kunitoki Audio Math Explainer
The rule: DSP math is small and composable — each concept builds directly on the last. Start with what the user needs, then connect it to code they're writing.
Step 1 — Identify the concept
| Topic | Core idea | Common "why" |
|---|---|---|
| Fourier transform / DFT / FFT | Decompose signal into frequency components | "Why does my FFT output look wrong?" |
| Convolution | Weighted moving average; LTI system response | "How does an IR reverb work?" |
| Z-transform | Frequency-domain analysis of discrete systems | "How do filter poles/zeros work?" |
| Biquad / IIR filters | Recursive difference equation with feedback | "Why does my filter ring / go unstable?" |
| FIR filters | Non-recursive weighted sum | "How do I design a linear-phase filter?" |
| Windowing | Reduce spectral leakage in DFT | "Why are FFT edges smeared?" |
| Sample rate / Nyquist | Highest representable frequency = sr/2 | "What causes aliasing?" |
| Decibels | Log scale for amplitude/power ratios | "How do I convert gain to dB?" |
If the user's question maps to multiple rows, start with the most fundamental one and build up.
Step 2 — Explain with three anchors
For each concept, give exactly these three things in order:
- Intuition — one sentence a musician could understand
- Math — the key formula (inline, no walls of derivation)
- Code — one-line pseudocode or a real function call from common audio libraries
Keep each anchor to 1–3 lines. Cut anything that doesn't directly answer the question.
Step 3 — Connect to the user's code
After the three anchors:
- Locate where the math appears in their implementation (coefficient calculation, loop structure, buffer size choice, etc.)
- Name the exact variable or line where theory becomes code
- If a value looks wrong, explain which part of the math it violates
Quick connection table
| Math concept | Where it shows up in code |
|---|---|
| DFT basis frequencies | bin_index * sr / N — each FFT bin |
| Convolution sum | FIR tap loop: y[n] = sum(h[k] * x[n-k]) |
| Z-plane pole radius | IIR feedback coefficients a1, a2 |
| Nyquist limit | if (freq > sampleRate / 2) clamp(...) |
| Window function | Multiply x[n] *= window[n] before FFT |
| dB conversion | gain_db = 20 * log10(amplitude) |
Output format
### [Concept name]
**Intuition:** [one sentence]
**Math:** `[formula]`
**Code:** `[one-liner]`
**In your code:** [where this appears and what it means for their specific question]
See
references/dsp-math-reference.mdfor full formula tables, gotchas, and property lists.