Filter::Filter (Parameters *param, FilterParams * pars)
{
this->param = param;
unsigned char Ftype = pars->Ptype;
unsigned char Fstages = pars->Pstages;
category = pars->Pcategory;
switch (category) {
case 1:
filter = new FormantFilter (param,pars);
break;
case 2:
filter = new SVFilter(param, Ftype, 1000.0f, pars->getq (), Fstages);
filter->outgain = dB2rap (pars->getgain ());
if (filter->outgain > 1.0f)
filter->outgain = sqrtf (filter->outgain);
break;
default:
filter = new AnalogFilter (param, Ftype, 1000.0f, pars->getq (), Fstages);
if ((Ftype >= 6) && (Ftype <= 8))
filter->setgain (pars->getgain ());
else
filter->outgain = dB2rap (pars->getgain ());
break;
};
};
Filter::Filter (FilterParams * pars)
{
unsigned char Ftype = pars->Ptype;
unsigned char Fstages = pars->Pstages;
category = pars->Pcategory;
interpbuf = new float[pars->intermediate_bufsize];
switch (category) {
case 1:
filter = new FormantFilter (pars, interpbuf);
break;
case 2:
filter = new SVFilter(Ftype, 1000.0f, pars->getq (), Fstages, pars->fSAMPLE_RATE, interpbuf);
filter->outgain = dB2rap (pars->getgain ());
if (filter->outgain > 1.0f)
filter->outgain = sqrtf (filter->outgain);
break;
default:
filter = new AnalogFilter (Ftype, 1000.0f, pars->getq (), Fstages, pars->fSAMPLE_RATE, interpbuf);
if ((Ftype >= 6) && (Ftype <= 8))
filter->setgain (pars->getgain ());
else
filter->outgain = dB2rap (pars->getgain ());
break;
};
};
Filter::Filter(FilterParams *pars, SynthEngine *_synth):
synth(_synth)
{
unsigned char Ftype = pars->Ptype;
unsigned char Fstages = pars->Pstages;
category = pars->Pcategory;
switch (category)
{
case 1:
filter = new FormantFilter(pars, synth);
break;
case 2:
filter = new SVFilter(Ftype, 1000.0f, pars->getq(), Fstages, synth);
filter->outgain = dB2rap(pars->getgain());
if (filter->outgain > 1.0f)
filter->outgain = sqrtf(filter->outgain);
break;
default:
filter = new AnalogFilter(Ftype, 1000.0f, pars->getq(), Fstages, synth);
if (Ftype >= 6 && Ftype <= 8)
filter->setgain(pars->getgain());
else
filter->outgain = dB2rap(pars->getgain());
break;
}
}
/*
* Envelope Output (dB)
*/
float Envelope::envout_dB()
{
float out;
if(linearenvelope != 0)
return envout();
if((currentpoint == 1) && (!keyreleased || (forcedrelase == 0))) { //first point is always lineary interpolated
float v1 = dB2rap(envval[0]);
float v2 = dB2rap(envval[1]);
out = v1 + (v2 - v1) * t;
t += inct;
if(t >= 1.0f) {
t = 0.0f;
inct = envdt[2];
currentpoint++;
out = v2;
}
if(out > 0.001f)
envoutval = rap2dB(out);
else
envoutval = MIN_ENVELOPE_DB;
}
else
out = dB2rap(envout());
return out;
}
Filter *Filter::generate(Allocator &memory, FilterParams *pars,
unsigned int srate, int bufsize)
{
assert(srate != 0);
assert(bufsize != 0);
unsigned char Ftype = pars->Ptype;
unsigned char Fstages = pars->Pstages;
Filter *filter;
switch(pars->Pcategory) {
case 1:
filter = memory.alloc<FormantFilter>(pars, &memory, srate, bufsize);
break;
case 2:
filter = memory.alloc<SVFilter>(Ftype, 1000.0f, pars->getq(), Fstages, srate, bufsize);
filter->outgain = dB2rap(pars->getgain());
if(filter->outgain > 1.0f)
filter->outgain = sqrt(filter->outgain);
break;
default:
filter = memory.alloc<AnalogFilter>(Ftype, 1000.0f, pars->getq(), Fstages, srate, bufsize);
if((Ftype >= 6) && (Ftype <= 8))
filter->setgain(pars->getgain());
else
filter->outgain = dB2rap(pars->getgain());
break;
}
return filter;
}
void
StompBox::changepar (int npar, int value)
{
switch (npar) {
case 0:
setvolume (value);
break;
case 1:
Phigh = value;
if( value < 0) highb = ((float) value)/64.0f;
if( value > 0) highb = ((float) value)/32.0f;
break;
case 2:
Pmid = value;
if( value < 0) midb = ((float) value)/64.0f;
if( value > 0) midb = ((float) value)/32.0f;
break;
case 3:
Plow = value;
if( value < 0) lowb = ((float) value)/64.0f;
if( value > 0) lowb = ((float) value)/32.0f;
break;
case 4:
Pgain = value;
gain = dB2rap(50.0f * ((float)value)/127.0f - 50.0f);
break;
case 5:
Pmode = value;
init_mode (Pmode);
break;
};
init_tone ();
};
void
RBFilter::setgain (float dBgain)
{
gain = dB2rap (dBgain);
if(!qmode) computefiltercoefs ();
else computefiltercoefs_hiQ ();
};
void SVFilter::init(float sample_rate, int type, REALTYPE Ffreq, REALTYPE Fq, unsigned char Fstages, float gain)
{
m_sample_rate = sample_rate;
stages = Fstages;
m_type = type;
freq = Ffreq;
q = Fq;
m_gain = 1.0;
m_outgain = 1.0;
needsinterpolation = 0;
firsttime = 1;
if (stages >= MAX_FILTER_STAGES)
{
stages = MAX_FILTER_STAGES;
}
cleanup();
setfreq_and_q(Ffreq,Fq);
m_outgain = dB2rap(gain);
if (m_outgain > 1.0)
{
m_outgain = sqrt(m_outgain);
}
}
void
CompBand::setlevel (int value)
{
Plevel = value;
level = dB2rap (60.0f * (float)value / 127.0f - 36.0f);
};
void
Sustainer::changepar (int npar, int value)
{
switch (npar) {
case 0:
Pvolume = value;
level = dB2rap(-30.0f * (1.0f - ((float) Pvolume/127.0f)));
break;
case 1:
Psustain = value;
fsustain = (float) Psustain/127.0f;
cratio = 1.25f - fsustain;
input = dB2rap (42.0f * fsustain - 6.0f);
cthresh = 0.25 + fsustain;
break;
};
};
void
Vocoder::changepar (int npar, int value)
{
float tmp = 0;
switch (npar)
{
case 0:
setvolume (value);
break;
case 1:
setpanning (value);
break;
case 2:
Pmuffle = value;
tmp = (float) Pmuffle;
tmp *= 0.0001f + tmp/64000;
alpha = ncSAMPLE_RATE/(ncSAMPLE_RATE + tmp);
beta = 1.0f - alpha;
break;
case 3:
Pqq = value;
tmp = (float) value;
adjustq(tmp);
break;
case 4:
Pinput = value;
input = dB2rap (75.0f * (float)Pinput / 127.0f - 40.0f);
break;
case 5:
Plevel = value;
level = dB2rap (60.0f * (float)Plevel / 127.0f - 40.0f);
break;
case 6:
Pring = value;
ringworm = (float) Pring/127.0f;
break;
};
};
void
Expander::Expander_Change (int np, int value)
{
switch (np) {
case 1:
Pthreshold = value;
tfactor = dB2rap (-((float) Pthreshold));
tlevel = 1.0f/tfactor;
break;
case 2:
Pshape = value;
sfactor = dB2rap ((float)Pshape/2);
sgain = expf(-sfactor);
break;
case 3:
Pattack = value;
a_rate = 1000.0f/((float)Pattack * fs);
break;
case 4:
Pdecay = value;
d_rate = 1000.0f/((float)Pdecay * fs);
break;
case 5:
setlpf(value);
break;
case 6:
sethpf(value);
break;
case 7:
Plevel = value;
level = dB2rap((float) value/6.0f);
break;
}
}
/*
* Effect output
*/
void Distorsion::out(const Stereo<float *> &smp)
{
int i;
REALTYPE l, r, lout, rout;
REALTYPE inputvol = pow(5.0, (Pdrive - 32.0) / 127.0);
if(Pnegate != 0)
inputvol *= -1.0;
if(Pstereo != 0) { //Stereo
for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
efxoutl[i] = smp.l[i] * inputvol * panning;
efxoutr[i] = smp.r[i] * inputvol * (1.0 - panning);
}
}
else {
for(i = 0; i < SOUND_BUFFER_SIZE; i++)
efxoutl[i] =
(smp.l[i] * panning + smp.r[i] * (1.0 - panning)) * inputvol;
;
}
if(Pprefiltering != 0)
applyfilters(efxoutl, efxoutr);
//no optimised, yet (no look table)
waveshapesmps(SOUND_BUFFER_SIZE, efxoutl, Ptype + 1, Pdrive);
if(Pstereo != 0)
waveshapesmps(SOUND_BUFFER_SIZE, efxoutr, Ptype + 1, Pdrive);
if(Pprefiltering == 0)
applyfilters(efxoutl, efxoutr);
if(Pstereo == 0)
for(i = 0; i < SOUND_BUFFER_SIZE; i++)
efxoutr[i] = efxoutl[i];
REALTYPE level = dB2rap(60.0 * Plevel / 127.0 - 40.0);
for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
lout = efxoutl[i];
rout = efxoutr[i];
l = lout * (1.0 - lrcross) + rout * lrcross;
r = rout * (1.0 - lrcross) + lout * lrcross;
lout = l;
rout = r;
efxoutl[i] = lout * 2.0 * level;
efxoutr[i] = rout * 2.0 * level;
}
}
FormantFilter::FormantFilter (FilterParams * pars)
{
numformants = pars->Pnumformants;
for (int i = 0; i < numformants; i++)
formant[i] = new AnalogFilter (4 /*BPF*/, 1000.0f, 10.0f, pars->Pstages);
cleanup ();
inbuffer = new float[PERIOD];
tmpbuf = new float[PERIOD];
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] = 1.0;
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 - ((float)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, ((float)pars->Pvowelclearness - 32.0f) / 48.0f);
sequencestretch = powf (0.1f, ((float)pars->Psequencestretch - 32.0f) / 48.0f);
if (pars->Psequencereversed)
sequencestretch *= -1.0f;
outgain = dB2rap (pars->getgain ());
oldinput = -1.0f;
Qfactor = 1.0f;
oldQfactor = Qfactor;
firsttime = 1;
};
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(FilterParams *pars, Allocator *alloc, unsigned int srate, int bufsize)
: Filter(srate, bufsize), memory(*alloc)
{
numformants = pars->Pnumformants;
for(int i = 0; i < numformants; ++i)
formant[i] = memory.alloc<AnalogFilter>(4 /*BPF*/, 1000.0f, 10.0f, pars->Pstages, srate, bufsize);
cleanup();
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;
}
// Effect output
void Distorsion::out(float *smpsl, float *smpsr)
{
float inputdrive = powf(5.0f, (Pdrive - 32.0f) / 127.0f);
if (Pnegate)
inputdrive *= -1.0f;
if (Pstereo) // Stereo
{
for (int i = 0; i < synth->p_buffersize; ++i)
{
efxoutl[i] = smpsl[i] * inputdrive * pangainL;
efxoutr[i] = smpsr[i] * inputdrive* pangainR;
}
}
else // Mono
for (int i = 0; i < synth->p_buffersize; ++i)
efxoutl[i] = inputdrive * (smpsl[i]* pangainL + smpsr[i]* pangainR) * 0.7f;
if (Pprefiltering)
applyfilters(efxoutl, efxoutr);
waveShapeSmps(synth->p_buffersize, efxoutl, Ptype + 1, Pdrive);
if (Pstereo)
waveShapeSmps(synth->p_buffersize, efxoutr, Ptype + 1, Pdrive);
if (!Pprefiltering)
applyfilters(efxoutl, efxoutr);
if (!Pstereo)
memcpy(efxoutr, efxoutl, synth->p_bufferbytes);
float level = dB2rap(60.0f * Plevel / 127.0f - 40.0f);
for (int i = 0; i < synth->p_buffersize; ++i)
{
float lout = efxoutl[i];
float rout = efxoutr[i];
float l = lout * (1.0f - lrcross) + rout * lrcross;
float r = rout * (1.0f - lrcross) + lout * lrcross;
lout = l;
rout = r;
efxoutl[i] = lout * 2.0f * level;
efxoutr[i] = rout * 2.0f * level;
}
}
//Effect output
void Distorsion::out(const Stereo<float *> &smp)
{
float inputvol = powf(5.0f, (Pdrive - 32.0f) / 127.0f);
if(Pnegate)
inputvol *= -1.0f;
if(Pstereo) //Stereo
for(int i = 0; i < buffersize; ++i) {
efxoutl[i] = smp.l[i] * inputvol * pangainL;
efxoutr[i] = smp.r[i] * inputvol * pangainR;
}
else //Mono
for(int i = 0; i < buffersize; ++i)
efxoutl[i] = (smp.l[i] * pangainL + smp.r[i] * pangainR) * inputvol;
if(Pprefiltering)
applyfilters(efxoutl, efxoutr);
waveShapeSmps(buffersize, efxoutl, Ptype + 1, Pdrive);
if(Pstereo)
waveShapeSmps(buffersize, efxoutr, Ptype + 1, Pdrive);
if(!Pprefiltering)
applyfilters(efxoutl, efxoutr);
if(!Pstereo)
memcpy(efxoutr, efxoutl, bufferbytes);
float level = dB2rap(60.0f * Plevel / 127.0f - 40.0f);
for(int i = 0; i < buffersize; ++i) {
float lout = efxoutl[i];
float rout = efxoutr[i];
float l = lout * (1.0f - lrcross) + rout * lrcross;
float r = rout * (1.0f - lrcross) + lout * lrcross;
lout = l;
rout = r;
efxoutl[i] = lout * 2.0f * level;
efxoutr[i] = rout * 2.0f * level;
}
}
void
setgain (goomf_synth_t * s, AnalogFilter * filter, float dBgain)
{
filter->gain = dB2rap (dBgain);
computefiltercoefs (s, filter);
};
void FormantFilter::init(float sample_rate, FilterParams *pars)
{
int i, j;
m_numformants = pars->Pnumformants;
for (i = 0 ; i < m_numformants ; i++)
{
m_formants[i].init(sample_rate, ZYN_FILTER_ANALOG_TYPE_BPF2, 1000.0, 10.0, pars->m_additional_stages, 0.0);
}
cleanup();
for (j = 0 ; j < FF_MAX_VOWELS ; j++)
{
for (i = 0 ; i < m_numformants ; i++)
{
m_formantpar[j][i].frequency = pars->getformantfreq(pars->Pvowels[j].formants[i].freq);
m_formantpar[j][i].amplitude = pars->getformantamp(pars->Pvowels[j].formants[i].amp);
m_formantpar[j][i].q_factor = pars->getformantq(pars->Pvowels[j].formants[i].q);
}
}
for (i = 0 ; i < FF_MAX_FORMANTS ; i++)
{
m_oldformantamp[i] = 1.0;
}
for (i = 0 ; i < m_numformants ; i++)
{
m_currentformants[i].frequency = 1000.0;
m_currentformants[i].amplitude = 1.0;
m_currentformants[i].q_factor = 2.0;
}
m_formantslowness = pow(1.0 - pars->Pformantslowness / 128.0, 3.0);
m_sequencesize = pars->Psequencesize;
if (m_sequencesize == 0)
{
m_sequencesize = 1;
}
for (i = 0 ; i < m_sequencesize ; i++)
{
m_sequence[i].nvowel = pars->Psequence[i].nvowel;
}
m_vowelclearness = pow(10.0, (pars->Pvowelclearness - 32.0) / 48.0);
m_sequencestretch = pow(0.1, (pars->Psequencestretch - 32.0) / 48.0);
if (pars->Psequencereversed)
{
m_sequencestretch *= -1.0;
}
m_outgain = dB2rap(pars->m_gain);
m_oldinput = -1.0;
m_Qfactor = 1.0;
m_oldQfactor = m_Qfactor;
m_firsttime = 1;
}
void
Compressor::Compressor_Change (int np, int value)
{
switch (np) {
case 1:
tthreshold = value;
thres_db = (float)tthreshold; //implicit type cast int to float
break;
case 2:
tratio = value;
ratio = (float)tratio;
break;
case 3:
toutput = value;
break;
case 4:
tatt = value;
att = cSAMPLE_RATE /(((float)value / 1000.0f) + cSAMPLE_RATE);
attr = att;
attl = att;
break;
case 5:
trel = value;
rel = cSAMPLE_RATE /(((float)value / 1000.0f) + cSAMPLE_RATE);
rell = rel;
relr = rel;
break;
case 6:
a_out = value;
break;
case 7:
tknee = value; //knee expressed a percentage of range between thresh and zero dB
kpct = (float)tknee/100.1f;
break;
case 8:
stereo = value;
break;
case 9:
peak = value;
break;
}
kratio = logf(ratio)/LOG_2; // Log base 2 relationship matches slope
knee = -kpct*thres_db;
coeff_kratio = 1.0 / kratio;
coeff_ratio = 1.0 / ratio;
coeff_knee = 1.0 / knee;
coeff_kk = knee * coeff_kratio;
thres_mx = thres_db + knee; //This is the value of the input when the output is at t+k
makeup = -thres_db - knee/kratio + thres_mx/ratio;
makeuplin = dB2rap(makeup);
if (a_out)
outlevel = dB2rap((float)toutput) * makeuplin;
else
outlevel = dB2rap((float)toutput);
}
void AnalogFilter::setgain(REALTYPE dBgain){
gain=dB2rap(dBgain);
computefiltercoefs();
};
void
Compressor::out (float *efxoutl, float *efxoutr)
{
int i;
for (i = 0; i < PERIOD; i++) {
float rdelta = 0.0f;
float ldelta = 0.0f;
//Right Channel
if(peak) {
if (rtimer > hold) {
rpeak *= 0.9998f; //The magic number corresponds to ~0.1s based on T/(RC + T),
rtimer--;
}
if (ltimer > hold) {
lpeak *= 0.9998f; //leaky peak detector.
ltimer --; //keeps the timer from eventually exceeding max int & rolling over
}
ltimer++;
rtimer++;
if(rpeak<fabs(efxoutr[i])) {
rpeak = fabs(efxoutr[i]);
rtimer = 0;
}
if(lpeak<fabs(efxoutl[i])) {
lpeak = fabs(efxoutl[i]);
ltimer = 0;
}
if(lpeak>20.0f) lpeak = 20.0f;
if(rpeak>20.0f) rpeak = 20.0f; //keeps limiter from getting locked up when signal levels go way out of bounds (like hundreds)
} else {
rpeak = efxoutr[i];
lpeak = efxoutl[i];
}
if(stereo) {
rdelta = fabsf (rpeak);
if(rvolume < 0.9f) {
attr = att;
relr = rel;
} else if (rvolume < 1.0f) {
attr = att + ((1.0f - att)*(rvolume - 0.9f)*10.0f); //dynamically change attack time for limiting mode
relr = rel/(1.0f + (rvolume - 0.9f)*9.0f); //release time gets longer when signal is above limiting
} else {
attr = 1.0f;
relr = rel*0.1f;
}
if (rdelta > rvolume)
rvolume = attr * rdelta + (1.0f - attr)*rvolume;
else
rvolume = relr * rdelta + (1.0f - relr)*rvolume;
rvolume_db = rap2dB (rvolume);
if (rvolume_db < thres_db) {
rgain = outlevel;
} else if (rvolume_db < thres_mx) {
//Dynamic ratio that depends on volume. As can be seen, ratio starts
//at something negligibly larger than 1 once volume exceeds thres, and increases toward selected
// ratio by the time it has reached thres_mx. --Transmogrifox
eratio = 1.0f + (kratio-1.0f)*(rvolume_db-thres_db)* coeff_knee;
rgain = outlevel*dB2rap(thres_db + (rvolume_db-thres_db)/eratio - rvolume_db);
} else {
rgain = outlevel*dB2rap(thres_db + coeff_kk + (rvolume_db-thres_mx)*coeff_ratio - rvolume_db);
limit = 1;
}
if ( rgain < MIN_GAIN) rgain = MIN_GAIN;
rgain_t = .4f * rgain + .6f * rgain_old;
};
//Left Channel
if(stereo) {
ldelta = fabsf (lpeak);
} else {
ldelta = 0.5f*(fabsf (lpeak) + fabsf (rpeak));
}; //It's not as efficient to check twice, but it's small expense worth code clarity
if(lvolume < 0.9f) {
attl = att;
rell = rel;
} else if (lvolume < 1.0f) {
attl = att + ((1.0f - att)*(lvolume - 0.9f)*10.0f); //dynamically change attack time for limiting mode
rell = rel/(1.0f + (lvolume - 0.9f)*9.0f); //release time gets longer when signal is above limiting
} else {
attl = 1.0f;
rell = rel*0.1f;
}
if (ldelta > lvolume)
lvolume = attl * ldelta + (1.0f - attl)*lvolume;
else
lvolume = rell*ldelta + (1.0f - rell)*lvolume;
lvolume_db = rap2dB (lvolume);
if (lvolume_db < thres_db) {
lgain = outlevel;
} else if (lvolume_db < thres_mx) { //knee region
eratio = 1.0f + (kratio-1.0f)*(lvolume_db-thres_db)* coeff_knee;
lgain = outlevel*dB2rap(thres_db + (lvolume_db-thres_db)/eratio - lvolume_db);
} else {
lgain = outlevel*dB2rap(thres_db + coeff_kk + (lvolume_db-thres_mx)*coeff_ratio - lvolume_db);
limit = 1;
}
if ( lgain < MIN_GAIN) lgain = MIN_GAIN;
lgain_t = .4f * lgain + .6f * lgain_old;
if (stereo) {
efxoutl[i] *= lgain_t;
efxoutr[i] *= rgain_t;
rgain_old = rgain;
lgain_old = lgain;
} else {
efxoutl[i] *= lgain_t;
efxoutr[i] *= lgain_t;
lgain_old = lgain;
}
if(peak) {
if(efxoutl[i]>0.999f) { //output hard limiting
efxoutl[i] = 0.999f;
clipping = 1;
}
if(efxoutl[i]<-0.999f) {
efxoutl[i] = -0.999f;
clipping = 1;
}
if(efxoutr[i]>0.999f) {
efxoutr[i] = 0.999f;
clipping = 1;
}
if(efxoutr[i]<-0.999f) {
efxoutr[i] = -0.999f;
clipping = 1;
}
//highly probably there is a more elegant way to do that, but what the hey...
}
}
}
void
setgain (ZEq10ban_t * s, AnalogFilter *filter, float dBgain)
{
filter->gain = dB2rap (dBgain);
computefiltercoefs (s,filter);
};
/*
* Effect output
*/
void
MBDist::out (float * smpsl, float * smpsr)
{
int i;
float l, r, lout, rout;
float inputvol = powf (5.0f, ((float)Pdrive - 32.0f) / 127.0f);
if (Pnegate != 0)
inputvol *= -1.0f;
if (Pstereo) {
for (i = 0; i < PERIOD; i++) {
efxoutl[i] = smpsl[i] * inputvol * 2.0f;
efxoutr[i] = smpsr[i] * inputvol * 2.0f;
};
} else {
for (i = 0; i < PERIOD; i++) {
efxoutl[i] =
(smpsl[i] + smpsr[i] ) * inputvol;
};
};
memcpy(lowl,efxoutl,sizeof(float) * PERIOD);
memcpy(midl,efxoutl,sizeof(float) * PERIOD);
memcpy(highl,efxoutl,sizeof(float) * PERIOD);
lpf1l->filterout(lowl);
hpf1l->filterout(midl);
lpf2l->filterout(midl);
hpf2l->filterout(highl);
if(volL> 0) mbwshape1l->waveshapesmps (PERIOD, lowl, PtypeL, PdriveL, 1);
if(volM> 0) mbwshape2l->waveshapesmps (PERIOD, midl, PtypeM, PdriveM, 1);
if(volH> 0) mbwshape3l->waveshapesmps (PERIOD, highl, PtypeH, PdriveH, 1);
if(Pstereo) {
memcpy(lowr,efxoutr,sizeof(float) * PERIOD);
memcpy(midr,efxoutr,sizeof(float) * PERIOD);
memcpy(highr,efxoutr,sizeof(float) * PERIOD);
lpf1r->filterout(lowr);
hpf1r->filterout(midr);
lpf2r->filterout(midr);
hpf2r->filterout(highr);
if(volL> 0) mbwshape1r->waveshapesmps (PERIOD, lowr, PtypeL, PdriveL, 1);
if(volM> 0) mbwshape2r->waveshapesmps (PERIOD, midr, PtypeM, PdriveM, 1);
if(volH> 0) mbwshape3r->waveshapesmps (PERIOD, highr, PtypeH, PdriveH, 1);
}
for (i = 0; i < PERIOD; i++) {
efxoutl[i]=lowl[i]*volL+midl[i]*volM+highl[i]*volH;
if (Pstereo) efxoutr[i]=lowr[i]*volL+midr[i]*volM+highr[i]*volH;
}
if (!Pstereo) memcpy(efxoutr, efxoutl, sizeof(float)* PERIOD);
float level = dB2rap (60.0f * (float)Plevel / 127.0f - 40.0f);
for (i = 0; i < PERIOD; i++) {
lout = efxoutl[i];
rout = efxoutr[i];
l = lout * (1.0f - lrcross) + rout * lrcross;
r = rout * (1.0f - lrcross) + lout * lrcross;
efxoutl[i] = l * 2.0f * level * panning;
efxoutr[i] = r * 2.0f * level * (1.0f -panning);
};
DCr->filterout (efxoutr);
DCl->filterout (efxoutl);
};
/*
* Effect output
*/
void
Valve::out (float * smpsl, float * smpsr, uint32_t period)
{
unsigned int i;
float l, r, lout, rout, fx;
if (Pstereo != 0) {
//Stereo
for (i = 0; i < period; i++) {
efxoutl[i] = smpsl[i] * inputvol;
efxoutr[i] = smpsr[i] * inputvol;
};
} else {
for (i = 0; i < period; i++) {
efxoutl[i] =
(smpsl[i] + smpsr[i] ) * inputvol;
};
};
harm->harm_out(efxoutl,efxoutr, period);
if (Pprefiltering != 0)
applyfilters (efxoutl, efxoutr, period);
if(Ped) {
for (i =0; i<period; i++) {
efxoutl[i]=Wshape(efxoutl[i]);
if (Pstereo != 0) efxoutr[i]=Wshape(efxoutr[i]);
}
}
for (i =0; i<period; i++) { //soft limiting to 3.0 (max)
fx = efxoutl[i];
if (fx>1.0f) fx = 3.0f - 2.0f/sqrtf(fx);
efxoutl[i] = fx;
fx = efxoutr[i];
if (fx>1.0f) fx = 3.0f - 2.0f/sqrtf(fx);
efxoutr[i] = fx;
}
if (q == 0.0f) {
for (i =0; i<period; i++) {
if (efxoutl[i] == q) fx = fdist;
else fx =efxoutl[i] / (1.0f - powf(2.0f,-dist * efxoutl[i] ));
otml = atk * otml + fx - itml;
itml = fx;
efxoutl[i]= otml;
}
} else {
for (i = 0; i < period; i++) {
if (efxoutl[i] == q) fx = fdist + qcoef;
else fx =(efxoutl[i] - q) / (1.0f - powf(2.0f,-dist * (efxoutl[i] - q))) + qcoef;
otml = atk * otml + fx - itml;
itml = fx;
efxoutl[i]= otml;
}
}
if (Pstereo != 0) {
if (q == 0.0f) {
for (i =0; i<period; i++) {
if (efxoutr[i] == q) fx = fdist;
else fx = efxoutr[i] / (1.0f - powf(2.0f,-dist * efxoutr[i] ));
otmr = atk * otmr + fx - itmr;
itmr = fx;
efxoutr[i]= otmr;
}
} else {
for (i = 0; i < period; i++) {
if (efxoutr[i] == q) fx = fdist + qcoef;
else fx = (efxoutr[i] - q) / (1.0f - powf(2.0f,-dist * (efxoutr[i] - q))) + qcoef;
otmr = atk * otmr + fx - itmr;
itmr = fx;
efxoutr[i]= otmr;
}
}
}
if (Pprefiltering == 0)
applyfilters (efxoutl, efxoutr, period);
if (Pstereo == 0) memcpy (efxoutr , efxoutl, period * sizeof(float));
float level = dB2rap (60.0f * (float)Plevel / 127.0f - 40.0f);
for (i = 0; i < period; i++) {
lout = efxoutl[i];
rout = efxoutr[i];
l = lout * (1.0f - lrcross) + rout * lrcross;
r = rout * (1.0f - lrcross) + lout * lrcross;
lout = l;
rout = r;
//efxoutl[i] = lout * 2.0f * level * panning;
//efxoutr[i] = rout * 2.0f * level * (1.0f -panning);
efxoutl[i] = lout * 2.0f * level * (1.0f -panning);
efxoutr[i] = rout * 2.0f * level * panning;
};
};
void SVFilter::setgain(float dBgain)
{
gain = dB2rap(dBgain);
computefiltercoefs();
}
/*
* Parameter control
*/
void Master::setPvolume(char Pvolume_){
Pvolume=Pvolume_;
volume=dB2rap((Pvolume-96.0)/96.0*40.0);
};
void SVFilter::setgain(REALTYPE dBgain)
{
m_gain = dB2rap(dBgain);
computefiltercoefs();
}
/*
* Effect output
*/
void
Distorsion::out (float * smpsl, float * smpsr)
{
int i;
float l, r, lout, rout;
float inputvol = powf (5.0f, ((float)Pdrive - 32.0f) / 127.0f);
if (Pnegate != 0)
inputvol *= -1.0f;
if (Pstereo != 0)
{ //Stereo
for (i = 0; i < PERIOD; i++)
{
efxoutl[i] = smpsl[i] * inputvol * 2.0f;
efxoutr[i] = smpsr[i] * inputvol * 2.0f;
};
}
else
{
for (i = 0; i < PERIOD; i++)
{
efxoutl[i] =
(smpsl[i] + smpsr[i] ) * inputvol;
};
};
if (Pprefiltering != 0)
applyfilters (efxoutl, efxoutr);
//no optimised, yet (no look table)
dwshapel->waveshapesmps (PERIOD, efxoutl, Ptype, Pdrive, 1);
if (Pstereo != 0)
dwshaper->waveshapesmps (PERIOD, efxoutr, Ptype, Pdrive, 1);
if (Pprefiltering == 0)
applyfilters (efxoutl, efxoutr);
if (Pstereo == 0) memcpy (efxoutr , efxoutl, PERIOD * sizeof(float));
if (octmix > 0.01f)
{
for (i = 0; i < PERIOD; i++)
{
lout = efxoutl[i];
rout = efxoutr[i];
if ( (octave_memoryl < 0.0f) && (lout > 0.0f) ) togglel *= -1.0f;
octave_memoryl = lout;
if ( (octave_memoryr < 0.0f) && (rout > 0.0f) ) toggler *= -1.0f;
octave_memoryr = rout;
octoutl[i] = lout * togglel;
octoutr[i] = rout * toggler;
}
blockDCr->filterout (octoutr);
blockDCl->filterout (octoutl);
}
float level = dB2rap (60.0f * (float)Plevel / 127.0f - 40.0f);
for (i = 0; i < PERIOD; i++)
{
lout = efxoutl[i];
rout = efxoutr[i];
l = lout * (1.0f - lrcross) + rout * lrcross;
r = rout * (1.0f - lrcross) + lout * lrcross;
if (octmix > 0.01f)
{
lout = l * (1.0f - octmix) + octoutl[i] * octmix;
rout = r * (1.0f - octmix) + octoutr[i] * octmix;
}
else
{
lout = l;
rout = r;
}
efxoutl[i] = lout * 2.0f * level * panning;
efxoutr[i] = rout * 2.0f * level * (1.0f -panning);
};
DCr->filterout (efxoutr);
DCl->filterout (efxoutl);
};