GOVERNMENT COLLEGE OF ENGINEERING,

BODINAYKKANUR
DEPARTMENT OF ELECTRICAL AND ELECTRONICS
ENGINEERING

EE8591 –DIGITAL SIGNAL PROCESSING

Year / Semester: III / V

UNIT – I

2 Marks Q & A

1. Consider the analog signal x(t) = 3cos50πt + 10 sin 300πt – cos 100πt.
What is the nyquist rate for this signal?
Here ωmax=300π
So, 2πfm=300π
Hence, Nyquist rate Fs=2fm=300

2. State Shannon’s sampling theorem.

A band limited continuous time signal with highest frequency (band
width) fm hertz , can be uniquely recovered from its samples provided
that the sampling rate fs is greater than or equal to 2fm samples per
second
Fs≥2fm

3. Given a continuous time signal x(t)= 2cos500πt. What is the Nyquist rate
and fundamental frequency of the signal?
ω =500π
2πf= 250π
f= 250Hz
Nyquist rate Fs=2fm= 2x250= 500Hz

4. Determine whether x[n]=u[n] is a power signal or an energy signal.

 The energy of a discrete time signal x(n) is defined as
2
∑
∞
E = =∞
n = −∞
x(n )

The average power of a discrete time signal x(n) is defined as

N 2
1
P = lim
N →∞
∑ x(n)
2 N + 1 n=− N
= 0.5
Here E= ∞ and P= Finite. Therefore the given signal is a power
signal.

5. What is the Nyquist rate for the signal xa(t)=3cos 600πt+2cos1800πt?

Solution: ω1=600π ω2=1800π
2πf1= 600π 2πf2= 1800π
f1= 300Hz f2= 900Hz
Nyquist rate Fs=2fm= 2x900= 1800Hz.

π 30n ⎞
6. Determine fundamental period of the signal Cos ⎛⎜ ⎟
⎝ 105 ⎠
⎛ 2π ⎞ 30π 105
Solution: Fundamental period, N= ⎜⎜ ⎟⎟m , Where ω o = = m,
⎝ ωo ⎠ 105 15
when m=1 & N = 7 periods.

7. What is a linear time invariant system?

An LTI system is one which possesses both Linerarity and Time-
invariance.
A system is linear if y1(n) = T[x1(n)] and y2(n) = T[x2(n)] then
T[a1 x1(n)+a2 x2(n)] = a1 y1(n)+ a2 y2(n)

8. What is meant by aliasing effect?

The superimposition of high frequency component on the low
frequency is known as “frequency aliasing” or “aliasing effect”.

9. Define Nyquist rate.

The frequency 2fm, which, under sampling theorem, must be exceeded
by the sampling frequency is known as the Nyquist rate.
10. What do you meant by sampling process?
Sampling is the conversion of a continuous –time signal (or analog
signal) into a discrete – time signal obtained by taking samples of the
continuous time signal (or analog signal) at discrete time instants.

11. How can aliasing be avoided?

To avoid aliasing the sampling frequency must be greater than twice
the highest frequency present in the signal.

12. What is an anti aliasing filter?

A filter is used to reject frequency signals before it is sampled to
reduce the aliasing is called an anti aliasing filter.

13. What is meant by Quantization error?

It is the difference between the quantized value and actual sample
value.
eq(n)=xq(n)-x(n)

14. What is a quantization level?

The value allows in a digital signal are called the quantization level.

15.Define resolution or quantization step size.

The distance between two successive level is called quantization step size or
resolution

16.Define unit sample response (impulse response) of a system and what is

its significance?
The response or output signal designated as h(n) , obtained from a discrete –
time system when the input signal is a unit sample sequence ( unit impulse) ,
is known as the unit sample response(impulse response).
The output y(n) of an LTI system for an input signal x(n) can be
obtained by convolving the impulse response h(n) and the input signal x(n).
y (n) = x(n) ∗ h(n)

∑
∞
= k =∞
x(k )h(n − k )
 17. What is the causality condition for an LTI system?
The necessary and sufficient condition for causality of an LTI system
is, its unit sample response h(n)=0 for negative values of n i.e., h(n)=
0 for n<0

18. What is the necessary and sufficient condition on the impulse response
for stability?
The necessary and sufficient condition guaranteeing the stability of a
linear time-invariant system is that its impulse response is absolutely
∞

summable. ∑ h (k ) < ∞ .
−∞

19. Define BIBO stable system?

Any relaxed system is said to be bounded input-bounded output
(BIBO) stable if and only if every bounded input yields a bounded
output. Mathematically, their exist some finite numbers, Mx and My
such that, x ( n ) ≤ Mx < ∞ and y ( n ) ≤ My < ∞

20. What are the classifications of discrete – time systems?

1. Static and Dynamic system. 2. Time – variant and time – invariant

system.
3. Linear and non – linear system. 4. Stable and Un-stable system.
5. Causal and non-causal system. 6. IIR and FIR system.

16marks

1. Discuss whether the following are energy or power signals.

(i)
(ii)

2. Explain the concept of quantization.

3. Check whether following are linear, time invariant, causal and stable.
(i)y(n) = x(n)+nx(n+1)
 (ii) y(n) = cos x(n)
(iii) y(n)= x(-n=5)

4. What is causality and stability of a system? Derive the necessary and

sufficient condition on the impulse response of the system for causality and
stability.

5. What is meant by energy and power signal? Determine whether the

following signals are energy or power or neither energy nor power signals.
n ⎛ πn π ⎞
π
x1(n) = ⎛⎜ ⎞⎟ u ( n ) x2(n) = sin ⎛⎜ n ⎞⎟ , x3(n) = e ⎝ 3 6 ⎠ , x4(n) = e 2 n u ( n)
1 j⎜ + ⎟

⎝2⎠ ⎝6 ⎠

6. A discrete time systems can be (i) Static or Dynamic, (ii) Linear or Non-
Linear, (iii) Time invariant or time varying & (iv) Stable or Unstable.
Examine the Following system with respect to the properties above y(n)=
x(n)+ nx(n+1)

7. Given y[n]= x[n2]. Determine whether the system is linear, time invariant,
memoryless and causal.

8. Determine whether the following is an energy signal or power signal.

⎛π ⎞
⎜ n⎟
a. x1[n]= 6cos ⎝ 2 ⎠
b. x2[n]= 3(0.5)n u(n).

9. Explain the digital signal processing system with necessary sketches and
give its merits and demerits

10. A discrete time systems can be

i. Static or dynamic
ii. Linear or non linear
iii. Time invariant or time varying
iv. Stable or unstable. Examine the following system with respect
to the properties above y(n) = x(n) + nx(n + 1)
 UNIT – II

2 Marks Q & A

1. What is meant by region of convergence?

The region of convergence (ROC) of X(z) is the set of all values of z
for which X(z) attains a finite value.
2. What are the properties of region of convergence?
The ROC is a ring or disk in the Z – plane centered at the origin.
The ROC cannot contains any poles.
The ROC of an LTI stable system contains the unit circle.
The ROC must be a connected region.
3. What are the properties of z- transform ?
i. Linearity: z [a1 x1 ( n) + a 2 x 2 ( n)] = a1 X 1 ( z ) + a 2 X 2 ( z )
⎡ m −1
m −i ⎤
ii. Shifting: (a) z[x(n + m)] = z ⎢ X ( z ) − ∑ x(i ) z ⎥
m

⎣ i =0 ⎦
1. (b) z [x ( n − m ) ] = z −m X (z)
m

[ m d
dz ⎠
]
iii. Multiplication: z n x(n) = ⎛⎜ − z ⎞⎟ X(z)
⎝
iv. Scaling in z- domain: z [a x ( n ) ] = X (a z )
n −1

v. Time reversal : z [x ( − n ) ] = X ( z
−1
)
vi. Conjugation: z [x ( n )] = X ( z )
∗ ∗ ∗

⎡ n ⎤
vii. Convolution: ⎢ ∑ h ( n − m ) r ( m ) ⎥ = H ( z ) R ( z )
z
⎣ m =0 ⎦
viii. Initial value: z [x (0) ] = Lt X ( z )
z→∞

ix. Final value: z [x (α ) ] = zLt

−1
(1 − z ) X ( z )
→1

4. What is the relationship between z-transform and DTFT?

∞

The z-transform of x(n) is given by X ( z ) = ∑ x(n) z

n = −∞
−n
……. (1)
Where z = re jω
Substituting z value in eqn (1) we get,
X ( re jω
)= ∑ x ( n ) r − n e − jω n ………………….. (2)
The Fourier transform of x(n) is given by
 ∞
X ( e jω ) = ∑ x ( n )e
n = −∞
− jω n
…………………… . (3)
Eqn (2) and Eqn (3) are identical, when r=1. In the z-plane this
corresponds to the locus of points on the unit circle z = 1 .
jω
Hence X ( e ) is equal to X(z) evaluated along unit circle , or
X ( e j ω ) = X ( z ) | z = e jω
jω
For X ( e ) to exist, the ROC of X(z) must include the unit circle.
5. Write the commutative and distributive properties of convolution.
Commutative Property: x(n)*h(n)= h(n)* x(n)
Distributive property: x(n)*[h1(n)+h2(n)]=[ x(n)* h1(n)]+[ x(n)* h2(n)]
6. Determine the Z-transform and ROC for the signal x(n)=δ(n-k)+
δ(n+k).
Solution: X(Z)= Z − k + Z + k , X(Z) will converge for all the values of Z,
except Z = 0 and ∞ .
7. Given a difference equation y(n)= x[n]+3x[n-1]+2y[n-1]. Determine the
system function H(Z).
Solution: On taking Z- Transform, Y(Z)= X(Z)+3Z-1 X(Z)+2Z-1 Y(Z)
Y(Z)[1-2Z-1]= X(Z)[1+3Z-1]
Y ( Z ) 1 + 3Z -1
H(Z)= =
X ( Z ) 1 - 2Z -1
n

8. Find the stability of the system whose impulse response h(n)= ⎛⎜ ⎞⎟ u (n)
1
⎝2⎠
∞
For Stable, ∑ |h(k)| < ∞
−∞

∑ |h(k)| =2< ∞
−∞

Therefore the system is stable.

9. State the convolution property of Z transform.
The convolution property states that the convolution of two sequences
in time domain is equivalent to multiplication of their Z transforms.

10. What are the computational savings in evaluation of DFT using radix-2
FFT?
Multiplications: N/2 logN2
Additions: N/ogN2
11. State the meaning of bit reversal in FFT algorithm.
DIT FFT: I/P – bit reversal order & O/P – normal order
DIF FFT: O/P – bit reversal order & I/P – normal order

12. State initial value theorem.

If x(n)and X(z) are z-Transform pairs, then, ,
x(0) = lim X(z)
z->∞
provided thet the first derivative of x(t) should be laplace
transformable.
13. State final value theorem.
If x(n)and X(z) are z-Transform pairs, then the final value of x(z) is
given as ,
x(∞)= Lim (1-z-1)X(z) have no pole on or outside the unit circle.
z->1

14. Define prewarping or prescaling.

For large frequency values the non linear compression that occurs in
the mapping of to w is more apparent .This compression causes the
transfer function at high frequency to be highly distorted when it is
translate to the w domain.This compression is being compensated by
introducing a prescaling or prewarpping to frequency Ωscale.For
bilinear transform Ωscale is converted into Ω* scale (i.e)
Ω*=2/Ts tan (ΩTs/2) (prewarped frequency)
15. Write the properties of frequency response of LTI system.
The frequency response is periodic function ω with a period of 2 π .

If h(n) is real then H ( ω )

i) is
symmetric and ∠ H ( ω ) is
antisymmetric.
ii) If h(n) is complex then the real part of
H (ω ) is antisymmteric over the interval
0 ≤ ω ≤ 2π .
iii) The frequency response is a continuous
function of ω .
16. Write short notes on the frequency response of first order system.
A first order system is characterized by the difference equation
y ( n ) = x ( n ) + ay ( n − 1)
The frequency response of first order system depends on the co efficient “a”
in the difference equation governing the LTI system. When the value of “a|”
 is in the range of 0<a<1, the first order system behave as a low pass filter.
When the value of “a” is in the range –1<a<0, the first order system behave
as a high pass filter.

17. Write a short note on the frequency response of second order system.
A second order system is characterized by the difference equation
y ( n ) = 2 r cos ω 0 y ( n − 1) − r 2 y ( n − 2 ) + x ( n ) − r cos ω 0 x ( n − 1)
The frequency response of second order system depends on the
parameters “r” and “ ω0 ” in the difference equation the LTI system.
When the value of r is in the range of 0<r<1, the second order system
behave as a resonant filter with center frequency ω0 . When the value
of r is varied from 0 to 1 , the sharpness of resonant peak increases.
18. Define discrete Fourier series.
Consider a sequence xp(n) with a period of N samples so that xp(n)=xp(n/N);
Then the discrete Fourier series of the sequence xp(n) is defined as
N −1
X p (k ) = ∑x
n=0
p ( n ) e − j 2 πkn / N

19. What are the two basic differences between the Fourier transform of a
discrete time signal with the Fourier transform of a continuous time
signal?
For a continuous signal, the frequency range extends
from − ∞to + ∞.
On the other hand, the frequency range of a discrete – time signal extends
from − π to + π ( or 0 to 2π ) .
The Fourier transform of a continuous signal involves integration , whereas ,
the Fourier transform of a discrete – time signal involves summation
process.
20. Find the Fourier transform of a sequence x(n) = 1 for − 2 ≤ n ≤ 2
= 0 otherwise.
Solution:
∞ 2
X (ω ) = ∑
n = −∞
x ( n ) e − jω n = ∑e
n = −2
− jω n

j 2ω jω − jω − j 2ω
=e +e +1+ e +e
16 Marks Questions
1. Find the Z- Transform of following :
 (a)x(n) = sin(nω0 )u(n)
(b) x(n) = cos(ωo n)u (n)
2. Determine the causal signal x(n) having the Z- Transform
1 + Z −1
X(Z) =
(a) 1 - Z −1 + 0.5Z − 2
1
X(Z)=
(b) 1 − Z + 0.5Z−2
−1

1 + 2z −1 + z −2
X(Z) =
(c) 1 + 4Z −1 + 4Z − 2

3. Obtain the linear convolution of x(n) = {3, 2, 1, 2}& {1, 2, 1, 2 }

4. A discrete time system is described b the following equations:

1 1

5. Determine x(n) for the given x(Z) with ROC

1 | | 2 2 | | 2

6. Find the Z- transform and its associated ROC for the following discrete time
1 1
signal x[n]= ( ) n u[n]+5 ( ) − n u[-n-1] (May/Jun 13)
3 2

7. A system is described bythe difference equation y(n)- 1/2y(n-

1)=5x(n).Determine the solution when the input x(n)=(1/5)nu(n) and the
initial condition is given by y(-1)=1 using Z transforms. (Nov/Dec’12)

8. Deterrmine the impulse response of the system described by the difference

equation y(n)=y(n-1)- 1/2y(n-2)+x(n=x(n-1) using Z transforms and discuss
its stability.

9. Evaluate the frequency response of the system described by the system

1
function H(Z)=
1 − 0.5 z −1
 1
10. Using the Z- Transform determine the response y[n] for n≥0 if y[n] = y[n-
2
1
1]+x[n], x[n]= ( ) n u[n], y(-1)=1.
3
UNIT – III

2 Marks Q & A

1. Draw the basic butterfly diagram for Radix 2 DITFFT

2. Write the DTFT for

a. x(n) = anu(n)
b. x(n)=4δ(n)+3δ(n-1)

3. Define DTFT pair for a discrete sequence

∞
DTFT of x(n)= X(w)= ∑ x ( n)e
n = −∞
− jwn

4. Find the 4-point DFT of the sequence x(n)={1,1}

X(K)= {2,1-j,0,1+j}

5. In eight point decimation in time(DIT), what is the gain of the signal

path that goes from x(7) to X(2)?
Answer: − W80W82

6. The first five points of the eight point DFT of a real valued sequence are
{0.25, 0.125-j0.3018,0,0.125-j0.0518,0}. Determine the remaining three
points.
Answer: X(5)= -0.125-j0.3018, X(6)=0, X(7)= -0.125-j0.0518

7. Find the discrete Fourier Transform for δ[n].

N −1
X(k) = ∑ x(n)e − j 2πnk / N =
n= 0
∑ δ (n)e
n=0
()
=1.

8. Draw the basic butterfly diagram for Radix 2 DIF FFT

 9. What is decimation – in – time algorithm?
The computation of 8 – point DFT using radix-2 FFT , involves three stages
of computations. Here N=8=23, therefore r=2 and m=3.

The given 8 – point sequence is decimated to 2- point sequences . For each 2

– point sequence, the 2-point DFT is computed . From the result of 2 – point DFT
the 4 – point DFT can be computed. From the result of 4-point DFT , the 8 – point

DFT can be computed.

10. What is FFT?

The term Fast Fourier Transform (FFT) usually refers to a class of algorithms
for efficiently computing the DFT.It makes use of the symmetry and periodicity
K
properties of twiddle factor W N to effectively reduce the DFT computation time.

It is based on the fundamental principle of decomposing the computation of DFT

of a sequence of length N into successively smaller discrete Fourier transforms.
The FFT algorithm provides speed increase factors , when compared with direct
computation of the DFT, of approximately 64 and 205 for 256 points and 1024 –
point transforms respectively.

11. Obtain the circular convolution the following sequences

x(n)={0,1,0,2 }; h(n)={ 2,0,1 }
Solution:
The circular convolution of the above sequences can be obtained by
using matrix method.
 ⎡ h ( 0 ) h ( 2 ) h (1) ⎤ ⎡ x ( 0 ) ⎤ ⎡ y (0 ) ⎤
⎢ h (1) h ( 0 ) h ( 2 ) ⎥ ⎢ x (1) ⎥ = ⎢ y (1 ) ⎥
⎢ ⎥⎢ ⎥ ⎢ ⎥
⎢⎣ h ( 2 ) h (1) h ( 0 ) ⎥⎦ ⎢⎣ x ( 2 ) ⎥⎦ ⎢⎣ y ( 2 ) ⎥⎦
⎡ 1 2 − 2⎤ ⎡1 ⎤ ⎡ 3 ⎤
⎢− 2 1 2 ⎥ ⎢ 2 ⎥ = ⎢⎢ 2 ⎥⎥
⎥ ⎢ ⎥
⎢
⎢⎣ 2 2 1 ⎥⎦ ⎢⎣ 1 ⎥⎦ ⎢⎣ − 1 ⎥⎦
y ( n ) = {3 , 2 , − 1}

12. What are the differences and similarities between DIF and DIT
algorithms?
Differences:
For DIT, the input is bit reversal while the output is in natural order,
whereas for DIF, the input is in natural order while the output is bit
reversed.
The DIF butterfly is slightly different from the DIT butterfly, the
difference being that the complex multiplication takes place after the
add-subtract operation in DIF.
Similarities: Both algorithms require same number of operations to
compute the DFT. Bot algorithms can be done in place and both need
to perform bit reversal at some place during the computation.

13. What is FFT?

The fast Fourier transforms (FFT) is an algorithm used to compute the
DFT. It makes use of the Symmetry and periodically properties of
twiddles factor WKN to effectively reduce the DFT computation time.
It is based on the fundamental principle of decomposing the
computation of the DFT of a sequence of length N into successively
smaller discrete Fourier transforms. The FFT algorithm provides
speed-increase factors, when compared with direct computation of the
DFT, of approximately 64 and 205 for 256-point and 1024-point
transforms, respectively.

14. What is the main advantage of FFT?

FFT reduces the computation time required to compute discrete
Fourier transform.

15. Relationship between DTFT and DFT and Z transform

 DTFT output is continuous in time where as DFT output is Discrete in
time. Whereas Z transform is order of discretization in z-domain

16. Distinguish between FIR and IIR filters.

S.No. FIR filter IIR filter

These filters can be easily designed to have These filters do not have
perfectly linear phase. linear phase.

FIR filters can be realized recursively and IIR filters can be realized
non-recursively. recursively.
Greater flexibility to control the shape of Less flexibility,usually
their magnitude response. limited to kind of filters.

Errors due to roundoff noise are less severe The roundoff noise in IIR
in FIR filters, mainly because feedback is not filters are more.
used.

17. Define the Discrete Fourier transformation of a given sequence x(n).

The N- point DFT of a sequence x(n) is
N −1
X (k ) = ∑ x (n )e − j 2 π kn / N

n=0 k= 0, 1, 2……..N-1.

18. Write the formula for N- point IDFT of a sequence X(k).

The N-point IDFT of a sequence X(k) is
N −1
1
x(n) =
N
∑ X ( k )e j 2πk / N

K =0 n= 0, 1 , 2 …..N-1 .

19. List any four proerties of DFT.

(a) Periodicity
If X(k) is N- point DFT of a finite duration sequence x(n)
then
x ( n + N ) = x ( n ) for all n
X ( k + N ) = X ( k ) for all k.
(b)Linearity
If X1(k)=DFT[x1(n)] and
X2(k)=DFT[x2(n)]
 then
DFT[a1x1(n)+a2x2(n)]=a1X1(k)+a2x2(k)
( c ) Time reversal of a sequence
If DFT {x(n)}=X(k),
then
DFT{x((-n))N}=DFT{x(N-n)}=X((-k))N=X(N-k)
(d)Circular time shifting of a sequence
If DFT {x(n)}=X(k),
then
− j 2 π kl / N
DFT{x((n-l))N}=X(k) e

20. IF N-point sequence x(n) has N- point DFT X(k) then what is the DFT of
the following?
(i)x∗ (n) (ii)x∗ (N − n) (iii)x((n − l))N (iv)x(n)e j 2π ln/ N
Solution:
( i ) DFT { x ∗ ( n )} = X ∗ ( N − k )
( ii ) DFT { x ∗ ( N − n )} = X ∗ ( k )
( iii ) DFT { x (( n − l )) N } = X ( k ) e − j 2 π kl / N
( iv ) DFT { x ( n ) e j 2 π ln/ N } = X (( k − l )) N

16 Marks Questions

1. Discuss various properties of DFT.

2. Perform circular convolution of two sequences

3. Develop a Radix-2, 8-point DIF FFT algorithm with neat flow chart.

4. Find the inverse DFT of X(k) = {1, 2, 3, 4}

5. An 8-point sequence is given by x (n) = {2, 2, 2, 2, 1, 1, 1, 1). Compute 8-

point DFT of X(n) by radix -2 DIT FFT. Also sketch the magnitude and
phase spectrum.

6. Compute the DFT of x(n)={1, 1, 0, 0}

7. Describe the following properties of DFT.

 (i) Time reversal (ii) Circular convolution

8. Obtain the circular convolution of x1(n)={1, 2, 2, 1}, x2(n)={1, 2, 3,1}

9. Describe the following properties of DFT.

(i) Time reversal (ii) Circular convolution

10. Find the inverse DFT of X(k)= {7,-√2-j√2,-j, √2-j√2,1, √2+j√2,j,- √2+j√2}

UNIT – IV

2 Marks Q & A

1. What are the properties of Chebyshev filter?

a) The magnitude response of the Chebyshev filter exhibits in ripple either in
pass band or in the stop band according to the type.
b) The magnitude response approaches the ideal response as the value of N
increases.
c) The Chebyshev type – 1 filters are all pole designs.
d) The poles of Chebyshev filter lies on an ellipse.
1
e) The normalized magnitude function has a value of 1+ ε 2
at the cutoff
frequency Ω c .

2. Distinguish between FIR and IIR filter.

Sl FIR filter IIR filter
.
N
o

1. These filters can be easily These filters do not have linear

designed to have perfectly linear phase.
phase.

2. FIR filters can be realized IIR filters are easily realized

recursively and non – recursively.
recursively.
3. Greater flexibility to Less flexibility, usually limited
control the shape of their to specific kind of filters.
magnitude response.

4. Error due to round off The rounds off noise in IIR

noise are less severe in FIR filters are more.
filters, mainly because
feedback is not used.

3. Write the condition for stability of digital filter

a. Choose the desired (ideal ) frequency response H d (ω ) of the filter.
b. Evaluate the Fourier series co-efficient of H d (ω ) which gives the
π
1
desired impulse response hd (n) . Where hd (n ) =
2π ∫π H
−
d (ω ) e jω n d ω

c. Truncate the infinite sequence hd (n)

to a finite duration sequence
h (n ) .
d. Take Z – transform of h ( n ) to get a non causal filter transfer
function H ( z ) of the FIR filter.
⎛ N −1 ⎞
e. Multiply H ( z ) by z − ⎜⎝ 2 ⎟⎠ to convert noncausal transfer function to
a realizable causal FIR filter transfer function.
⎛ N −1 ⎞ ⎡ N −1
⎤
∑ h ( n ) (z )
−⎜ ⎟ 2
H (z) = z ⎝ 2 ⎠ ⎢h (0) + n
+ z −n ⎥
f. ⎢ n =1
⎥
⎢⎣ ⎥⎦
4. What is Gibbs phenomenon?
One possible way of finding an FIR filter that
jω
approximates H ( e ) would be to truncate the infinite
⎡ N − 1⎤
Fourier series at n= ± ⎢⎣ 2 ⎥⎦ .The abrupt truncation of
the series will lead to oscillation both in passband and in
stopband. This phenomenon is known as Gibbs
phenomenon.
5. What is window and why it is necessary?
jω
One possible way of finding an FIR filter that approximates H ( e )
⎡ N − 1⎤
would be to truncate the infinite Fourier series at n= ± ⎢⎣ 2 ⎥⎦ . The
 abrupt truncation of the series will lead to oscillation both in
passband and in stopband. These oscillations can be reduced through
the use of less abrupt truncation of the Fourier series. This can be
achieved by multiplying the infinite impulse response with a finite
weighing w ( n ) , called a window.
6. Mention any two procedures for digitizing the transfer function of an
analog filter.
The two important procedures for digitizing the transfer function of an analog filter
are

(i)Impulse invariance method. (ii)Bilinear transformation method.

7. What is frequency warping?

In bilinear transformation the relation between analog and
digital frequencies is nonlinear. When the s-plane is mapped
into z-plane using bilinear transformation , this nonlinear
relationship introduces distortion in frequency axis, which is
called frequency warping.
8. Explain the technique of prewarping.
In IIR filter design using bilinear transformation the specified
digital frequencies are converted to analog equivalent
frequencies, which are called prewarp frequencies. Using the
prewarp frequencies, the analog filter transfer function is
designed and then it is transformed to digital filter transfer
function.
9. Compare the impulse invariant and bilinear transformations.
S Impulse Invariant Bilinear transformation
l. transformation
N
o

1 It is many – to – one It is one – to – one mapping.

. mapping

2 The relation between analog The relation between analog and digital
. and digital frequency is frequency is nonlinear.
 linear.

3 To prevent the problem of There is no problem of aliasing and so the

. aliasing the analog filters analog filter need not be band limited.
should be band limited.

The magnitude and phase Due to the effect of warping, the phase
response of analog filter can response of analog filter cannot be
4
be preserved by choosing preserved . But the magnitude response can
.
low sampling time or high be preserved by prewarping.
sampling frequency.

10. What is meant by linear phase response of a filter?

The maintenance of a constant gain between the cut-off frequencies.
1 + 0 . 8 z −1
11. Is the given transfer function H(z)= represents low pass filter
1 − 0.9 z −1
or high pass filter?
Answer: Low pass filter

12. The impulse response of an analog filter is shown in below figure. Let
h(n)= ha(nT). Where T=1sec. Determine the System function.

nt , 0<t <5
Solution : ha(nT)= {
10 − nT , 5 < t < 10
5 10
H(Z) = ∑ nZ −n + ∑ (10 − n)Z −n
n =0 n =5

13. What is prewarping? Why it is employed?

In IIR filter design using bilinear transformation , the conversion of
the specified digital frequencies to analog frequencies is called
prewarping.
The prewarping is necessary to eliminate the effect of warping on
amplitude response.
14. What are the advantages and disadvantages of digital filters?
Advantages:
a) High thermal stability due to absence of resistors , inductors and
capacitors.
b) The performance characteristics like accuracy, dynamic range,
stability and tolerance can be enhanced by increasing the length of
the registers.
c) The digital filters are programmable.
d) Multiplexing and adaptive filtering are possible.
Disadvantages:

a) The bandwidth of the discrete signal is limited by the sampling frequency.

b) The performance of the digital filter depends on the hardware used to
implement the filter.
15. What is impulse invariant transformation?
The transformation of analog filter to digital filter without modifying
the impulse response of the filter is called impulse invariant
transformation (i.e., in this transformation the impulse response of the
digital filter will be sampled version of the impulse response of the
analog filter.)
16. What is the main objective of impulse invariant transformation?
The objective of this method is to develop an IIR filter transfer
function whose impulse is the sampled version of the impulse
response of the analog filter. Therefore the frequency response
characteristics of the analog filter are preserved.
17. Write the impulse invariant transformation used to transform real poles
with and without multiplicity.
The impulse invariant transformation used to transform real poles (at
s = - pi) without multiplicity is
1 1
is transforme d to → − pi T −1
s + pi 1− e z
The impulse invariant transformation used to transform multiple real pole (at
s = - pi) is
1 (−1) m −1 d m −1 1
is transforme d to →
(s + pi )m m −1 − pi T −1
( m − 1) dp i 1 − e z
 18. What is the relation between digital and analog frequency in impulse
invariant transformation.
The relation between analog and digital frequency in impulse
invariant transformation is given by
Digital frequency, ω = Ω T
Where, Ω - Analog frequency and T - Sampling time period

19. What is Bilinear transformation?

The Bilinear transformation is a conformal mapping that transforms the s-plane to
z-plane . In this mapping the imaginary axis of s-plane is mapped into the unit
circle in z-plane, the left half of s-plane is mapped into interior of unit circle in z-
plane and the right half of s-plane is mapped into exterior of unit circle in z-plane .
The Bilinear mapping is a one – to-one mapping and it is accomplished when
2 1 − z −1
s=
T 1 + z1

20. What is the relation between digital and analog frequency in Bilinear
transformation?
In Bilinear transformation , the digital frequency and analog
frequency are related by the equation,
ΩT
Digital frequency, ω
−1
= 2 tan or
2
2 ω
Analog frequency Ω = tan
T 2
Where, Ω - Analog frequency & T - Sampling time period

16 Marks Questions

1. Obtain the Direct form –I, Direct form – II, Cascade form and Parallel form
structure for the system described by
(a ) y (n) = − 0.1y (n − 1) + 0.2 y (n − 2) + 3 x(n) + 3.6 x(n − 1) + 0.6 x(n − 2)
(b) y (n) = 0.5 y (n − 1) + 0.25 y (n − 2) + x(n) + x(n − 1)

2. Design an FIR linear phase, digital filter approximating the ideal frequency
response
 ⎧ π
⎪⎪1, for ω ≤
6
H d (ω ) = ⎨
⎪0, π
for ≤ ω ≤π
⎪⎩ 6

3. Determine the coefficients of a 11- tap filter based on the window method
with a Hamming window

4. Design a FIR filter having following specification

⎧ π
≤ ω ≤π
Hd e( )jw ⎪1,
=⎨
for
4
⎪⎩0, otherwise

and given that N=11. Using (a) Hanning Window (b) Hamming Window

2
5. Convert analog filter H a ( S ) = into digital filter by means of
(s + 1)(s + 2)
bilinear transformation when T = 1 sec

6. Convert the analog filter with system function

s + 0.1
H a(S ) = into digital IIR filter by means of Impulse invariant
( s + 0.1) 2 + 9
transformation

7. Design (a) Butterworth and (b)Chebyshev analog high pass filter that will
pass all signals of radian frequencies greater than200rad/sec with no more
than 2 dB attenuation and have a stop band attenuation of greater than 20
dB for all Ω less than 100rad/sec.

8. Obtain the cascade realization of

2 + Z −1 + Z −2
H ( z) =
⎛ 1 −1 ⎞⎛ 1 −1 ⎞⎛ 1 −1 ⎞
⎜1 + Z ⎟⎜1 − Z ⎟⎜1 + Z ⎟
⎝ 2 ⎠⎝ 4 ⎠⎝ 8 ⎠

9. Design a chebyshev filter for the following specification using (a) bilinear
transformation (b) Impulse invariance method.
10. Realize the following system functions using a minimum number of
multipliers
1 −1 3 − 2 1 −3
(a) H ( z ) = 1 + z + z + z + z −4
2 4 2
1 1
(b) H ( z ) = 1 + z −1 + z − 2 + z −3
2 2
1 −1
2
) 1
(c) H ( z ) = (1 + z + z − 2 (1 + z −1 + z − 2
4
)

UNIT – V

2 Marks Q & A

1. Give the special features of DSP processors. (Nov/Dec’11)

(i) Harmonics can be analyzed using Fourier analysis.
(ii) Generation of pulses
(iii) Discretizing the waveform
2. What is the function of parallel logic unit DSP processor? (Nov/Dec’12)
The parallel logic unit is a second logic unit, that execute logic
operations on data Without affecting contents of accumulator.
3. Define Period gram. (May/Jun’12)
Periodic analysis of the waveform can be analyzed.
4. What is meant by bit reversed addressing mode? What is the
application for which this addressing mode is preferred? (Nov/Dec’13)
Bit-reverse addressing is a special type of indirect addressing. It uses one of
the auxiliary registers (AR0−AR7) as a base pointer of an array and uses
temporary register 0 (T0) as an index register. When you add T0 to the
auxiliary register using bit-reverse addressing, the address is generated in a
 bit-reversed fashion, with the carry propagating from left to right instead of
from right to left.
Application: Bit-reversed addressing, a special addressing mode useful for
calculating FFTs

5. Compare the RISC and CISC processors. (Nov/Dec’13)

CISC  RISC 

Emphasis on hardware  Emphasis on software 

Includes multi‐clock  Single‐clock,
complex instructions  reduced instruction only 

Memory‐to‐memory:  Register to register: 
"LOAD" and "STORE"  "LOAD" and "STORE" 
incorporated in instructions  are independent instructions 

Small code sizes,  Low cycles per second, 
high cycles per second  large code sizes 

Transistors used for storing  Spends more transistors 
complex instructions  on memory registers 

If it reads as above (i.e. as CISC  If it reads as above (i.e. as RISC 
computer),  computer), 

it means a computer that has  it means a computer that has 
a Complex Instruction Set Chip as its  a Reduced Instruction Set Chip as its 
cpu.  cpu 

6. Mention one important feature of Harvard architecture. (May/Jun’13)

(i) High Compatibility, (ii) RISC Processor
7. What is the advantage of Pipelining? (May/Jun’13)
 It provides sequential flow of execution with one after the other
process without any interruption.

8. What are the different buses of TMS320C54X and their functions?

The C5X architecture has four buses and their functions are as
follows:

a. Program bus (PB):

It carries the instruction code and immediate operands from program
memory Space to the CPU.

b. Program address bus (PAB):

It provides addresses to program memory space for both reads and
writes.

c. Data read bus (DB):

It interconnects various elements of the CPU to data memory space.

d. Data read address bus (DAB):

It provides the address to access the data memory space.

9. Briefly explain about Harvard architecture.

The principal feature of Harvard architecture is that the program and the data
memories lie in two separate spaces, permitting full overlap of instruction
fetch and execution called pipelining. Thus reducing the operation time.
Typically these types of instructions would involve their distinct type.
Instruction fetch
Instruction decode
Instruction execute
 

10. How DSP processor works faster than general purpose processors?
Digital signal processors are basically high speed microprocessors with hard
ware architecture and instruction set optimized for DSP operations. These
processors make extensive use of parallelism, Harvard architecture,
pipelining and dedicated hardware whenever possible to perform time
consuming operations

11. What is meant by pipeline technique? What are the different stages in
pipelining?
 The pipeline technique is used to allow overall instruction executions to
overlap. That is where all four phases operate in parallel. By adapting this
technique, execution speed is increased.

The four stages are

(i)Fetch
(ii)Decode
(iii)Read
(iv) Execution
 

12. What is the function of parallel logic unit in DSP processor

The parallel logic unit (PLU) can directly set, clear, test, or toggle multiple
bits in control/status register pr any data memory location. The PLU
provides a direct logic operation path to data memory values without
affecting the contents of the ACC or the PREG.

13. State the intended applications of DaVinci Digital Media processors

Image compression, Image coding ,speech compression, multirate signal
filters

14. List out the typical features of Digital Signal Processor

a. Fixed-point processor (TMS320C5000, 56000...) or floating point
processor (TMS320C67, 96000...)
b. Architecture optimized for intensive computation. For instance the
TMS320C67 can do 1000 Million floating point operations a second
(1 GIGA Flop).
c. Narrow address bus supporting a only limited amounts of memory.
d. Specialized addressing modes to efficiently support signal processing
operations (circular addressing for filters, bit-reverse addressing for
Fast Fourier Transforms…etc.)
e. Narrow data formats (16 bits or 32 bits typical).
f. Many specialized peripherals integrated on the chip (serial ports,
memory,
g. Low power consumption & Low cost.

15. What is Barrel Shifter?

The Barrel Shifter is used for scaling operations such as prescaling an input
data-memory operand or the accumulator value before an ALU operation;
 performing a logical or arithmetic shift of the accumulator value;
normalising the accumulator; postscaling the accumulator before storing the
accumulator value into data memory.

16. What are the elements present in CPU of 54X processor?

(i) 40-Bit Arithmetic Logic Unit (ALU)
(ii) Two 40- Bit Accumulator Registers
(iii) Barrel Shifter
(iv) Multiply / Accumulate Block
(v) 16-Bit Temporary Register (T)
(vi) 16-Bit Transition Register
(vii) Compare, select and store unit
(viii) Exponent Encoder
17. List the various registers used with ARAU.
Eight auxiliary registers (AR0 –AR7)
Auxiliary register pointer (ARP)

18. What are the factors that influence selection of DSPs?

1. Architectural features 2.Execution speed 3.Type of arithmetic
4.Word length
19. What is pipeline depth?
The number of pipeline stages is referred to as the pipeline depth.
20. What is the pipeline depth of TMS320C50, TM 320C54x?
TMS320C50 – 4 TM 320C54x – 6

16 Marks Questions

1. Explain Von Neumann, Harvard architecture and modified Harvard

architecture for the computer.
2. Write short notes on auxiliary registers
3. Write short notes on circular addressing mode

4. Write short notes on memory mapped register addressing

5. Explain the advantages and disadvantages of VLIW architecture

6. Draw the block diagram of Harvard architecture and explain
7. Explain about pipelining in DSP
8. Explain the addressing modes of a DSP processor.
9. Describe in detail the architecture of TMS 320C 54 DSP processor and state
the main features of this processor.
10. Enumerate various applications of DSP processors.

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