keep the second HPF output, 
// the compiler will only generate the needed code.
//
// #### Usage
//
// ```
// _ : sallenKeyOnePoleHPF(normFreq) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
//---------------------------------------------------------------------
declare sallenKeyOnePoleHPF author "Eric Tarr";
declare sallenKeyOnePoleHPF license "MIT-style STK-4.3 license";
sallenKeyOnePoleHPF(normFreq) = sallenKeyOnePole(normFreq) : !,_;


//------------------`(ve.)sallenKey2ndOrder`-----------------
// Sallen-Key generic 2nd order filter that produces the LPF, BPF and HPF outputs. 
//
// This is a 2nd-order Sallen-Key state-variable filter. The idea is that by 
// "tapping" into different points in the circuit, different filters 
// (LPF,BPF,HPF) can be achieved. See Figure 4.6 of 
// <https://www.willpirkle.com/706-2/>
//
// This is also a good example of the next step for generalizing the Faust 
// programming approach used for all these VA filters. In this case, there are 
// three things to calculate each recursive step (`y`,`s1`,`s2`). For each thing, the 
// circuit is only calculated up to that point. 
//
// Comparing the LPF to BPF, the output signal (`y`) is calculated similarly. 
// Except, the output of the BPF stops earlier in the circuit. Similarly, the 
// states (`s1` and `s2`) only differ in that `s2` includes a couple more terms 
// beyond what is used for `s1`. 
//
// #### Usage
//
// ```
// _ : sallenKey2ndOrder(normFreq,Q) : _,_,_
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare sallenKey2ndOrder author "Eric Tarr";
declare sallenKey2ndOrder license "MIT-style STK-4.3 license";
sallenKey2ndOrder(normFreq,Q) = _<:(s1,s2,ylpf,ybpf,yhpf) : !,!,_,_,_ 
letrec{
  's1 = -(s2):-(s1*FBs1):*(alpha0):*(g*2):+(s1);
  's2 = -(s2):-(s1*FBs1):*(alpha0):*(g):+(s1):*(g*2):+(s2);
  // Compute the LPF, BPF, HPF outputs
  'ylpf = -(s2):-(s1*FBs1):*(alpha0):*(g*2):+(s1):*(g):+(s2);
  'ybpf = -(s2):-(s1*FBs1):*(alpha0):*(g):+(s1);
  'yhpf = -(s2):-(s1*FBs1):*(alpha0); 
}
with{
  freq = 2*(10^(3*normFreq+1));
  wd = 2*ma.PI*freq;
  T = 1/ma.SR;
  wa = (2/T)*tan(wd*T/2);
  g = wa*T/2; 
  G = g/(1.0 + g);
  R = 1/(2*Q);
  FBs1 = (2*R+g);
  alpha0 = 1/(1 + 2*R*g + g*g);
};

//------------------`(ve.)sallenKey2ndOrderLPF`-----------------
// Sallen-Key 2nd order lowpass filter (see description above). 
// Specialize the generic implementation: keep the first LPF output,
// the compiler will only generate the needed code.
//
// #### Usage
//
// ```
// _ : sallenKey2ndOrderLPF(normFreq,Q) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare sallenKey2ndOrderLPF author "Eric Tarr";
declare sallenKey2ndOrderLPF license "MIT-style STK-4.3 license";
sallenKey2ndOrderLPF(normFreq,Q) = sallenKey2ndOrder(normFreq,Q) : _,!,!;


//------------------`(ve.)sallenKey2ndOrderBPF`-----------------
// Sallen-Key 2nd order bandpass filter (see description above). 
// Specialize the generic implementation: keep the second BPF output, 
// the compiler will only generate the needed code.
//
// #### Usage
//
// ```
// _ : sallenKey2ndOrderBPF(normFreq,Q) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare sallenKey2ndOrderBPF author "Eric Tarr";
declare sallenKey2ndOrderBPF license "MIT-style STK-4.3 license";
sallenKey2ndOrderBPF(normFreq,Q) = sallenKey2ndOrder(normFreq,Q) : !,_,!;


//------------------`(ve.)sallenKey2ndOrderHPF`-----------------
// Sallen-Key 2nd order highpass filter (see description above). 
// Specialize the generic implementation: keep the third HPF output, 
// the compiler will only generate the needed code.
//
// #### Usage
//
// ```
// _ : sallenKey2ndOrderHPF(normFreq,Q) : _
// ```
//
// Where:
//
// * `normFreq`: normalized frequency (0-1)
// * `Q`: q
//---------------------------------------------------------------------
declare sallenKey2ndOrderHPF author "Eric Tarr";
declare sallenKey2ndOrderHPF license "MIT-style STK-4.3 license";
sallenKey2ndOrderHPF(normFreq,Q) = sallenKey2ndOrder(normFreq,Q) : !,!,_;


//=========================================Effects========================================
//========================================================================================

//--------------------------`(ve.)wah4`-------------------------------
// Wah effect, 4th order.
// `wah4` is a standard Faust function.
//
// #### Usage
//
// ```
// _ : wah4(fr) : _
// ```
//
// Where:
//
// * `fr`: resonance frequency in Hz
//
// #### Reference
//
// <https://ccrma.stanford.edu/~jos/pasp/vegf.html>
//------------------------------------------------------------
wah4(fr) = 4*moog_vcf((3.2/4),fr:si.smooth(0.999));

//------------------------`(ve.)autowah`-----------------------------
// Auto-wah effect.
// `autowah` is a standard Faust function.
//
// #### Usage
//
// ```
// _ : autowah(level) : _
// ```
//
// Where:
//
// * `level`: amount of effect desired (0 to 1).
//------------------------------------------------------------
autowah(level,x) = level * crybaby(an.amp_follower(0.1,x),x) + (1.0-level)*x;

//--------------------------`(ve.)crybaby`-----------------------------
// Digitized CryBaby wah pedal.
// `crybaby` is a standard Faust function.
//
// #### Usage
//
// ```
// _ : crybaby(wah) : _
// ```
//
// Where:
//
// * `wah`: "pedal angle" from 0 to 1
//
// #### Reference
//
// <https://ccrma.stanford.edu/~jos/pasp/vegf.html>
//------------------------------------------------------------
crybaby(wah) = *(gs) : fi.tf2(1,-1,0,a1s,a2s)
with {
  Q  = pow(2.0,(2.0*(1.0-wah)+1.0)); // Resonance "quality factor"
  fr = 450.0*pow(2.0,2.3*wah);       // Resonance tuning
  g  = 0.1*pow(4.0,wah);             // gain (optional)

  // Biquad fit using z = exp(s T) ~ 1 + sT for low frequencies:
  frn = fr/ma.SR; // Normalized pole frequency (cycles per sample)
  R = 1 - ma.PI*frn/Q; // pole radius
  theta = 2*ma.PI*frn; // pole angle
  a1 = 0-2.0*R*cos(theta); // biquad coeff
  a2 = R*R;                // biquad coeff

  // dezippering of slider-driven signals:
  s = 0.999; // smoothing parameter (one-pole pole location)
  a1s = a1 : si.smooth(s);
  a2s = a2 : si.smooth(s);
  gs =  g  : si.smooth(s);

  //tf2 = component("filters.lib").tf2;
};

// end jos section
/************************************************************************
************************************************************************
FAUST library file, GRAME section

Except where noted otherwise, Copyright (C) 2003-2017 by GRAME,
Centre National de Creation Musicale.
----------------------------------------------------------------------
GRAME LICENSE

This program is free software; you can redistribute it and/or modify
it under the terms of the GNU Lesser General Public License as
published by the Free Software Foundation; either version 2.1 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 Lesser General Public License for more details.

You should have received a copy of the GNU Lesser General Public
License along with the GNU C Library; if not, write to the Free
Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA
02111-1307 USA.

EXCEPTION TO THE LGPL LICENSE : As a special exception, you may create a
larger FAUST program which directly or indirectly imports this library
file and still distribute the compiled code generated by the FAUST
compiler, or a modified version of this compiled code, under your own
copyright and license. This EXCEPTION TO THE LGPL LICENSE explicitly
grants you the right to freely choose the license for the resulting
compiled code. In particular the resulting compiled code has no obligation
to be LGPL or GPL. For example you are free to choose a commercial or
closed source license or any other license if you decide so.
************************************************************************
************************************************************************/

//----------------------------`(ve.)vocoder`-------------------------
// A very simple vocoder where the spectrum of the modulation signal
// is analyzed using a filter bank.
// `vocoder` is a standard Faust function.
//
// #### Usage
//
// ```
// _ : vocoder(nBands,att,rel,BWRatio,source,excitation) : _
// ```
//
// Where:
//
// * `nBands`: Number of vocoder bands
// * `att`: Attack time in seconds
// * `rel`: Release time in seconds
// * `BWRatio`: Coefficient to adjust the bandwidth of each band (0.1 - 2)
// * `source`: Modulation signal
// * `excitation`: Excitation/Carrier signal
//------------------------------------------------------------
declare oneVocoderBand author "Romain Michon";
oneVocoderBand(band,bandsNumb,bwRatio,bandGain,x) = x : fi.resonbp(bandFreq,bandQ,bandGain) with {
        bandFreq = 25*pow(2,(band+1)*(9/bandsNumb));
        BW = (bandFreq - 25*pow(2,(band)*(9/bandsNumb)))*bwRatio;
        bandQ = bandFreq/BW;
};

vocoder(nBands,att,rel,BWRatio,source,excitation) = source <: par(i,nBands,oneVocoderBand(i,nBands,BWRatio,1) :
	an.amp_follower_ar(att,rel) : _,excitation : oneVocoderBand(i,nBands,BWRatio)) :> _ ;

//########################################################################################
/************************************************************************
FAUST library file, further contributions section

All contributions below should indicate both the contributor and terms
of license.  If no such indication is found, "git blame" will say who
last edited each line, and that person can be emailed to inquire about
license disposition, if their license choice is not already indicated
elsewhere among the libraries.  It is expected that all software will be
released under LGPL, STK-4.3, MIT, BSD, or a similar FOSS license.
************************************************************************/

// end further further contributions section