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Polish GUI, add pattern retrig option, and rework voice DSP
This commit is contained in:
Armin
parent
8ed279e84c
commit
4732240cc7
7 changed files with 546 additions and 129 deletions
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@ -9,6 +9,14 @@ namespace monostep
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class SynthVoice
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{
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public:
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enum class FilterType
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{
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lp12 = 0, hp12, bp12, notch12,
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lp24, hp24, bp24, notch24,
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lp48, hp48, bp48, notch48, // 48 dB/octave = "2 x 24"
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numFilterTypes
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};
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struct Params
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{
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Waveform waveA = Waveform::saw;
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@ -21,15 +29,22 @@ public:
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float osc2Phase = 0.0f;
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float cutoff = 7000.0f;
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float resonance = 0.15f;
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FilterType filterType = FilterType::lp12;
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float attack = 0.005f;
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float decay = 0.3f;
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float sustain = 0.7f;
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float release = 0.4f;
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float fAttack = 0.005f;
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float fDecay = 0.3f;
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float fSustain = 0.7f;
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float fRelease = 0.4f;
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float fAmount = 1.0f; // filter-env cutoff modulation in octaves
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float glide = 0.12f;
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float master = 0.9f;
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float drive = 0.0f;
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float ringMod = 0.0f;
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float detuneRatio = 1.0f;
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float detuneRatio = 1.0f; // OSC2 (inverted detune)
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float detuneRatioA = 1.0f; // OSC1 (direct detune)
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float mix = 0.5f;
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};
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@ -42,7 +57,8 @@ public:
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void prepare (double sr)
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{
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sampleRate = (float) sr;
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accentSmoothCoef = 1.0f - std::exp (-1.0f / (0.003f * sampleRate));
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accentAttackCoef = 1.0f - std::exp (-1.0f / (0.003f * sampleRate));
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accentReleaseCoef = 1.0f - std::exp (-1.0f / (0.050f * sampleRate));
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reset();
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}
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@ -51,9 +67,15 @@ public:
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phaseA = phaseB = 0.0f;
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currentFreq = targetFreq = 440.0f;
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smoothing = 0.0f;
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filterLow = filterBand = 0.0f;
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for (int i = 0; i < 4; ++i)
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{
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filterLow[i] = 0.0f;
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filterBand[i] = 0.0f;
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}
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env = 0.0f;
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envStage = Stage::idle;
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fenv = 0.0f;
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fenvStage = Stage::idle;
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gate = false;
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accentLevel = 1.0f;
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accentSmooth = 1.0f;
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@ -92,6 +114,9 @@ public:
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if (envStage != Stage::idle && envStage != Stage::release)
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envStage = Stage::release;
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if (fenvStage != Stage::idle && fenvStage != Stage::release)
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fenvStage = Stage::release;
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}
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}
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@ -109,6 +134,37 @@ private:
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void retriggerEnvelope()
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{
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envStage = Stage::attack;
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fenvStage = Stage::attack;
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}
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void updateFilterEnvelope()
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{
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const float sr = sampleRate;
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switch (fenvStage)
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{
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case Stage::attack:
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fenv += 1.0f / (params.fAttack * sr);
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if (fenv >= 1.0f) { fenv = 1.0f; fenvStage = Stage::decay; }
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break;
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case Stage::decay:
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fenv -= (1.0f - params.fSustain) / (params.fDecay * sr);
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if (fenv <= params.fSustain) { fenv = params.fSustain; fenvStage = Stage::sustain; }
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break;
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case Stage::sustain:
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fenv = params.fSustain;
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break;
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case Stage::release:
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fenv -= 1.0f / (params.fRelease * sr);
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if (fenv <= 0.0f) { fenv = 0.0f; fenvStage = Stage::idle; }
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break;
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case Stage::idle:
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break;
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}
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}
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void updateEnvelope()
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@ -155,27 +211,83 @@ private:
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}
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}
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float processFilter (float input)
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float processFilter (float input, float cutoff)
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{
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const float f = 2.0f * std::sin (juce::MathConstants<float>::pi * params.cutoff / sampleRate);
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const float q = 1.0f / (1.0f + params.resonance * 9.0f);
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const int numStages = numFilterStages();
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const float f = 2.0f * std::sin (juce::MathConstants<float>::pi * cutoff / sampleRate);
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// Distribute resonance across the pole stages (each stage gets res/stages) so the
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// composite resonance matches the 12 dB case instead of stacking into self-oscillation
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// near Nyquist. The 12 dB case (one stage) is unchanged.
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const float q = 1.0f / (1.0f + params.resonance * 9.0f / (float) numStages);
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filterLow += f * filterBand;
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const float high = input - filterLow - q * filterBand;
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filterBand += f * high;
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// Always cascade the LP response (low -> low): this keeps the integrator chain
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// unconditionally stable for every pole count. The requested response (LP/HP/BP/Notch)
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// is then tapped from the final stage, which preserves the 12/24/48 slope while
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// avoiding the ringing/unstability of cascading raw high/band signals.
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float sig = input;
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float low = 0.0f, high = 0.0f, band = 0.0f;
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return filterLow;
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for (int s = 0; s < numStages; ++s)
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{
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low = filterLow[s] + f * filterBand[s]; // new low (old band)
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high = sig - low - q * filterBand[s]; // new high (new low, old band)
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band = filterBand[s] + f * high; // new band
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// Safety clamp: prevents the resonant state from diverging into NaN when a
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// high pole count is driven at max resonance near Nyquist. Normal signals
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// (accents peak ~[1 + res*4]) never reach this bound.
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band = juce::jlimit (-16.0f, 16.0f, band);
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low = juce::jlimit (-16.0f, 16.0f, low);
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filterLow[s] = low;
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filterBand[s] = band;
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sig = low; // cascade LP response
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}
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return filterTap (low, high, band); // tap from last stage
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}
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int numFilterStages() const
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{
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switch (params.filterType)
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{
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case FilterType::lp12: case FilterType::hp12:
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case FilterType::bp12: case FilterType::notch12: return 1;
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case FilterType::lp24: case FilterType::hp24:
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case FilterType::bp24: case FilterType::notch24: return 2;
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case FilterType::lp48: case FilterType::hp48:
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case FilterType::bp48: case FilterType::notch48: return 4;
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default: return 1;
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}
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return 1;
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}
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float filterTap (float low, float high, float band) const
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{
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switch (params.filterType)
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{
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case FilterType::lp12: case FilterType::lp24: case FilterType::lp48: return low;
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case FilterType::hp12: case FilterType::hp24: case FilterType::hp48: return high;
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case FilterType::bp12: case FilterType::bp24: case FilterType::bp48: return band;
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case FilterType::notch12: case FilterType::notch24: case FilterType::notch48: return low + high;
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default: return low;
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}
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return low;
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}
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float renderSample()
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{
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updateEnvelope();
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updateFilterEnvelope();
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updateGlide();
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// Smooth the accent gain so it ramps in/out instead of clicking.
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accentSmooth += (accentLevel - accentSmooth) * accentSmoothCoef;
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// Accent gains fast attack / slow release, so the boost fades out smoothly
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// at the tail of an accented note instead of cutting off hard-edged.
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accentSmooth += (accentLevel - accentSmooth)
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* (accentLevel > accentSmooth ? accentAttackCoef : accentReleaseCoef);
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const float incA = currentFreq / sampleRate;
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const float incA = (currentFreq * params.detuneRatioA) / sampleRate;
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const float incB = (currentFreq * params.detuneRatio) / sampleRate;
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const float a = renderWave (params.waveA, phaseA, incA);
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@ -190,7 +302,9 @@ private:
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out = (1.0f - params.ringMod) * out + params.ringMod * (a * b);
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const float preFilter = out;
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out = processFilter (out);
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const float envCutoff = juce::jlimit (20.0f, 20000.0f,
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params.cutoff * std::pow (2.0f, params.fAmount * fenv));
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out = processFilter (out, envCutoff);
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// Accented steps also get a touch of the unfiltered signal so they cut
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// through the mix (presence), not just a volume bump.
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@ -215,15 +329,18 @@ private:
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float targetFreq = 440.0f;
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float smoothing = 0.0f;
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float filterLow = 0.0f;
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float filterBand = 0.0f;
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float filterLow[4] = {}; // SVF stage states (up to 4 poles for 48 dB)
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float filterBand[4] = {};
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float env = 0.0f;
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Stage envStage = Stage::idle;
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float fenv = 0.0f;
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Stage fenvStage = Stage::idle;
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bool gate = false;
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float accentLevel = 1.0f;
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float accentSmooth = 1.0f;
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float accentSmoothCoef = 0.01f;
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float accentAttackCoef = 0.0f; // fast charge when an accent starts
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float accentReleaseCoef = 0.01f; // slow release -> boost fades out at note end
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};
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} // namespace monostep
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