DSP 101
M7 · L1
Module 7 — FIR Filter Design
What is an FIR Filter?

Finite Impulse Response filters — the difference equation, the impulse response, unconditional stability, and the celebrated linear-phase property that makes FIR indispensable.

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DSP 101
M7 · L1
The Core Idea
Weighted Sum, No Feedback

An FIR filter outputs a weighted sum of the current and M past input samples. No feedback — the output never re-enters the calculation. This one structural choice delivers all of FIR's best properties.

FIR Difference Equation
y[n] = \sum_{k=0}^{M} b_k\, x[n-k]
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DSP 101
M7 · L1
The Filter IS Its Impulse Response
h[n] = the Coefficients

For an FIR filter, the impulse response h[n] equals the coefficient sequence b_k — nothing more. The output is just the convolution of the input with this finite sequence.

M+1
Taps (filter length)
M+1
MACs per sample
0
Poles (no feedback)
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DSP 101
M7 · L1
Stability Guarantee
Always Stable

No poles → no instability. BIBO stability requires all poles inside the unit circle. FIR filters have only zeros — their poles are trivially at the origin. Any coefficient values you choose produce a stable filter.

Contrast with IIR
IIR filters use feedback and have poles that can escape the unit circle — especially dangerous when finite-word-length quantization nudges a pole outward.
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DSP 101
M7 · L1
Frequency Response
A Polynomial in e−jω

The frequency response is the DTFT of h[n] evaluated on the unit circle — a polynomial in e^{−jω}. Longer filters (more terms) → sharper transitions, flatter passbands.

Frequency Response
H(e^{j\omega}) = \sum_{k=0}^{M} b_k\, e^{-j\omega k}
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DSP 101
M7 · L1
The Celebrated Property
Linear Phase — No Distortion

Symmetric coefficients (h[n] = h[M−n]) → linear phase: ∠H(e^{jω}) = −(M/2)ω. Every frequency is delayed by the same M/2 samples — waveform shape is preserved exactly.

  • Symmetric h[n] → zero phase distortion (Type I & II)
  • Antisymmetric h[n] → 90° shift + linear phase (Type III & IV)
  • Symmetry also halves unique multiplications needed
  • IIR filters cannot achieve exact linear phase
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DSP 101
M7 · L1
Where FIR Filters Live
FIR in the Real World
  • Audio EQ & crossovers — linear phase preserves music fidelity
  • Pulse shaping (RRC) in 4G/5G modems & Wi-Fi
  • Anti-aliasing & anti-imaging in sample-rate converters
  • ECG/EEG processing — waveform shape must not be distorted
  • Very long filters: overlap-add FFT method for efficiency
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DSP 101
Quick check

Check what stuck

Four questions from this lesson. Answer to see why — the explanation appears whether you were right or wrong. Nothing is scored or saved.

Question 1 of 0
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8 / 9
DSP 101
M7 · L1
Key Takeaways
What You Learned
  • FIR = weighted sum of M+1 input samples, zero feedback
  • Impulse response h[n] = coefficient sequence b_k
  • No poles → unconditionally BIBO stable for any coefficients
  • Symmetric h[n] → linear phase → no waveform distortion
  • Cost: M+1 MACs/sample; long filters use overlap-add FFT
  • Used everywhere: audio, comms, multirate, biomedical
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