|
- /*
- ZynAddSubFX - a software synthesizer
-
- PADnoteParameters.cpp - Parameters for PADnote (PADsynth)
- Copyright (C) 2002-2005 Nasca Octavian Paul
- Author: Nasca Octavian Paul
-
- This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
- as published by the Free Software Foundation.
-
- This program is distributed in the hope that it will be useful,
- but WITHOUT ANY WARRANTY; without even the implied warranty of
- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- GNU General Public License (version 2 or later) for more details.
-
- You should have received a copy of the GNU General Public License (version 2)
- along with this program; if not, write to the Free Software Foundation,
- Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
-
- */
- #include <cmath>
- #include "PADnoteParameters.h"
- #include "FilterParams.h"
- #include "EnvelopeParams.h"
- #include "LFOParams.h"
- #include "../Synth/Resonance.h"
- #include "../Synth/OscilGen.h"
- #include "../Misc/WavFile.h"
- #include <cstdio>
-
- #include <rtosc/ports.h>
- #include <rtosc/port-sugar.h>
- using namespace rtosc;
-
-
- #define rObject PADnoteParameters
- static const rtosc::Ports realtime_ports =
- {
- rRecurp(FreqLfo, "Frequency LFO"),
- rRecurp(AmpLfo, "Amplitude LFO"),
- rRecurp(FilterLfo, "Filter LFO"),
- rRecurp(FreqEnvelope, "Frequency Envelope"),
- rRecurp(AmpEnvelope, "Amplitude Envelope"),
- rRecurp(FilterEnvelope, "Filter Envelope"),
- rRecurp(GlobalFilter, "Post Filter"),
-
- //Volume
- rToggle(PStereo, "Stereo/Mono Mode"),
- rParamZyn(PPanning, "Left Right Panning"),
- rParamZyn(PVolume, "Synth Volume"),
- rParamZyn(PAmpVelocityScaleFunction, "Amplitude Velocity Sensing function"),
-
- rParamZyn(Fadein_adjustment, "Adjustment for anti-pop strategy."),
-
- //Punch
- rParamZyn(PPunchStrength, "Punch Strength"),
- rParamZyn(PPunchTime, "UNKNOWN"),
- rParamZyn(PPunchStretch, "How Punch changes with note frequency"),
- rParamZyn(PPunchVelocitySensing, "Punch Velocity control"),
-
- //Filter
- rParamZyn(PFilterVelocityScale, "Filter Velocity Magnitude"),
- rParamZyn(PFilterVelocityScaleFunction, "Filter Velocity Function Shape"),
-
- //Freq
- rToggle(Pfixedfreq, "Base frequency fixed frequency enable"),
- rParamZyn(PfixedfreqET, "Equal temeperate control for fixed frequency operation"),
-
- rParamI(PDetune, "Fine Detune"),
- rParamI(PCoarseDetune, "Coarse Detune"),
- rParamZyn(PDetuneType, "Magnitude of Detune"),
-
- {"sample#64:ifb", rProp(internal) rDoc("Nothing to see here"), 0,
- [](const char *m, rtosc::RtData &d)
- {
- PADnoteParameters *p = (PADnoteParameters*)d.obj;
- const char *mm = m;
- while(!isdigit(*mm))++mm;
- unsigned n = atoi(mm);
- p->sample[n].size = rtosc_argument(m,0).i;
- p->sample[n].basefreq = rtosc_argument(m,1).f;
- p->sample[n].smp = *(float**)rtosc_argument(m,2).b.data;
-
- //XXX TODO memory managment (deallocation of smp buffer)
- }},
- //weird stuff for PCoarseDetune
- {"detunevalue:", rMap(unit,cents) rDoc("Get detune value"), NULL,
- [](const char *, RtData &d)
- {
- PADnoteParameters *obj = (PADnoteParameters *)d.obj;
- d.reply(d.loc, "f", getdetune(obj->PDetuneType, 0, obj->PDetune));
- }},
- {"octave::c:i", rProp(parameter) rDoc("Octave note offset"), NULL,
- [](const char *msg, RtData &d)
- {
- PADnoteParameters *obj = (PADnoteParameters *)d.obj;
- if(!rtosc_narguments(msg)) {
- int k=obj->PCoarseDetune/1024;
- if (k>=8) k-=16;
- d.reply(d.loc, "i", k);
- } else {
- int k=(int) rtosc_argument(msg, 0).i;
- if (k<0) k+=16;
- obj->PCoarseDetune = k*1024 + obj->PCoarseDetune%1024;
- }
- }},
- {"coarsedetune::c:i", rProp(parameter) rDoc("Coarse note detune"), NULL,
- [](const char *msg, RtData &d)
- {
- PADnoteParameters *obj = (PADnoteParameters *)d.obj;
- if(!rtosc_narguments(msg)) {
- int k=obj->PCoarseDetune%1024;
- if (k>=512) k-=1024;
- d.reply(d.loc, "i", k);
- } else {
- int k=(int) rtosc_argument(msg, 0).i;
- if (k<0) k+=1024;
- obj->PCoarseDetune = k + (obj->PCoarseDetune/1024)*1024;
- }
- }},
-
- };
- static const rtosc::Ports non_realtime_ports =
- {
- //Harmonic Source Distribution
- rRecurp(oscilgen, "Oscillator"),
- rRecurp(resonance, "Resonance"),
-
- //Harmonic Shape
- rOption(Pmode, rMap(min, 0), rMap(max, 2), rOptions(bandwidth,discrete,continious),
- "Harmonic Distribution Model"),
- rOption(Php.base.type, rOptions(Gaussian, Rectanglar, Double Exponential),
- "Harmonic profile shape"),
- rParamZyn(Php.base.par1, "Harmonic shape distribution parameter"),
- rParamZyn(Php.freqmult, "Frequency multiplier on distribution"),
- rParamZyn(Php.modulator.par1, "Distribution modulator parameter"),
- rParamZyn(Php.modulator.freq, "Frequency of modulator parameter"),
- rParamZyn(Php.width, "Width of base harmonic"),
- rOption(Php.amp.mode, rOptions(Sum, Mult, Div1, Div2),
- "Amplitude harmonic multiplier type"),
-
- //Harmonic Modulation
- rOption(Php.amp.type, rOptions(Off, Gauss, Sine, Flat),
- "Type of amplitude multipler"),
- rParamZyn(Php.amp.par1, "Amplitude multiplier parameter"),
- rParamZyn(Php.amp.par2, "Amplitude multiplier parameter"),
- rToggle(Php.autoscale, "Autoscaling Harmonics"),
- rOption(Php.onehalf,
- rOptions(Full, Upper Half, Lower Half),
- "Harmonic cutoff model"),
-
- //Harmonic Bandwidth
- rOption(Pbwscale,
- rOptions(Normal,
- EqualHz, Quater,
- Half, 75%, 150%,
- Double, Inv. Half),
- "Bandwidth scaling"),
-
- //Harmonic Position Modulation
- rOption(Phrpos.type,
- rOptions(Harmonic, ShiftU, ShiftL, PowerU, PowerL, Sine,
- Power, Shift),
- "Harmonic Overtone shifting mode"),
- rParamZyn(Phrpos.par1, "Harmonic position parameter"),
- rParamZyn(Phrpos.par2, "Harmonic position parameter"),
- rParamZyn(Phrpos.par3, "Harmonic position parameter"),
-
- //Quality
- rOption(Pquality.samplesize,
- rOptions(16k (Tiny), 32k, 64k (Small), 128k,
- 256k (Normal), 512k, 1M (Big)),
- "Size of each wavetable element"),
- rOption(Pquality.basenote,
- rOptions( C-2, G-2, C-3, G-3, C-4,
- G-4, C-5, G-5, G-6,),
- "Base note for wavetable"),
- rOption(Pquality.smpoct,
- rOptions(0.5, 1, 2, 3, 4, 6, 12),
- "Samples per octave"),
- rParamI(Pquality.oct, rLinear(0,7),
- "Number of octaves to sample (above the first sample"),
-
- {"Pbandwidth::i", rProp(parameter) rLinear(0,1000) rDoc("Bandwith Of Harmonics"), NULL,
- [](const char *msg, rtosc::RtData &d) {
- PADnoteParameters *p = ((PADnoteParameters*)d.obj);
- if(rtosc_narguments(msg)) {
- p->setPbandwidth(rtosc_argument(msg, 0).i);
- } else {
- d.reply(d.loc, "i", p->Pbandwidth);
- }}},
-
- {"bandwidthvalue:", rMap(unit, cents) rDoc("Get Bandwidth"), NULL,
- [](const char *, rtosc::RtData &d) {
- PADnoteParameters *p = ((PADnoteParameters*)d.obj);
- d.reply(d.loc, "f", p->setPbandwidth(p->Pbandwidth));
- }},
-
-
- {"nhr:", rProp(non-realtime) rDoc("Returns the harmonic shifts"),
- NULL, [](const char *, rtosc::RtData &d) {
- PADnoteParameters *p = ((PADnoteParameters*)d.obj);
- const unsigned n = p->synth.oscilsize / 2;
- float *tmp = new float[n];
- *tmp = 0;
- for(unsigned i=1; i<n; ++i)
- tmp[i] = p->getNhr(i);
- d.reply(d.loc, "b", n*sizeof(float), tmp);
- delete[] tmp;}},
- {"profile:i", rProp(non-realtime) rDoc("UI display of the harmonic profile"),
- NULL, [](const char *m, rtosc::RtData &d) {
- PADnoteParameters *p = ((PADnoteParameters*)d.obj);
- const int n = rtosc_argument(m, 0).i;
- if(n<=0)
- return;
- float *tmp = new float[n];
- float realbw = p->getprofile(tmp, n);
- d.reply(d.loc, "b", n*sizeof(float), tmp);
- d.reply(d.loc, "i", realbw);
- delete[] tmp;}},
- };
-
- const rtosc::Ports &PADnoteParameters::non_realtime_ports = ::non_realtime_ports;
- const rtosc::Ports &PADnoteParameters::realtime_ports = ::realtime_ports;
-
-
- const rtosc::MergePorts PADnoteParameters::ports =
- {
- &non_realtime_ports,
- &realtime_ports
- };
-
-
- PADnoteParameters::PADnoteParameters(const SYNTH_T &synth_, FFTwrapper *fft_)
- :Presets(), synth(synth_)
- {
- setpresettype("Ppadsynth");
-
- fft = fft_;
-
- resonance = new Resonance();
- oscilgen = new OscilGen(synth, fft_, resonance);
- oscilgen->ADvsPAD = true;
-
- FreqEnvelope = new EnvelopeParams(0, 0);
- FreqEnvelope->ASRinit(64, 50, 64, 60);
- FreqLfo = new LFOParams(70, 0, 64, 0, 0, 0, 0, 0);
-
- AmpEnvelope = new EnvelopeParams(64, 1);
- AmpEnvelope->ADSRinit_dB(0, 40, 127, 25);
- AmpLfo = new LFOParams(80, 0, 64, 0, 0, 0, 0, 1);
-
- GlobalFilter = new FilterParams(2, 94, 40);
- FilterEnvelope = new EnvelopeParams(0, 1);
- FilterEnvelope->ADSRinit_filter(64, 40, 64, 70, 60, 64);
- FilterLfo = new LFOParams(80, 0, 64, 0, 0, 0, 0, 2);
-
- for(int i = 0; i < PAD_MAX_SAMPLES; ++i)
- sample[i].smp = NULL;
-
- defaults();
- }
-
- PADnoteParameters::~PADnoteParameters()
- {
- deletesamples();
- delete (oscilgen);
- delete (resonance);
-
- delete (FreqEnvelope);
- delete (FreqLfo);
- delete (AmpEnvelope);
- delete (AmpLfo);
- delete (GlobalFilter);
- delete (FilterEnvelope);
- delete (FilterLfo);
- }
-
- void PADnoteParameters::defaults()
- {
- Pmode = 0;
- Php.base.type = 0;
- Php.base.par1 = 80;
- Php.freqmult = 0;
- Php.modulator.par1 = 0;
- Php.modulator.freq = 30;
- Php.width = 127;
- Php.amp.type = 0;
- Php.amp.mode = 0;
- Php.amp.par1 = 80;
- Php.amp.par2 = 64;
- Php.autoscale = true;
- Php.onehalf = 0;
-
- setPbandwidth(500);
- Pbwscale = 0;
-
- resonance->defaults();
- oscilgen->defaults();
-
- Phrpos.type = 0;
- Phrpos.par1 = 64;
- Phrpos.par2 = 64;
- Phrpos.par3 = 0;
-
- Pquality.samplesize = 3;
- Pquality.basenote = 4;
- Pquality.oct = 3;
- Pquality.smpoct = 2;
-
- PStereo = 1; //stereo
- /* Frequency Global Parameters */
- Pfixedfreq = 0;
- PfixedfreqET = 0;
- PDetune = 8192; //zero
- PCoarseDetune = 0;
- PDetuneType = 1;
- FreqEnvelope->defaults();
- FreqLfo->defaults();
-
- /* Amplitude Global Parameters */
- PVolume = 90;
- PPanning = 64; //center
- PAmpVelocityScaleFunction = 64;
- AmpEnvelope->defaults();
- AmpLfo->defaults();
- Fadein_adjustment = FADEIN_ADJUSTMENT_SCALE;
- PPunchStrength = 0;
- PPunchTime = 60;
- PPunchStretch = 64;
- PPunchVelocitySensing = 72;
-
- /* Filter Global Parameters*/
- PFilterVelocityScale = 64;
- PFilterVelocityScaleFunction = 64;
- GlobalFilter->defaults();
- FilterEnvelope->defaults();
- FilterLfo->defaults();
-
- deletesamples();
- }
-
- void PADnoteParameters::deletesample(int n)
- {
- if((n < 0) || (n >= PAD_MAX_SAMPLES))
- return;
-
- delete[] sample[n].smp;
- sample[n].smp = NULL;
- sample[n].size = 0;
- sample[n].basefreq = 440.0f;
- }
-
- void PADnoteParameters::deletesamples()
- {
- for(int i = 0; i < PAD_MAX_SAMPLES; ++i)
- deletesample(i);
- }
-
- /*
- * Get the harmonic profile (i.e. the frequency distributio of a single harmonic)
- */
- float PADnoteParameters::getprofile(float *smp, int size)
- {
- for(int i = 0; i < size; ++i)
- smp[i] = 0.0f;
- const int supersample = 16;
- float basepar = powf(2.0f, (1.0f - Php.base.par1 / 127.0f) * 12.0f);
- float freqmult = floor(powf(2.0f,
- Php.freqmult / 127.0f
- * 5.0f) + 0.000001f);
-
- float modfreq = floor(powf(2.0f,
- Php.modulator.freq / 127.0f
- * 5.0f) + 0.000001f);
- float modpar1 = powf(Php.modulator.par1 / 127.0f, 4.0f) * 5.0f / sqrt(
- modfreq);
- float amppar1 =
- powf(2.0f, powf(Php.amp.par1 / 127.0f, 2.0f) * 10.0f) - 0.999f;
- float amppar2 = (1.0f - Php.amp.par2 / 127.0f) * 0.998f + 0.001f;
- float width = powf(150.0f / (Php.width + 22.0f), 2.0f);
-
- for(int i = 0; i < size * supersample; ++i) {
- bool makezero = false;
- float x = i * 1.0f / (size * (float) supersample);
-
- float origx = x;
-
- //do the sizing (width)
- x = (x - 0.5f) * width + 0.5f;
- if(x < 0.0f) {
- x = 0.0f;
- makezero = true;
- }
- else
- if(x > 1.0f) {
- x = 1.0f;
- makezero = true;
- }
-
- //compute the full profile or one half
- switch(Php.onehalf) {
- case 1:
- x = x * 0.5f + 0.5f;
- break;
- case 2:
- x = x * 0.5f;
- break;
- }
-
- float x_before_freq_mult = x;
-
- //do the frequency multiplier
- x *= freqmult;
-
- //do the modulation of the profile
- x += sinf(x_before_freq_mult * 3.1415926f * modfreq) * modpar1;
- x = fmod(x + 1000.0f, 1.0f) * 2.0f - 1.0f;
-
-
- //this is the base function of the profile
- float f;
- switch(Php.base.type) {
- case 1:
- f = expf(-(x * x) * basepar);
- if(f < 0.4f)
- f = 0.0f;
- else
- f = 1.0f;
- break;
- case 2:
- f = expf(-(fabs(x)) * sqrt(basepar));
- break;
- default:
- f = expf(-(x * x) * basepar);
- break;
- }
- if(makezero)
- f = 0.0f;
-
- float amp = 1.0f;
- origx = origx * 2.0f - 1.0f;
-
- //compute the amplitude multiplier
- switch(Php.amp.type) {
- case 1:
- amp = expf(-(origx * origx) * 10.0f * amppar1);
- break;
- case 2:
- amp = 0.5f
- * (1.0f
- + cosf(3.1415926f * origx * sqrt(amppar1 * 4.0f + 1.0f)));
- break;
- case 3:
- amp = 1.0f
- / (powf(origx * (amppar1 * 2.0f + 0.8f), 14.0f) + 1.0f);
- break;
- }
-
- //apply the amplitude multiplier
- float finalsmp = f;
- if(Php.amp.type != 0)
- switch(Php.amp.mode) {
- case 0:
- finalsmp = amp * (1.0f - amppar2) + finalsmp * amppar2;
- break;
- case 1:
- finalsmp *= amp * (1.0f - amppar2) + amppar2;
- break;
- case 2:
- finalsmp = finalsmp
- / (amp + powf(amppar2, 4.0f) * 20.0f + 0.0001f);
- break;
- case 3:
- finalsmp = amp
- / (finalsmp
- + powf(amppar2, 4.0f) * 20.0f + 0.0001f);
- break;
- }
- ;
-
- smp[i / supersample] += finalsmp / supersample;
- }
-
- //normalize the profile (make the max. to be equal to 1.0f)
- float max = 0.0f;
- for(int i = 0; i < size; ++i) {
- if(smp[i] < 0.0f)
- smp[i] = 0.0f;
- if(smp[i] > max)
- max = smp[i];
- }
- if(max < 0.00001f)
- max = 1.0f;
- for(int i = 0; i < size; ++i)
- smp[i] /= max;
-
- if(!Php.autoscale)
- return 0.5f;
-
- //compute the estimated perceived bandwidth
- float sum = 0.0f;
- int i;
- for(i = 0; i < size / 2 - 2; ++i) {
- sum += smp[i] * smp[i] + smp[size - i - 1] * smp[size - i - 1];
- if(sum >= 4.0f)
- break;
- }
-
- float result = 1.0f - 2.0f * i / (float) size;
- return result;
- }
-
- /*
- * Compute the real bandwidth in cents and returns it
- * Also, sets the bandwidth parameter
- */
- float PADnoteParameters::setPbandwidth(int Pbandwidth)
- {
- this->Pbandwidth = Pbandwidth;
- float result = powf(Pbandwidth / 1000.0f, 1.1f);
- result = powf(10.0f, result * 4.0f) * 0.25f;
- return result;
- }
-
- /*
- * Get the harmonic(overtone) position
- */
- float PADnoteParameters::getNhr(int n)
- {
- float result = 1.0f;
- const float par1 = powf(10.0f, -(1.0f - Phrpos.par1 / 255.0f) * 3.0f);
- const float par2 = Phrpos.par2 / 255.0f;
-
- const float n0 = n - 1.0f;
- float tmp = 0.0f;
- int thresh = 0;
- switch(Phrpos.type) {
- case 1:
- thresh = (int)(par2 * par2 * 100.0f) + 1;
- if(n < thresh)
- result = n;
- else
- result = 1.0f + n0 + (n0 - thresh + 1.0f) * par1 * 8.0f;
- break;
- case 2:
- thresh = (int)(par2 * par2 * 100.0f) + 1;
- if(n < thresh)
- result = n;
- else
- result = 1.0f + n0 - (n0 - thresh + 1.0f) * par1 * 0.90f;
- break;
- case 3:
- tmp = par1 * 100.0f + 1.0f;
- result = powf(n0 / tmp, 1.0f - par2 * 0.8f) * tmp + 1.0f;
- break;
- case 4:
- result = n0
- * (1.0f
- - par1)
- + powf(n0 * 0.1f, par2 * 3.0f
- + 1.0f) * par1 * 10.0f + 1.0f;
- break;
- case 5:
- result = n0
- + sinf(n0 * par2 * par2 * PI
- * 0.999f) * sqrt(par1) * 2.0f + 1.0f;
- break;
- case 6:
- tmp = powf(par2 * 2.0f, 2.0f) + 0.1f;
- result = n0 * powf(1.0f + par1 * powf(n0 * 0.8f, tmp), tmp) + 1.0f;
- break;
- case 7:
- result = (n + Phrpos.par1 / 255.0f) / (Phrpos.par1 / 255.0f + 1);
- break;
- default:
- result = n;
- break;
- }
-
- const float par3 = Phrpos.par3 / 255.0f;
-
- const float iresult = floor(result + 0.5f);
- const float dresult = result - iresult;
-
- return iresult + (1.0f - par3) * dresult;
- }
-
- //Transform non zero positive signals into ones with a max of one
- static void normalize_max(float *f, size_t len)
- {
- float max = 0.0f;
- for(unsigned i = 0; i < len; ++i)
- if(f[i] > i)
- max = f[i];
- if(max > 0.000001f)
- for(unsigned i = 0; i < len; ++i)
- f[i] /= max;
- }
-
- //Translate Bandwidth scale integer into floating point value
- static float Pbwscale_translate(char Pbwscale)
- {
- switch(Pbwscale) {
- case 0: return 1.0f;
- case 1: return 0.0f;
- case 2: return 0.25f;
- case 3: return 0.5f;
- case 4: return 0.75f;
- case 5: return 1.5f;
- case 6: return 2.0f;
- case 7: return -0.5f;
- default: return 1.0;
- }
- }
-
- /*
- * Generates the long spectrum for Bandwidth mode (only amplitudes are generated; phases will be random)
- */
-
- //Requires
- // - bandwidth scaling power
- // - bandwidth
- // - oscilator harmonics at various frequences (oodles of data)
- // - sampled resonance
- void PADnoteParameters::generatespectrum_bandwidthMode(float *spectrum,
- int size,
- float basefreq,
- float *profile,
- int profilesize,
- float bwadjust)
- {
- float harmonics[synth.oscilsize];
- memset(spectrum, 0, sizeof(float) * size);
- memset(harmonics, 0, sizeof(float) * synth.oscilsize);
-
- //get the harmonic structure from the oscillator (I am using the frequency amplitudes, only)
- oscilgen->get(harmonics, basefreq, false);
-
- //normalize
- normalize_max(harmonics, synth.oscilsize / 2);
-
- //Constants across harmonics
- const float power = Pbwscale_translate(Pbwscale);
- const float bandwidthcents = setPbandwidth(Pbandwidth);
-
- for(int nh = 1; nh < synth.oscilsize / 2; ++nh) { //for each harmonic
- const float realfreq = getNhr(nh) * basefreq;
- if(realfreq > synth.samplerate_f * 0.49999f)
- break;
- if(realfreq < 20.0f)
- break;
- if(harmonics[nh - 1] < 1e-4)
- continue;
-
- //compute the bandwidth of each harmonic
- const float bw =
- ((powf(2.0f, bandwidthcents / 1200.0f) - 1.0f) * basefreq / bwadjust)
- * powf(realfreq / basefreq, power);
- const int ibw = (int)((bw / (synth.samplerate_f * 0.5f) * size)) + 1;
-
- float amp = harmonics[nh - 1];
- if(resonance->Penabled)
- amp *= resonance->getfreqresponse(realfreq);
-
- if(ibw > profilesize) { //if the bandwidth is larger than the profilesize
- const float rap = sqrt((float)profilesize / (float)ibw);
- const int cfreq =
- (int) (realfreq
- / (synth.samplerate_f * 0.5f) * size) - ibw / 2;
- for(int i = 0; i < ibw; ++i) {
- const int src = i * rap * rap;
- const int spfreq = i + cfreq;
- if(spfreq < 0)
- continue;
- if(spfreq >= size)
- break;
- spectrum[spfreq] += amp * profile[src] * rap;
- }
- }
- else { //if the bandwidth is smaller than the profilesize
- const float rap = sqrt((float)ibw / (float)profilesize);
- const float ibasefreq = realfreq / (synth.samplerate_f * 0.5f) * size;
- for(int i = 0; i < profilesize; ++i) {
- const float idfreq = (i / (float)profilesize - 0.5f) * ibw;
- const int spfreq = (int) (idfreq + ibasefreq);
- const float fspfreq = fmodf((float)idfreq + ibasefreq, 1.0f);
- if(spfreq <= 0)
- continue;
- if(spfreq >= size - 1)
- break;
- spectrum[spfreq] += amp * profile[i] * rap
- * (1.0f - fspfreq);
- spectrum[spfreq + 1] += amp * profile[i] * rap * fspfreq;
- }
- }
- }
- }
-
- /*
- * Generates the long spectrum for non-Bandwidth modes (only amplitudes are generated; phases will be random)
- */
- void PADnoteParameters::generatespectrum_otherModes(float *spectrum,
- int size,
- float basefreq)
- {
- float harmonics[synth.oscilsize];
- memset(spectrum, 0, sizeof(float) * size);
- memset(harmonics, 0, sizeof(float) * synth.oscilsize);
-
- //get the harmonic structure from the oscillator (I am using the frequency amplitudes, only)
- oscilgen->get(harmonics, basefreq, false);
-
- //normalize
- normalize_max(harmonics, synth.oscilsize / 2);
-
- for(int nh = 1; nh < synth.oscilsize / 2; ++nh) { //for each harmonic
- const float realfreq = getNhr(nh) * basefreq;
-
- //take care of interpolation if frequency decreases
- if(realfreq > synth.samplerate_f * 0.49999f)
- break;
- if(realfreq < 20.0f)
- break;
-
-
- float amp = harmonics[nh - 1];
- if(resonance->Penabled)
- amp *= resonance->getfreqresponse(realfreq);
- const int cfreq = realfreq / (synth.samplerate_f * 0.5f) * size;
-
- spectrum[cfreq] = amp + 1e-9;
- }
-
- //In continous mode the spectrum gets additional interpolation between the
- //spectral peaks
- if(Pmode != 1) { //continous mode
- int old = 0;
- for(int k = 1; k < size; ++k)
- if((spectrum[k] > 1e-10) || (k == (size - 1))) {
- const int delta = k - old;
- const float val1 = spectrum[old];
- const float val2 = spectrum[k];
- const float idelta = 1.0f / delta;
- for(int i = 0; i < delta; ++i) {
- const float x = idelta * i;
- spectrum[old + i] = val1 * (1.0f - x) + val2 * x;
- }
- old = k;
- }
- }
- }
-
- /*
- * Applies the parameters (i.e. computes all the samples, based on parameters);
- */
- void PADnoteParameters::applyparameters()
- {
- applyparameters([]{return false;});
- }
-
- void PADnoteParameters::applyparameters(std::function<bool()> do_abort)
- {
- if(do_abort())
- return;
- unsigned max = 0;
- sampleGenerator([&max,this]
- (unsigned N, PADnoteParameters::Sample &smp) {
- delete[] sample[N].smp;
- sample[N] = smp;
- max = max < N ? N : max;
- },
- do_abort);
-
- //Delete remaining unused samples
- for(unsigned i = max; i < PAD_MAX_SAMPLES; ++i)
- deletesample(i);
- }
-
- //Requires
- // - Pquality.samplesize
- // - Pquality.basenote
- // - Pquality.oct
- // - Pquality.smpoct
- // - spectrum at various frequencies (oodles of data)
- void PADnoteParameters::sampleGenerator(PADnoteParameters::callback cb,
- std::function<bool()> do_abort)
- {
- const int samplesize = (((int) 1) << (Pquality.samplesize + 14));
- const int spectrumsize = samplesize / 2;
- float *spectrum = new float[spectrumsize];
- const int profilesize = 512;
- float profile[profilesize];
-
-
- const float bwadjust = getprofile(profile, profilesize);
- float basefreq = 65.406f * powf(2.0f, Pquality.basenote / 2);
- if(Pquality.basenote % 2 == 1)
- basefreq *= 1.5f;
-
- int samplemax = Pquality.oct + 1;
- int smpoct = Pquality.smpoct;
- if(Pquality.smpoct == 5)
- smpoct = 6;
- if(Pquality.smpoct == 6)
- smpoct = 12;
- if(smpoct != 0)
- samplemax *= smpoct;
- else
- samplemax = samplemax / 2 + 1;
- if(samplemax == 0)
- samplemax = 1;
-
- //prepare a BIG FFT
- FFTwrapper *fft = new FFTwrapper(samplesize);
- fft_t *fftfreqs = new fft_t[samplesize / 2];
-
- //this is used to compute frequency relation to the base frequency
- float adj[samplemax];
- for(int nsample = 0; nsample < samplemax; ++nsample)
- adj[nsample] = (Pquality.oct + 1.0f) * (float)nsample / samplemax;
- for(int nsample = 0; nsample < samplemax; ++nsample) {
- if(do_abort())
- goto exit;
- const float basefreqadjust =
- powf(2.0f, adj[nsample] - adj[samplemax - 1] * 0.5f);
-
- if(Pmode == 0)
- generatespectrum_bandwidthMode(spectrum,
- spectrumsize,
- basefreq * basefreqadjust,
- profile,
- profilesize,
- bwadjust);
- else
- generatespectrum_otherModes(spectrum, spectrumsize,
- basefreq * basefreqadjust);
-
- //the last samples contains the first samples
- //(used for linear/cubic interpolation)
- const int extra_samples = 5;
- PADnoteParameters::Sample newsample;
- newsample.smp = new float[samplesize + extra_samples];
-
- newsample.smp[0] = 0.0f;
- for(int i = 1; i < spectrumsize; ++i) //randomize the phases
- fftfreqs[i] = FFTpolar(spectrum[i], (float)RND * 2 * PI);
- //that's all; here is the only ifft for the whole sample;
- //no windows are used ;-)
- fft->freqs2smps(fftfreqs, newsample.smp);
-
-
- //normalize(rms)
- float rms = 0.0f;
- for(int i = 0; i < samplesize; ++i)
- rms += newsample.smp[i] * newsample.smp[i];
- rms = sqrt(rms);
- if(rms < 0.000001f)
- rms = 1.0f;
- rms *= sqrt(262144.0f / samplesize);//262144=2^18
- for(int i = 0; i < samplesize; ++i)
- newsample.smp[i] *= 1.0f / rms * 50.0f;
-
- //prepare extra samples used by the linear or cubic interpolation
- for(int i = 0; i < extra_samples; ++i)
- newsample.smp[i + samplesize] = newsample.smp[i];
-
- //yield new sample
- newsample.size = samplesize;
- newsample.basefreq = basefreq * basefreqadjust;
- cb(nsample, newsample);
- }
- exit:
-
- //Cleanup
- delete (fft);
- delete[] fftfreqs;
- delete[] spectrum;
- }
-
- void PADnoteParameters::export2wav(std::string basefilename)
- {
- applyparameters();
- basefilename += "_PADsynth_";
- for(int k = 0; k < PAD_MAX_SAMPLES; ++k) {
- if(sample[k].smp == NULL)
- continue;
- char tmpstr[20];
- snprintf(tmpstr, 20, "_%02d", k + 1);
- std::string filename = basefilename + std::string(tmpstr) + ".wav";
- WavFile wav(filename, synth.samplerate, 1);
- if(wav.good()) {
- int nsmps = sample[k].size;
- short int *smps = new short int[nsmps];
- for(int i = 0; i < nsmps; ++i)
- smps[i] = (short int)(sample[k].smp[i] * 32767.0f);
- wav.writeMonoSamples(nsmps, smps);
- }
- }
- }
-
- void PADnoteParameters::add2XML(XMLwrapper *xml)
- {
- xml->setPadSynth(true);
-
- xml->addparbool("stereo", PStereo);
- xml->addpar("mode", Pmode);
- xml->addpar("bandwidth", Pbandwidth);
- xml->addpar("bandwidth_scale", Pbwscale);
-
- xml->beginbranch("HARMONIC_PROFILE");
- xml->addpar("base_type", Php.base.type);
- xml->addpar("base_par1", Php.base.par1);
- xml->addpar("frequency_multiplier", Php.freqmult);
- xml->addpar("modulator_par1", Php.modulator.par1);
- xml->addpar("modulator_frequency", Php.modulator.freq);
- xml->addpar("width", Php.width);
- xml->addpar("amplitude_multiplier_type", Php.amp.type);
- xml->addpar("amplitude_multiplier_mode", Php.amp.mode);
- xml->addpar("amplitude_multiplier_par1", Php.amp.par1);
- xml->addpar("amplitude_multiplier_par2", Php.amp.par2);
- xml->addparbool("autoscale", Php.autoscale);
- xml->addpar("one_half", Php.onehalf);
- xml->endbranch();
-
- xml->beginbranch("OSCIL");
- oscilgen->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("RESONANCE");
- resonance->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("HARMONIC_POSITION");
- xml->addpar("type", Phrpos.type);
- xml->addpar("parameter1", Phrpos.par1);
- xml->addpar("parameter2", Phrpos.par2);
- xml->addpar("parameter3", Phrpos.par3);
- xml->endbranch();
-
- xml->beginbranch("SAMPLE_QUALITY");
- xml->addpar("samplesize", Pquality.samplesize);
- xml->addpar("basenote", Pquality.basenote);
- xml->addpar("octaves", Pquality.oct);
- xml->addpar("samples_per_octave", Pquality.smpoct);
- xml->endbranch();
-
- xml->beginbranch("AMPLITUDE_PARAMETERS");
- xml->addpar("volume", PVolume);
- xml->addpar("panning", PPanning);
- xml->addpar("velocity_sensing", PAmpVelocityScaleFunction);
- xml->addpar("fadein_adjustment", Fadein_adjustment);
- xml->addpar("punch_strength", PPunchStrength);
- xml->addpar("punch_time", PPunchTime);
- xml->addpar("punch_stretch", PPunchStretch);
- xml->addpar("punch_velocity_sensing", PPunchVelocitySensing);
-
- xml->beginbranch("AMPLITUDE_ENVELOPE");
- AmpEnvelope->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("AMPLITUDE_LFO");
- AmpLfo->add2XML(xml);
- xml->endbranch();
-
- xml->endbranch();
-
- xml->beginbranch("FREQUENCY_PARAMETERS");
- xml->addpar("fixed_freq", Pfixedfreq);
- xml->addpar("fixed_freq_et", PfixedfreqET);
- xml->addpar("detune", PDetune);
- xml->addpar("coarse_detune", PCoarseDetune);
- xml->addpar("detune_type", PDetuneType);
-
- xml->beginbranch("FREQUENCY_ENVELOPE");
- FreqEnvelope->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("FREQUENCY_LFO");
- FreqLfo->add2XML(xml);
- xml->endbranch();
- xml->endbranch();
-
- xml->beginbranch("FILTER_PARAMETERS");
- xml->addpar("velocity_sensing_amplitude", PFilterVelocityScale);
- xml->addpar("velocity_sensing", PFilterVelocityScaleFunction);
-
- xml->beginbranch("FILTER");
- GlobalFilter->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("FILTER_ENVELOPE");
- FilterEnvelope->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("FILTER_LFO");
- FilterLfo->add2XML(xml);
- xml->endbranch();
- xml->endbranch();
- }
-
- void PADnoteParameters::getfromXML(XMLwrapper *xml)
- {
- PStereo = xml->getparbool("stereo", PStereo);
- Pmode = xml->getpar127("mode", 0);
- Pbandwidth = xml->getpar("bandwidth", Pbandwidth, 0, 1000);
- Pbwscale = xml->getpar127("bandwidth_scale", Pbwscale);
-
- if(xml->enterbranch("HARMONIC_PROFILE")) {
- Php.base.type = xml->getpar127("base_type", Php.base.type);
- Php.base.par1 = xml->getpar127("base_par1", Php.base.par1);
- Php.freqmult = xml->getpar127("frequency_multiplier",
- Php.freqmult);
- Php.modulator.par1 = xml->getpar127("modulator_par1",
- Php.modulator.par1);
- Php.modulator.freq = xml->getpar127("modulator_frequency",
- Php.modulator.freq);
- Php.width = xml->getpar127("width", Php.width);
- Php.amp.type = xml->getpar127("amplitude_multiplier_type",
- Php.amp.type);
- Php.amp.mode = xml->getpar127("amplitude_multiplier_mode",
- Php.amp.mode);
- Php.amp.par1 = xml->getpar127("amplitude_multiplier_par1",
- Php.amp.par1);
- Php.amp.par2 = xml->getpar127("amplitude_multiplier_par2",
- Php.amp.par2);
- Php.autoscale = xml->getparbool("autoscale", Php.autoscale);
- Php.onehalf = xml->getpar127("one_half", Php.onehalf);
- xml->exitbranch();
- }
-
- if(xml->enterbranch("OSCIL")) {
- oscilgen->getfromXML(xml);
- xml->exitbranch();
- }
-
- if(xml->enterbranch("RESONANCE")) {
- resonance->getfromXML(xml);
- xml->exitbranch();
- }
-
- if(xml->enterbranch("HARMONIC_POSITION")) {
- Phrpos.type = xml->getpar127("type", Phrpos.type);
- Phrpos.par1 = xml->getpar("parameter1", Phrpos.par1, 0, 255);
- Phrpos.par2 = xml->getpar("parameter2", Phrpos.par2, 0, 255);
- Phrpos.par3 = xml->getpar("parameter3", Phrpos.par3, 0, 255);
- xml->exitbranch();
- }
-
- if(xml->enterbranch("SAMPLE_QUALITY")) {
- Pquality.samplesize = xml->getpar127("samplesize", Pquality.samplesize);
- Pquality.basenote = xml->getpar127("basenote", Pquality.basenote);
- Pquality.oct = xml->getpar127("octaves", Pquality.oct);
- Pquality.smpoct = xml->getpar127("samples_per_octave",
- Pquality.smpoct);
- xml->exitbranch();
- }
-
- if(xml->enterbranch("AMPLITUDE_PARAMETERS")) {
- PVolume = xml->getpar127("volume", PVolume);
- PPanning = xml->getpar127("panning", PPanning);
- PAmpVelocityScaleFunction = xml->getpar127("velocity_sensing",
- PAmpVelocityScaleFunction);
- Fadein_adjustment = xml->getpar127("fadein_adjustment", Fadein_adjustment);
- PPunchStrength = xml->getpar127("punch_strength", PPunchStrength);
- PPunchTime = xml->getpar127("punch_time", PPunchTime);
- PPunchStretch = xml->getpar127("punch_stretch", PPunchStretch);
- PPunchVelocitySensing = xml->getpar127("punch_velocity_sensing",
- PPunchVelocitySensing);
-
- xml->enterbranch("AMPLITUDE_ENVELOPE");
- AmpEnvelope->getfromXML(xml);
- xml->exitbranch();
-
- xml->enterbranch("AMPLITUDE_LFO");
- AmpLfo->getfromXML(xml);
- xml->exitbranch();
-
- xml->exitbranch();
- }
-
- if(xml->enterbranch("FREQUENCY_PARAMETERS")) {
- Pfixedfreq = xml->getpar127("fixed_freq", Pfixedfreq);
- PfixedfreqET = xml->getpar127("fixed_freq_et", PfixedfreqET);
- PDetune = xml->getpar("detune", PDetune, 0, 16383);
- PCoarseDetune = xml->getpar("coarse_detune", PCoarseDetune, 0, 16383);
- PDetuneType = xml->getpar127("detune_type", PDetuneType);
-
- xml->enterbranch("FREQUENCY_ENVELOPE");
- FreqEnvelope->getfromXML(xml);
- xml->exitbranch();
-
- xml->enterbranch("FREQUENCY_LFO");
- FreqLfo->getfromXML(xml);
- xml->exitbranch();
- xml->exitbranch();
- }
-
- if(xml->enterbranch("FILTER_PARAMETERS")) {
- PFilterVelocityScale = xml->getpar127("velocity_sensing_amplitude",
- PFilterVelocityScale);
- PFilterVelocityScaleFunction = xml->getpar127(
- "velocity_sensing",
- PFilterVelocityScaleFunction);
-
- xml->enterbranch("FILTER");
- GlobalFilter->getfromXML(xml);
- xml->exitbranch();
-
- xml->enterbranch("FILTER_ENVELOPE");
- FilterEnvelope->getfromXML(xml);
- xml->exitbranch();
-
- xml->enterbranch("FILTER_LFO");
- FilterLfo->getfromXML(xml);
- xml->exitbranch();
- xml->exitbranch();
- }
- }
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