Short notes

Sampling Theorem Short Notes

The ADC0804 in a data acquisition system sampling a 1 kHz audio signal needs a sample rate of at least 2 kHz to reconstruct the signal — but in practice, the anti-aliasing filter before the ADC must roll off sharply below half the sample rate, which is why real audio systems like CD players use 44.1 kHz sampling even though the highest audible frequency is 20 kHz. That practical oversampling margin is the engineering interpretation of the Nyquist-Shannon sampling theorem.

EEE, ECE, EI

How it works

The sampling theorem states that a band-limited signal with highest frequency component f_max can be perfectly reconstructed from its samples if the sampling frequency f_s ≥ 2f_max. The Nyquist rate is exactly 2f_max; the Nyquist interval (maximum allowable sampling period) is T_s = 1/(2f_max). In the frequency domain, sampling x(t) at rate f_s produces a spectrum X_s(f) that is a sum of shifted copies of X(f) at integer multiples of f_s. If f_s > 2f_max, these copies do not overlap and x(t) is recoverable by ideal low-pass filtering with cutoff f_s/2.

Key points to remember

Aliasing occurs when f_s < 2f_max, causing spectral copies to overlap and making reconstruction impossible without distortion. An anti-aliasing filter (low-pass, cutoff at f_s/2) must be applied before sampling to prevent this. The ideal reconstruction filter is a low-pass filter with cutoff f_s/2 and gain T_s, implemented approximately by a sample-and-hold circuit followed by a smoothing filter. Practical ADCs like the ADC0804 have a built-in sample-and-hold; its aperture time of about 1 µs limits the maximum signal frequency for distortion-free sampling. Oversampling by a factor of 4× relaxes anti-aliasing filter requirements significantly.

Exam tip

The examiner always asks you to state the sampling theorem, define aliasing, and draw the spectrum of a sampled signal showing overlapping copies — sketch both the aliasing case and the correctly sampled case on the same diagram.

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