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# Linear Systems

July 2016

## Signals & Systems Tutorial Sheet

Every accomplishment starts with the decision to try it
1. A linear time-invariant (LTI) system has input x(t) = u(t)+2u(t2)+u(t5), where
u(t) refers to the standard unit step function. The impulse response of a system is
h(t) = e2t u(1 t).
(a) Sketch the x(t) and h(t)
(b) Is this system causal ?
(c) Is this system stable ?
(d) Obtain and sketch the response of the system to a unit step input.
(e) Obtain the response y(t) of the system to the given input x(t).
(f) Simplify the expression for y(t) when (i )t < 0 (ii) t > 0 and sketch y(t) neatly
in these two intervals.
2. You know that the Fourier Series coefficients of a periodic function, x(t), is given by,

1Z
ci =

2kt

x(t) exp j
dt
T

(1)

## and the Fourier series expansion is given by,

x(t) =

2kt
ci exp j

(2)

(a) Begin by calculating the Fourier series for a square wave voltage signal that goes
from 0 to 1 V and has a period of T seconds and duty cycle of 50%. In other
words, x(t) is the periodic repetition of the following function:

1 |t| < T /4
xp (t) =

## 0 T /4 < |t| < T

1

(b) Notice that the Fourier Series coefficient are symmetric. Additionally, the even
terms are zero. This is expected from the symmetry of the signal. Now consider
another square wave voltage, c(t), which has a zero average and a peak to peak
value of 2 V. What is its Fourier Series coefficients?
(c) Now that you have the Fourier Series coefficients of c(t), write down the Fourier
Transform. Plot the spectrum.
3. Consider the schematic shown in Figure 1.

VLO(t)
VRF(t)

VO(t)
RL

## Figure 1: A Mixer Model

The schematic models a mixer, a circuit used in RF transceivers. As you can see, it
consists of a switch controlled by a voltage VLO (t) which turns the switch on and off
at a frequency of LO (assume a 50% duty cycle).
4. The input to the system is a high frequency RF signal. It has the following spectrum:

1
-RF

RF

## Figure 2: RF Signal Spectrum

Were using the circuit as a receiver. With this information and the exercises done in
the previous sections,
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## (a) Sketch the output spectrum of the circuit.

(b) Find the conversion gain of the mixer. To do so, calculate the amplitude of the
desired spectral component and divide it by the amplitude of the input spectrum.
5. Let x(t) = sinc(t) + sinc(t2). Find the value of the inner product hsinc(t 2), x(t)i.
(Hint: Use Parsevals Theorem). Here sinc(t) =

sin t
t .

## orthogonal to each other. Can they be used as basis vector ?

6. Given an LTI system with frequency response
H(ej ) = e

j
4

, < <

obtain the output of the system, y[n], for the input x[n] = cos( 3n
2 ).
7. Given an LTI system difference equation y[n] + 0.5y[n 1] = x[n], with x[n] = (1)n ,
the maximum value of the output y[n] is. (Hint: Think if you can make x[n] look like
an eigenfunction of an LTI system.)
8. Find the Fourier Transform of e|t| .
9. Determine whether each of the following statement is true or false. Justify tour answer.
(i) An odd and imaginary always hs and odd and real spectrum.
(ii) An even and real signal always has an even and real spectrum.
(iii) An odd and real signal always has odd and imaginary spectrum.
(iv) An even and imaginary signal always has even and real spectrum.
10. The continuous time real signal x(t) is defined as
x(t) = x1 (t){x2 (t) x3 (t)} + x4 (t)
The Fourier transforms of the signals are such that
X1 (j) = 0, || > 1
X2 (j) = 0, || > 2
X3 (j) = 0, || > 3
X4 (j) = 0, || > 4
Given that |1 | > |4 | , |1 | < |2 | + |3 | , |2 | < |3 |. If signal x(t) is sampled to
obtain s[n]. what is the minimum sampling frequency required to recover x(t) from
s[n]?
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11. The signal x(t) = cos(2f1 t)+cos(2f2 t) is subjected to ideal uniform sampling at the
rate of 18 kHz. All frequencies, including f1 and f2 are expressed in kiloHertz(kHz)
and time t is measured in milliseconds. The sampled signal is subjected to the
action of an adjustable ideal bandpass filter, with a real valued impulse response,
having a frequency response of 1 in its passband and 0 on the rest of the frequency
axis (stopband). The phase response of the bandpass filter can be taken to be zero
for all frequencies.
(a) In the first experiment, f1 = 3, f2 = 5; the passband is from 4 to 16 kHz. Obtain
the output of the bandpass filter, expressing it as a sum of sinuoids, specifying
the frequencies.
(b) In the second experiment, f1 = 7, f2 = 10; the passband is from 4 to 12 kHz.
Obtain the output of the bandpass filter and express it as a product of two terms,
the first term being a pure sinusoid of an appropriate frequency, the second, a
sum of two sinusoids in which the frequency of one sinusoid is twice that of the
other.
12. For each of the pole-zero plots given below, plot the magnitude response of the spectrum. Do this graphically (i.e, without writing any math).

Pole-Zero Map
3

-1

-1

-2

-3
-3

-2

-1

1
-1

Pole-Zero Map
3

-1

-1

-2

-3
-3

-2

-1

1
-1

Pole-Zero Map
3

-1

-1

-2

-3
-3

-2

-1

1
-1

Pole-Zero Map
3

-1

-1

-2

-3
-3

-2

-1

1
-1

Pole-Zero Map
3

-1

-1

-2

-3
-3

-2

-1

1
-1

Pole-Zero Map
3

-1

-1

-2

-3
-3

-2

-1

## Real Axis (seconds

-1

Brain Teasers
1. Three systems A,B and C have the inputs and outputs indicated in Table. Determine
whether each system could be LTI. If your answer is yes, specify whether there could
be more than one LTI system with the given input-output pair. Explain your answer.

System
System A

Input

1 n

Output

1 n

System B

ejn/7 u[n]

3ejn/7 u[n]

System C

ejn/7

2ejn/7

sin n
2
h[n] =
n

## (a) Examine if the system is BIBO stable or not.

(b) Does its Fourier Transform exists ? If Yes, then what is it ?

## 3. In question 5 we proved that sinc(t n) for n Z can be used as basis vectors.

Suppose we sample a continuous time signal x(t) having bandwidth W at sampling
frequency Fs to get x[nTs ] (here Fs > 2W ). After sampling, in frequency domain we
get the copies of the signal at the multiples of Fs . If we want to reconstruct the signal
back from the sampled version, we need to use a LPF (Ideal) of bandwidth W . Can
you represent this reconstructed signal with sinc(t n) as their basis vectors.
4. Effective number of bits (ENOB) are the number of bits of ADC required to achieve
the required SQNR. Find the ENOB for SQNR = 65 dB.
Now lets go into the frequency domain analysis of the signal and the quantization
noise. When the signal is sampled at Fs and quantized using an N-bit ADC, the
quantization noise is present all over the frequencies from 0 Fs /2. The actual signal
occupies just a small bandwidth in the low frequency region. Now, according to
the Parsevals Theorem, Power in time domain is equal to the power in frequency
domain. Hence the SQNR is also equal to ratio of power of the signal to the power of
the quantization noise, in frequency domain. Can you describe a procedure by which
you can increase the SQNR (Hint: You have to fiddle with the Fs and use a filter). If
you get the answer then congratulations !! you have just learned the concept of
5. Suppose a signal x[n] is transmitted from a source and some noise v[n] is added to it
in the transmission medium. At the destination you have to design a filter to h[n] to
recover back the clean signal x[n]. Comment on the ways in which you can design the
filter if the signal and the noise spectrum are(i) non-overlapping, (ii) overlapping.
6. In this problem we look at diffusion. Diffusion is a process that occurs in many systems. Diffusion is why you can smell good food thats cooking. For us electrical
engineers, we are interested in how diffusion works in semiconductors. Diffusion is
also a process used in IC fabrication. We look into this application.
In IC fabrication, we can dope a semiconductor by first implanting a large concentration of dopant atoms in the silicon wafer. We then anneal the wafer and let the
atoms diffuse into the crystal. Consider the time depended diffusion equation. Suppose c(x, t) (atoms per unit volume) is the concentration profile of the dopant atoms
as a function of the diffusion time (t) and the position on the wafer (x). The situation
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is shown in Figure 3

N atoms/unit area

x
Figure 3: Initial concentration profile

2c

c
t

=D

(3)

x2

## where D is the diffusion constant.

We wont look into how we can solve the equation. Heres the solution:

2
x

exp
c(x, t) =

4Dt
Dt
N

(4)

where N is the initial number of atoms per unit area and is quite large. The concentration profile is shown in Figure 4
Time Dependent Diffusion

1.0

t1>0
t2>t1

Concentration

0.8

0.6

0.4

0.2

0.0
10

0
Position

## Figure 4: Concentration profile after anneal

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What will the concentration profile look like at some time t > 0 if the initial concentration profile is as shown in Figure 5

## Initial concentration c(0, 0)

x
Figure 5: Initial concentration profile

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