Sampling
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Sampling is the process of converting a continuous-time signal into a discrete-time signal by measuring its value at regular intervals.
Mathematical Model
Ideal sampling multiplies the continuous signal $x(t)$ by a Dirac comb:
$$ x_s(t) = x(t) \cdot \sum_{n=-\infty}^{\infty} \delta(t - nT_s) $$where $T_s$ is the sampling period and $f_s = 1/T_s$ is the sampling frequency.
Effect in Frequency Domain
Sampling causes the spectrum to become periodic (replicated at multiples of $f_s$):
$$ X_s(f) = \frac{1}{T_s} \sum_{k=-\infty}^{\infty} X(f - k f_s) $$Nyquist-Shannon Sampling Theorem
A bandlimited signal with maximum frequency $f_{max}$ can be perfectly reconstructed from its samples if:
$$ f_s > 2 f_{max} $$The minimum sampling rate $2 f_{max}$ is called the Nyquist rate.
See: Whittaker-Shannon sampling theorem
Aliasing
When $f_s < 2 f_{max}$, the replicated spectra overlap. High frequencies “fold back” and appear as lower frequencies — this is aliasing.
Aliasing cannot be corrected after sampling. Prevention requires:
- Anti-aliasing filter: Low-pass filter before sampling
- Oversampling: Sample faster than strictly necessary
Reconstruction
Ideal reconstruction uses a low-pass filter (sinc interpolation):
$$ x(t) = \sum_{n=-\infty}^{\infty} x[n] \, \text{sinc}\left(\frac{t - nT_s}{T_s}\right) $$See: Sinc function
Practical Considerations
- Real signals are never perfectly bandlimited
- Anti-aliasing filters have finite rolloff
- Reconstruction filters are causal approximations
- Quantization adds another source of error