/*
    FormantFilter.cpp - formant filters

    Original ZynAddSubFX author Nasca Octavian Paul
    Copyright (C) 2002-2009 Nasca Octavian Paul
    Copyright 2009, James Morris
    Copyright 2009-2011, Alan Calvert

    This file is part of yoshimi, which is free software: you can redistribute
    it and/or modify it under the terms of the GNU Library General Public
    License as published by the Free Software Foundation; either version 2 of
    the License, or (at your option) any later version.

    yoshimi 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 (version 2 or
    later) for more details.

    You should have received a copy of the GNU General Public License along with
    yoshimi; if not, write to the Free Software Foundation, Inc., 51 Franklin
    Street, Fifth Floor, Boston, MA  02110-1301, USA.

    This file is derivative of ZynAddSubFX original code, modified March 2011
*/

#include <fftw3.h>

#include "Misc/SynthEngine.h"
#include "DSP/FormantFilter.h"

FormantFilter::FormantFilter(FilterParams *pars, SynthEngine *_synth):
    synth(_synth)
{
    numformants = pars->Pnumformants;
    for (int i = 0; i < numformants; ++i)
        formant[i] = new AnalogFilter(4/*BPF*/, 1000.0f, 10.0f, pars->Pstages, synth);
    cleanup();
    inbuffer = (float*)fftwf_malloc(synth->bufferbytes);
    tmpbuf = (float*)fftwf_malloc(synth->bufferbytes);

    for (int j = 0; j < FF_MAX_VOWELS; ++j)
        for (int i = 0; i < numformants; ++i)
        {
            formantpar[j][i].freq = pars->getformantfreq(pars->Pvowels[j].formants[i].freq);
            formantpar[j][i].amp = pars->getformantamp(pars->Pvowels[j].formants[i].amp);
            formantpar[j][i].q = pars->getformantq(pars->Pvowels[j].formants[i].q);
        }
    for (int i = 0; i < FF_MAX_FORMANTS; ++i)
        oldformantamp[i] = 1.0f;
    for (int i = 0; i < numformants; ++i)
    {
        currentformants[i].freq = 1000.0f;
        currentformants[i].amp = 1.0f;
        currentformants[i].q = 2.0f;
    }

    formantslowness = powf(1.0f - (pars->Pformantslowness / 128.0f), 3.0f);

    sequencesize = pars->Psequencesize;
    if (sequencesize == 0)
        sequencesize = 1;
    for (int k = 0; k < sequencesize; ++k)
        sequence[k].nvowel = pars->Psequence[k].nvowel;

    vowelclearness = powf(10.0f, (pars->Pvowelclearness - 32.0f) / 48.0f);

    sequencestretch = powf(0.1f, (pars->Psequencestretch - 32.0f) / 48.0f);
    if (pars->Psequencereversed)
        sequencestretch *= -1.0f;

    outgain = dB2rap(pars->getgain());

    oldinput = -1.0f;
    Qfactor = pars->getq();
    oldQfactor = Qfactor;
    firsttime = 1;
}


FormantFilter::~FormantFilter()
{
    for (int i = 0; i < numformants; ++i)
        delete(formant[i]);
    fftwf_free(inbuffer);
    fftwf_free(tmpbuf);
}


void FormantFilter::cleanup()
{
    for (int i = 0; i < numformants; ++i)
        formant[i]->cleanup();
}


void FormantFilter::setpos(float input)
{
    int p1, p2;

    if (firsttime)
        slowinput = input;
    else
        slowinput = slowinput * (1.0f - formantslowness) + input * formantslowness;

    if ((fabsf(oldinput-input) < 0.001f) && (fabsf(slowinput - input) < 0.001f) &&
            (fabsf(Qfactor - oldQfactor) < 0.001f))
    {
        //	oldinput=input; daca setez asta, o sa faca probleme la schimbari foarte lente
        firsttime = 0;
        return;
    } else
        oldinput = input;

    float pos = input * sequencestretch;
    pos -= floorf(pos);

    p2 = (int)(pos * sequencesize);
    p1 = p2 - 1;
    if (p1 < 0)
        p1 += sequencesize;

    pos = pos * sequencesize;
    pos -= floorf(pos);

    pos = (atanf((pos * 2.0f - 1.0f) * vowelclearness) / atanf(vowelclearness) + 1.0f) * 0.5f;

    p1 = sequence[p1].nvowel;
    p2 = sequence[p2].nvowel;

    if (firsttime)
    {
        for (int i = 0; i < numformants; ++i)
        {
            currentformants[i].freq =
                formantpar[p1][i].freq * (1.0f - pos) + formantpar[p2][i].freq * pos;
            currentformants[i].amp =
                formantpar[p1][i].amp * (1.0f - pos) + formantpar[p2][i].amp * pos;
            currentformants[i].q =
                formantpar[p1][i].q * (1.0f - pos) + formantpar[p2][i].q * pos;
            formant[i]->setfreq_and_q(currentformants[i].freq,
                                      currentformants[i].q * Qfactor);
            oldformantamp[i] = currentformants[i].amp;
        }
        firsttime = 0;
    } else {
        for (int i = 0; i < numformants; ++i)
        {
            currentformants[i].freq =
                currentformants[i].freq * (1.0f - formantslowness)
                + (formantpar[p1][i].freq
                    * (1.0f - pos) + formantpar[p2][i].freq * pos)
                * formantslowness;

            currentformants[i].amp =
                currentformants[i].amp * (1.0f - formantslowness)
                + (formantpar[p1][i].amp * (1.0f - pos)
                   + formantpar[p2][i].amp * pos) * formantslowness;

            currentformants[i].q =
                currentformants[i].q * (1.0f - formantslowness)
                    + (formantpar[p1][i].q * (1.0f - pos)
                        + formantpar[p2][i].q * pos) * formantslowness;

            formant[i]->setfreq_and_q(currentformants[i].freq,
                                      currentformants[i].q * Qfactor);
        }
    }
    oldQfactor = Qfactor;
}


void FormantFilter::setfreq(float frequency)
{
    setpos(frequency);
}


void FormantFilter::setq(float q_)
{
    Qfactor = q_;
    for (int i = 0; i <numformants; ++i)
        formant[i]->setq(Qfactor * currentformants[i].q);
}


void FormantFilter::setfreq_and_q(float frequency, float q_)
{
   Qfactor = q_;
    setpos(frequency);
}


void FormantFilter::filterout(float *smp)
{
    memcpy(inbuffer, smp, synth->sent_bufferbytes);
    memset(smp, 0, synth->sent_bufferbytes);

    for (int j = 0; j < numformants; ++j)
    {
        for (int k = 0; k < synth->sent_buffersize; ++k)
            tmpbuf[k] = inbuffer[k] * outgain;
        formant[j]->filterout(tmpbuf);

        if (aboveAmplitudeThreshold(oldformantamp[j], currentformants[j].amp))
            for (int i = 0; i < synth->sent_buffersize; ++i)
                smp[i] += tmpbuf[i]
                          * interpolateAmplitude(oldformantamp[j],
                                                  currentformants[j].amp, i,
                                                  synth->sent_buffersize);
        else
            for (int i = 0; i < synth->sent_buffersize; ++i)
                smp[i] += tmpbuf[i] * currentformants[j].amp;
        oldformantamp[j] = currentformants[j].amp;
    }
}
