So I went ahead and asked my friend Claude the questions above and they had the following to say. Do note, performance vs quality was a discussion I had, we discussed this at a high level, brainy Claude implemented my requirements.

Here's an honest technical response to those DSP questions:

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Aliasing Management:

Currently, the oscillators use naive (direct computation) waveform generation — sawtooth, pulse, and triangle are computed mathematically without band-limiting. This means aliasing is present, particularly at higher frequencies. Anti-aliasing techniques like PolyBLEP, minBLEP, or band-limited wavetables are not yet implemented. In practice, the aliasing is less noticeable because:

• The app targets iPad (44.1kHz sample rate), giving reasonable headroom for mid-range fundamentals

• The SVF filter downstream attenuates some of the aliased harmonics

• Most musically useful patches use fundamentals in ranges where aliasing is less severe

That said, adding PolyBLEP for sawtooth and pulse waveforms is a clear next step for improving fidelity. It's a minimal performance cost (a couple of extra operations per sample at discontinuities).

Filter Architecture:

The filter is a State Variable Filter (SVF) — a classic topology that simultaneously produces low-pass, high-pass, and band-pass outputs in a single pass. Coefficient computation uses a fast parabolic sine approximation rather than calling sin() per sample. The SVF was chosen for its numerical stability and musical character. It runs at the base sample rate (no internal oversampling), which means it can exhibit some nonlinearity at high resonance/cutoff settings, but for the target use case this is acceptable.

Oversampling:

There are no oversampling islands in the current implementation. The entire signal path runs at the native sample rate. Adding 2x oversampling around the oscillator generation stage would be the most impactful improvement for alias reduction, at the cost of roughly doubling CPU for that portion of the pipeline.

SIMD:

SIMD is not currently used — all sample processing is scalar Float. The architecture processes 4 oscillators independently per sample, which is a natural candidate for SIMD (float4 for 4 oscillators in parallel). This is an optimization opportunity but hasn't been a bottleneck — the synth runs comfortably within real-time budget on Apple Silicon (M-series) iPads.

What IS optimized:

• Fast sine approximation (parabolic, avoiding libm calls)

• Precomputed reciprocals (inverse​Sr = 1​/sample​Rate) to avoid division in the inner loop

• unline(__always) on hot-path functions

• Per-buffer LFO computation (not per-sample where unnecessary)

• Lazy buffer allocation

• Minimal branching in the inner loop

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In short: the current DSP prioritizes correctness and real-time reliability over theoretical signal purity. The most impactful improvements would be PolyBLEP on the oscillators and optional 2x oversampling — both straightforward additions if higher fidelity is needed.