monostep/Source/dsp/SynthVoice.h

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#pragma once
#include <JuceHeader.h>
#include "WaveTables.h"
namespace monostep
{
class SynthVoice
{
public:
enum class FilterType
{
lp12 = 0, hp12, bp12, notch12,
lp24, hp24, bp24, notch24,
lp48, hp48, bp48, notch48, // 48 dB/octave = "2 x 24"
numFilterTypes
};
struct Params
{
Waveform waveA = Waveform::saw;
Waveform waveB = Waveform::square;
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float osc1Coarse = 0.0f;
float osc2Coarse = 0.0f;
float osc1Fine = 0.0f;
float osc2Fine = 0.0f;
float osc1Phase = 0.0f;
float osc2Phase = 0.0f;
float cutoff = 7000.0f;
float resonance = 0.15f;
FilterType filterType = FilterType::lp12;
float attack = 0.005f;
float decay = 0.3f;
float sustain = 0.7f;
float release = 0.4f;
float fAttack = 0.005f;
float fDecay = 0.3f;
float fSustain = 0.7f;
float fRelease = 0.4f;
float fAmount = 1.0f; // filter-env cutoff modulation in octaves
float glide = 0.12f;
float master = 0.9f;
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float drive = 0.0f;
float ringMod = 0.0f;
float detuneRatio = 1.0f; // OSC2 (inverted detune)
float detuneRatioA = 1.0f; // OSC1 (direct detune)
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float mix = 0.5f;
};
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void addPhase (float phaseOffset)
{
phaseA = frac (phaseA + phaseOffset);
phaseB = frac (phaseB + phaseOffset);
}
void prepare (double sr)
{
sampleRate = (float) sr;
accentAttackCoef = 1.0f - std::exp (-1.0f / (0.003f * sampleRate));
accentReleaseCoef = 1.0f - std::exp (-1.0f / (0.050f * sampleRate));
reset();
}
void reset()
{
phaseA = phaseB = 0.0f;
currentFreq = targetFreq = 440.0f;
smoothing = 0.0f;
for (int i = 0; i < 4; ++i)
{
filterLow[i] = 0.0f;
filterBand[i] = 0.0f;
}
env = 0.0f;
envStage = Stage::idle;
fenv = 0.0f;
fenvStage = Stage::idle;
gate = false;
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accentLevel = 1.0f;
accentSmooth = 1.0f;
}
void setParams (const Params& p) { params = p; }
bool isActive() const { return gate; }
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void setAccentLevel (float level) { accentLevel = level; }
void noteOn (float freqHz, bool legato)
{
targetFreq = freqHz;
if (gate && legato)
{
const float time = std::max (params.glide, 0.001f);
smoothing = 1.0f - std::exp (-1.0f / (time * sampleRate));
}
else
{
currentFreq = targetFreq;
smoothing = 0.0f;
retriggerEnvelope();
}
gate = true;
}
void noteOff()
{
if (gate)
{
gate = false;
if (envStage != Stage::idle && envStage != Stage::release)
envStage = Stage::release;
if (fenvStage != Stage::idle && fenvStage != Stage::release)
fenvStage = Stage::release;
}
}
void render (juce::AudioBuffer<float>& buffer, int numSamples)
{
auto* out = buffer.getWritePointer (0);
for (int i = 0; i < numSamples; ++i)
out[i] = renderSample();
}
private:
enum class Stage { idle, attack, decay, sustain, release };
void retriggerEnvelope()
{
envStage = Stage::attack;
fenvStage = Stage::attack;
}
void updateFilterEnvelope()
{
const float sr = sampleRate;
switch (fenvStage)
{
case Stage::attack:
fenv += 1.0f / (params.fAttack * sr);
if (fenv >= 1.0f) { fenv = 1.0f; fenvStage = Stage::decay; }
break;
case Stage::decay:
fenv -= (1.0f - params.fSustain) / (params.fDecay * sr);
if (fenv <= params.fSustain) { fenv = params.fSustain; fenvStage = Stage::sustain; }
break;
case Stage::sustain:
fenv = params.fSustain;
break;
case Stage::release:
fenv -= 1.0f / (params.fRelease * sr);
if (fenv <= 0.0f) { fenv = 0.0f; fenvStage = Stage::idle; }
break;
case Stage::idle:
break;
}
}
void updateEnvelope()
{
const float sr = sampleRate;
switch (envStage)
{
case Stage::attack:
env += 1.0f / (params.attack * sr);
if (env >= 1.0f) { env = 1.0f; envStage = Stage::decay; }
break;
case Stage::decay:
env -= (1.0f - params.sustain) / (params.decay * sr);
if (env <= params.sustain) { env = params.sustain; envStage = Stage::sustain; }
break;
case Stage::sustain:
env = params.sustain;
break;
case Stage::release:
env -= 1.0f / (params.release * sr);
if (env <= 0.0f) { env = 0.0f; envStage = Stage::idle; }
break;
case Stage::idle:
break;
}
}
void updateGlide()
{
if (smoothing > 0.0f)
{
currentFreq += (targetFreq - currentFreq) * smoothing;
if (std::fabs (targetFreq - currentFreq) < 0.01f)
{
currentFreq = targetFreq;
smoothing = 0.0f;
}
}
}
float processFilter (float input, float cutoff)
{
const int numStages = numFilterStages();
const float f = 2.0f * std::sin (juce::MathConstants<float>::pi * cutoff / sampleRate);
// Distribute resonance across the pole stages (each stage gets res/stages) so the
// composite resonance matches the 12 dB case instead of stacking into self-oscillation
// near Nyquist. The 12 dB case (one stage) is unchanged.
const float q = 1.0f / (1.0f + params.resonance * 9.0f / (float) numStages);
// Always cascade the LP response (low -> low): this keeps the integrator chain
// unconditionally stable for every pole count. The requested response (LP/HP/BP/Notch)
// is then tapped from the final stage, which preserves the 12/24/48 slope while
// avoiding the ringing/unstability of cascading raw high/band signals.
float sig = input;
float low = 0.0f, high = 0.0f, band = 0.0f;
for (int s = 0; s < numStages; ++s)
{
low = filterLow[s] + f * filterBand[s]; // new low (old band)
high = sig - low - q * filterBand[s]; // new high (new low, old band)
band = filterBand[s] + f * high; // new band
// Safety clamp: prevents the resonant state from diverging into NaN when a
// high pole count is driven at max resonance near Nyquist. Normal signals
// (accents peak ~[1 + res*4]) never reach this bound.
band = juce::jlimit (-16.0f, 16.0f, band);
low = juce::jlimit (-16.0f, 16.0f, low);
filterLow[s] = low;
filterBand[s] = band;
sig = low; // cascade LP response
}
return filterTap (low, high, band); // tap from last stage
}
int numFilterStages() const
{
switch (params.filterType)
{
case FilterType::lp12: case FilterType::hp12:
case FilterType::bp12: case FilterType::notch12: return 1;
case FilterType::lp24: case FilterType::hp24:
case FilterType::bp24: case FilterType::notch24: return 2;
case FilterType::lp48: case FilterType::hp48:
case FilterType::bp48: case FilterType::notch48: return 4;
default: return 1;
}
return 1;
}
float filterTap (float low, float high, float band) const
{
switch (params.filterType)
{
case FilterType::lp12: case FilterType::lp24: case FilterType::lp48: return low;
case FilterType::hp12: case FilterType::hp24: case FilterType::hp48: return high;
case FilterType::bp12: case FilterType::bp24: case FilterType::bp48: return band;
case FilterType::notch12: case FilterType::notch24: case FilterType::notch48: return low + high;
default: return low;
}
return low;
}
float renderSample()
{
updateEnvelope();
updateFilterEnvelope();
updateGlide();
// Accent gains fast attack / slow release, so the boost fades out smoothly
// at the tail of an accented note instead of cutting off hard-edged.
accentSmooth += (accentLevel - accentSmooth)
* (accentLevel > accentSmooth ? accentAttackCoef : accentReleaseCoef);
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const float incA = (currentFreq * params.detuneRatioA) / sampleRate;
const float incB = (currentFreq * params.detuneRatio) / sampleRate;
const float a = renderWave (params.waveA, phaseA, incA);
const float b = renderWave (params.waveB, phaseB, incB);
phaseA = frac (phaseA + incA);
phaseB = frac (phaseB + incB);
float out = (1.0f - params.mix) * a + params.mix * b;
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if (params.ringMod > 0.0f)
out = (1.0f - params.ringMod) * out + params.ringMod * (a * b);
const float preFilter = out;
const float envCutoff = juce::jlimit (20.0f, 20000.0f,
params.cutoff * std::pow (2.0f, params.fAmount * fenv));
out = processFilter (out, envCutoff);
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// Accented steps also get a touch of the unfiltered signal so they cut
// through the mix (presence), not just a volume bump.
if (accentSmooth > 1.0f)
out += (accentSmooth - 1.0f) * 0.2f * preFilter;
if (params.drive > 0.0f)
{
const float wet = std::tanh (out * (1.0f + params.drive * 9.0f));
out = (1.0f - params.drive) * out + params.drive * wet;
}
return out * env * params.master * accentSmooth;
}
float sampleRate = 44100.0f;
Params params;
float phaseA = 0.0f;
float phaseB = 0.0f;
float currentFreq = 440.0f;
float targetFreq = 440.0f;
float smoothing = 0.0f;
float filterLow[4] = {}; // SVF stage states (up to 4 poles for 48 dB)
float filterBand[4] = {};
float env = 0.0f;
Stage envStage = Stage::idle;
float fenv = 0.0f;
Stage fenvStage = Stage::idle;
bool gate = false;
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float accentLevel = 1.0f;
float accentSmooth = 1.0f;
float accentAttackCoef = 0.0f; // fast charge when an accent starts
float accentReleaseCoef = 0.01f; // slow release -> boost fades out at note end
};
} // namespace monostep