The Sampling Theorem

Sources:

  1. B. P. Lathi & Roger Green. (2018). Chapter 8: Sampling: The Bridge From Continuous To Discrete. Signal Processing and Linear Systems (3rd ed., pp. 775-784). Oxford University Press.

In this article, we show that a real signal x(t) whose spectrum is bandlimited to B Hz, i.e., X(ω)=0 for |ω|>2πB), can be reconstructed exactly (without any error) from its samples taken uniformly at a rate fs>2B samples per second. In other words, the minimum sampling frequency is fs>2B Hz.

NOTE: In some literature, fs=2B is be used, which is not correct.

Notation

Symbol Meaning
fs The sampling frequency in Hz.
x¯(t) The sampled signal x¯(t) whereas the original signal is x(t).
δT(t) The impulse train signal, consisting of unit impulses repeating periodically every T seconds,

The sampling theorem

To prove the sampling theorem, consider a signal x(t) (Fig. 8.1a) whose spectrum (Fig. 8.1b) is bandlimited to B Hz.

Figure 8.1

For convenience, spectra are shown as functions of ω as well as of f (hertz).

Sampling x(t) at a rate of fs Hz can be accomplished by multiplying x(t) by an impulse train δT(t) (Fig. 8.1c), consisting of unit impulses repeating periodically every T seconds, where T=1/fs. The schematic of a sampler is shown in Fig. 8.1d. The resulting sampled signal x¯(t) is shown in Fig. 8.1e.

The sampled signal consists of impulses spaced every T seconds (the sampling interval). The nth impulse, located at t=nT, has a strength x(nT), the value of x(t) at t=nT. x¯(t)=x(t)δT(t)=nx(nT)δ(tnT)

Because the impulse train δT(t) is a periodic signal of period T, it can be expressed as a trigonometric Fourier series: δT(t)=1T[1+2cosωst+2cos2ωst+2cos3ωst+]ωs=2πT=2πfs

Therefore, x¯(t)=x(t)δT(t)=1T[x(t)+2x(t)cosωst+2x(t)cos2ωst+2x(t)cos3ωst+]

To find X¯(ω), the Fourier transform of x¯(t), we take the Fourier transform of the right-hand side of this equation, term by term.

  1. The transform of the first term in the brackets is X(ω).

  2. The transform of the second term 2x(t)cosωst is X(ωωs)+X(ω+ωs). This represents spectrum X(ω) shifted to ωs and ωs.

  3. Similarly, the transform of the third term 2x(t)cos2ωst is X(ω2ωs)+X(ω+2ωs), which represents the spectrum X(ω) shifted to 2ωs and 2ωs, and so on to infinity.

This result means that the spectrum X¯(ω) consists of X(ω) repeating periodically with period ωs=2π/Trad/s, or fs=1/T Hz, as depicted in Fig. 8.1f. Therefore,

X¯(ω)=1Tn=X(ωnωs)

If we are to reconstruct x(t) from x¯(t), we should be able to recover X(ω) from X¯(ω). This recovery is possible if there is no overlap between successive cycles of X¯(ω). Figure 8.1f indicates that this requires (1)fs>2B

Also, the sampling interval T=1/fs. Therefore, T<12B

The corresponding sampling interval T=1/2B is called the Nyquist interval for x(t).

Thus, as long as the sampling frequency fs>2B (in hertz), X¯(ω) consists of nonoverlapping repetitions of X(ω), and x(t) can be recovered from its samples x¯(t) by passing the sampled signal x¯(t) through an ideal lowpass filter having a bandwidth of any value between B and fsB Hz.

Practical sampling

In proving the sampling theorem, we assumed ideal samples obtained by multiplying a signal x(t) by an impulse train that is physically unrealizable. In practice, we multiply a signal x(t) by a train of pulses of finite width, depicted in Fig. 8.3c. The sampler is shown in Fig. 8.3d. The sampled signal x¯(t) is illustrated in Fig. 8.3e.

Figure 8.3

We wonder whether it is possible to recover or reconstruct x(t) from this x¯(t). Surprisingly, the answer is affirmative, provided the sampling rate is not below the Nyquist rate. The signal x(t) can be recovered by lowpass filtering x¯(t) as if it were sampled by impulse train.

The plausibility of this result becomes apparent when we consider the fact that reconstruction of x(t) requires the knowledge of the Nyquist sample values. This information is available or built into the sampled signal x¯(t) in Fig. 8.3e because the nth sampled pulse strength is x(nT). To prove the result analytically, we observe that the sampling pulse train pT(t) depicted in Fig. 8.3c, being a periodic signal, can be expressed as a trigonometric Fourier series pT(t)=C0+n=1Cncos(nωst+θn)ωs=2πT

Thus, x¯(t)=x(t)pT(t)=x(t)[C0+n=1Cncos(nωst+θn)]=C0x(t)+n=1Cnx(t)cos(nωst+θn)

The sampled signal x¯(t) consists of C0x(t),C1x(t)cos(ωst+θ1),C2x(t)cos(2ωst+θ2), Note that the first term C0x(t) is the desired signal and all the other terms are modulated signals with spectra centered at ±ωs,±2ωs,±3ωs,, as illustrated in Fig. 8.3f.

Clearly the signal x(t) can be recovered by lowpass filtering of x¯(t), as shown in Fig. 8.3d. As before, it is necessary that ωs>4πB( or fs>2B).