Unit - 5

Sampling and Reconstruction

5.1 Sampling theorem and its implications

Sampling theorem states that “The continues form of a time-variant signal

can be represented in the discrete form of a signal with help of samples and
the sampled can be recovered to original form when the sampling signal
frequency Fs having the more excellent frequency value than or equal to
twice the input signal frequency Fm.

Fs ≥ 2Fm

If the sampling frequency (Fs) equals twice the input signal frequency (Fm),
then such a condition is called the Nyquist Criteria. When sampling
frequency equals twice the input signal frequency is known as “Nyquist
rate”.

Fs=2Fm

If the sampling frequency (Fs) is less than twice the input signal frequency,
such criteria called an Aliasing effect.

Fs<2Fm

5.2 Spectra of sampled signals

Consider a continuous time signal x(t). The spectrum of x(t) is a band

limited to fm Hz that is the spectrum of x(t) is zero for |ω|>ωm.
Sampling of input signal x(t) can be obtained by multiplying x(t) with an
impulse train δ(t) of period Ts. The output of multiplier is a discrete signal
called sampled signal which is represented with y(t) in the following
diagrams:
Figure 1. Spectrum
of the sampled
signal

The sampled signal

y(t) = x(t) . δ(t) -------
--------------------------
(1)

The Fourier series

representation
of δ(t) is given by

δ (t) = ao + an cosnwst + bn sin nwst) ------------------------------------(2)

Where ao = 1/Ts dt = 1/Ts δ(0) = 1/Ts

An = 2/Ts cos nws t . Dt = 2/T2 δ(0) cosnws0= 2/T

Bn= 2/Ts sin nws t.dt = 2/T2 δ(0) sin ws0 =0

Therefore δ (t) = 1/Ts + cosnwst +0)

The sampled signal

y(t) = x(t). δ(t)

= x(t) [1/Ts + 2/Ts cos nwst)]

= 1/Ts [ x(t) + 2 cosnwst ) x(t)]

y(t) = 1/Ts [ x(t) + 2 cos wst . x(t) + 2 cos 2wst. x(t) + 2 cos 3ws(t) . x(t)
+………..]

Taking Fourier transform on both the sides we get

Y(w) = 1/Ts [ X(w) + X(w-ws) + X(w+ws) + X(w-2ws) + X(w+2ws) +…..]

Y(w) = 1/Ts (w-nws) where n = 0, 1, …………

Figure 2. Sampled
spectrum with
different conditions

Key Takeaways:

The sampling
theorem specifies
the minimum-
sampling rate at
which a continuous-
time signal needs to
be
uniformly sampled so that the original signal can be recovered entirely or
reconstructed by these samples alone

5.3 Reconstruction, Ideal Interpolator

The reconstructed analog signal involves weighing each individual discrete sample
by a sinc function shifted-in-time by the sampling period. The sinc function:

P (t) = sin (π/Ts (t-nTs)]/ π/Ts (t-nTs) -------------------------------------------- (1)

Is known as the ideal interpolation filter expressed in the frequency domain as:

Applying
the ideal

interpolation filter in eel(2) to the spectrum of a non-aliased discrete-time signal

results in recovering the original analog signal without any loss of information, as
shown in Figure .

Figure 3. Recovered sampled spectrum of original analog signal

To start with filtering the spectrum of the discrete-time signal by the interpolation filter
eel (2), that is:

(F/Fs) = X(F/Fs) P(F)-----------------------------------------------------------(3)

Multiplication in the frequency domain is convolution in the time domain,

* p(t) = sin( (t-nTs))/ (t-nTs) ------(4)

5.4 Zero Order Hold

The continuous - time signal is multiplied with a periodic impulse train, referred to
as Sampling Function. A sampled signal is then obtained as shown in figure below.
Figure 4. Sampled Zero Order Hold

p(t) = (t-nts) -------------------------------(1)

The ideally sampled signal up (t) is the product of the impulse train p (t) and the
analog signal XC (t) which is written as

Xp(t) = xo(t) . p(t) ------------------------------------(2)

Xp(t) = δ(t-nts) = δ(t-nts) ------------------------(3)

Xp(t) = δ(t-nts)--------------------------------(4)

The ZOH Sampled Signal xZOH (t) can be regarded as the convolution of ho(t) and a
sampled signal xp(t)

xzoH(t) = ho(t) * xp(t)-------------------------------------(5)

= ho(t) * [ δ(t-nts)]

= ho(t-nts)-----------------------------------------(6)

5.5 First Order Hold

Consider the signal

 A continuous signal x1(t) can be recovered by

x 1(t) can be recovered by

x1(t) = h1(t) * xs(t)

Which is the linear interpolation of the sample train x(mTs) . This interpolation
corresponds a low pass filtering in frequency domain by

H1(w) = F[h1(t)] = 4/w2 Ts sin2 (wsT/2)

Figure 5. First Order Hold

Key Takeaways:

The reconstruction process consists of replacing each sample by a sinc function,

centered at the time of the sample and scaled by the sample value x(nT) times 2f c/
fs and adding all the functions so created

5.6 Aliasing and its effects

When the signal is converted back into a continuous time signal, it will exhibit a
phenomenon called aliasing. Aliasing is the presence of unwanted components in
the reconstructed signal. These components were not present when the original
signal was sampled.
In addition, some of the frequencies in the original signal may be lost in the
reconstructed signal. Aliasing occurs because signal frequencies can overlap if the
sampling frequency is too low. Frequencies "fold" around half the sampling
frequency - which is why this frequency is often referred to as the folding frequency.

Key Takeaways:

Aliasing occurs when a signal is sampled at a less than twice the highest frequency
present in the signal

5.7 Relation between continuous and discrete systems

Let's start sampling a continuous-time signal, as shown in this graph:

Figure 6. Sampled signal

Mathematically, the relationship

between the discrete-time signal and
the continuous-time signal is given
by:

x[n] = xc(nT) -------------------------------

------------(1)

The sampling frequency is fs =

1/T (in Hz) orΩs= 2π/T (in radians per second).

The discrete-time Fourier transform of x[n] is related to the continuous-time Fourier

transform of xc(t) as follows:
X(w) = 1/T (w/T +2πk/T)

There are two key pieces to this equation.

The first is a scaling relationship between w and Ω : w=ΩT . This means that the
sampling frequency in the continuous-time Fourier transform, Ωs , becomes the
frequency 2π in the discrete-time Fourier transform.

The discrete-time frequency w=π corresponds to half the sampling frequency,

or Ωs/2.

The second key piece of the equation is that there are an infinite number of copies
of Xc(w/T) spaced by 2π.

Suppose Xc(Ω )looks like this:

Figure 7. Spectrum Xc(Ω)

Note that Xc(Ω) equals zero for all

frequencies |Ω| >Ωo. This means the
continuous-time signal is band-limited. The frequency Ωo is called the bandwidth of
the signal.

The discrete-time Fourier transform of xc=x[nT] looks like this:

Figure 8. Discrete spectrum

Where wo=ΩoT . One period

of X(w) is shown:
 Figure 9. Scaled version

For this example, then, X(w) between -π and π looks just

like a scaled version of Xc(Ω).

Key Takeaways:

If a continuous time signal x(t) that has been sampled each T seconds to produce
a discrete time signal x(n) .

5.8 Introduction to the application of Signal And System Theory,

filtering, feedback control systems

Modulation for communication

A message carrying signal has to be transmitted over a distance and to establish a

reliable communication, it needs to take the help of a high frequency signal which
should not affect the original characteristics of the message signal.

The characteristics of the message signal, if changed, the message contained in it

also alters. Hence care must be taken of the message signal. A high frequency signal
can travel up to a longer distance, without getting affected by external disturbances.

We take the help of such high frequency signal which is called as a carrier signal to
transmit our message signal. Such a process is simply called as Modulation.

Modulation is the process of changing the parameters of the carrier signal, in

accordance with the instantaneous values of the modulating signal.
Filtering

The concept of filtering is a direct consequence of the fact that for linear, time-
invariant systems the Fourier transform of the output is the Fourier transform of the
input multiplied by the frequency response, that is the Fourier transform of the
impulse response.

Because of this, the frequency content of the output is the frequency content of the
input shaped by this frequency response. Frequency-selective filters attempt to
exactly pass some bands of frequencies and exactly reject others. Frequency-shaping
filters more generally try to reshape the signal spectrum by multiplying the input
spectrum by some specified shaping.

Ideal frequency-selective filters, such as lowpass, highpass, and bandpass filters, are
useful abstractions mathematically but are not precisely implementable.

Feedback Control Systems

A feedback control system is a system whose output is controlled using its

measurement as a feedback signal. This feedback signal is compared with a reference
signal to generate an error signal which is filtered by a controller to produce the
system's control input. The block diagram below illustrates a general feedback
system.

Figure 10. Feedback system

The controlled system is called the plant, and its LTI model is the transfer function P (s
). The disturbed output of the plant is y (t) and its noisy measurement is my(t )
corrupted by the measurement noise n (t ) . The error between the desired output dy(t
) (or reference) and my(t ) is the measured error, denoted as me(t ) . The output
disturbance is the signal d(to ) and the output measurement noise is n(t ) .

Gs (m) models the feedback measurement sensor dynamics. The actuator (e.g., a
valve) modeled by G (s ) is the device that translates a control signal from the
controller K (s ) into an action on the plant input. The input disturbance signal d(ti )
disturbs the control signal from the actuator to the plant input.