Signals & Systems Guide
Signals & Systems Guide
x x
The rectangular function is the result of an
ON - OFF switching operation of a constant
x x
voltage source in an electrical circuit.
Signals & Systems Signals & Systems
Hint:
The sum of harmonic signals
y(t) = x1(t) + x2(t) + x3(t) + - - - - - is periodic with
overall period
T = LCM (T1, T2, T3, ….)
• A discrete signal x(n) is periodic
if x x[n] = x[n + N] ;
where N periodic of x[n]
Hint:
04. Consider the D.T. signal These signals are sampled with a sampling 11.
t t
period of T = 0.25 seconds to obtain discrete A) A signal x(t) = 2 cos (150 t + 300) is sampled at (m) t
t t
x(n) = time signals x1[n] and x2[n], respectively. Which 200Hz. Find the fundamental period of discrete
one of the following statements is true? signal? (n) y(n) = ex(n)
Find the values of M and n0 so that (a) The energy of x1[n] is greater than the B) A periodic discrete time signal x(n) is given by
x(n) = u[Mn – n0 ] energy of x2[n]. x(n) = cos (3 n) + sin (7 n) + cos (2.5 n). The 13. Test the following systems for time - invariance?
(b) The energy of x2[n] is greater than energy of term sin (7 n) in x(n) corresponds to (a) y(t) = tx(t) + 3
05. Sketch the wave forms of the following signals? x2[n]. (a) 14th harmonic (b) 7th harmonic (b) y(t) = ex(t)
(a) x(t) = u(t+1) – 2 u(t) + u(t–1) (c) x1[n] and x2[n] have equal energies. (c) 6th harmonic (d) 5th harmonic (c) y(t) = x(t) cos3t
(b) x(t) = r(t+2) – r (t+1) – r(t–1) + r(t–2) (d) Neither x1[n] nor x2 (d) y(t) = sin{x(t)}
where r(t) is unit ramp function evaluate the average value xav, the energy E (e) y(t) = x t
t
08. Find the conjugate anti-symmetric part of (f) y(t) = x2(t)
x value xrms. (g) y(t) = x(2t)
(h) y(n) = 2x(n) x(n)
x(t) and x(t) Signal 1 x(t) Signal 2
4 (i) y(n) = x(n + 2) – x(7 – n)
06. Determine whether the following signals are its even part xe(t) for t 0 only; that is x(t) & xe(t) 1 t
energy (or) power signals? for t < 0 are not given. Complete the plots of x(t) 7 t t t t
t -5 -2 -2 2 5
t
(a) x(t) = e t u(t) & x0(t). –2 2 t
x(t) Signal 3 t t
(b) x(t) = A t
2
1
2 t
15. Check whether the following systems are (e) y(n) = x[n]x[n – 1] 24. Statement (I): A memory less system is causal
t t
causal (or) non causal? (c) t t (f) y(n) = nx(n) Statement (II): A system is causal if the output
t t t
(a) y(t) = (2t + 3) x(t) (g) y(n) = x[n] – x[n – 1] at any time depends only on values of input at
t t
(b) y(t) = x2(t) (d) t t
t t (h) that time and in the past.
(c) y(t) = x(t)sin5t
(d) y(t) = x{sin(t)} 18. Match the following
t t Causality: Stability:
h(t) = 0; t < 0
x t x
h(n) = 0; n < 0
(a) 0 (b) 1/2
(c) 3/2 (d) 1
Memory less : Invertibility & Inverse:
h(t) = 0 for t 0 h(t) hinv(t) = (t)
h(n) = 0 for n 0 h(n) hinv(n) = (n) 03. An L.T.I system is having impulse response
h(t) = u(–t–1) for which the input signal
t = 4 & t = 0.5?
Signals & Systems Signals & Systems
04. The impulse response of a continuous time 10. An Input signal x(t t 18. Two discrete -time signals x[n] and h[n] are
13. Given x t and t t
system is given by h(t) = (t 1) + (t 3). The the system with impulse response both non-zero only for n = 0,1,2, and are zero
then x t t is
value of the step response at t = 2 is t t otherwise. It is given that x[0] = 1, x[1] = 2,
(a) 10 t
(a) 0 (b) 1 x[2]=1, h[0] = 1, Let y[n] be the linear
Find the output, for the values of (b) t
(c) 2 (d) 3 convolution of x[n] and h[n]. Given that
i) T = 4 y[1] = 3 and y[2] = 4, the value of the expression
ii) T = 2 t (10y[3] + y[4]) is _______
05. t x t (c)
(b) The impulse response of a system is (c) x(- ) and h(- ) response
h(t) = t u(t). For an input u(t – 1), the output (d) x(t) and h(-t)
to be stable is _______
is (a) |a|<1, |b| < 1
07. Explain the difference between each of the (b) |a|>1, |b| < 1
following operations? (c) |a|>1, |b| > 1
(a) [e t u(t)] (t – 1) (d) |a|<1, |b| > 1
Discrete Convolution
(b) 20. Given h(t) = e t u(t) + e t u(–t). For what values of
15. A linear system with Input x(n) & output y(n)
related as and system is stable?
(c) e tu(t) (t – 1) t
impulse response is t (a) < 0, <0
where g(n) = u(n) – u(n – 4). Find y(n) when (b) < 0, >0
08. Let x(t) = u(t–3) – u(t – 5) & h(t) = e–3tu(t).
input x(t) = u(t)? x(n) = (n – 2) (c) > 0, >0
(d) > 0, <0
Find x t t
t 16. The I.R of a D.T LTI system is given by
t
12. Suppose that x t h(n)=(0.5)n u(n) of the input is
09. Signals that replicate under self-convolution Find the output at n = 1 & n = 4?
t x t
include the impulse, Sinc, Gaussian, and
y t 17. Given x = [a, b, c, d] as the Input to an LTI
Lorentzian. For each of the following known Let y(t) = x(t) h(t). Ifhas to contain
t
only three discontinuities, the value of is system produces an output
results, determine the constant A using the
(a) 1 y = [x, x, x, x, … repeated N times].
area property of convolution.
(b) 2 The impulse response of the system is
(a) ( t) ( t) = A ( t) (a)
(c) 0.5
(b) Sinc( t) Sinc( t) = ASinc( t) (a) Find I.R. of overall system.
(d) – 1
(b) u(n) – u(n – N) (b) Is this system causal ? Under
(c)
(c) u(n) – u(n–N–1) What condition the system is stable.
(d) (d)
t t t
Signals & Systems Signals & Systems
22. Consider a D.T system ‘s1’, with I.R 28. If step responses of 2 L.T.I systems are s1(t) & However, musical instruments, such as a piano,
h(n) = (1/5)nu(n) s2(t) respectively, how the cascaded step are made of many strings all vibrating at once.
(a) Find ‘A’ such that h(n) – Ah(n–1) = (n) response sc(t) is related interms of s1(t) & The question that intrigued Fourier was: How do
Historical perspective:
(b) Using result from part (a), determine the I.R s2(t)? you evaluate the waveforms from a number of
Jean Baptiste Joseph Fourier (1768 – 1830)
g(n) of an LTI system s2 which is inverse of s1 strings all vibrating at once? As a product of his
Joseph Fourier was born in Auxerre, France
research, Fourier realized that the sound heard
23. For the interconnected system shown in Fig.
Key for Practice Questions by the ear is actually the arithmetic sum of
1830.
sponse. each of the individual waveforms. This is called
Motivation:
the principle of superposition.
01. (a) Ans: 04. Ans: (b) • Representation of continuous time, periodic
signals in the frequency domain • Representing CT signals as superposition of
05. Ans: z(t) = y(t + a) complex exponentials leads to frequency –
24. Determine whether each of the following of planets and their satellites, vibration of domain characterizations. eg :- A human ear is
06. a. Ans: x(t – 2), b. Ans: x t oscillators, electric power distribution, beating sensitive to audio signals within the frequency
statements are TRUE (or) FALSE.
of the heart, vibration of vocal chords, etc. range 20Hz to 20 kHz. Typically, musical note
Justify your answer.
07. a. Ans: t , b. Ans: e , Introduction: occupies a much wider frequency range.
(1) The cascade of a non causal LTI system
c. Ans: e–(t–1). u(t–1) • The Fourier series is named after the French Therefore the human ear processor frequency
with casual one is necessarily noncausal
mathematician Joseph Fourier. components within the audible range &
(2) If an LTI system is causal, it is stable 08. Ans: In this chapter we will consider approximating rejects other frequency components. In such
(3) If h(t) is the I.R of an LTI system which is
a function by a linear combination of basis applications, frequency-domain analysis
periodic & nonzero, the system is unstable 11. b). Ans: (c) 13. Ans: (a) provides a convenient means of solving for the
functions, which are simple functions that can
(4) The inverse of a causal system is always be generated in a laboratory. Joseph Fourier response of L.T.I. systems to arbitrary input.
14. Ans: (d)
causal (1768–1830) developed the mathematical
• By using F.S, a non-sinusoidal periodic function
15. Ans: y(n) = g (n 4) theory of heat conduction using a set of
25. If the unit step response of a system is trigonometric (sine and cosine) series of the
16. Ans: y(1) = 1, y(4) = 5/8 functions.
, then its unit impulse response is form we now call Fourier series. He established
_______ 17. Ans: (a) 19. Ans: (d) 20. Ans: (b) that an arbitrary mathematical function can • Sinusoidal signals arise in describing motion of
(a) be represented by its Fourier series. planets & periodic behavior of earth’s climate.
21. Ans: u u is causal
(b) • Fourier series and the Fourier transform are A.C. sources generate sinusoidal voltages &
for any value of , and stable if | | < 1,
(c) basics to mathematics and science, especially currents.
and any value of .
(d) to the theory of communications. For example,
• There are 2 reasons for evaluating the F.S.
22. Ans: 23. Ans: a phoneme in a speech signal is smooth and
1. To obtain an expression for f(t) that applies
26. Find the step response of the system if the wavy. A linear combination of a few sinusoidal
everywhere, rather than only over a single
n 24. Ans: 1-false, 2-false, 3-true, 4-false. functions would approximate a segment of
impulse response is h(n) = (0.5) u(n)
period.
speech within some error tolerance.
25. Ans: (a) 2. To obtain phasors, which indirectly tell how
27. An LTI system with Input u(n) produces the • Fourier and a number of his contemporaries
much power is available at each harmonic
output as were interested in the study of vibrating
26. Ans: u of the waveform.
strings. In the simple case of just one naturally
the input nu(n)?
vibrating string the analysis is straightforward:
27. Ans: u(n 1)
the vibration is described by a sine wave.
Signals & Systems Signals & Systems
3.1 ANALOGY BETWEEN VECTORS & SIGNALS Length of the component Examples (b) For orthonormality 2T = 1 T = 1/2
• Signals are not just like vectors. = (c) x(t) = C1 (t)
1
(t) + C2 2
(t) + C3 3
A vector can be represented as a sum of Example 01:
its components, depending on the choice x t t t
For the three continuous functions shown in
of coordinate system. A signal can also be
represented as a sum of its components.
• We know that an arbitrary M-dimensional • 2 vectors are orthogonal if inner (or)
vector can be represented in terms of scalar product
M orthogonal co-ordinates.
• If we consider 2 basis vectors t t
Component of a Signal
If we approximate f by CX,
t t t t t t t t t
Error in the approximation e = f – CX
Minimize the error vector, such that f and X
are approximated
Signals & Systems Signals & Systems
-2 0 - 0 0 0 2 n 0
half. Therefore, any function that is even-harmonic is actually a regular periodic function whose period 0
has been labeled twice what it should be. In other words, there is nothing special about even-harmonic
t
functions. t Where t t
• Shifting a signal left/right in time does not affect whether or not it is odd-harmonic.
• Shifting a signal up/down (adding a DC offset) does not affect whether it is odd-harmonic, other than
adding a term in the Fourier series at Zero frequency. Convergence Of F.S: (Dirichlet Conditions)
• An odd-harmonic function does not have to be odd. (1) x(t) is absolutely integrable i.e.,
x t t
(2) x(t
t t minima
even
odd (3) The number of discontinuities in x(t) must be
t t
3.4 Properties of F.S
even and odd - harmonic 1.
odd-harmonic Linearity: x1(t) Cn
x2(t) dn
then x1(t) + x2(t) Cn + dn
t t t
2. Time Shift: x t t
odd and odd - harmonic even and odd - harmonic w/ DC offset
When we shift in the time-domain, it changes
the phase of each harmonic in proportion to its
frequency n 0.
3. Frequency Shift:
x(t) Cn
t
then x t
4. Time-Scaling:
x t x t
Time-Compressing by changes frequency
from 0 to 0
Signals & Systems Signals & Systems
12. The rms value of the periodic waveform shown Exponential F.S 19. Let x(t) be a periodic signal with period T and
n
16. For the periodic signal
t t Let y(t) = x(t–t0) + x(t + t0
x t
y(t) is dn.
? If dn= 0 odd n then t0 can be
(a) T/8 (b) T/4
17. Obtain the E.F.S. representation of periodic
? (c)T/2 (d) 2T
a peak voltage of 10 V. Its average value and 23. The magnitude & phase spectra of a periodic
the peak value of the fundamental component signal x(t
are respectively given by :
____
Signals & Systems Signals & Systems
(b) For the periodic signal x(t) shown below with 28. The systems S1 & S2
following properties
Key for Practice Questions
period T = 8 s, the power in the 10th harmonic is
sin t 5 cos ( t + 30 )
S1
01. Ans: 0
(1/4) sin10t cos (5t)
S2
02. Ans: (d)
2. Increasing exponential
4.1 Introduction 1
• Fourier Transform (F.T.) provides a frequency domain description of time domain signals and is extension 0 t
of F.S to non-periodic signals.
• CTFT expresses signals as linear combination of complex Sinusoids
• Transformation makes the analysis of signal much easier because certain features which may be 3. C.T. impulse function: (t) 1
obscure in one form may be obvious in other form
1
• Spectrum of F.T. is continuous whereas spectrum of F.S is discrete.
• F.T (or) spectrum of a signal x(t) is t
0
x t t ........(1) 0
• x t t x t t t
x t
| X( ) |
AT
Convergence of F.T.
x(t)
A
x t t
can be considered to have F.T. if impulse functions are permitted in the transform.
3. x(t T/2 0 T/2 t 0
A A
interval.
1/a |X( )|
1 /2 0
/4
-
-a 0 a
0 t
- /4
- /2 X( )
Signals & Systems Signals & Systems
Spectral Width
Operational Properties of the Fourier Transform
• A broadband signal is the one, which spectrum is distributed over a wide range of frequencies as it is
Property x(t) X(f)
shown in Fig. 4(a)
• A bandlimited signal is limited in the frequency domain with some maximum frequency as it is shown in (1). Similarity X(t) x( f) x( )
Fig. 4(b).
(2). Time Scaling x( t)
• A narrowband signal has a spectrum that is localized about a frequency f0 that is illustrated in Fig. 4(c).
• A baseband signal has a spectral contents in a narrow range close to zero (Fig. 4(d)). Accordingly, a (3). Folding x( t) X( f) X( )
spectrum beginning at 0 Hz and extending contiguously over an increasing frequency range is called a
baseband spectrum. (4). Time Shift x(t ) e X(f) e X( )
(c) j2 t
(d) (5). Frequency shift e x(t) X(f ) X( )
(a)
(6). Convolution x(t) h(t) X(f) H(f) X( ) H( )
(11). Integration x t t
t t
t t t
Signals & Systems Signals & Systems
Some Useful Fourier Transform Pairs 4.2 DISTORTIONLESS TRANSMISSION • If the gain is not constant over the required
In several application such as signal frequency range, we have amplitude distortion.
Entry x(t) X(f) X( )
If the phase shift is not linear with frequency, we
1 (t) 1 1
have phase distortion as the signal undergoes
over communication channel, we require that
2 rect(t) sinc(f) different delays for different frequencies.
the output waveform be a replica of the Input
waveform.
3 tri(t) sinc2(f)
physically unrealizable in the sense that their
• Transmission is said to be distortionless if the characteristics can not be achieved with a
4 Sinc(t) rect(f)
input and output have identical waveshapes
9 e
k tp( ) is the delay occurring at a
Single frequency. As the signal propagates
10
from source to destination the amount of delay
caused is known as phase delay.
11 sgn(t)
|H( )|= k
t
12 u(t)
H( ) = ( ) = – t0
t
13 e cos(2 t) u(t) delay. A steady Sinusoidal signal doesn’t carry
information. Information can be transmitted
only by applying some appropriate change to
t
14 e sin(2 t) u(t)
Sinusoidal wave. Suppose that a slowly varying
signal is multiplied by a Sinusoidal wave so that
resulting modulated wave consists of a narrow
15 t Slope = –t0
group of frequencies. When this modulated
wave is transmitted through the channel, we
16 • For distortion less transmission, magnitude
x t
response must be a constant, phase response of Input and received signal. This is known as
must be a linear function of with slope –t0, envelope (or) group delay (True signal delay)
where t0 is delay in output with respective to
input. t
Signals & Systems Signals & Systems
Properties of H.T:
t 1. H.T. doesn’t change the domain of a signal
t t t t 2. H.T. doesn’t alter the amplitude spectrum of a
signal
1.5v 1.5v tp(f) = tg(f) for
3. If x t is H.T. of x(t), then H.T. of x t is – x(t)
4. x(t) and x t are orthogonal to each other.
0v 1.5v
1.5v 1.5v Example : Find H.T. of
Low pass
(1) x(t) = cos 0t
(2) x(t) = sin t
0
0v 1.5v
1.5v 1.5v (3) x t t t
High pass
4.3 HILBERT TRANSFORM: (H.T) (4) x(t)cos(2 fct), where x(t) represents a
0v 1.5v The Hilbert transform is an operation that shifts signal band limited to B, fc > B
1.5v 1.5v
Band pass the phase of x(t) by – /2, while the amplitude Example :
spectrum of the signal remains unaltered. x(t) = cost and
0v 1.5v h(t) = (1/ ), then the output y(t) is
1.5v 1.5v An ideal H.T. is an all pass 900 phase shifter.
Band reject
H.T. is used in number of application such as
analyzer; shown at the right are the waveforms that we would observe with an oscilloscope. The
spectrum and waveform at the top pertain to the input signal, and those below pertain, respectively,
to the low-pass, high-pass. Band-pass, and band-reject outputs. For instance, if we send Vi(t) through a 4.4 CORRELATION
Frequency response of the H.T.
c
somewhere between 4 0 and 16 0
= –jsgn( ) • It provides a measure of the similarity between
and thus passed, but the third component is multiplied by 0 and is thus blocked: the result is
2 waveforms as the function of search
parameter.
• An application of correlation to signal detection
Example : in a radar, where a signal pulse is transmitted in
A channel has the frequency response order to detect a suspect target. If a target is
• Auto correlation function of an energy signal 06. The magnitude of F.T. X( ) of a function 11. The F.T of a triangular pulse f(t
Practice Questions
x(t) is ) of other is F( ) =
function y(t ?
x t x t t x t x t t X( ) and Y( ) are zero for all . The magnitude
01. If x(t) is a voltage waveform, then what are the f(t)
units of X(f) ? and frequency units are identical in both the 1
• ACF of power signals is
y(t) can be expressed in
02. For the signal x(t terms of x(t) as ______
(a) X(0)
(b) 3 -1 0 t
1
f1(t) f2(t)
1
Properties of ACF: 1.5
–100 0 100 -50 0 50
1. ACF is an even function of i.e.,
Rx( ) = Rx( )
t
2. ACF at origin indicates either energy (or) power (a) x (b) x t
-1/2 0 1/2 t 0 2 t
in the signal
t
3. Maximum value of ACF occurs at origin i.e., (c) x t (d) x 12. If x(t) as shown in Fig. (a) has Fourier transform
|Rx( )| |Rx(0)| X(f), then the Fourier transform of g(t) as shown
4. Rx( ) = x( ) x( ) 07. Find F.T. of the following signals: in Fig.(b) is
(i) x(t) = 1
Rx( ) Sx( Properties of F.T
(ii) x t
6. For an LTI system t
Linearity, duality & scaling (iii) x t
Y( ) =X( )H( ) t
|Y( )|2 = |X( )|2 |H( )|2 (iv) x t
t
SY( ) = SX( ) |H( )|2 03. Find the F.T of the signals
i) x t [Two sided exponential]
08. Let x(t) = rect( t – ½ )
ii) x(t) = Sgn(t) [Signum function]
{where rect(x) = 1 for –1/2 x 1/2 } then if
x
2
sinc(x) = x
output S.D = [input S.D] [|H( )| ] 04. The F.T. of a function g(t) is given as
F.T. of x(t) + x(–t) will be given by ____
. Find g(t)?
14. Find the F.T. of y(t) = Sinc(t) cos10 t The output, y(t) is equal to 23. Given t t
25. The input signal x t t t 1
< 2
t is applied to the system shown in
15. i) The inverse F.T of X(4 +3) in terms of x(t) x t
is_______
ii) I.F.T of X( ) = 2 ( )+ ( 4 ) Find the F. T of the following signals?
H( )
+ ( + 4 ) is 1
(a) 1+ cos 4 t (b) (1– cos 4 t)
(c) 2 (1– cos 4 t) (d) 2 (1+cos 4 t)
(a) x(t) (b) x t
16. The Fourier spectrum X(f) of a signal x(t) is (c) – x(t) (d) x(t) cos( ) - f 0 f
-3 -1 1 3 f
0 28. (a) Find the output of a system having impulse
(c) response t t when the
4 21. Find the F.T. of te input applied is x(t) = cos t
t
of
f t
-3 3 (b) Let t , and h(t
0
22. Given x(t) X( ), express the F.T. of the following to g(t). If g(t) applied as input to h(t), then
(d) None of these
signals in terms of X( ) ? the Fourier transform of the output is
Convolution (a) e (b) e
18. A signal x(t) with bandwidth B is put on a carrier (i) x1(t) = x(2 – t) + x(– t – 2 )
(ii) x2(t) = x( 3t – 6 ) (c) e (d) e
cos( ct) with c >> B. The modulated signal x(t)
cos( c
t) is then applied to a system shown in 24. An L.T.I system is having Impulse response
(iii) x t x t
t
t
t t for which the input applied is
t x t
(iv) x t
t x(t) = cos2t + sin 6t ?
Signals & Systems Signals & Systems
29. The time function x(t) corresponding to the 39. For a linear phase channel, what is tp & tg? 43. Consider an LTI system with magnitude
36. For the signal t
Fourier spectrum: t , determine the response
essential B.W. B Hz of g(t) such that the energy
is
contained in the spectral components of g(t) of with R = 1 k
(a) real and even symmetric
frequencies below B Hz is 99% of signal energy
(b) conjugate odd symmetric i) Let H(f) denote the frequency response of RC and phase response Arg[H(f)] = –2f.
in g(t)?
(c) conjugate even symmetric be the highest frequency component If the input to the system is
1
(d) real and odd symmetric 37. The magnitude of the Fourier transform of a
1
,
signal x(t Then the average power of the output signal
then f1 (in Hz) is _____
30. Find the F.T. of The energy of the signal is y(t) is ______
(a) y(t) = x(t)cos t? (a) 327.8 (b) 163.9
0 |X(f)|
t 10-6 (c) 52.2 (d) 104.4
t
(b) x t
t (ii) Let tg
f2 = 100 Hz, then tg(f2 ) in msec, is ____
following steady - state outputs. Tell what kind
t -104 104 f(Hz) (a) 0.717 (b) 7.17
31. Find the F.T. of t 0 of distortion has occurred?
t
(c) 71.7 (d) 4.505
(a) (b)
Parseval’s Power theorem 10
41. The input to a channel is a bandpass signal. It
(c) (d) 10
is obtained by linearly modulating a sinusoidal
32. Find the energy in the signal Distortionless Transmission carrier with a single-tone signal. The output of
t
x t t the channel due to this input is given by
38. Consider a transmission system H( ) with
magnitude and phase response as shown y t t t
?
The group delay (tg) & the phase delay (tp) in
X( )
x(t) = 2Cos 10 t + Sin 26 t is given to the system seconds, of the channel are
the output will be ________ 6
(a) tg = 10 6, tp = 1.56 (b) tg = 1.56, tp = 10
8 6 8
(c) tg = 10 , tp = 1.56×10 (d) tg = 10 , tp = 1.56
Correlation Sampling Theorem 55. A signal represented by x(t) = 5cos(400 t) 57. A signal x(t) = 6cos10 t is sampled at a rate
is sampled at a rate of 300Hz. The resulting of 14 Hz to recover the original signal, cut-off
45. Find the auto correlation and power in the
51. Find the Nyquist rate & Nyquist interval for each
signal x(t) = 6 cos(6 t + /3)
of the following signals? with cut-off frequency of 150 Hz. Which of the (a) 5 < fc < 9 (b) 9
3t t ? (c) 10 (d) 14
46. Find the ACF of x(t) = e u(t)
(a) x t t (a) 100 Hz
t
and input (b) x t t (b) 100 Hz, 150 Hz 58. The spectrum of a bandlimited signal after
2t
x(t) = e u(t) (c) 50 Hz, 100Hz
(c) x3(t) = 5cos1000 t cos4000 t
(a) Find the ESD of the output? (d) 20,100,150Hz sampling interval is
(b) Show that total energy in the output is (d) x
one-third of the input energy? 56. The frequency spectrum of a signal is shown in
(e) x t t t
48. Let g(t) = exp(–8t)u(t) t of 1 msec, then the frequency spectrum of the
x(t) = convolution of g(t) with it self and 100 f(Hz)
(f) x t t 0 100 150 350 400 600
t sampled signal will be _________
y( ) = correlation of g(t) with it self
U (f) (a) 1msec (b) 2msec
i) The value of x(t) at t = 1/16 is
(c) 4msec (d) 8msec
(a) (b) 52. Let x(t) be a signal with Nyquist rate .
o
Determine the Nyquist rate for each of the
(c) (d) 1/(16e) 59. Let x(t) = 2 cos(800 t) + cos(1400 t) and x(t) is
following signals. f (kHz)
–1 0 1 sampled with the rectangular pulse train shown
ii) The energy density spectrum at f = 0 is (a) x(t) + x(t–1) (b) x t
t
(a) 8 (b) 1/8 (c) x(3t) (d) x(t) cos o
t (a) present in the sampled signal in the frequency
(c) 1/64 (d) 64 range 2.5kHz to 3.5kHz.
53. Two signals x1(t) & x2(t) are band limited to 2 kHz
iii) y( ) value at = 0, is f
(a) (b) following signals?
(a) x1(2t) (b)
(c) (d) 16
(b) x2(t – 3)
(c) x1(t) + x2(t)
49. Find the C.C.F of x(t) = e tu(t) and
3t
(d) x1(t)x2(t)
y(t) = e u(t) ? f
(e) x1(t) x2(t)
(f) x1(t) cos(1000 t) (c)
50. The signal x(t) = sinc10t is the input to a system
with frequency response
54. A signal x(t) = 100 cos(24 103t) is ideally
sampled with a sampling period of 50 sec (a) 2.7, 3.4
Find the output energy? (b) 3.3, 3.6
and then passed through an ideal low pass
f (c) 2.6, 2.7, 3.3, 3.4, 3.6
the following frequencies is/are present at the (d) 2.7, 3.3
(d)
?
(a) 12 kHz only
(b) 8 kHz only
(c) 12 kHz and 9 kHz
(d) 12 kHz and 8 kHz
f
Signals & Systems Signals & Systems
60. Hilbert Transform & Complex Envelope 65. Statement (I): Aliasing occurs when the
Key for Practice Questions
(i) The output y(t) of the following system is to be sampling frequency is less than twice the
sampled, so as to reconstruct it from its samples 61. The complex envelope of the bandpass signal maximum frequency in the signal.
01. Ans: Volt/Hz
uniquely. The required minimum sampling Statement (II): Aliasing is a reversible process.
t
x t t ,
rate is t 02. (a). Ans: 7, (b). Ans: 4
66. Statement (I): Sampling in one domain makes
centered about f = Hz, is
the signal to be periodic in the other domain.
04. Ans: t t
t Statement (II): Multiplication in one domain is
(a)
t the convolution in the other domain. 05. Ans: 0 06. Ans: (d) 12.Ans: (a)
t
(b)
t t
15. i. Ans: x ii. Ans: (a)
t
(c)
t 17. Ans: (b)
t
(a) 1000 samples/s (d) 18. Ans: (d)
t
(b) 1500 samples/s
(c) 2000 samples/s 62. A modulated signal is given by 24. Ans: cos(2t)
(d) 3000 samples/s s(t) = e atcos[( c )t]u(t), where a, c are
t
positive constants, and c . The complex 25. a). Ans: t ,
envelope of s(t) is given by
t t t
(ii) The signal is ideally sampled (a) exp( at) exp[j( c )t]u(t) b). Ans: t t
at a sampling frequency of 15 Hz. The (b) exp( at t)u(t)
t)u(t) c). Output = Input
with impulse response t .
t t (d) exp[j( c
)t]
26. Ans: a is non - periodic, b is periodic, c may
be periodic.
63. The input 4sinc(2t) is fed to a Hilbert transformer
(a) t to obtain y(t
27. a). Ans: y t t,
t
(b) t
t b). Ans: y t
(c) t t
c). Ans:
t
(d) t Here The value (accurate to
t d). Ans:0
two decimal places) of is __________
28. (a). Ans: cos (t 1), (b). Ans: d
36. Ans:
51. Ans: -a
a) 200 Hz, b) 400 Hz, c) 5 KHz, • The primary role of the L.T in engineering is
0 t 5.2 Properties of L.T
d) 2a e) 120 Hz. the transient & stability analysis of Causal L.T.I
systems.
1. Linearity:
52. Ans: 2. x
a). , b). , c). 3 , d). 3 , • L.T provides a broader characterization of If x1(t) X1(s) with ROC = R1
0 0 0 0
j
systems & their interaction with signals than is x2(t) X2(s) with ROC = R2
possible with F.T. Then a x1(t) + b x2(t) aX1(s) + bX2(s)
0
55. Ans: (a)
t with ROC = R1 R2
• In addition to its simplicity, many design
56. Ans: (b) -1 -a
2. Time-shifting:
have been developed in L.T. domain.
57. Ans: (a) x(t) X(s), ROC = R
x(t) = est then x t t
58. Ans: (c) 3. x
(where s = +j then the output is y(t) = estH(s) with ROC = R
60. (ii) Ans: (a) where H(s) is transfer function of the system.
j
3. Shift in S-domain
• L.T. of a general signal x(t) is 1
x(t) X(s) with ROC = R
a
x t t --------- (1) then x t
0 t
= F.T {x(t)e t} with ROC = R + Re(s0)
t 4. x
• e may be decaying (or) growing depending 4. Time-reversal:
on whether ‘ ’ is +ve (or) –ve. j x(t) X(s)
x t
then x(–t) X(–s), ROC = – R
0
5.1 Region of Convergence (R.O.C) of L.T: t 5. Differentiation in time:
The range of values of ‘S’ for which eq(1) is a
x(t) X(s) with ROC = R
|x(t)e dt < , is known as x t
then with ROC = R
R.O.C of L.T. t
Example: (b) Find the values of ‘A’ &‘t0’ such that the
For an L.T.I system described by the transfer L.T. G(s) of g(t) = Ae 5tu( t t0) has same 09. Let g(t) = x(t) + x(–t) where
function algebraic form as X(s). What is the R.O.C
x(t) = e–tu(t) and
due to corresponding to G(s)?
0
(a) 5cos(2t+30 )
(b) 10sin(2t+450) Find and ?
Signals & Systems Signals & Systems
16. Consider an LTI system with impulse response 23. Consider a system with transfer function
5t
h(t) = e u(t). If the output of the system is . Find the steady-state
y(t) = e t u(t) e 5t u(t) then the input, x(t), is given response when the input applied is 8cos2t?
by
3t 3t
(a) e u(t) (b) 2e u(t)
5t 5t
(c) e u(t) (d) 2e u(t)
Signals & Systems Signals & Systems
Causality & Stability 31. The Laplace transform of a causal signal 23. Ans: y(t) = 2.8288 cos(2t + 45 )
. The value of the signal y(t) at
27. Consider an LTI system for which we are given t = 0.1 sec is _______ 24. Ans: (a) • The DTFT describes the spectrum of discrete
the following information
signals & formalizes that discrete signals have
and x(t) = 0, t > 0 and output is 27. Ans:
periodic spectra. The frequency range for a
(a) > 1
Key for Practice Questions discrete signal is unique over ( – , + ) (or)(0, 2 )
y (b) t
x
(a) Find T.F & R.O.C.? 01. (1)
28. Ans:
(b) Find the output if the input is , it can’t be both stable x
(2) =
x(t) = e3t t using part (a)? and causal.
(3) No ROC, No L.T
• X(ej ) is decomposition of x(n) into its frequency
(4) No ROC, No L.T
28. Consider an LTI system with input x(t) and components.
(5) No ROC, No L.T
output y(t) related as
t t t
t
t
t
t 02. Ans: Re[ ] = 3, Img[ ] any value 6.1 Convergence of DTFT
t
Find the T.F. of inverse system. Does a stable &
Causal inverse system exist? 04. Ans: y
x
29. Which one of the following statements is NOT 05. Ans: > 5
TRUE for a continuous time causal and stable A = 1, to = 1 • Some sequences are not absolutely summable,
LTI system? but they are square summable.
(a) All the poles of the system must lie on the 06. Ans:
• There are signals that are neither absolutely
left side of the j axis.
(b) Zeros of the system can lie anywhere in 07. Ans: (a) 08. Ans: (c)
DTFT.
the s-plane.
(c) All the poles must lie within |s| = 1. 09. Ans:
anu(n), |a| < 1
(d) All the roots of the characteristic equation
must be located on the left side of the j 10. Ans:
(n) 1
axis.
Periodicity property:
30. The output of a causal all-pass system is
+2 )
y(t) = e 2tu(t), for which the system function is 12. Ans: y(t) = (t) X(e j( ) = X(ej )
6.2 Oversampling and Sampling Rate Conversion the same signal sampled at NS Hz extends only to This describes Yp(F) as a scaled (compressed)
Practice Questions
B/NS Hz, and the spectrum of the signal sampled version of the periodic spectrum Xp(F) and leads to
designed to operate at different sampling rates at S/M Hz extends farther out to BM/S Hz. After an N-folds spectrum replication. The spectrum of the
because of the advantages it offers. For example, 01. Determine whether or not the DT systems with
interpolated signal shows N compressed images in
oversampling the analog signal prior to digital rate changes are typically made by manipulating these frequency response are causal?
the principal period |F| 0.5 with the central image
processing can reduce the errors (Sinc distortion) the signal samples (and not re-sampling the analog
occupying the frequency range |F| 0.5/N. This is (a)
caused by zero-order-hold sampling devices. Since signal). The key to the process lies in interpolation
and decimation. exactly analogous to the Fourier series result where
operations in one sampling interval, oversampling spectrum zero interpolation produced replication
(b)
can impose an added computational burden. Zero Interpolation and Spectrum Compression: (compression). The spectrum of the interpolated
It is for this reason that the sampling rate is often A property of the DTFT that forms the basis for signal signal y(n) shows N compressed images per period
(c) H( ) = e –j3 + e +j2
reduced (by decimation) before performing interpolation and sampling rate conversion is that centered at
M fold zero interpolation of a discrete-time signal
x 02. (a) Let x(n) = (1/2)n u(n), y(n) = x2(n) and
before reconstruction. Oversampling prior to signal x[n] leads to an M-fold spectrum compression and
Y(e j ) be the F.T of y(n). Then Y(ej0) is
reconstruction allows us to relax the stringent replication, as illustrated in Figure (ii). (b) Given X(ej ) = cos3(3
Figure (i) shows the spectrum of sampled relation. If the spectrum of the original signal x[n] x
The zero-interpolated signal y[n] = x[n/N] is
signals obtained from a band-limited analog signal nonzero only if n = kN, k = 0, 2 ......(i.e., if n is an extends to |F| 0.5/M in the central period, the (c) What is the d.c & high - frequency gain of
whose spectrum is X(f) at a sampling rate S, a higher integer multiple of N). The DTFT Yp(F) of y[n] = y[kN] spectrum of the decimated signal extends over
rate NS, and a lower rate S/M. may be expressed as |F| 0.5, and there is no aliasing or overlap
between the spectral images. 03. (i) Find the inverse DTFT of
X(f) Sampling rate =S X(ej ) = 1 + 2 cos + 3 cos2
SX(f)
y y
f f (ii) The DTFT of x(n) = 2 [n+3] 3 [n 3],
B B S can be expressed in the form
Sampling rate =NS Sampling rate =S/M = x X(ej ) = asin(b ) + cejd Find a, b, c & d.
NSX(f) SX(f)M
(b) A discrete system with input x(n) & output Convolution 16. An input x(n) with length 3 is applied to a LTI 21. Let h[n] be the impulse response of a discrete-
y(n) are related as system having an impulse response h(n) of
y(n) = x(n) + ( 1)n x(n) length 5, and Y( ) is the DTFT of the output y(n) response is given by ; ;
10. Consider x
If the input spectrum X(ej ) is shown below of the system. If |h(n)| L & |x(n)| B n, the ; and h[n] = 0 for n < 0 and n > 2.
Find the output if the impulse response is
maximum value of Y(0) can be …. Let H ( ) be the Discrete-Time Fourier transform
the o/p spectrum values at =0& =
(a) 15 LB (b) 12 LB (DTFT) of h[n], where is the normalized angular
are
(c) 8 LB (d) 7 LB frequency in radians. Given that H( 0)= 0 and
11. An L.T.I system is having impulse response 0< 0
< , the value of 0
(in radians) is equal to
X(ej )
1 ______.
equation y(n) = x(n) + 2x(n 1) + x(n 2)
(a) Obtain the magnitude & phase response? 22. Find the energy in the signal
(b) Find the o/p when the input is x
Find the output when input applied is
/2 0 /2 x ?
jn /4
x(n) = e ?
18. The impulse response of a causal linear phase 23. Find the value of
Scaling & Frequency Differentiation 12. For each of the following pair of signals
determine whether or not the system is LTI if
by h(0) = 2, h(1)= 3, h(2) = 0, h(3) = 3 and
06. Find the I.F.T of ?
h(4)= k. The value of k and slope of the phase
u u curve are respectively quantities without calculating DTFT?
(a) 2, –2 (b) –2, –2
07. Given the signal x .
(c) 0, 2 (d) –3, 2
Then Fourier transform of x[2n] is
(a) 3 + 2cos 2 + 4 cos 19. The frequency response of a discrete LTI system
j /4
(b) 3 + 2cos is H(ej ) = e , < .
(c) 3 + 2cos2 and the amplitude response Then the output of the system due to the input,
(d) 3 + 2cos + 2cos( /2) blocks the frequency f = 1/3 & passes the x is
frequency f = 1/8 with unity gain. What is the
? (a)
08. The spectrum of signal x[n] is X(F) = 2tri(5F).
Sketch X(F) and the spectra of the following
14. Consider the system described by the (b)
signals and explain how they are related to
equation y(n) = ay(n–1) + bx(n) + x(n–1), where (a) X(ej0) (b) X(ej )
X(F). (c)
i. y[n] = x[n/2] (c) (d)
‘a’ & ‘b’ such that
ii. d[n] = x[2n]
(d)
|H(ej )| = 1 ?
x
20. A signal x (e)
x(n) = 1, for n = 1, 3 & 5
09. Find the F.T of x u y(n) = x(n) + x(n–1)+ x(n–2). Then the values of = 2, for n = 0 & 4
& , such that the input x(n) = 1 + 4 cos (n ) = 0, elsewhere (f)
results in the output y(n) = 4, are Its phase spectrum at = 0.25 is
(a) = 2, =1 (b) = 1, =2 (a) 0.25 (b) – 0.5
(g)
(c) = =1 (d) = 1, = 2 (c) 0.5 (d)
Signals & Systems Signals & Systems
25. Given h(n) = [1, 2, 2], f(n) is obtained by Z.T. of standard Signals:
Key for Practice Questions
convolving h(n) with itself and g(n) by
u
correlating h(n) with itself. Which one of the
01. (a). Ans: Non causal 7.1 INTRODUCTION TO Z-TRANSFORM
following statements is TRUE?
(b). Ans: causal
(a) f(n) is causal and its maximum value is 9 • Discrete-time counterpart of L.T. is Z.T.
(c). Ans: Non causal
(b) f(n) is non-causal • For a D.T.L.T.I system with impulse response h(n),
(c) g(n) is causal and its maximum value is 9 the response y(n) of the system to a complex
02. (a). Ans: (b).Ans: 1
(d) g(n) is non causal and maximum value is 9 exponential Input of the form zn is y(n) = znH(z) n
(c). Ans: DC gain = 10, HF gain = 2 0 1 2 3 4
where H(z) is known as transfer function of the
26. A continuous time signal x(t system.
remove frequency component in the range 04. Ans: y Z.T. of a general D.T. signal x(n) is
5kHz f 10 kHz. The maximum frequency
------------ (1)
present in x(t) is 20 kHz. Find the minimum
(b). Ans: 1 where z = rej Complex Variable
23. Ans:
(n) 1; ROC : entire z-plane
25. Ans: (d) • The primary role of Z.T. in engineering are the 1
u(n) 1/1 z ; |z| > 1
26. Ans: fS = 40 kHz, study of system characteristics & the derivation
of computational structures for implementing
27. Ans: (a)
discrete systems on computers.
Signals & Systems Signals & Systems
35. Consider the following statements regarding a Realization Structures 42. In the IIR
linear discrete - time system For which of the following cases, the system will (a) Band pass (b) All pass
39. A DF having T.F. using transit from stable to unstable condition? (c) Low pass (d) High pass
DFI and DFII realizations of IIR the number
1. The system is stable. of delay units required in DFI and DFII are,
the range of k for system stability is ____
2. The initial value h(0) of the impulse respectively …….
response is 4. (a) 6 & 6 (b) 6 & 3 x(n) + y(n)
+ Z-1+ Z-2
3. The steady-state output is zero for a (c) 3 & 3 (d) 3 & 2 +
sinusoidal discrete time input of frequency
equal to one-fourth the sampling 40. 2 systems H1(z) & H2(z) are connected in k
frequency. cascaded as shown below. The overall output
(a) 0.1 < a < 0.5 (a) (b)
y(n) is the same as the input x(n) with a one
Which of these statements are correct? (b) 0.5 < a < 1.5
unit delay. The transfer function of the second
(a) 1, 2 and 3 (b) 1 and 2 (c) 1.5 < a < 2.5 (c) (d)
system H2(z) is
(c) 1 and 3 (d) 2 and 3 (d) 2 < a <
37. Statement (I): The system function values of a0, a1 & a2 are respectively _________ Find b, c & d such that SFG is a realisation
of H(z)
is not causal. Input Output
Statement (II): If the numerator of H(z) is of a0 1 Common Data for Questions 44 & 45
z
lower order than the denominator, the system a1 1
may be causal. z -1
a2
38. Statement (I): 3 z -1
(a) 1.0, 0.7 and 0.13 X Y
Z-transform approach is used to analyze the
(b) 0.13, 0.7 and 1.0
discrete time systems and is also called as pulse
(c) 1.0, 0.7 and 0.13
transfer function approach.
(d) 0.13, 0.7 and 1.0
Statement (II):
-1
The sampled signal is assumed to be a train of
impulses whose strengths, or areas, are equal
to the continuous time signal at the sampling 44. The transfer function
instants. (a) (b)
(c) (d)
Signals & Systems Signals & Systems
map into the inside of the unit circle in the Where T represents the sampling interval and
20. (a). Ans: (b). Ans: y
y(n) y(nT). The analog differentiator with
output dy(t)/dt has the system function
21. Ans: (a)
H(s) = s, while the digital system that produces
y y
23. Ans: 0 24. Ans: the output has the system function
Low-pass Analog s
transformation mapping
.
29. (a). Ans: 0.5 <|z|< 2 analog Analog Digital
prototype HP(s) Filter H(s) filter H( )
(b). Ans: No Therefore ……. (1)
C = 1 rad/s
(c). |z|< 0.5,|z|>3, ,
th
For k derivative …… (2)
Let us consider the mapping of points from the 8.3 IIR Filter Design by Bilinear Transformation 2. Frequency warping does not distort Example:
s-plane to the z-plane implied by the relation
The bilinear transformation is a conformal mapping
z = esT “rearrangement” of equal values will preserve a 3-dB bandwidth of 0.2 , using the bilinear
that transforms the j -axis into the unit circle in
If we substitute s = +j and express the complex the z-plane only once, thus avoiding aliasing of
variable z in polar form as z = rej frequency components. Furthermore, all points bands are mapped into equiripple bands.
rej = e T ej T
Where c
is the 3-dB bandwidth of the analog
clearly, we must have
magnitude response. Thus, a continuous-
r=eT mapped into corresponding points outside the unit
time differentiator cannot be converted Sol:
= T circle in the z-plane.
to a discrete-time one using the bilinear at c
= 0.2 . In the frequency domain of the
Consequently, < 0 implies that 0 < r < 1 and …………… (1) c
= 0.2 corresponds to
transformation.
> 0 implies that r > 1. When = 0, we have
From equation (1) we note that if
To avoid warping we are using pre-warping that is
r < 1, then < 0,
the unit circle in z. r > 1, then > 0,
However, the mapping of the j -axis into the unit When r = 1, then = 0,
circle is not one-to-one. Since is unique over the point s = into the point z = –1.
………………..(2) domain. By applying bilinear transformation
range (– , ) the mapping = T implies that the We conclude form the previous discussion that the
interval – /T /T maps into the corresponding Which show that we will get .
values of – . Even the frequency interval with piecewise-constant magnitude responses,
=0 =0
T 3 /T also maps into the interval – . …….. (3)
Thus the mapping from analog frequency to the Types of Analog Filters:
– –
frequency variable in the digital domain is many
Example:
2. chebyshev
The non-linear relationship between and
sampling. 3. Elliptical (cauer)
in eq (2) is known as frequency warping. The
j j
bilinear transformation converts H(j ) to H(e ) by
8.4 Butterworth Filter
compressing the continuous time frequency axis Into a digital IIR
according to eq (3).
z - plane s-plane characterized by the magnitude-squared
resonant frequency of r
= /2
z = esT frequency response
Sol:
………… (1)
= 4. This frequency
is to be mapped into = /2 by selecting the
c
is its –3 dB frequency.
value of T.
0 Eq (1) can be adjusted as
– …….. (2)
Unit -
circle
– The poles of H(s) H(–s) occur on a circle of radius c
1. We note that for less than about 0.3 , the
at equally spaced points.
relation between and is approximately
Due to the presence of aliasing, the impulse From eq (2) poles can be obtained by using
linear (recall that tan for small ). Thus, any
invariance method is appropriate for the design of shape of magnitude response in this range is , k = 0, 1, ….., n –1
preserved.
Signals & Systems Signals & Systems
From eq (1) we can verify Few observations can be made from the above
|H( )|2 |H( )|2
1. At = , for all n function:
1 i. For | | 1, |H (j )|2 varies between 1 and
1 1
2. From eq (1) it is clear that |H( )|2 decreases
2 . This is the pass-band description of
1
monotonically as increases.
Poles of Poles of |H(j )|. i.e.,
|H( )|2 H(s) H(–s)
Ideal characteristic
N=4 and, ripple in pass-band =
1.0 p p
ii. At = 1, always. This can be easily
n odd type I n even type I
n =18 = 1 eq (2) can
n =14 be as follows:
0.5 n =10 The chebyshev approximation is implemented
Poles of with the help of chebyshev polynomials. They are
n =6 Poles of
n =2 H(–s) or
H(s)
N=5
0 c
Thus at cutoff frequency = 1, magnitude is
Example:
For n = 0 C0( ) = cos(0) = 1
Determine the order of a lowpass Butterworth iii. In the stopband | | > 1, and .
For n = 1 C1( ) = cos(1cos–1 ( )) =
Hence eq (2) can be written as,
The higher order chebyshev polynomials are
more close to ideal characteristics.
an attenuation of 40 dB at 1000 Hz. obtained by following recursive formula.
The order & cut off frequency of the Butterworth
Sol: The critical frequencies are the –3 dB Cn ( ) = 2 Cn–1 ( ) – Cn 2 ( )
or
frequency c
and the stop-band frequency
which are n Chebyshev polynomials The above equation can be written in decibels
s
0 1 also i.e.,
c
= 1000
1
…… (3) s
= 2000
2
2 2 –1
3
3 4 –3 = –20 log10 [ Cn ( )]
4 2
4 8 –8 +1
……… (4) As is large Cn( ) is mainly represented by its
5 3
5 16 – 20 +5
n
2n-1 n for large
6 4 2
8.5 Chebyshev Filter 6 32 – 48 + 18 –1 Hence the above equation can be written as
n
Butterworth polynomials |H(j( )| in dB = –20 log10 [ .2n-1 ]
Some of the properties of these polynomials are = –20 log – 20 log 2n-1 – 20log n
Type-I
Order Factors 1. |Cn( )| 1 for all | | 1 = –20 log – 20(n – 1) log 2 – 20n log
exhibit equiripple behavior in the pass-band and
1 s+1 2. Cn(1) = 1 for all n = –20 log – 6(n –1) – 20n log (norm)
a monotonic characteristic in the stop-band. On
2 3. All the roots of the polynomials Cn( ) occurs in
the other hand, the family of type-II chebyshev
3 the interval –1 1
The following points can be noted about transfer
4
a monotonic behavior in the pass-band and an The squared magnitude function of the chebyshev
5
equiripple behavior in the stop-band. The zeros of
6
……….. (2)
ii. The numerator is constant and there are no
s-plane.
Signals & Systems Signals & Systems
Band-pass a2 = (K – 1) / (K + 1)
The constant K in the numerator in the above
equation is adjusted to provide a loss of 0 dB at
the pass band minima.
x
Blackman
………. (1)
Kaiser I
………….. (2)
Comparison of window characteristics
An FIR
Location of zeros of a linear phase FIR The unit sample response corresponding to Hd( ) is
design using rectangular (a), Hamming (b),
Blackman (c) and Kaiser window with = 4 for a
length of 61 samples.
z3 z1
1
z2 z2
z
z Design of Hilbert Transformers
1
z3 imparts a 90o phase shift on the signal at its input.
Unit Hence the frequency response of the ideal Hilbert
circle
frequency response
Hd( ) = j –
Signals & Systems Signals & Systems
N b0 b1 b2 b3 b4 b5 b6 b7 b8 b9 N b0 b1 b2 b3 b4 b5 b6 b7 b8 b9
1 2.8627752 1 1.0023773
2 1.5162026 1.4265245 2 0.7079478 0.6448996
3 0.7156938 1.5348954 1.2529130 3 0.2505943 0.9283480 0.5972404
4 0.3790506 1.0254553 1.7168662 1.1973856 4 0.1769869 0.4047679 1.1691176 0.5815799
5 0.1789234 0.7525181 1.3095747 1.9373675 1.1724909 5 0.626391 0.4079421 0.5488626 1.4149847 0.5744296
6 0.0947626 0.4323669 1.1718613 1.5897635 2.1718446 1.1591761 6 0.0442497 0.1634299 0.6990977 0.6906098 1.6628481 0.5706979
7 0.0447309 0.2820722 0.7556511 1.6479029 1.8694079 2.4126510 1.1512176 7 0.0156621 0.1461530 0.3000167 1.0518448 0.8314411 1.6628481 0.5684201
8 0.0236907 0.1525444 0.5735604 1.1485894 2.1840154 2.1492173 2.6567498 1.1460801 8 0.0110617 0.0564813 0.3207646 0.4718990 1.4666990 0.9719473 2.1607148 0.5669476
9 0.0111827 0.0941198 0.3408193 0.9836199 1.6113880 2.7814990 2.4293297 2.9027337 1.425705
9 0.0039154 0.0475900 0.1313851 0.5834984 0.6789075 1.9438443 1.1122863 2.4101346 0.5659234
10 0.0059227 0.2372688 0.2372688 0.6269689 1.5274307 2.1442372 3.4409268 2.7097415 3.1498757 1.1400664
10 0.0027654 0.0180313 0.1277560 0.2492043 0.9499208 0.9210659 2.4834205 1.2526467 2.6597378 0.5652218
N b0 b1 b2 b3 b4 b5 b6 b7 b8 b9
1 1.9652267
2 1.1025103 1.0977343
3 0.4913067 1.2384092 0.9883412
4 0.2756276 0.7426194 1.4539248 0.9528114
5 0.1228267 0.5805342 0.9743961 1.6888160 0.9368201
6 0.0689069 0.3070808 0.9393461 1.2021409 1.9308256 0.9282510
7 0.0307066 0.2136715 0.5486192 1.3575440 1.4287930 2.1760778 0.9231228
8 0.0172267 0.1073447 0.4478257 0.8468243 1.8369024 1.6551557 2.4230264 0.9198113
9 0.0067767 0.0706048 0.2441864 0.7863109 1.2016071 2.3781188 1.8814798 2.6709468 0.9175474
10 0.0043067 0.0344971 0.1824512 0.4553892 1.2444914 1.6129856 2.9815094 2.1078524 2.9194657 0.9159320
N b0 b1 b2 b3 b4 b5 b6 b7 b8 b9
1 1.3075603
2 0.6367681 0.8038164
3 0.3268901 1.0221903 0.7378216
4 0.2057651 0.5167981 1.2564819 0.7162150
5 0.0817225 0.4593491 0.6934770 1.4995433 0.7064606
6 0.0514413 0.2102706 0.7714618 0.8670149 1.7458587 0.7012257
7 0.0204228 0.1660920 0.3825056 1.1444390 1.0392203 1.9935272 0.6978929
8 0.0128603 0.0729373 0.3587043 0.5982214 1.5795807 1.2117121 2.2422529 0.6960646
9 0.0051076 0.0543756 0.1684473 0.6444677 0.8568648 2.0767479 1.3837474 2.4912897 0.6946793
10 0.0032151 0.0233347 0.1440057 0.3177560 1.0389104 1.1585287 2.6362507 1.5557424 2.7406032 0.6936904
Signals & Systems Signals & Systems
08. An IIR 13. Find H(z) and H(F) for each sequence and
Practice Questions
whose cutoff frequency is known establish the type of FIR
checking values of H(F) at F = 0 and F = 0.5
(or peak-to-peak ripple): 0.5 dB
(a)
sampling instants? Should it? Explain. a sampling rate of 2 Hz. sequence (assuming the smallest length for
(b) Use H(s) and the bilinear transformation to (a) 1-dB ripple in the pass-band 0 | | 0.3 h[n] if the sequence is to be:
(b) At least 60 dB attenuation in the (a) type 1
. gain at f = 20 kHz matches the gain of H(s) stop-band 0.35 | | . Use the bilinear
(b) type 2
at = 3 rad/s. The sampling frequency is transformation.
80 kHz. (c) type 3
invariance at a sampling frequency of 2Hz.
(b) Will the impulse response h[n] match the 10. Determine the system function H(z) of the (d) type 4
.
impulse response h(t
?
the sampling instants? Should it? Explain.
(b) Use H(s) and the bilinear transformation (a) ripple in the pass-band locations are in keeping with linear-phase and
(c) Will the step response s[n] match the step 0 | | 0.24
response s(t 1 kHz such that its gain at f0 = 250 Hz (b) At least 50 dB attenuation in the stop-band
sampling instants? Should it? Explain. matches the gain of H(s) at a = 1 rad/s. 0.35 | | .
? Use the bilinear transformation. (a) zero location: z = 0.5ej0.25
(b) zero location: z = ej0.25
. 11. A linear phase FIR (c) zero location: z = 1, z = ej0.25
. Find the remaining zeros? (d) zero locations: z = 0.5, z = –1; odd symmetry
ramp invariance at a sampling frequency 1 dB (e) zero locations: z = 0.5, z = 1, z = –1; even
symmetry
of 2Hz. 12. Transfer function of a IVth order linear phase
Stopband edge: 40 dB –1
(b) Will the impulse response h[n] match the F.I.R + 3z–2)G(z)
Stopband edge: 6 kHz 16. Design the symmetric FIR
impulse response h(t then G(z) is _____
Sample rate: 24 kHz frequency response is given as
the sampling instants? Should it? Explain. (a) 3 + 2z–1 + z–2
(c) Will the step response s[n] match the step bilinear transformation on an analog system (b)
function. Determine what order Butterworth, use Hamming window.
response s(t (c)
Chebyshev analog design must be used
sampling instants? Should it? Explain. (d) 1 + 2z–1 + 3z–2
implementation.
Signals & Systems Signals & Systems
x n = 0, 1 ………. (N – 1) (a) Picket - fence Effect: at frequencies other than f0 because the
(c) x periodic extension of the sampled portion of
The picket - fence effect is caused by the
approximation of the continuous frequency the periodic signal doesn’t match the original
• The DFT & its IDFT are also periodic with period (d) x
signal but describes a different signal altogether
Note: For direct calculation of N point DFT, we (b) Spectral leakage: We obtain kf = NF0 = 0.5. Thus F0 corresponds
2 Leakage is present in the DFT results if a periodic
require N complex multiplications and to the fractional index kf = 0.5 & the largest
signal x(t) is not sampled for an integer number DFT components should appear at the integer
N(N 1) additions.
of periods. The DFT shows nonzero components indices closest to kf at k = 0 (d .c), k = 1(2Hz)
Signals & Systems Signals & Systems
Example: Example:
Let x(n) = {1, 2, 3, 4, 5} and h(n) = {1,1,1} using Find convolution of x(n) = {1, 2, 3, 4, 5} and h(n) = {1, 1, 1} using Overlap Save method?
y(n)? Sol: First add (N – 1) = 2 zeros to
Sol: L = 6 & N = 3. x0(n) = {1, 2, 3} x(n) = {0, 0, 1, 2, 3, 3, 4, 5}
h(n) = {1, 1, 1} x1(n) = {3, 4, 5} Take M = 2N – 1 = 5, section into K overlapping segments of length M(5)
y0(n) = x0(n) * h(n) = {1, 3, 6, 5, 3} x0(n) = {0, 0, 1, 2, 3}
Zero pad h(n) to length M = 5 samples
y1(n) = x1(n) * h(n) = {3, 7, 12, 9, 5}
x1(n) = {2, 3, 3, 4, 5}
Shifting & superposition results in the required
h(n) = {1, 1, 1, 0, 0}
convolution
x2(n) = {4, 5, 0, 0, 0}
y(n) = y0(n) + y1[n – 3]
x0(n) h(n) = {5, 3, 1, 3, 6}
x1(n) h(n) = {11, 10, 8, 10, 12}
y0(n) x2(n) h(n) = {4, 9, 9, 5, 0}
To minimize leakage sample for the longest y(n) = {1, 3, 6, 8, 10, 12, 9, 5}
9.3 FFT
duration possible or for integer periods.
Fast algorithm reduce the problem of calculating an N-point DFT to that of calculating many smaller-
Convolution of Long Sequences 2. Overlap-Save method:
size DFTs. The computation is carried out separately on even - indexed and odd-indexed samples to
1. Overlap-Add method If L > N and we zero - pad the second sequence
reduce the computational effort. All algorithms allocate for computed results. The less the storage
2. Overlap-Save method to length L, their periodic convolution has
1) samples are
Many FFT algorithms reduce storage requirements by performing computations in place by storing
Some times we have to process a long stream contaminated by wraparound, and the rest
results in the same memory locations that previously held the data.
correspond to the regular convolution. eg. Let
response is much shorter than that of incoming L = 16 & N = 7.
data. The convolution of a short sequence h(n) If we pad N by 9 zeros, their regular convolution
3 stages in an 8-point DIT-FFT:
of length N with a very long sequence x(n) of has 31(or 2L 1) samples with 9 trailing zeros
length L > > N can involve large amount of (L N = 9). For periodic convolution 15 samples
computation & memory. (L 1 = 15) are wrapped around. Since the last
nine [or L
1. Overlap - Add method: of the periodic convolution are contaminated
Suppose h(n) is of length N, and the length of by wraparound, which is the basis idea of this
x(n) is L = mN (if not, we can always zero pad it method.
to this length). We partition x[n] into m segments First, we add (N – 1) leading zeros to the longer
x0(n), x1(n) …xm-1 sequence x(n) & section it into k overlapping
the regular convolution of each section with {by N – 1} segments of length M. Typically, we
h(n) to give partial results y0(n), y1(n), … ym-1(n). choose M 2N.
y(n) = y0[n] + y1[n–N] + ym-1[n–(m–1)N] Next, we zero-pad h(n) { with trailing zeros} to
Since each regular convolution contains
(2N – 1) samples, we zero – pad h(n) and each of h(n) with each section of x(n). Finally, we
section xk
yk[n] using the FFT. Splitting x(n) into equal – from each convolution & glue(concatenate)
length segments is not a strict requirement. the results to give the required convolution.
Signals & Systems Signals & Systems
(4) The list obtained in step3 now becomes the between spectral samples?
then
x(n).
(a) 10. Find the 10-point inverse DFT of (C) Let y(n) be an 8 point sequence for
20. Let be the 8-point DFT
0 n 7 and Y(k) be 8 point DFT.
(b) of a real signal x[n]
(c) + + + + If (a) Determine X[k] in its entirety.
(d) [ ] 11. A signal x(t) is bandlimited to 10 kHz is sampled (b) What is the DFT Y[k] of the signal
express y(n) in terms of x(n). y[n] = (–1)n x[n]?
with a sampling frequency of 20 kHz. The DFT of
07. Given x(n) = {1, –2, 3, –4, 5, –6}, without N = 1000 samples x(n) is then calculated. (c) What is the DFT G[k] of the zero-interpolated
16. The four point DFTS of x(n) and y(n) is
(a) What is the spacing between the spectral signal g[n] = x[n/2]
X(K) = {22, – 4 + j2, – 6, – 4 – j2} and
samples? Y(K) = {8, – 2 – j2, 0, – 2 + j2}. The circular
(a) X(0) (b) (c) X(3) (d) (b) To what analog frequency does the index convolution of x(n) and y(n) is w(n), then w(2)
k = 150 correspond? What about k = 800? 21. The Discrete Fourier Transform (DFT) of the
is
(e) 4-point sequence
(a) 38 (b) 30
x[n] = {x[0], x[2], x[3]} = { 3, 2, 3, 4} is
(c) 12 (d) 60
08. X(k) is the discrete Fourier Transform of a 6-point x[n] shown in Fig., whose six-point DFT is X[k]. X[k] = {X[0], X[1], X[2], X[3]}
real sequence x(n). If Q[k] = X[2k], k = 0, 1, 2 represents the 3-point = {12,2j, 0, 2j}.
17. Consider x[n] = {1, 2, 3, 4} with DFT
If X(0) = 9 + j0, X(2) = 2 + j2, DFT, then q [n] is X(K) = {10, 2+2j, 2, 2 2j}. If the sampling rate
If X1[k] is the DFT of the 12-point sequence
X(3) = 3 – j0, X(5) = 1 – j1, x(0) is x1[n]= {3, 0, 0, 2, 0, 0, 3, 0, 0, 4, 0, 0} the value
is 10Hz
(a) 3 (b) 9 (i) Determine the sampling period, time index of is
(c) 15 (d) 18 and the sampling instant for a digital sample
x(3) in time domain 22. Assume that a complex multiply takes 1 s and
(ii) Determine the frequency resolution, that the amount of time to compute a DFT is
09.
(i) The two 8-point sequences x1(n) & x2(n) shown frequency bin number and frequency for determined by the amount of time it takes to
1
(k) and X2(k) respectively. perform all of the multiplications.
(a) {5, 3, 2} (a) How much time does it take to compute a
(b) {4, 3, 2} 18. We wish to sample a signal of 1-s duration, and 1024-point DFT directly?
(c) {2, 3, 4} band-limited to 100Hz, in order to compute (b) How much time is required if an FFT is used
(d) {1, 2, 3} its spectrum. The spectral spacing should not
13. Given x[n] = {A, 2, 3, 4, 5, 6, 7, B} is having 8 exceed 0.5 Hz. Find the minimum number N
of samples needed and the actual spectral 23. Speech data is sampled at a rate of 10 kHz is
?
Find the relation between X1(k) & X2(k)? spacing f if we use
(a) The DFT (b) The radix-2 FFT the computations required involve collecting
(ii). Consider the sequence x 14. x[n].
blocks of 1024 speech values and computing
y(n) whose six-point DFT is
Then the value of x is ____ a 1024-point DFT and a 1024 point inverse DFT.
, 19. We wish to sample the signal,
If it takes 1 s for each real multiply, how much
Where X(k) is the six-point DFT of x(n). x(t) = cos(50 t) + sin(200 t) at 800 Hz and
15. Let x[n] be a real 8 point sequence and let X(k) time remains for processing the data after the
be its 8 point DFT DFT and inverse DFT are computed?
(A) Evaluate signal x[n]
(a) Let N = 100. At what indices would you
x
expect to see the spectral peaks? Will the
(B) Let w(n) be a 4 point sequence for peaks occur at the frequencies of x(t)?
0 n 3 and W(k) be its 4 point DFT. (b) Let N = 128. At what indices would you
If W(k) = X(k) + X(k+4), express w(n) in expect to see the spectral peaks? Will the
terms of x(n) peaks occur at the frequencies of x(t)?
Signals & Systems Signals & Systems
s
09. (i) Ans: X2(k) = ( 1)kX1(k) 6 Time Shift x(t ) u(t ), >0 e X(s)
7 Times-t t x(t)
(ii) Ans: y(n) = x((n 4))6
= {2, 1, 0, 0, 4, 3}
8 tn x(t)
2
16. Ans: (a) 10 x (t) s X(s) – sx(0–) – x (0–)
(n)
11 x (t) snX(s) – sn–1 x(0–) –….– xn-1(0–)
12 Integral x t t
Switched periodic,
14 xp(t)u(t) x t
x1(t)
Laplace Transform Theorems
15 Initial value x
16 Final value x t
Signals & Systems Signals & Systems