//#################################### reverbs.lib ########################################
// A library of reverb effects. Its official prefix is `re`.
//
// #### References
// * <https://github.com/grame-cncm/faustlibraries/blob/master/reverbs.lib>
//########################################################################################

ma = library("maths.lib");
ba = library("basics.lib");
de = library("delays.lib");
ro = library("routes.lib");
si = library("signals.lib");
fi = library("filters.lib");
os = library("oscillators.lib");
it = library("interpolators.lib");
ef = library("misceffects.lib");
sp = library("spats.lib");
aa = library("aanl.lib");

declare name "Faust Reverb Library";
declare version "1.4.0";

//########################################################################################
/************************************************************************
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!)

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

//=============================Schroeder Reverberators======================================
//==========================================================================================

//------------------------------`(re.)jcrev`------------------------------
// This artificial reverberator take a mono signal and output stereo
// (`satrev`) and quad (`jcrev`). They were implemented by John Chowning
// in the MUS10 computer-music language (descended from Music V by Max
// Mathews).  They are Schroeder Reverberators, well tuned for their size.
// Nowadays, the more expensive freeverb is more commonly used (see the
// Faust examples directory).
//
// `jcrev` reverb below was made from a listing of "RV", dated April 14, 1972,
// which was recovered from an old SAIL DART backup tape.
// John Chowning thinks this might be the one that became the
// well known and often copied JCREV.
//
// `jcrev` is a standard Faust function.
//
// #### Usage
//
// ```
// _ : jcrev : _,_,_,_
// ```
//------------------------------------------------------------
jcrev = *(0.06) : allpass_chain <: comb_bank : mix_mtx with {
  rev1N = fi.rev1;
  rev12(len,g) = rev1N(2048,len,g);
  rev14(len,g) = rev1N(4096,len,g);
  allpass_chain =
    fi.rev2(512,347,0.7) :
    fi.rev2(128,113,0.7) :
    fi.rev2(64, 37,0.7);
  comb_bank =
    rev12(1601,.802),
    rev12(1867,.773),
    rev14(2053,.753),
    rev14(2251,.733);
    mix_mtx = _,_,_,_ <: psum, -psum, asum, -asum : _,_,_,_ with {
    psum = _,_,_,_ :> _;
    asum = *(-1),_,*(-1),_ :> _;
  };
};


//------------------------------`(re.)satrev`------------------------------
// This artificial reverberator take a mono signal and output stereo
// (`satrev`) and quad (`jcrev`).  They were implemented by John Chowning
// in the MUS10 computer-music language (descended from Music V by Max
// Mathews).  They are Schroeder Reverberators, well tuned for their size.
// Nowadays, the more expensive freeverb is more commonly used (see the
// Faust examples directory).
//
// `satrev` was made from a listing of "SATREV", dated May 15, 1971,
// which was recovered from an old SAIL DART backup tape.
// John Chowning thinks this might be the one used on his
// often-heard brass canon sound examples, one of which can be found at
// <https://ccrma.stanford.edu/~jos/wav/FM-BrassCanon2.wav>.
//
// #### Usage
//
// ```
// _ : satrev : _,_
// ```
//------------------------------------------------------------
satrev = *(0.2) <: comb_bank :> allpass_chain <: _,*(-1) with {
  rev1N = fi.rev1;
  rev11(len,g) = rev1N(1024,len,g);
  rev12(len,g) = rev1N(2048,len,g);
  comb_bank =
    rev11(778,.827),
    rev11(901,.805),
    rev11(1011,.783),
    rev12(1123,.764);
  rev2N = fi.rev2;
  allpass_chain =
    rev2N(128,125,0.7) :
    rev2N(64, 42,0.7) :
    rev2N(16, 12,0.7);
};

//======================Feedback Delay Network (FDN) Reverberators========================
//========================================================================================

//--------------------------------`(re.)fdnrev0`---------------------------------
// Pure Feedback Delay Network Reverberator (generalized for easy scaling).
// `fdnrev0` is a standard Faust function.
//
// #### Usage
//
// ```
// <1,2,4,...,N signals> <:
// fdnrev0(MAXDELAY,delays,BBSO,freqs,durs,loopgainmax,nonl) :>
// <1,2,4,...,N signals>
// ```
//
// Where:
//
// * `N`: 2, 4, 8, ...  (power of 2)
// * `MAXDELAY`: power of 2 at least as large as longest delay-line length
// * `delays`: N delay lines, N a power of 2, lengths preferably coprime
// * `BBSO`: odd positive integer = order of bandsplit desired at freqs
// * `freqs`: NB-1 crossover frequencies separating desired frequency bands
// * `durs`: NB decay times (t60) desired for the various bands
// * `loopgainmax`: scalar gain between 0 and 1 used to "squelch" the reverb
// * `nonl`: nonlinearity (0 to 0.999..., 0 being linear)
//
// #### Reference
//
// <https://ccrma.stanford.edu/~jos/pasp/FDN_Reverberation.html>
//------------------------------------------------------------
fdnrev0(MAXDELAY, delays, BBSO, freqs, durs, loopgainmax, nonl)
  = (si.bus(2*N) :> si.bus(N) : delaylines(N)) ~
    (delayfilters(N,freqs,durs) : feedbackmatrix(N))
with {
  N = ba.count(delays);
  NB = ba.count(durs);
//assert(count(freqs)+1==NB);
  delayval(i) = ba.take(i+1,delays);
  dlmax(i) = MAXDELAY; // must hardwire this from argument for now
//dlmax(i) = 2^max(1,nextpow2(delayval(i))) // try when slider min/max is known
//           with { nextpow2(x) = ceil(log(x)/log(2.0)); };
// -1 is for feedback delay:
  delaylines(N) = par(i,N,(de.delay(dlmax(i),(delayval(i)-1))));
  delayfilters(N,freqs,durs) = par(i,N,filter(i,freqs,durs));
  feedbackmatrix(N) = bhadamard(N);
  vbutterfly(n) = si.bus(n) <: (si.bus(n):>bus(n/2)) , ((si.bus(n/2),(si.bus(n/2):par(i,n/2,*(-1)))) :> si.bus(n/2));
  bhadamard(2) = si.bus(2) <: +,-;
  bhadamard(n) = si.bus(n) <: (si.bus(n):>si.bus(n/2)) , ((si.bus(n/2),(si.bus(n/2):par(i,n/2,*(-1)))) :> si.bus(n/2))
                 : (bhadamard(n/2) , bhadamard(n/2));

  // Experimental nonlinearities:
  // nonlinallpass = apnl(nonl,-nonl);
  // s = nonl*PI;
  // nonlinallpass(x) = allpassnn(3,(s*x,s*x*x,s*x*x*x)); // filters.lib
     nonlinallpass = _; // disabled by default (rather expensive)

  filter(i,freqs,durs) = fi.filterbank(BBSO,freqs) : par(j,NB,*(g(j,i)))
                         :> *(loopgainmax) / sqrt(N) : nonlinallpass
  with {
    dur(j) = ba.take(j+1,durs);
    n60(j) = dur(j)*ma.SR; // decay time in samples
    g(j,i) = exp(-3.0*log(10.0)*delayval(i)/n60(j));
        // ~ 1.0 - 6.91*delayval(i)/(SR*dur(j)); // valid for large dur(j)
  };
};


//-------------------------------`(re.)zita_rev_fdn`-------------------------------
// Internal 8x8 late-reverberation FDN used in the FOSS Linux reverb `zita-rev1`
// by Fons Adriaensen <fons@linuxaudio.org>.  This is an FDN reverb with
// allpass comb filters in each feedback delay in addition to the
// damping filters.
//
// #### Usage
//
// ```
// si.bus(8) : zita_rev_fdn(f1,f2,t60dc,t60m,fsmax) : si.bus(8)
// ```
//
// Where:
//
// * `f1`: crossover frequency (Hz) separating dc and midrange frequencies
// * `f2`: frequency (Hz) above f1 where T60 = t60m/2 (see below)
// * `t60dc`: desired decay time (t60) at frequency 0 (sec)
// * `t60m`: desired decay time (t60) at midrange frequencies (sec)
// * `fsmax`: maximum sampling rate to be used (Hz)
//
// #### Reference
//
// * <http://www.kokkinizita.net/linuxaudio/zita-rev1-doc/quickguide.html>
// * <https://ccrma.stanford.edu/~jos/pasp/Zita_Rev1.html>
//------------------------------------------------------------
zita_rev_fdn(f1,f2,t60dc,t60m,fsmax) =
  ((si.bus(2*N) :> allpass_combs(N) : feedbackmatrix(N)) ~
   (delayfilters(N,freqs,durs) : fbdelaylines(N)))
with {
  N = 8;

  // Delay-line lengths in seconds:
  apdelays = (0.020346, 0.024421, 0.031604, 0.027333, 0.022904,
              0.029291, 0.013458, 0.019123); // feedforward delays in seconds
  tdelays = (0.153129, 0.210389, 0.127837, 0.256891, 0.174713,
             0.192303, 0.125000, 0.219991); // total delays in seconds
  tdelay(i) = floor(0.5 + ma.SR*ba.take(i+1,tdelays)); // samples
  apdelay(i) = floor(0.5 + ma.SR*ba.take(i+1,apdelays));
  fbdelay(i) = tdelay(i) - apdelay(i);
  // NOTE: Since SR is not bounded at compile time, we can't use it to
  // allocate delay lines; hence, the fsmax parameter:
  tdelaymaxfs(i) = floor(0.5 + fsmax*ba.take(i+1,tdelays));
  apdelaymaxfs(i) = floor(0.5 + fsmax*ba.take(i+1,apdelays));
  fbdelaymaxfs(i) = tdelaymaxfs(i) - apdelaymaxfs(i);
  nextpow2(x) = ceil(log(x)/log(2.0));
  maxapdelay(i) = int(2.0^max(1.0,nextpow2(apdelaymaxfs(i))));
  maxfbdelay(i) = int(2.0^max(1.0,nextpow2(fbdelaymaxfs(i))));

  apcoeff(i) = select2(i&1,0.6,-0.6);  // allpass comb-filter coefficient
  allpass_combs(N) =
    par(i,N,(fi.allpass_comb(maxapdelay(i),apdelay(i),apcoeff(i)))); // filters.lib
  fbdelaylines(N) = par(i,N,(de.delay(maxfbdelay(i),(fbdelay(i)))));
  freqs = (f1,f2); durs = (t60dc,t60m);
  delayfilters(N,freqs,durs) = par(i,N,filter(i,freqs,durs));
  feedbackmatrix(N) = ro.hadamard(N);

  staynormal = 10.0^(-20); // let signals decay well below LSB, but not to zero

  special_lowpass(g,f) = si.smooth(p) with {
    // unity-dc-gain lowpass needs gain g at frequency f => quadratic formula:
    p = mbo2 - sqrt(max(0,mbo2*mbo2 - 1.0)); // other solution is unstable
    mbo2 = (1.0 - gs*c)/(1.0 - gs); // NOTE: must ensure |g|<1 (t60m finite)
    gs = g*g;
    c = cos(2.0*ma.PI*f/float(ma.SR));
  };

  filter(i,freqs,durs) = lowshelf_lowpass(i)/sqrt(float(N))+staynormal
  with {
    lowshelf_lowpass(i) = gM*low_shelf1_l(g0/gM,f(1)):special_lowpass(gM,f(2));
    low_shelf1_l(G0,fx,x) = x + (G0-1)*fi.lowpass(1,fx,x); // filters.lib
    g0 = g(0,i);
    gM = g(1,i);
    f(k) = ba.take(k,freqs);
    dur(j) = ba.take(j+1,durs);
    n60(j) = dur(j)*ma.SR; // decay time in samples
    g(j,i) = exp(-3.0*log(10.0)*tdelay(i)/n60(j));
  };
};

// Stereo input delay used by zita_rev1 in both stereo and ambisonics mode:
zita_in_delay(rdel) = zita_delay_mono(rdel), zita_delay_mono(rdel) with {
  zita_delay_mono(rdel) = de.delay(8192,ma.SR*rdel*0.001) * 0.3;
};

// Stereo input mapping used by zita_rev1 in both stereo and ambisonics mode:
zita_distrib2(N) = _,_ <: fanflip(N) with {
   fanflip(4) = _,_,*(-1),*(-1);
   fanflip(N) = fanflip(N/2),fanflip(N/2);
};


//----------------------------`(re.)zita_rev1_stereo`---------------------------
// Extend `zita_rev_fdn` to include `zita_rev1` input/output mapping in stereo mode.
// `zita_rev1_stereo` is a standard Faust function.
//
// #### Usage
//
// ```
// _,_ : zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax) : _,_
// ```
//
// Where:
//
// `rdel`  = delay (in ms) before reverberation begins (e.g., 0 to ~100 ms)
// (remaining args and refs as for `zita_rev_fdn` above)
//------------------------------------------------------------
zita_rev1_stereo(rdel,f1,f2,t60dc,t60m,fsmax) =
   zita_in_delay(rdel)
 : zita_distrib2(N)
 : zita_rev_fdn(f1,f2,t60dc,t60m,fsmax)
 : output2(N)
with {
 N = 8;
 output2(N) = outmix(N) : *(t1),*(t1);
 t1 = 0.37; // zita-rev1 linearly ramps from 0 to t1 over one buffer
 outmix(4) = !,ro.butterfly(2),!; // probably the result of some experimenting!
 outmix(N) = outmix(N/2),par(i,N/2,!);
};


//-----------------------------`(re.)zita_rev1_ambi`---------------------------
// Extend `zita_rev_fdn` to include `zita_rev1` input/output mapping in
// "ambisonics mode", as provided in the Linux C++ version.
//
// #### Usage
//
// ```
// _,_ : zita_rev1_ambi(rgxyz,rdel,f1,f2,t60dc,t60m,fsmax) : _,_,_,_
// ```
//
// Where:
//
// `rgxyz` = relative gain of lanes 1,4,2 to lane 0 in output (e.g., -9 to 9)
//   (remaining args and references as for zita_rev1_stereo above)
//------------------------------------------------------------
zita_rev1_ambi(rgxyz,rdel,f1,f2,t60dc,t60m,fsmax) =
   zita_in_delay(rdel)
 : zita_distrib2(N)
 : zita_rev_fdn(f1,f2,t60dc,t60m,fsmax)
 : output4(N) // ambisonics mode
with {
  N = 8;
  output4(N) = select4 : *(t0),*(t1),*(t1),*(t1);
  select4 = _,_,_,!,_,!,!,! : _,_,cross with { cross(x,y) = y,x; };
  t0 = 1.0/sqrt(2.0);
  t1 = t0 * 10.0^(0.05 * rgxyz);
};

// 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.
************************************************************************
************************************************************************/

//-------------------`(re.)vital_rev`------------------------------------------
// A port of the reverb from the Vital synthesizer. All input parameters
// have been normalized to a continuous [0,1] range, making them easy to modulate.
// The scaling of the parameters happens inside the function.
//
// #### Usage
//
// ```
// _,_ : vital_rev(prelow, prehigh, lowcutoff, highcutoff, lowgain, highgain, chorus_amt, chorus_freq, predelay, time, size, mix) : _,_ 
// ```
//
// Where:
//
// * `prelow`: In the pre-filter, this is the cutoff frequency of a high-pass filter (hence a low value).
// * `prehigh`: In the pre-filter, this is the cutoff frequency of a low-pass filter (hence a high value).
// * `lowcutoff`: In the feedback filter stage, this is the cutoff frequency of a low-shelf filter.
// * `highcutoff`: In the feedback filter stage, this is the cutoff frequency of a high-shelf filter.
// * `lowgain`: In the feedback filter stage, this is the gain of a low-shelf filter.
// * `highgain`: In the feedback filter stage, this is the gain of a high-shelf filter.
// * `chorus_amt`: The amount of chorus modulation in the main delay lines.
// * `chorus_freq`: The LFO rate of chorus modulation in the main delay lines.
// * `predelay`: The amount of pre-delay time.
// * `time`: The decay time of the reverb.
// * `size`: The size of the room.
// * `mix`: A wetness value to use in a final dry/wet mixer.
//-----------------------------------------------------------------------------
vital_rev(_prelow, _prehigh, _lowcutoff, _highcutoff, _lowgain, _highgain, _chorus_amt, _chorus_freq, _predelay, _time, _size, _mix) = ef.dryWetMixerConstantPower(wetAmount, reverb) : gainMakeup
with {
    /* Copyright 2013-2019 Matt Tytel
    *
    * vital 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.
    *
    * vital 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 vital. If not, see <http://www.gnu.org/licenses/>.
    */

    /*
    Adapted from https://github.com/mtytel/vital/blob/main/src/synthesis/effects/reverb.cpp
    C++ author: Matt Tytel
    Faust/Python author: David Braun
    This is not an official product related to Vital.

    David's analysis of the code:
    This technique is an example of a Feedback Delay Network (FDN) (https://ccrma.stanford.edu/~jos/pasp/FDN_Reverberation.html).
    Specifically, the feedback matrix is a 16x16 Hadamard matrix that has been efficiently implemented.

    This matrix is the sum of three matrices:
    * an identity matrix
    * other_feedback
    * adjacent_feedback

    Although other_feedback and adjacent_feedback are not orthogonal,
    when all three matrices are summed together, they are orthogonal and form a Hadamard matrix (ignoring a gain factor).
    The orthogonality makes it a valid FDN matrix.

    Note that other_feedback and adjacent_feedback are both 16x16 matrices.
    Each is cleverly factorized into the product of a 16x4 matrix and a 4x16 matrix,
    resulting in a 16x16 matrix.

    Let's use numpy to explain the matrices and eventually show that the final result is Hadamard:

    ```python
    import numpy as np
    from scipy.linalg import block_diag

    # Make identity matrix
    identity = np.eye(16)

    # Make adjacent_matrix matrix
    x = np.ones((4,))
    A = block_diag(x,x,x,x)
    assert A.shape == (4,16)
    adjacent_matrix = (A.T @ A)*-.5
    assert adjacent_matrix.shape == (16,16)

    # Make other_feedback
    A = np.eye(4)
    A = np.concatenate([A,A,A,A], axis=1)
    assert A.shape == (4,16)
    other_feedback = (A.T @ A)*-.5 + np.full((16,16), 1/4)
    assert other_feedback.shape == (16,16)

    final_matrix = identity + adjacent_matrix + other_feedback

    # assert that final_matrix is orthogonal
    assert ((final_matrix @ final_matrix.T) == identity).all()

    # assert that (final_matrix*4) is a Hadamard matrix (see Wikipedia)
    N = 16
    assert np.round(np.linalg.det(final_matrix*4)) == N**(N/2)
    ```

    Other notes:
    Vital's reverb is similar to dm.zita_rev1.
    In this Faust port of Vital, I placed the allpasses, and matrix in the
    "forward" stage. I placed the filters, decay and delay in the "feedback" stage.
    Similarly, in Zita Rev, the forward stage has the allpasses and matrix, while the
    feedback stage has the filters and delay.
    */

    DELAY_QUALITY = 3; // [1-3] where 3 is best, and 1 is linear interpolation

    kBaseSampleRate = 44100; // it's ok if ma.SR is not 44100, but don't change this
    kT60Amplitude = 0.001;
    kAllpassFeedback = 0.6;

    kMaxChorusDrift = 2500.0; // samples
    kMinDecayTime = 0.1; // seconds
    kMaxDecayTime = 100.0; // seconds
    kMaxChorusFrequency = 16.0; // Hz

    kPreDelayMaxSec = 0.3; // seconds

    kNetworkContainers = 4; // the number of delay lines used for allpassDelay
    kContainerSize = 4; // how long the allpassDelay waveforms below are

    kMinSizePower = -3;
    kMaxSizePower = 1; // decrease this to shorten the delay lines and reduce memory cost.

    clampremap(from1, from2, to1, to2) = aa.clip(from1, from2) : it.remap(from1, from2, to1, to2);

    low_pre_cutoff_frequency = _prelow : clampremap(0, 1, 16, 135) : ba.midikey2hz;
    high_pre_cutoff_frequency = _prehigh : clampremap(0, 1, 16, 135) : ba.midikey2hz;

    low_cutoff_frequency = _lowcutoff : clampremap(0, 1, 16, 135) : ba.midikey2hz;
    high_cutoff_frequency = _highcutoff : clampremap(0, 1, 16, 135) : ba.midikey2hz;
    low_gain = _lowgain : clampremap(0, 1, -24, 0);
    high_gain = _highgain : clampremap(0, 1, -24, 0);

    chorus_amount = _chorus_amt : aa.clip(0,1) : quadScale : _*kMaxChorusDrift*sample_rate_ratio*size_mult;
    chorus_frequency = _chorus_freq : exp(it.remap(0, 1, -8, 3)) : min(kMaxChorusFrequency);

    pre_delay_samples = _predelay : clampremap(0, 1, 0, kPreDelayMaxSec) : _*ma.SR;
    decay_samples = _time : exp(it.remap(0, 1, -6, 6)) : aa.clip(kMinDecayTime, kMaxDecayTime) : _*kBaseSampleRate;
    size = _size : aa.clip(0, 1);

    wetAmount = _mix : aa.clip(0, 1);

    quadScale(x) = x*x;

    kNetworkSize = kNetworkContainers * kContainerSize;

    containerBus = si.bus(kContainerSize);
    networkBus = si.bus(kNetworkSize);

    size_mult = pow(2, it.interpolate_linear(size, kMinSizePower, kMaxSizePower));

    sample_rate_ratio = ma.SR/kBaseSampleRate;

    kAllpassMaxDelay = 1001; // manually inspect values below and take max value
    // allpassDelay(i) where `i` corresponds to 0 to (kNetworkContainers-1)
    allpassDelay(0, i) = ba.take(i+1, (1001, 799, 933, 876));
    allpassDelay(1, i) = ba.take(i+1, (895, 807, 907, 853));
    allpassDelay(2, i) = ba.take(i+1, (957, 1019, 711, 567));
    allpassDelay(3, i) = ba.take(i+1, (833, 779, 663, 997));

    kFeedbackMaxDelay = (11329 + kMaxChorusDrift)*sample_rate_ratio*pow(2, kMaxSizePower); // pow(2, kMaxSizePower) is the max size_mult
    // feedbackDelayHelp(i) where `i` corresponds to 0 to (kNetworkContainers-1)
    feedbackDelayHelp(0, i) = ba.take(i+1, (6753.2, 9278.4, 7704.5, 11328.5));
    feedbackDelayHelp(1, i) = ba.take(i+1, (9701.12, 5512.5, 8480.45, 5638.65));
    feedbackDelayHelp(2, i) = ba.take(i+1, (3120.73, 3429.5, 3626.37, 7713.52));
    feedbackDelayHelp(3, i) = ba.take(i+1, (4521.54, 6518.97, 5265.56, 5630.25));

    feedbackDelay(i, j) = feedbackDelayHelp(i, j) : _ * size_mult * sample_rate_ratio;

    getDecay(i, j) = pow(kT60Amplitude, feedbackDelay(i, j) / (decay_samples*sample_rate_ratio));

    pre_delay = sp.stereoize(de.fdelayltv(N, kPreDelayMaxSec*ma.SR, safe_delay_samples))
    with {
        safe_delay_samples = pre_delay_samples : max((N-1)/2);
        N = DELAY_QUALITY;
    };

    pre_filter = sp.stereoize(fi.highpass(1, low_pre_cutoff_frequency) : fi.lowpass(1, high_pre_cutoff_frequency) : _/kNetworkContainers);

    // mathematical hard-syncing phasor (see `phasor_imp` in `faustlibraries/oscillators.lib`)
    m_lfo_sine(freq, phase) = ((select2(hard_reset, +(freq/ma.SR), phase) : ma.decimal) ~ _) : sin(_*2*ma.PI)
    with {
        hard_reset = (1-1'); // To correctly start at `phase` at the first sample
    };

    phase_offset(i) = i / kNetworkSize;
    real_part(phase_offset) = m_lfo_sine(chorus_frequency, phase_offset + .25);  // "real" cosine part
    imag_part(phase_offset) = m_lfo_sine(chorus_frequency, phase_offset);  // "imaginary" sine part

    lfo1(i) = phase_offset(i) : real_part;
    lfo2(i) = phase_offset(i) : imag_part;

    mydelay(delay_samples) = de.fdelayltv(N, kFeedbackMaxDelay, safe_delay_samples)
    with {
        safe_delay_samples = delay_samples : max((N-1)/2);
        N = DELAY_QUALITY;
    };

    delay = par(i, kContainerSize, mydelay(feedbackDelay(0, i) + lfo1(i)*chorus_amount)),
            par(i, kContainerSize, mydelay(feedbackDelay(1, i) - lfo1(i)*chorus_amount)),
            par(i, kContainerSize, mydelay(feedbackDelay(2, i) + lfo2(i)*chorus_amount)),
            par(i, kContainerSize, mydelay(feedbackDelay(3, i) - lfo2(i)*chorus_amount));

    allpass(j) = par(i, kContainerSize, fi.allpass_comb(kAllpassMaxDelay, allpassDelay(j, i), kAllpassFeedback));
    allpasses = par(i, kNetworkContainers, allpass(i));

    filters = par(i, kNetworkSize,
        (fi.lowshelf(1, low_gain, low_cutoff_frequency) : fi.highshelf(1, high_gain, high_cutoff_frequency))
    );

    other_feedback = total_rows <: (containerBus :> _/kContainerSize <: containerBus), par(i, kContainerSize, _*-.5) :> containerBus <: networkBus
    with {
        total_rows = networkBus :> containerBus;
    };

    adjacent_feedback = par(i, kNetworkContainers, containerBus :>_) : par(i, kNetworkContainers, _*s) <: par(i, kNetworkContainers, _<: containerBus)
    with {
        s = -.5*kNetworkContainers/kContainerSize;
    };

    decay = par(i, kNetworkContainers, par(j, kContainerSize, _*getDecay(i, j)));

    matrix =  networkBus <: networkBus, other_feedback, adjacent_feedback :> networkBus;

    reverb =
    // PRE
    pre_delay : pre_filter <:

    // BODY
    (networkBus, networkBus :> allpasses : matrix)
    ~
    // FEEDBACK
    (filters : decay : delay)

    // POST
    // todo: It's odd that we have to swap the channels with ro.crossnn.
    // This might be covering up some other mistake.
    :> par(i, 2, _*2/kNetworkContainers) : ro.crossnn(1);

    // dryWetMixerConstantPower causes an "insertion loss" when wet is not 50%.
    // So a choice of 0% wet would actually have less volume than not using the reverb at all.
    // We can makeup for this insertion loss by undoing the sqrt(2) division inside dryWetMixerConstantPower.
    gainMakeup = _*sqrt(2), _*sqrt(2);
};

declare vital_rev author "David Braun";
declare vital_rev license "GPL-3.0";

//===============================Freeverb===================================
//==========================================================================

//----------------------------`(re.)mono_freeverb`-------------------------
// A simple Schroeder reverberator primarily developed by "Jezar at Dreampoint" that
// is extensively used in the free-software world. It uses four Schroeder allpasses in
// series and eight parallel Schroeder-Moorer filtered-feedback comb-filters for each
// audio channel, and is said to be especially well tuned.
//
// `mono_freeverb` is a standard Faust function.
//
// #### Usage
//
// ```
// _ : mono_freeverb(fb1, fb2, damp, spread) : _
// ```
//
// Where:
//
// * `fb1`: coefficient of the lowpass comb filters (0-1)
// * `fb2`: coefficient of the allpass comb filters (0-1)
// * `damp`: damping of the lowpass comb filter (0-1)
// * `spread`: spatial spread in number of samples (for stereo)
//
// #### License
// While this version is licensed LGPL (with exception) along with other GRAME
// library functions, the file freeverb.dsp in the examples directory of older
// Faust distributions, such as faust-0.9.85, was released under the BSD license,
// which is less restrictive.
//------------------------------------------------------------
declare mono_freeverb author "Romain Michon";

mono_freeverb(fb1, fb2, damp, spread) = _ <: par(i,8,lbcf(combtuningL(i)+spread,fb1,damp))
	:> seq(i,4,fi.allpass_comb(1024, allpasstuningL(i)+spread, -fb2))
with {
 
    // Filters parameters
    combtuningL(0) = adaptSR(1116);
    combtuningL(1) = adaptSR(1188);
    combtuningL(2) = adaptSR(1277);
    combtuningL(3) = adaptSR(1356);
    combtuningL(4) = adaptSR(1422);
    combtuningL(5) = adaptSR(1491);
    combtuningL(6) = adaptSR(1557);
    combtuningL(7) = adaptSR(1617);

    allpasstuningL(0) = adaptSR(556);
    allpasstuningL(1) = adaptSR(441);
    allpasstuningL(2) = adaptSR(341);
    allpasstuningL(3) = adaptSR(225);
    
    // Lowpass Feedback Combfilter:
    // <https://ccrma.stanford.edu/~jos/pasp/Lowpass_Feedback_Comb_Filter.html>
    lbcf(dt, fb, damp) = (+ : @ (max(0, (dt - 1)))) ~ (*(1-damp) : (+ ~ *(damp)) : *(fb)) : mem;
     
    origSR = 44100;
    adaptSR(val) = val*ma.SR/origSR : int;

};


//----------------------------`(re.)stereo_freeverb`-------------------------
// A simple Schroeder reverberator primarily developed by "Jezar at Dreampoint" that
// is extensively used in the free-software world. It uses four Schroeder allpasses in
// series and eight parallel Schroeder-Moorer filtered-feedback comb-filters for each
// audio channel, and is said to be especially well tuned.
//
// #### Usage
//
// ```
// _,_ : stereo_freeverb(fb1, fb2, damp, spread) : _,_
// ```
//
// Where:
//
// * `fb1`: coefficient of the lowpass comb filters (0-1)
// * `fb2`: coefficient of the allpass comb filters (0-1)
// * `damp`: damping of the lowpass comb filter (0-1)
// * `spread`: spatial spread in number of samples (for stereo)
//------------------------------------------------------------
declare stereo_freeverb author "Romain Michon";

stereo_freeverb(fb1, fb2, damp, spread) = + <: mono_freeverb(fb1, fb2, damp, 0), mono_freeverb(fb1, fb2, damp, spread);

//########################################################################################
/************************************************************************
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.
************************************************************************/

//===============================Dattorro Reverb============================
//==========================================================================

//-------------------------------`(re.)dattorro_rev`-------------------------------
// Reverberator based on the Dattorro reverb topology. This implementation does
// not use modulated delay lengths (excursion).
//
// #### Usage
//
// ```
// _,_ : dattorro_rev(pre_delay, bw, i_diff1, i_diff2, decay, d_diff1, d_diff2, damping) : _,_
// ```
//
// Where:
//
// * `pre_delay`: pre-delay in samples (fixed at compile time)
// * `bw`: band-width filter (pre filtering); (0 - 1)
// * `i_diff1`: input diffusion factor 1; (0 - 1)
// * `i_diff2`: input diffusion factor 2;
// * `decay`: decay rate; (0 - 1); infinite decay = 1.0
// * `d_diff1`: decay diffusion factor 1; (0 - 1)
// * `d_diff2`: decay diffusion factor 2;
// * `damping`: high-frequency damping; no damping = 0.0
//
// #### Reference
//
// <https://ccrma.stanford.edu/~dattorro/EffectDesignPart1.pdf>
//------------------------------------------------------------
declare dattorro_rev author "Jakob Zerbian";
declare dattorro_rev licence "MIT-style STK-4.3 license";

dattorro_rev(pre_delay, bw, i_diff1, i_diff2, decay, d_diff1, d_diff2, damping) = 
    si.bus(2) : + : *(0.5) : predelay : bw_filter : diffusion_network <: ((si.bus(4) :> _,_) ~ (reverb_network : ro.cross(2)))
with {
    // allpass using delay with fixed size
    allpass_f(t, a) = (+ <: @(t),*(a)) ~ *(-a) : mem,_ : +;

    // input pre-delay and diffusion
    predelay = @(pre_delay);
    bw_filter = *(bw) : +~(mem : *(1-bw));
    diffusion_network = allpass_f(142, i_diff1) : allpass_f(107, i_diff1) : allpass_f(379, i_diff2) : allpass_f(277, i_diff2);

    // reverb loop
    reverb_network = par(i, 2, block(i)) with {
        d = (672, 908, 4453, 4217, 1800, 2656, 3720, 3163);
        block(i) = allpass_f(ba.take(i+1, d),-d_diff1) : @(ba.take(i+3, d)) : damp : 
            allpass_f(ba.take(i+5, d), d_diff2) : @(ba.take(i+5, d)) : *(decay)
        with {
            damp = *(1-damping) : +~*(damping) : *(decay);
        };
    };
};


//-------------------------------`(re.)dattorro_rev_default`-------------------------------
// Reverberator based on the Dattorro reverb topology with reverb parameters from the
// original paper.
// This implementation does not use modulated delay lengths (excursion) and
// uses zero length pre-delay.
//
// #### Usage
//
// ```
// _,_ : dattorro_rev_default : _,_
// ```
//
// #### Reference
//
// <https://ccrma.stanford.edu/~dattorro/EffectDesignPart1.pdf>
//------------------------------------------------------------
declare dattorro_rev_default author "Jakob Zerbian";
declare dattorro_rev_default license "MIT-style STK-4.3 license";

dattorro_rev_default = dattorro_rev(0, 0.9995, 0.75, 0.625, 0.5, 0.7, 0.5, 0.0005);

//===============================JPverb and Greyhole Reverbs============================
//======================================================================================

jp_gh_rev = environment {

    diffuser_aux(angle, g, scale1, scale2, size, block) = si.bus(2) <: ((si.bus(2):par(i,2,*(c_norm))
        : ((si.bus(4) :> si.bus(2)
            : block
            : rotator(angle)
            : (de.fdelay1a(8192, ma.primes(size*scale1):smooth_init(0.9999,ma.primes(size*scale1)) -1),
               de.fdelay1a(8192, ma.primes(size*scale2):smooth_init(0.9999,ma.primes(size*scale2)) -1)))
        ~ par(i,2,*(-s_norm))) : par(i,2,mem:*(c_norm)))
        ,
        par(i,2,*(s_norm)))
        :> si.bus(2)
        with {
            rotator(angle) = si.bus(2) <: (*(c),*(-s),*(s),*(c)) :(+,+) : si.bus(2)
            with {
                c = cos(angle);
                s = sin(angle);
            };
            c_norm = cos(g);
            s_norm = sin(g);
        };

    diffuser(angle, g, scale1, scale2, size) = diffuser_aux(angle,g,scale1,scale2,size,si.bus(2));

    // Nested version
    diffuser_nested(1, angle, g, scale, size) = diffuser_aux(angle,g,scale,scale+10,size,si.bus(2));
    diffuser_nested(N, angle, g, scale, size) = diffuser_aux(angle,g,scale,scale+10,size,diffuser_nested(N-1,angle,g,scale+13,size));

    smooth_init(s,default) = *(1.0 - s) : + ~ (+(default*init(1)):*(s)) with { init(value) = value - value'; };

    invSqrt2 = 1/sqrt(2);
    jpverb(t60, damp, size, early_diff, mod_depth, mod_freq, low, mid, high, low_cutoff, high_cutoff)
        = ((si.bus(4) :> (de.fdelay4(512, depth + depth*os.oscrs(mod_freq) + 5),de.fdelay4(512, depth + depth*os.oscrc(mod_freq) + 5))
            : par(i,2,si.smooth(damp))
            : diffuser(ma.PI/4,early_diff,55,240,size)
            : diffuser(ma.PI/4,early_diff,215,85,size)
            : diffuser(ma.PI/4,early_diff,115,190,size)
            : diffuser(ma.PI/4,early_diff,175,145,size)
        )~(seq(i,5,diffuser(ma.PI/4,invSqrt2,10+30*i,110+30*i,size))
            : par(i,2,de.fdelay4(512, depth + (-1^i)*depth*os.oscrc(mod_freq)+5)
            : de.fdelay1a(8192,(ma.primes(size*(54+150*i))
            : smooth_init(0.995,ma.primes(size*(54+150*i)))) -1))
            : seq(i,5,diffuser(ma.PI/4,invSqrt2,125+30*i,25+30*i,size))
            : par(i,2,de.fdelay4(8192, depth + (-1^i)*depth*os.oscrs(mod_freq)+5)
            : de.fdelay1a(8192,(ma.primes(size*(134-100*i))
            : smooth_init(0.995,ma.primes(size*(134-100*i)))) -1))
            : par(i,2,fi.filterbank(5,(low_cutoff,high_cutoff)):(_*(high),_*(mid),_*(low)) :> _)
            : par(i,2,*(fb))))
        with {
            depth = 50*mod_depth;
            calib = 1.7; // Calibration constant given by t60 in seconds when fb = 0.5
            total_length = calib*0.1*(size*5/4 -1/4);
            fb = 10^(-3/((t60)/(total_length)));
        };

    greyhole(dt, damp, size, early_diff, feedback, mod_depth, mod_freq)
        = (si.bus(4) :> seq(i,3,diffuser_nested(4,ma.PI/2,(-1^i)*diff,10+19*i,size))
            : par(i,2,si.smooth(damp_interp)))
        ~((de.fdelay4(512, 10 + depth + depth*os.oscrc(freq)),de.fdelay4(512, 10 + depth + depth*os.oscrs(freq)))
            : (de.sdelay(65536,44100/2,floor(dt_constrained)),de.sdelay(65536,44100/2,floor(dt_constrained)))
            : par(i,2,*(fb)))
        with {
            fb = feedback:linear_interp;
            depth = ((ma.SR/44100)*50*mod_depth):linear_interp;
            freq = mod_freq:linear_interp;
            diff = early_diff:linear_interp;
            dt_constrained = min(65533,ma.SR*dt);
            damp_interp = damp:linear_interp;
            linear_interp(x) = (x+x')/2;
        };

};

//-------------------------------`(re.)jpverb`-------------------------------
// An algorithmic reverb (stereo in/out), inspired by the lush chorused sound 
// of certain vintage Lexicon and Alesis reverberation units. 
// Designed to sound great with synthetic sound sources, rather than sound like a realistic space.
//
// #### Usage
//
// ```
// _,_ : jpverb(t60, damp, size, early_diff, mod_depth, mod_freq, low, mid, high, low_cutoff, high_cutoff) : _,_
// ```
//
// Where:
//
// * `t60`: approximate reverberation time in seconds ([0.1..60] sec) (T60 - the time for the reverb to decay by 60db when damp == 0 ). Does not effect early reflections
// * `damp`: controls damping of high-frequencies as the reverb decays. 0 is no damping, 1 is very strong damping. Values should be in the range ([0..1])
// * `size`: scales size of delay-lines within the reverberator, producing the impression of a larger or smaller space. Values below 1 can sound metallic. Values should be in the range [0.5..5]
// * `early_diff`: controls shape of early reflections. Values of 0.707 or more produce smooth exponential decay. Lower values produce a slower build-up of echoes. Values should be in the range ([0..1])
// * `mod_depth`: depth ([0..1]) of delay-line modulation. Use in combination with `mod_freq` to set amount of chorusing within the structure
// * `mod_freq`: frequency ([0..10] Hz) of delay-line modulation. Use in combination with `mod_depth` to set amount of chorusing within the structure
// * `low`: multiplier ([0..1]) for the reverberation time within the low band
// * `mid`: multiplier ([0..1]) for the reverberation time within the mid band
// * `high`: multiplier ([0..1]) for the reverberation time within the high band
// * `low_cutoff`: frequency (100..6000 Hz) at which the crossover between the low and mid bands of the reverb occurs
// * `high_cutoff`: frequency (1000..10000 Hz) at which the crossover between the mid and high bands of the reverb occurs
//
// #### Reference
//
// <https://doc.sccode.org/Overviews/DEIND.html>
//------------------------------------------------------------
declare jpverb author "Julian Parker, bug fixes and minor interface changes by Till Bovermann";
declare jpverb license "MIT license";

jpverb(t60, damp, size, early_diff, 
		mod_depth, mod_freq, 
		low, mid, high, 
		low_cutoff, high_cutoff)
    = jp_gh_rev.jpverb(t60, damp, size, early_diff, mod_depth, mod_freq, low, mid, high, low_cutoff, high_cutoff);


//-------------------------------`(re.)greyhole`-------------------------------
// A complex echo-like effect (stereo in/out), inspired by the classic Eventide effect of a similar name. 
// The effect consists of a diffuser (like a mini-reverb, structurally similar to the one used in `jpverb`)
// connected in a feedback system with a long, modulated delay-line. 
// Excels at producing spacey washes of sound.
//
// #### Usage
//
// ```
// _,_ : greyhole(dt, damp, size, early_diff, feedback, mod_depth, mod_freq) : _,_
// ```
//
// Where:
//
// * `dt`: approximate reverberation time in seconds ([0.1..60 sec])
// * `damp`: controls damping of high-frequencies as the reverb decays. 0 is no damping, 1 is very strong damping. Values should be between ([0..1])
// * `size`: control of relative "room size" roughly in the range ([0.5..3])
// * `early_diff`: controls pattern of echoes produced by the diffuser. At very low values, the diffuser acts like a delay-line whose length is controlled by the 'size' parameter. Medium values produce a slow build-up of echoes, giving the sound a reversed-like quality. Values of 0.707 or greater than produce smooth exponentially decaying echoes. Values should be in the range ([0..1])
// * `feedback`: amount of feedback through the system. Sets the number of repeating echoes. A setting of 1.0 produces infinite sustain. Values should be in the range ([0..1])
// * `mod_depth`: depth ([0..1]) of delay-line modulation. Use in combination with `mod_freq` to produce chorus and pitch-variations in the echoes
// * `mod_freq`: frequency ([0..10] Hz) of delay-line modulation. Use in combination with `mod_depth` to produce chorus and pitch-variations in the echoes
//
// #### Reference
//
// <https://doc.sccode.org/Overviews/DEIND.html>
//------------------------------------------------------------
declare greyhole author "Julian Parker, bug fixes and minor interface changes by Till Bovermann";
declare greyhole license "MIT license";

greyhole(dt, damp, size, early_diff, feedback, mod_depth, mod_freq) 
	= jp_gh_rev.greyhole(dt, damp, size, early_diff, feedback, mod_depth, mod_freq);

//===============================Keith Barr Allpass Loop Reverb=======================
//======================================================================================

//----------------------------`(re.)kb_rom_rev1`---------------------------
// Reverberator based on Keith Barr's all-pass single feedback loop reverb topology. Originally designed for the Spin Semiconductor FV-1 chip, this code is an adaptation of the rom_rev1.spn file, part of the Spin Semiconductor Free DSP Programs available on the Spin Semiconductor website.  
// It was submitted by Keith Barr himself and written in Spin Semiconductor Assembly, a dedicated assembly language for programming the FV-1 chip.  
//
// In this topology, when multiple delays and all-pass filters are placed in a loop, sound injected into the loop will recirculate, increasing the density of any impulse as the signal successively passes through the all-pass filters. 
// The result, after a short period of time, is a wash of sound, completely diffused into a natural reverb tail.  
//
// The reverb typically has a mono input (as from a single source) but benefits from a stereo output, providing the listener with a fuller, more immersive reverberant image.
//
// #### Usage
//
// ```
// _,_ : kb_rom_rev1(rt, damp) : _,_
// ```
//
// Where:
//
// * `rt`: coefficent of the decay of the reverb (0-1)
// * `damp`: coefficient of the lowpass filters (0-1)
//
// #### Reference
//
// * <https://www.spinsemi.com/programs.php#:~:text=Keith%20Barr-,rom_rev1.spn,-ROM%20reverb%202>
// * <https://www.spinsemi.com/knowledge_base/effects.html#Reverberation>
// * <https://www.spinsemi.com/knowledge_base/inst_syntax.html>
//------------------------------------------------------------
declare kb_rom_rev1 author "Luca Spanedda";
declare kb_rom_rev1 license "GPL-3.0";

kb_rom_rev1(rt, damp) = aploop
with {
    // Allpass
    // (t,g) = give: delay in samples, feedback gain 0-1
    apf(t, g) = _ : (+ : _ <: @ (t  - 1), * (- g)) ~ * (g) : mem, _ : + : _; 

    // modulated Allpass filter
    apfMod(mod, t, tMod, g) = _ : (+ : _ <: delaymod(mod, t - 1, tMod), * (- g)) ~ * (g) : mem, _ : + : _;

    // from Original Sample Rate (origSR), (samples) to current Sample Rate
    adaptSR(origSR, samples) = (samples * ma.SR / origSR) : max(ma.EPSILON, int); 

    // delay modulated : mod = mod source +/- 1, t = del in samps, tMod = mod in samps
    delaymod(mod, t, tMod) = de.fdelay(tMax, modIndx)
    with{
        tMax = t + tMod;
        modIndx = t + mod * tMod;
    };

    // Onepole, g = give amplitude 0 to +/- 1 (open - close) to the delayed signal 
    op(b1) = _ * (1 - abs(b1)) : + ~ * (b1);

    // complete allpass loop
    aploop(l, r) = (_ : loopSec(0) <: ((loopSec(1) <: ((_ : loopSec(2) <: loopSec(3), _), _)), _)) ~ _ : ro.cross(4) <: outTaps
    with {
        // input allpass sections
        apSec(0) = apf(adaptSR(32768, 156), 0.5) : apf(adaptSR(32768, 223), 0.5) : apf(adaptSR(32768, 332), 0.5) : apf(adaptSR(32768, 548), 0.5);
        apSec(1) = apf(adaptSR(32768, 186), 0.5) : apf(adaptSR(32768, 253), 0.5) : apf(adaptSR(32768, 302), 0.5) : apf(adaptSR(32768, 498), 0.5);

        // allpass loop sections
        loopSec(0) = _ @ (adaptSR(32768, 4568) - 1) : _ * rt : _ + (l : apSec(0)) : apfMod(os.osc(0.5), adaptSR(32768, 1251), adaptSR(32768, 20), 0.6) : 
	    apf(adaptSR(32768, 1751), 0.6) : op(damp) : op(- 0.05);
        loopSec(1) = _ @ adaptSR(32768, 5859) : _ * rt : apf(adaptSR(32768, 1443), 0.6) : apf(adaptSR(32768, 1343), 0.6) : op(damp) : op(- 0.05);
        loopSec(2) = _ @ adaptSR(32768, 4145) : _ * rt : _ + (r : apSec(1)) : apfMod(os.osc(0.5), adaptSR(32768, 1582), adaptSR(32768, 20), 0.6) : 
	    apf(adaptSR(32768, 1981), 0.6) : op(damp) : op(- 0.05);
        loopSec(3) = _ @ adaptSR(32768, 3476) : _ * rt : apf(adaptSR(32768, 1274), 0.6) : apf(adaptSR(32768, 1382), 0.6) : op(damp) : op(- 0.05);

        // output delay taps
        outTaps = ((_ * 1.5 @ adaptSR(32768, 2630), _ * 1.2 @ adaptSR(32768, 1943), _ * 1.0 @ adaptSR(32768, 3200), 
	    _ * 0.8 @ adaptSR(32768, 4016)) :> +),
    	((_ * 1.0 @ adaptSR(32768, 2420), _ * 0.8 @ adaptSR(32768, 2631), _ * 1.5 @ adaptSR(32768, 1163), 
	    _ * 1.2 @ adaptSR(32768, 3330)) :> +);
    };
};

// end further further contributions section