Realistic sampling of signals:
Sampling a continuous-time signal in practice is realised by multiplying the signal with a train of pulses. Ideal impulses (Dirac deltas) are a mathematical abstraction and cannot be produced exactly in hardware. In real systems we use a periodic pulse train (for example rectangular pulses or saw-tooth derived gating pulses) generated by time-base circuits such as those used in a cathode-ray oscilloscope (CRO). Multiplication by such a pulse train produces a sequence of gated copies of the continuous-time signal; this is referred to here as realistic sampling.
A common hardware method uses two complementary pulse trains synchronized so that when one is high the other is low. By gating the input signal with one of these pulse trains through an electronic multiplier or switch, we obtain a sampled output that is non-zero only during the ON intervals of the chosen pulse train.
Consider the schematic where the signal x(t) is multiplied by a pulse train p1(t) while the complementary train p2(t) is low during those intervals. The multiplication output is present only when p1(t) is ON; when p2(t) is ON the output is zero. This is an implementable version of the ideal mathematical sampling operation.
In this arrangement the sampled signal is the product x(t) p(t), where p(t) is a periodic pulse train with period T (sampling period) and fundamental frequency ωs = 2π/T. The pulse width (ON duration) is denoted τ, and the duty cycle is α = τ/T.
A periodic pulse train that satisfies Dirichlet's conditions can be expanded in a complex Fourier series. If p(t) has period T and complex Fourier coefficients Ck, then
p(t) = Σk=-∞∞ Ck ej k ωs t, where ωs = 2π/T.
The Fourier coefficients are given by the period integral
Ck = (1/T) ∫0T p(t) e-j k ωs t dt.
For the constant (DC) coefficient (k = 0) we get the average value of p(t) over one period:
C0 = (1/T) ∫0T p(t) dt = α (for a rectangular pulse train whose amplitude during the ON interval is 1).
For a rectangular pulse of width τ centred within the period (integration limits chosen symmetrically), the kth coefficient simplifies to:
Ck = (τ/T) e-j k ωs τ/2 sinc( k ωs τ/2 ), where sinc(x) = sin(x)/x and ωs = 2π/T.
The magnitude of the coefficients has an envelope given by the sinc function:
The envelope |Ck| therefore decreases with |k| according to the sinc shape. The central coefficient (k = 0) equals the duty cycle α = τ/T and is the largest term in magnitude.
From the coefficient expression we can observe the following:
In the limiting case as
When the continuous-time signal x(t) is multiplied by the periodic pulse train p(t), we can use the Fourier series of p(t) to find the spectrum of the sampled signal.
Write the pulse train as p(t) = Σk Ck ej k ωs t. The sampled signal is xs(t) = x(t) p(t) = x(t) Σk Ck ej k ωs t.
Taking the Fourier transform and using the frequency-translation property gives
Xs(ω) = Σk=-∞∞ Ck X(ω - k ωs), where X(ω) is the Fourier transform of x(t).
Thus the spectrum of the sampled signal is a sum of frequency-shifted copies of the original spectrum X(ω), each copy weighted (modulated) by the corresponding Fourier series coefficient Ck.
If x(t) is band-limited with maximum frequency fm (or bandwidth fn), then reconstruction without overlap (aliasing) requires that the shifted copies do not overlap in frequency. In frequency (Hz) terms the sampling frequency fs = 1/T must satisfy
fs > 2 fn,
so that the copies centred at integer multiples of fs are separated. Under this condition the original spectrum can be recovered theoretically by passing xs(t) through an ideal low-pass filter whose passband selects the central copy (k = 0).
The faithful reconstruction condition is therefore
where fn denotes the highest frequency component (bandwidth) of the band-limited input signal.
The analysis above used a rectangular pulse train as an example, but the same reasoning applies to any periodic waveform p(t) that has a convergent Fourier series. The sampled spectrum will still be a sum of shifted copies of X(ω), with weights equal to the Fourier series coefficients of p(t). The central copy is weighted by the DC coefficient C0 (the average of p(t)).
If the periodic waveform has a non-zero average (i.e., C0 ≠ 0) and its fundamental frequency is greater than twice the bandwidth of the signal, then the central copy exists and an ideal low-pass filter can recover the original signal from the sampled representation.
If the periodic waveform has zero average (for example, certain symmetric waveforms with positive and negative lobes that cancel), then C0 = 0. In that case the central copy is suppressed and an ideal low-pass filter cannot directly recover the original signal; some other reconstruction method or preconditioning would be required.
Replacing rectangular pulses by triangular pulses or other shapes does not change the fundamental structure: the sampled spectrum remains a sum of shifted versions of X(ω) weighted by the Fourier coefficients of the chosen pulse shape. The coefficient magnitudes (the envelope) change according to the pulse shape: triangular pulses lead to a faster roll-off of coefficients (roughly the square of the sinc envelope for rectangular pulses), which affects the relative amplitudes of the shifted spectral copies and the level of out-of-band energy.
Sampling with a realistic periodic pulse train is modelled by multiplication in time, which yields a spectrum formed by shifted copies of the original signal spectrum weighted by the Fourier series coefficients of the pulse train. The duty cycle and shape of the pulses determine the coefficients (sinc-like envelopes for rectangular pulses). The usual aliasing condition applies: sampling frequency must exceed twice the signal bandwidth to avoid spectral overlap. Any periodic waveform with a non-zero average and sufficient fundamental frequency may be used for sampling and subsequent ideal low-pass reconstruction; if the average is zero the central spectral copy is suppressed and simple low-pass reconstruction is not possible.
Key terms: sampling, pulse train, duty cycle, Fourier series, Fourier coefficients, sinc envelope, band-limited signal, aliasing, sampling frequency, ideal low-pass reconstruction.
| 1. What is realistic sampling of signals? | ![]() |
| 2. Why is realistic sampling important in signal processing? | ![]() |
| 3. What factors should be considered for realistic sampling of signals? | ![]() |
| 4. How does realistic sampling affect signal reconstruction? | ![]() |
| 5. What are some common techniques used for realistic sampling of signals? | ![]() |
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