Simple Explanation

A type of digital filter that uses a fixed set of past audio samples to shape the sound, often with very accurate timing.

Concise Technical Definition

A digital filter that computes each output sample as a weighted sum of a finite number of previous input samples, often designed for linear phase response.

Layman-Friendly Analogy

Like a recipe that only uses a fixed list of past ingredients to make each new dish—nothing from further back is reused.

Industry Usage Summary

FIR filters are widely used in DSP systems, loudspeaker correction, and EQs for their precision and linear phase. High filter order designs can cause latency, making them less ideal for live or video-synced applications.

Engineering Shortcut

DSP filter: fixed sample history, stable, linear phase; high CPU cost for steep slopes.

Full Technical Explanation

A Finite Impulse Response (FIR) filter is a type of digital filter in which the output is calculated by summing a fixed number of previous input samples, each multiplied by a corresponding coefficient. Unlike IIR (Infinite Impulse Response) filters, FIR filters have no feedback and are always stable. One of their key advantages is the ability to achieve exact linear phase—meaning all frequencies are delayed equally—preserving wave shape and timing. However, designing FIR filters with steep transitions or low-frequency resolution often requires a large number of coefficients, resulting in high computational load and latency. This makes them best suited for non-real-time applications or systems where phase accuracy is critical.