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UNIT I INTRODUCTION

(2 Mark Questions and Answers)


1. Define DSP.
DSP - Digital Signal Processing.
It is defined as changing or analyzing information which is measured as discrete time
sequences.
2. List out the basic elements of DSP.
* signal in
* Analog to Digital converter
* Digital Signal processor
* Digital to Analog converter
* signal out
3. Mention the advantages of DSP.
* Veracity
* Simplicity
* Repeatability
4. Give the applications of DSP.
* Telecommunication spread spectrum, data communication
* Biomedical ECG analysis, Scanners.
* Speech/audio speech recognition
* Military SONAR, RADAR
5. Define Signal.
Signal is a physical quantity that varies with respect to time , space or any other
independent variable.
(Or)
It is a mathematical representation of the system
Eg y(t) = t. and x(t)= sin t.
6. Define system.
A set of components that are connected together to perform the particular task.
E.g.; Filters
7. What are the major classifications of the signal?
(i) Discrete time signal
(ii) Continuous time signal
8. Define discrete time signals and classify them.
Discrete time signals are defined only at discrete times, and for these signals, the
independent variable takes on only a discrete set of values.
Classification of discrete time signal:
1. Periodic and A periodic signal
2. Even and Odd signal
9. Define continuous time signals and classify them.

Continuous time signals are defined for a continuous of values of the Independent
variable. In the case of continuous time signals the independent Variable is continuous.
For example:
(i) A speech signal as a function of time
(ii) Atmospheric pressure as a function of altitude
Classification of continuous time signal:
(i) Periodic and A periodic signal
(ii) Even and Odd signal
10. Define discrete time unit step &unit impulse.
Discrete time Unit impulse is defined as
[n]= {1, n=0; }
{ 0, otherwise.}
Unit impulse is also known as unit sample.
Discrete time unit step signal is defined by
U[n]={0,n=0 }
{1,n>= 0 }

11. Define even and odd signal.


A discrete time signal is said to be even when,
x[-n]=x[n].
The continuous time signal is said to be even when,
x(-t)= x(t)
For example, Cosine wave is an even signal. The discrete time signal is said to be odd
when
x[-n]= -x[n]
The continuous time signal is said to be odd when
x(-t)= -x(t)
Odd signals are also known as nonsymmetrical signal. Sine wave signal is an odd signal.
12. Define Energy and power signal.
A signal is said to be energy signal if it have finite energy and zero power. A
signal is said to be power signal if it have infinite energy and finite power. If the above
two conditions are not satisfied then the signal is said to be Neither energy nor power
signal
13. What is analog signal?
The analog signal is a continuous function of independent variables. The analog
Signal is defined for every instant of independent variable and so magnitude of
Independent variable is continuous in the specified range. here both the independent
Variable and magnitude are continuous.
14. What is digital signal?
The digital signal is same as discrete signal except that the magnitude of signal is
Quantized
15. What are the different types of signal representations?
a. Graphical representation

b. Functional representations
c. Tabular representation
d. Sequence representation.
16. Define periodic and non periodic discrete time signals?
If the discrete time signal repeated after equal samples of time then it is called
periodic signal. When the discrete time signal x[n] satisfies the condition x[n+N]=x(n),
then it is called periodic signal with fundamental period N samples. If x(n) * x(n+N) then
it is called non periodic signals.
17. State the classification of discrete time signals.
The types of discrete time signals are
* Energy and power signals
* Periodic and A periodic signals
* Symmetric (Even) and Ant symmetric (Odd) signals

18. Define energy and power signal.


If E is finite i.e. 0<E< , then x (n) is called energy signal.
If P is finite i.e. 0<P< , then the signal x(n) is called a power signal.
19. What are all the blocks are used to represent the CT signals by its samples?
* Sampler
* Quantizer
20. Define sampling process.
Sampling is a process of converting Ct signal into Dt signal.
21. Mention the types of sampling.
* Up sampling
*Down sampling
22. What is meant by quantizer?
It is a process of converting discrete time continuous amplitude into discrete time
discrete amplitude.
23. Define system function?
The ratio between z transform of out put signal y(z) to z transform of input signal
x(z) is called system function of the particular system.
24. List out the types of quantization process.
* Truncation
* Rounding
25. Define truncation.
Truncating the sequence by multiplying with window function to get the finite
value
26. State sampling theorem.
The sampling frequency must be at least twice the maximum frequency present in

the signal.
That is Fs = > 2fm
Where, Fs = sampling frequency Fm
= maximum frequency
27. Define nyquist rate.
It is the minimum rate at which a signal can be sampled and still reconstructed
from its samples. Nyquist rate is always equal to 2fm.
28. Define aliasing or folding.
The superimposition of high frequency behavior on to the low frequency behavior
is referred as aliasing. This effect is also referred as folding.
29. What is the condition for avoid the aliasing effect?
To avoid the aliasing effect the sampling frequency must be twice the maximum
frequency present in the signal.
30. What is meant by interpolation?
It is also referred as up sampling. that is , increasing the sampling rate.
31. What is an anti-aliasing filter?
A filter that is used to reject high frequency signals before it is sampled to remove
the aliasing of unwanted high frequency signals is called an ant aliasing filter.
32. Mention the types of sample/hold?
* zero order hold
* first order hold
33. What is meant by sampling rate?
Sampling rate = number of samples / second.
34. What is meant by step response of the DT system?
The output of the system y(n) is obtained for the unit step input u(n) then it is said
to be step response of the system.
35. Define Transfer function of the DT system.
The Transfer function of DT system is defined as the ratio of Z transform of the
system output to the input. That is , H(z)=Y(z)/X(z).
36. Define impulse response of a DT system.
The impulse response is the output produced by DT system when unit impulse is
applied at the input. The impulse response is denoted by h(n). The impulse response h(n) is
obtained by taking inverse Z transform from the transfer function H(z).
37. What are the properties of convolution?
I. Commutative
ii. Associative.
iii. Distributive

38. State the Commutative properties of convolution?


Commutative property of Convolution is,
x(n)*h(n) = h(n)*x(n)
39. State the Associative properties of convolution
Associative Property of convolution is,
[x(n)*h1n)]*h2(n)=x(n)*[h1(n)*h2(n)]
40. State Distributive properties of convolution The
Distributive Property of convolution is,
{x(n)*[h1(n)+ h2(n)]}= [x(n)*h1(n) + x(n)*h2(n)]

41. Define causal LTI DT system.


For a LTI system to be causal if h(n)=0, for n<0.
42. What are the steps involved in calculating convolution sum?
The steps involved in calculating sum are
Folding
Shifting
Multiplication
Summation
43. What is the condition for stable LTI DT system?
A LTI system is stable if, Here, the summation is absolutely sum able.
44. Define a causal system.
The causal system generates the output depending upon present &past inputs
only. A causal system is non anticipatory.
45. What is meant by linear system?
A linear system should satisfy superposition principle. A linear system should
satisfy F[ax1(n)+bx2(n)] = a y 1(n)+by2(n)
Where, y1(n)=F[x1(n)]
y2(n)=F[x2(n)]
46. Define time invariant system.
1 .A system is time invariant if the behavior and characteristics of the system are
fixed over time.
2. A system is time invariant if a time shift in the input signal results in an
identical time shift in the output signal.
3. For example, a time invariant system should produce y(t-t0)as the output when
x(n-no) is the input.

UNIT II DISCRETE
DISCRETE TIME SYSTEM ANALYSIS
1. Define z transform?
The Z transform of a discrete time signal x(n) is defined as,

-j

Where, z is a complex variable. In polar form z=re

2. What is meant by ROC?


The region of convergence (ROC) is defined as the set of all values of z for which
X(z) converges.
3. Explain about the roc of causal and anti
anti-causal infinite sequences?
For causal system the roc is exterior to the circle of radius r. For
anti causal system it is interior to the circle of radius r.
4. Explain about the roc of causal and anti causal finite ssequences
For causal system the roc is entire z plane except z=0. For
anti causal system it is entire z plane except z=
z= .

5. What are the properties of ROC?


a. The roc is a ring or disk in the z plane centered at the origin.
b. The roc cannot contain any pole.
c. The roc must be a connected region
d. The roc of an LTI stable system contains the unit circle.
6. Explain the linearity property of the z transform
If z{x1(n)}=X1(z) and z{X2(n)}=x2(z) then, z{ax1(n)+bx2(n)}=aX1(z)+bX2(z)
a&b are constants.
7. State the time shifting property of the z transform If
k
z{x(n)}=X(z) then z{x(n
z{x(n-k)}=z- X(z)

8. State the scaling property of the z transform


n
-1
If z{x(n)}=X(z) then z{a x(n)}=X(a z)
9. State the time reversal property of the z transform If
z{x(n)}=X(z) then z{x(
z{x(-n)}=X(z-1)

10. Explain convolution property of the z transform


If z{x(n)}=X(z) & z{h(n)}=H(z) then, z{x(n)*h(n)}=X(z)H(z)
11. Define system function?
The ratio between z transform of out put signal y(z) to z transform of input signal
x(z) is called system function of the particular system.
12. What are the conditions of stability of a causal system?
All the poles of the system are with in the unit circle.
The sum of impulse response for all values of n is bounded.

13. What are the different methods of evaluating inverse ztransform? It can be evaluated using several methods.
Long division method
Partial fraction expansion method
Residue method
Convolution method
14. What is the need for Z-transform?
Z-transform is used for analysis the both periodic and a periodic signals.
15. Give the Z-transform of unit sample sequence
(n). Z[ (n) ] = 1
16. Define zeros.
The zeros of the system H(z) are the values of z for which H(z) = 0.
17. Define poles.
The poles of the system H(z) are the values of z for which H(z) = .
18. What is the z-transform of A (n-m)?
Z [ A (n-m) ] =1.
19. Find Z transform of x(n)={1,2,3,4}?
x(n)= {1,2,3,4}
X(z)= x(n)z-n
1+2z-1+3z-2+4z-3.
1+2/z+3/z2+4/z3
20. State the convolution properties of Z transform?
The convolution property states that the convolution of two sequences in time
domain is equivalent to multiplication of their Z transforms
.
21. What z transform of (n-m)? By time shifting
property
Z[A (n-m)]=AZ-m sinZ[ (n)] =1
22. Obtain the inverse z transform of X(z)=1/z-a,|z|>|a|?
Given X(z)=z-1/1-az-1
By time shifting property
X(n)=an.u(n-1)

UNIT III DISCRETE TRANSFORMS


1. List any four properties of DFT
a. Periodicity
b. Linearity
c. Time reversal
d. Circular time shift
e. Duality
f. Circular convolution
g. Symmetry
h. Circular symmetry
2. State periodicity property with respect to DFT.
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.
3. State periodicity property with respect to DFT.
If X1(k) and X2(k) are N-point DFTs of finite duration sequences x1(n) and x2(n), then
DFT [a X1(n) + b X2(n)] = a X1(k) + b X2(k), a, b are constants.
4. State time reversal property with respect to DFT.
If DFT[x(n)] =X(k), then
DFT[x((-n))N] = DFT[x(N-n)] = X((-k))N = X(N-k)
5. Define circular convolution.
Let x1(n) and x2(n) are finite duration sequences both of length n with DFTs x1(k) and
x2(k). If X3(k) = X1(k) X2(k), then the sequence X3(k) can be obtained by circular
convolution.
6. What is the need for DFT?
DFT is used for analysis the both periodic and a periodic signals.
7. What is zero padding? What are its uses?
Let the sequence x (n) has a length L. If we want to find the N-point DFT (N>L) of the
sequence x(n), we have to add (N-L) zeros to the sequence x(n). This is known as Zero
padding.
The uses of zero padding are
1) We can get better display of the frequency spectrum.
2) With zero padding the DFT can be used in linear filtering.
8. Why FFT is needed?
The direct evaluation of DFT requires N complex multiplications and N - N complex
additions. Thus for large values of N direct evaluation of the DFT is difficult. By using FFT
algorithm the number of complex computations can be reduced. So we use FFT
9. What is FFT?
The Fast Fourier Transform is an algorithm used to compute the DFT. It makes use of
the symmetry and periodicity properties of twiddle factor to effectively reduce the DFT
computation time. It is based on the fundamental principle of decomposing the mutation of
DFT of a sequence of length N into successively smaller DFTs.
10. How many multiplications and additions are required to compute N point DFT using
Radix-2 FFT?
The number of multiplications and additions required to compute N point DFT
Using radix-2 FFT are N log2 N and N/2 log2 N respectively.
11. What is meant by radix-2 FFT?
The FFT algorithm is most efficient in calculating N point DFT. If the number
of output points N can be expressed as a power of 2 that is N=2M, where M is an
integer, then this algorithm is known as radix-2 algorithm.
12. What is DIT algorithm?
Decimation-In-Time algorithm is used to calculate the DFT of a N point
sequence. The idea is to break the N point sequence into two sequences, the DFTs of

which can be combined to give the DFt of the original N point sequence. This algorithm
is called DIT because the sequence x(n) is often spitted into smaller sub- sequences.
13. What DIF algorithm?
It is a popular form of the FFT algorithm. In this the output sequence X(k) is
divided into smaller and smaller sub-sequences , that is why the name Decimation In
Frequency.
14. What are the applications of FFT algorithm?
The applications of FFT algorithm includes
1) Linear filtering
2) Correlation
3) Spectrum analysis
15. Distinguish between linear convolution and circular convolution of two sequences.
S. No
1.

Linear convolution
If x(n) is a sequence of L number of samples and h(n)
with M number of samples, after convolution y(n) will
have N=L+M-1 samples.

Circular convolution
If x(n) is a sequence of L
number of samples and h(n)
with M samples, after
convolution y(n) will have
N=max(L,M) samples.

2.

It can be used to find the response of a linear filter.

3.

Zero padding is not necessary to find the response of a


linear filter.

It cannot be used to find the


response of a filter.
Zero padding is necessary
to find the response
of a filter.

16. What are the differences and similarities between DIF and DIT
algorithms? Differences:
1) The input is bit reversed while the output is in natural order for
DIT, whereas for DIF the output is bit reversed while the input is in natural
order.
17. 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. Both
algorithms can be done in place and both need to perform bit reversal at some place during
the computation.
18. What is meant by radix-2 FFT?

If the number of output points N can be expressed as a power of 2, i.e., N = 2


M is an integer then this algorithm is known as radix-2 algorithm.

Where

19. What is DIT radix-2 algorithm?


The radix 2 DIT FFT is an efficient algorithm for computing DFT. The idea is
to break N point sequence in to two sequences, the DFT of which can be combined to
give DFT of the original N-point sequence. Initially the N point sequence is divided
in to two N/2 point sequences, on the basis of odd and even and the DFTs of them are
evaluated and combined to give N-point sequence. Similarly the N/2 DFT s are
divided and expressed in to the combination of N/4 point DFTs. This process is
continued until we left with 2-point DFTs.
20. What is DIF radix-2 algorithm?
1. The radix 2 DIFFFT is an efficient algorithm for computing DFT in this the out put
sequence x(k) is divided in to smaller and smaller.
2. The idea is to break N point sequence in to two sequences ,x1(n) and x2(n)
consisting of the first N/2 points of x(n)and last N/2 points of x(n) respectively. Then we find
N/2 point sequences f(n) and g(nSimilarly).
3. The N/2 DFT s are divided and expressed in to the combination of N/4 point DFT s.
This process is continued until we left with 2-point DFTs.
21. What are the differences between DIT and DIF algorithms?
* For DIT the input is bit reversed and the output is in natural order, and in DIF the

input is in natural order and output is bit reversed.


* In butterfly the phase factor is multiplied before the add and subtract operation but in
DIF it is multiplied after add-subtract operation.
22. What is meant by in place in DIT and DFT algorithm?
An algorithm that uses the same location to store both the input and output sequence is
called in-place algorithm.
23. Differentiate DTFT and DFT
DTFT output is continuous in time where as DFT output is Discrete in time.
24. Differentiate between DIT and DIF algorithm
DIT Time is decimated and input is bi reversed format output in natural order DIF
Frequency is decimated and input is natural order output is bit reversed Format.
How many stages are there for 8 point DFT 3 stages
25. How many multiplication terms are required for doing DFT by expressional?
Expression Method and FFT method
2
Expression N FFT - N /2 log N
26. What is DFT?
It is a finite duration discrete frequency sequence which is obtained by sampling one
period of Fourier transform. Sampling is done at N equally spaced points over the period
extending from = 0 to = 2.
27. What is the DFT of unit impulse (n) ? The
DFT of unit impulse (n) is unity.

28. Why the result of circular and linear convolution is not same ?
Circular convolution contains same number of samples as that of x(n) and h(n),
while in linear convolution, number of samples in the result(N) are,
N =L + M 1.
Where, L = Number of samples in x(n).
M = Number of samples in h(n).
That is why the result of linear and circular convolution is not same.
29. How to obtain same result from linear and circular convolution?
* Calculate the value of N, that means number of samples contained in linear
convolution.
* By doing zero padding make the length of every sequence equal to number of samples
contained in linear convolution.
* Perform the circular convolution. The result of linear and circular convolution will be
same.
30. How will you perform linear convolution from circular convolution?
* Calculate the value of N, that means number of samples contained in linear
convolution.
* By doing zero padding make the length of every sequence equal to number of samples
contained in linear convolution.
* Perform the circular convolution. The result of linear and circular convolution will be
same.
31. What methods are used to do linear filtering of long data sequences?
* Overlap save method.
* Overlap add method.
32. What is the disadvantage of direct computation of DFT?
2
2
complex multiplication and N N
For the computation of N-point DFT, N
complex additions are required. If the value
of N is large then the number of
computations will go into lakhs. This proves inefficiency of direct DFT computation.
33. What is the way to reduce number of arithmetic operations during DFT computation?
Numbers of arithmetic operations involved in the computation of DFT are greatly
reduced by using different FFT algorithms as follows,

Radix-2 FFT algorithm.


- Radix-2 Decimation In Time (DIT) algorithm.
- Radix-2 Decimation In Frequency (DIF) algorithm.
Radix-4 FFT algorithm.
34. What are the properties of twiddle factor?
* Twiddle factor is periodic.
* Twiddle factor is symmetric.
35. What is up sampling process and what its effect?
Addition of one zero after each sample in x(n) is called up sampling process. Due to this
process, the entire DFT repeats one time.
36. How linear filtering is done using FFT?
Correlation is the basic process of doing linear filtering using FFT. The correlation is
nothing but the convolution with one of the sequence, folded. Thus, by folding the sequence
h(n), we can compute the linear filtering (convolution) using FFT.

UNIT IV DESIGN OF DIGITAL FILTERS


1. What is a digital filter?
A digital filter is a device that eliminates noise and extracts the signal of interest from
other signals.
2. Analog filters are composed of which parameters?
* pass band
* stop band
* Cut-off frequency
3. Define pass band.
It passes certain range of frequencies. In this, attenuation is zero.
4. Define stop band.
It suppresses certain range of frequencies. In this, attenuation is infinity. 5.
What is mean by cut-off frequency?
This is the frequency which separates pass band and stop band.
6. What is the difference between analog and digital filters?
Analog filters are designed using analog components (R,L,C) while digital filters are
implemented using difference equation and implemented using software.
7. What are the basic types of analog filters?
* Low pass filter - LPF
* High pass filter - HPF
* Band pass filter - BPF
* Band stop filter - BSF
8. What is the condition for digital filter to be realize?
The impulse response of filter should be causal, h(n) = 0 for n<0.
9. Why ideal frequency selective filters are not realizable?
Ideal frequency selective filters are not realizable because they are non-causal. That is,
its impulse response is present for negative values of n also.
10. For IIR filter realization what is required?
Present, past, future samples of input and past values of output are required. 11.
Why IIR systems are called recursive systems?
Because the feedback connection is present from output side to input
12. Which types of structures are used to realize IIR systems?
* Direct form structure
* Cascade form structure
* Parallel form structure
13. Why direct form-II structure is preferred most and why?
The numbers of delay elements are reduced in direct form-II structure compared to
direct form-I structure. That means the memory locations are reduced in direct form-II
structure.

14. Why direct form-I and direct form-II are called as direct form structures?
The direct form-I and direct form-II structures are obtained directly from the
corresponding transfer function without any rearrangements. So these structures are called as
direct form structures.
15. What is advantage of direct form structure?
Implementation of direct form is very easy.

16. Give the disadvantage of direct form structure?


Both direct form structures are sensitive to the effects of quantization errors in the
coefficients. So practically not preferred
17. What is the use of transpose operation?
If two digital structures have the same transfer function then they are called as
equivalent structures. By using the transpose operation, we can obtain equivalent structure from
a given realization structure.
18. What is transposition or flow graph reversal theorem?
If we reverse the directions of all branch transmittances and interchange input and
output in the flow graph then the system transfer function remains unchanged.
19. How a transposed structure is obtained?
* Reverse all signal flow graph directions.
* Change branching nodes into adders and vice-versa.
* Interchange input and output.
20. Why feed back is required in IIR systems?
It is required to generate infinitely long impulse response in IIR systems.
21. Write the expression for order of Butterworth filter?

22. Write the expression for the order of chebyshev filter?

23. Write the various frequency transformations in analog domain?

24. Write the steps in designing chebyshev filter?


1. Find the order of the filter.
2. Find the value of major and minor axis.
3. Calculate the poles.
4. Find the denominator function using the above poles.
5. The numerator polynomial value depends on the value of n.
If n is odd: put s=0 in the denominator polynomial.
2 1/2
If n is even put s=0 and divide it by (1+e )

25. Write down the steps for designing a Butterworth filter?

26. State the equation for finding the poles in chebyshev filter.

27. State the steps to design digital IIR filter using bilinear method.

28. What is warping effect or frequency warping?


For smaller values of w there exist linear relationship between w and .but for Larger
values of w the relationship is nonlinear. This introduces distortion in the Frequency axis. This
effect compresses the magnitude and phase response. This Effect is called warping effect.
29. Write a note on pre warping or pre scaling.
The effect of the non linear compression at high frequencies can be compensated. When
the desired magnitude response is piecewise constant over frequency, this Compression can be
compensated by introducing a suitable rescaling or prewar Ping the critical frequencies.
30. Give the bilinear transform equation between s plane and z plane s=2/T
(z-1/z+1)
31. Why impulse invariant method is not preferred in the design of IIR filters other Than
low pass filter?
In this method the mapping from s plane to z plane is many to one. Thus there is an
infinite number of poles that map to the same location in the z plane, producing an aliasing
effect. It is inappropriate in designing high pass filters. Therefore this method is not much
preferred.
32. What is meant by impulse invariant method?
In this method of digitizing an analog filter, the impulse response of the resulting digital
filter is a sampled version of the impulse response of the analog filter. For
e.g. if the transfer function is of the form, 1/s-p, then
-pT -1
H (z) =1/1-e z
33. What do you understand by backward difference?
One of the simplest methods of converting analog to digital filter is to approximate the
differential equation by an equivalent difference equation.
d/dt(y(t)/t=nT=(y(nT)-y(nT-T))/T
34. What are the properties of chebyshev filter?
1. The magnitude response of the chebyshev filter exhibits ripple either in the stop band
or the pass band.
2. The poles of this filter lies on the ellipse.
35. Give the Butterworth filter transfer function and its magnitude characteristics for
Different orders of filter.

36. Give the equation for the order N, major, minor axis of an ellipse in case of
chebyshev filter?

37. How can you design a digital filter from analog filter?
Digital filter can de designed from analog filter using the following methods
1. Approximation of derivatives
2. Impulse invariant method (IIM)
3. Bilinear transformation (BLT)
38. write down bilinear transformation.
s=2/T (z-1/z+1)

39. What is a disadvantage of BLT method?


The mapping is non-linear and because of this, frequency warping effect takes
place.
40. List the Butterworth polynomial for various orders. N
Denominator polynomial
1 S+1
2
2 S +.707s+1
2
3 (s+1)(s +s+1)
2
2
4 (s +.7653s+1)(s +1.84s+1)
2
2
5 (s+1)(s +.6183s+1)(s +1.618s+1)
2
2
2
6 (s +1.93s+1)(s +.707s+1)(s +.5s+1)
2
2
2
7 (s+1)(s +1.809s+1)(s +1.24s+1)(s +.48s+1)
41. Differentiate Butterworth and Chebyshev filter. Butterworth
damping factor 1.44 and chebyshev is 1.06
Butterworth is flat response .but chebyshev is damped response.

42. What is filter?


Filter is a frequency selective device ,which amplify particular range of frequencies and
attenuate particular range of frequencies.
43. What are the types of digital filter according to their impulse response?
IIR(Infinite impulse response )filter
FIR(Finite Impulse Response)filter.
44. How phase distortion and delay distortion are introduced?
1. The phase distortion is introduced when the phase characteristics of a filter is
Nonlinear with in the desired frequency band
2. The delay distortion is introduced when the delay is not constant with in the
Desired frequency band
45. Define IIR filter.
The filters designed by considering all the infinite samples of impulse response are
called IIR filter.
46. What is the limitation of approximation of derivative method?
It is suitable only for designing of low pass and band pass IIR digital filters with
relatively small resonant frequencies.
47. What are the reasons to use elliptic filters?
It has smallest transition bandwidth and also it is more efficient.

UNIT V DSP HARDWARE


1. What are the factors that influence selection of DSPs?
* Architectural features
* Execution speed
* Type of arithmetic
* Word length
2. What are the classification digital signal processors? The
digital signal processors are classified as
(i)
General purpose digital signal processors.
(ii)
Special purpose digital signal processors.
3. What are the applications of PDSPs?
Digital cell phones, automated inspection, voicemail, motor control, video
conferencing, noise cancellation, medical imaging, speech synthesis, satellite
communication etc.
4. Give some examples for fixed point DSPs.
TM32OC50, TMS320C54, TMS320C55, ADSP-219x, ADSP-219xx..
5. Give some example for floating point DSPs?

TMS320C3x, TMS320C67x, ADSP-21xxx

6. What is pipelining?
Pipelining a processor means breaking down its instruction into a series of discrete
pipeline stages which can be completed in sequence by specialized hardware.
7. What is pipeline depth?
The number of pipeline stages is referred to as the pipeline depth.
8. What are the advantages of VLIW architecture?
Advantages of VLIW architecture
a. Increased performance
b. Better compiler targets
c. Potentially easier to program
d. Potentially scalable
e. Can add more execution units; allow more instructions to be packed into the
VLIW instruction.
9. What are the disadvantages of VLIW architecture?
Disadvantages of VLIW architecture
f. New kind of programmer/compiler complexity
g. Program must keep track of instruction scheduling
h. Increased memory use
i. High power consumption
10. What is the pipeline depth of TMS320C50 and TMS320C54x?
TMS320C50 4
TMS320C54x 6

11. What are the different buses of TMS320C5x?


The C5x architecture has four buses
j. Program bus (PB)
k. Program address bus (PAB)
l. Data read bus (DB)
m. Data read address bus (DAB)
12. Give the functions of program bus?
The program bus carries the instruction code and immediate operands from program
memory to the CPU.
13. Give the functions of program address bus?
The program address bus provides address to program memory space for both read and
write.
14. Give the functions of data read bus?
The data read bus interconnects various elements of the CPU to data memory space.
16. Give the functions of data read address bus?
The data read address bus provides the address to access the data memory space.
17. What are the different stages in pipelining?
n. The fetch phase
o. The decode phase
p. Memory read phase
q. The execute phase

18. List the various registers used with ARAU. Eight


auxiliary registers (AR0 AR7) Auxiliary
register pointer (ARP)
Unsigned 16-bit ALU

19. What are the elements that the control processing unit of C5x consists of The central
processing unit consists of the following elements:
r. Central arithmetic logic unit (CALU)
s. Parallel logic unit (PLU)

t. Auxiliary register arithmetic unit (ARAU)


u. Memory mapped registers
v. Program controller
20. What is the function of parallel logic unit?
The parallel logic unit is a second logic unit that executes logic operations on data
without affecting the contents of accumulator.
21. List the on chip peripherals in C5x.
The on-chip peripherals interfaces connected to the C5x CPU include
w. Clock generator
x. Hardware timer
y. Software programmable wait state generators
z. General purpose I/O pins
aa. Parallel I/O ports
bb. Serial port interface
cc. Buffered serial port
dd. Time-division multiplexed (TDM) serial port
ee. Host port interface
ff. User unmask able interrupts
22. What are the arithmetic instructions of C5x?
ADD, ADDB, ADDC, SUB, SUBB, MPY, MPYU
23. What are the shift instructions?
ROR, ROL, ROLB, RORB, BSAR.
24. What are the general purpose I/O pins?
Branch control input (BIO) External flag
(XF)
25. What are the logical instructions of C5x? AND,
ANDB, OR, ORB, XOR, XORB

26. What are load/store instructions?


LACB, LACC, LACL, LAMM, LAR, SACB, SACH, SACL, SAR, SAMM.
27. Mention the addressing modes available in TMS320C5X processor?
1. Direct addressing mode
2. Indirect addressing mode
3. Circular addressing mode
4. Immediate addressing
5. Register addressing
6. Memory mapped register addressing
7.
28. Give the features of DSPs?
* Architectural features
* Execution speed
* Type of arithmetic
* Word length
29. What is function of NOP instruction?
* NOP- No operation
* Perform no operation.
30. What is function of ZAC instruction? ZAC
Zero accumulator
Clear the contents of accumulator to zero.

31. Give the function of BIT instruction. BIT


Test bit
Copy the specified bit of the data memory value to the TC bit in ST1.
32. Mention the function of B instruction.
B Branch conditionally.Branch to the specified program memory address. Modify the
current AR and ARP as specified.

33. What is use of ADD instruction?


ADD Add to accumulator with shift.
Add the content of addressed data memory location or an immediate value of
accumulator, if a shift is specified, left-shift the data before the add. During shifting, low-order
bits are Zero-filled, and high-order bits are sign extended if SXM=1.
34. Give the advantages of DSPs?
Architectural features, Execution speed, Type of arithmetic, Word length
35. Give the applications of DSP Processors?
Digital cell phones, automated inspection, voicemail, motor control, video conferencing,
noise cancellation, medical imaging, speech synthesis, satellite communication etc.

UNIT I INTRODUCTION
PART A
1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
11.
12.
13.
14.

15.
16.
17.
18.
19.
20.
21.
22.

What is signal and Signal processing?


List the advantages of Digital Signal processing.
Mention few applications of Digital Signal processing.
Classify discrete time signals.
What are Energy and Power signals?
What do you mean by periodic and Aperiodic signals?
When a signal is said to be symmetric and Anti symmetric?
What are deterministic and random signals?
What are the elementary signals?
What are the different types of representation of discrete time signals?
Draw the basic block diagram of digital signal processing of analog signals.
What are the basic time domain operations of discrete time signal?
What is the significance of unit sample response of a system?
Classify discrete time systems.
n
-n
Whether the system defined by the impulse response h(n) = 2 u(-n) + 2 u(n) is causal ? Justify
your answer.
-n
Compute the energy of the signal x(n) = 2 u(n)
n
Compute the energy of the signal x(n) = (0.5) u(n)
Define convolution.
List out the properties of convolution.
What do you mean by BIBO stable?
What is linear time invariant system?
Compute the convolution of x(n) = {1,2,1,-1} and h(n) = {1,2,1,-1} using

tabulation method.

23.
24.
25.
26.
27.
28.
29.

Check whether the system defined by h (n) = [5 (1/2) + 4 (1/3) ] u(n) is stable.
Differentiate between analog, discrete, quantized and digital signals.
Differentiate between analog
and digital signals.
Differentiate between one dimensional and two dimensional signal with an example for each.
Name any four elementary time domain operations for discrete time signals.
2
For the signal f (t) = 5 cos (5000t) + sin (3000t), determine the minimum sampling rate for
recovery without aliasing.
2 = 3000 = 2F2
1 = 5000 = 2F1
F1 = 2.5 kHz
Fmax = 2.5 kHz

F2=1.5 kHz

According to sampling theorem Fs 2 Fmax


So, Fs = 5 kHz

30. For the signal f (t) = cos (4000t) + 2 sin (6000t), determine the minimum sampling rate for
recovery without aliasing.
1 = 4000 = 2F1
2 = 6000 = 2F2
F1 = 2 kHz

F2 =3 kHz

Fmax = 3 kHz
According to sampling theorem Fs 2 Fmax
So, Fs = 6 kHz
31. What is sampling?
32. State sampling theorem and what is Nyquist frequency?
Sampling theorem - Fs 2 Fmax
Nyquist frequency or Nyquist rate FN = 2 Fmax
33.
34.
35.
36.
37.
38.
39.
40.
41.

What is known as aliasing?


Define the criteria to perform sampling process without aliasing.
Differentiate between anti aliasing and anti imaging filters.
What are the effects of aliasing?
What is anti aliasing filter? What is the need for it?
Draw the basic block diagram of a digital processing of an analog signal.
Draw the basic structure of linear constant difference equation.
What is sample and Hold circuit?
If a minimum signal to noise ratio (SQR) of 33 dB is desired, how many bits per code word are
required in a linearly quantized system?
SQR =1.76 +6.02b
SQR given is 33 dB
1.76+6.02b = 33 dB
6.02 b = 31.24
b = 5.18 = 6 bits

42. Determine the number of bits required in computing the DFT of a 1024 point sequence with
an SQR of 30 dB
10
The size of the sequence is N = 1024 = 2
2
2
2b
2
SQR is X / q = 2 / N
2
2
2b
2
10 log [X / q ] = 10 log [2 / N ]
2
20
N =2
2
2
2b
20
10 log [X / q ] = 10 log [2 / 2 ]
2b - 20
SNR = 10 log [2
] = 30 dB
3(2b-20) =30
b=15 bits is the precision for both multiplication and addition
43. Determine the system described by the equation y (n) = n x (n) is linear or not.
44. What is the total energy of the discrete time signal x(n) which takes the value of unity at n =1,0,1?
45. Draw the signal x(n) = u(n) u(n-3)
PART - B
1. For each of the following systems, determine whether the system is static stable, causal, linear
and time invariant
x(n)
a. y(n) = e
b. y(n) = ax(n) +b
n
c. y(n) = k=n0 x(k)
n+1
d. y(n) = k= - x(k )
2
e. y(n) = n x (n)
f. y(n) = x(-n+2)
g. y(n) = nx(n)
h. y(n) = x(n) +C
i. y(n) = x(n) x(n-1)
j. y(n) = x(-n)
k. y(n) = x(n) where x(n) = [x(n+1) x(n)]
l. y(n) = g(n) x(n)

m.
n.
o.
p.

y(n) = x(n )
2
y(n) = x (n)
y(n) = cos x(n)
y(n) = x(n) cos 0n

2. Compute the linear convolution of h(n) = {1,2,1} and x(n) ={1,-3,0,2,2}


3. Explain the concept of Energy and Power signals and determine whether the following are energy
or power signals
n
a. x(n) = (1/3) u(n)
n
b. x(n) = sin ( / 4)
4. The unit sample response h(n) of a system is represented by
2
h (n) = n u(n+1) 3 u(n) +2n u(n-1) for -5 n 5. Plot the unit sample response.
5. State and prove sampling theorem. How do you recover continuous signals from its
samples? Discuss the various parameters involved in sampling and
reconstruction.
6. What is the input x(n) that will generate an output sequence
y(n) = {1,5,10,11,8,4,1} for a system with impulse response h(n) = {1,2,1}
n

7. Check whether the system defined by h(n) = [5 (1/2) +4(1/3) ] u(n) is stable?
8. Explain the analog to digital conversion process and reconstruction of analog signal from digital
signal.
9. What are the advantages and disadvantages of digital signal processing compared with analog
signal processing?
10. Classify and explain different types of signals.
11. Explain the various elementary discrete time signals.
12. Explain the different types of mathematical operations that can be performed on a discrete time
signal.
13. Explain the different types of representation of discrete time signals.
14. Determine whether the systems having the following impulse responses are causal and stable
n
a. h(n) = 2 u(-n)
b. h(n) = sin n / 2
c. h(n) = sin n + (n)
2n
d. h(n) = e u(n-1)
15. For the given discrete time signal
x (n) = { -0.5,0.5, for n = -2, -1
1,
n=0
3, 2, 0.4

n > 0}

Sketch the following a) x (n-3), b) x (3-n) c) x (2n) d) x (n/2) e) [x (n) + x (-n)] / 2


16.
17.
18.
19.

Find the convolution of x (n) = a u (n), a < 1 with h(n) = 1for 0 n N-1
Draw the analog, discrete, quantized and digital signal with an example.
Explain the properties of linearity and stability of discrete time systems with examples.
The impulse response of a linear time invariant system is h (n) = {1, 2, 1,-1}.

Determine the response of the system to the input signal x (n) = {1, 2, 3, 1}.

Determine whether or not each of the following signals are periodic. If a signal is
Periodic specify its fundamental time period.
i.
ii.
iii.
iv.

x(t) = 2 cos 3 t
x(t) = sin 15 t + sin 20 t
x(n) = 5 sin 2n
x(n) = cos (n/8) cos (n / 8)

**********************************************************************

UNIT - II DISCRETE TIME SYSTEM ANALYSIS


PART - A
1. Define Z-transform
2. Define ROC in Z-transform
3. Determine Z-transform of the sequence x(n)= {2,1,-1,0, 3}

4.
5.
6.
7.
8.
9.
10.
11.
12.
13.
14.

Determine Z-transform of x(n) = - 0.5 u (-n-1)


n
Find Z- transform of x(n) = - b u(-n-1) and its ROC
n
Find Z- transform of x(n) = a u(n) and its ROC
What are the properties of ROC in Z- transform?
State the initial value theorem of Z- transforms.
State the final value theorem of Z- transforms.
-1
Obtain the inverse Z transform of X(Z) = log ( 1 + Z ) for _Z _ < 1
Obtain the inverse Z transform of X(Z) = log ( 1-2z) for _Z _ < 1/2
What is the condition for stability in Z-domain?
Mention the basic factors that affect thr ROC of z- transform.
Find the z- transform of a digita limpulse signal and digital step signal
PART - B

1. Determine the Z-transform and ROC of


n
a. x(n) = r cos n u(n)
2 n
b. x(n) = n a u(n)
n
n
c. x(n) = -1/3 (-1/4) u(n) 4/3 (2) u(-n-1)
n
n
n
d. x(n) = a u(n) + b u(n) + c u(-n-1) , |a | < | b| < |c|
e. x(n) = cos n u(n)
f. x(n) = sin 0n . u(n)
n
g. x(n) = a u(n)
n
n
h. x(n) = [ 3 (2 ) 4 (3 )] u(n)
2. Find the inverse Z-transform of
3
a. X(z) = z (z+1) / (z-0.5)
-1
-1
-2
b. X(z) = 1+3z / 1 + 3z + 2z
-1
-2
c. H(z) = 1 / [1 - 3z + 0.5z ]
|z | > 1
2
d. X(z) = [z (z - 4z +5)] / [(z-3) (z-2) ( z-1)] for ROC |2 | < | z| < |3|, |z| > 3, |z|< 1
3. Determine the system function and pole zero pattern for the system described by
difference equation y (n) -0.6 y (n-1) +0.5 y (n-2) = x (n) 0.7 x (n-2)
4. Determine the pole zero plot for the system described by the difference
equation y (n) 3/4 y (n-1) +1/8 y (n-2) = x(n) x(n-1)
5. Explain the properties of Z-transform.
6. Perform the convolution of the following two sequences using Z-transforms. x(n) =
n
n
0.2 u(n) and h(n) = (0.3) u(n)
7. A causal LTI system has an impulse response h(n) for which the Z-transform is given
-1

-1

-1

by H(z) = (1+z ) / [(1 + 1/2z ) (1 + 1/4z ). What is the ROC of H (z)? Is the system
stable? Find the Z-transform X (z) of an input x (n) that will produce the output y(n) =
n
n
-1/3 (-1/4) u(n) 4/3 (2) u(-n-1).Find the impulse response h (n) of the system.
8. Solve the difference equation y(n) -3y(n-1) 4y(n-2) = 0, n 0 ,y(-1) = 5
9. Compute the response of the system y(n) = 0.7 y(n-1)-0.12y(n-2) +x(n-1)+
(n-2)to the input x(n) = n u(n)
10. What is ROC? Explain with an example.
11. A causal LTI IIR digital filter is characterized by a constant co-efficient difference equation given
by y(n) = x(n-1)-1.2x(n-2)+x(n-3)+1.3 y(n-1) 1.04 y(n-2)+0.222y(n-3),obtain its transfer
function.
12. Determine the system function and impulse response of the system described by the
difference equation y(n) = x(n) +2x(n-1)- 4x(n-2) + x(n-3)
13. Solve the difference equation y(n) - 4y(n-1) - +4 y(n-2) = x(n) x(n-1) with the initial
condition y(-1) = y(-2) = 1
14. Find the impulse response of the system described by the difference equation y(n) = 0.7 y(n-1)
-0.1 y(n-2) +2 x(n) x(n-2)
n
n
15. Determine the z- transform and ROC of the signal x (n) = [3 (2 ) 4 (3 )] u(n).

16. State and prove convolution theorem in z-transform.


17. Given x(n) = (n) + 2 (n-1) and y(n) = 3 (n+1) + (n)- (n-1). Find x(n) * y(n) and X(z).Y(z).

*********************************************************************
UNIT - III DISCRETE TRANSFORMS
PART A
Compute the DFT of x(n) = (n no)
State and prove the Parsevals relation of DFT.
What do you mean by the term bit reversal as applied to DFT?
Define discrete Fourier series.
Draw the basic butterfly diagram of DIF FFT algorithm.
n
Compute the DFT of x(n) = a
State the time shifting and frequency shifting properties of DFT.
What is twiddle factor? What are its properties?
Draw the basic butterfly diagram of DIT FFT algorithm.
Determine the 3 point circular convolution of x(n) = {1,2,3} and h(n) = {0.5,0,1}
If an N-point sequence x(n) has N-point DFT of X(K) then what is the DFT of the following
*
*
j2ln/N
i) x (n) ii) x (N-n) iii) x((n-l))N iv) x(n) e
12. What is FFT and what are its advantages?
13. Distinguish between DFT and DTFT (Fourier transform)
14. What is the basic operation of DIT FFT algorithm?
15. What is zero padding? What are its uses?
16. State and prove Parsevals relation for DFT.
17. Draw the flow graph of radix 2 DIF - FFT algorithm for N= 4
18. What do you mean by bit reversal in DFT?
20. Write the periodicity and symmetry property of twiddle factor.

1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
11.

21. Give the relationship between z-domain and frequency domain.


22. Distinguish between discrete Fourier series and discrete Fourier transform.
23. What is the relationship between Fourier series co-efficient of a periodic
sequence and DFT?
24. What is the circular frequency shifting property of DFT?
25. Establish the relation between DFT and z-transform.
26. Define DFT pair.
27. Define overshoot.
28. Define Gibbs phenomenon.
29. How many multiplications and additions are required to compute N-point DFT using
radix 2 FFT?
PART B
1.

Perform circular convolution of the sequence using DFT and IDFT technique
x1(n) = {2, 1,2,1} x2 (n) = {0,1,2,3}

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

(8)
(8)

3. From the first principles obtain the signal flow graph for computing 8 point DFT using radix-2
DIT FFT algorithm. Using the above compute the DFT of sequence x(n) =
{0.5,0.5,0.5,0.5,0,0,0,0}
(16)
4. State and prove the circular convolution property of DFT.Compute the circular
convolution of x(n) = {0,1,2,3,4} and h(n) = {0,1,0,0,0}
(8)
5. Perform circular convolution of the sequence using DFT and IDFT technique
x1(n) = {1,1,2,1} x2 (n) = {1,2,3,4}

(8)

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


(8)
7. From the first principles obtain the signal flow graph for computing 8 point DFT using radix-2
DIF-FFT algorithm. An 8 point sequence is given by x(n)={2,2,2,2,1,1,1,1} compute
its 8 point DFT of x(n) by radix-2 DIF-FFT (16)
8. Compute 5 point circular convolution of x1(n) = (n) + (n-1)- (n-2) - (n-3) and x2(n) =
(n) (n-2)+ (n-4)
(8)
9. Explain any five properties of DFT.
(10)
10. Derive DIF FFT algorithm. Draw its basic butterfly structure and compute the DFT x(n) = (n
(16)
1) using radix 2 DIF FFT algorithm.
11. Perform circular convolution of the sequence using DFT and IDFT technique
x1(n) = {0,1,2,3} x2 (n) = {1,0,0,1}

(8)

12. Compute the DFT of the sequence x (n) = 1/3 (n) 1/3 (n-1) +1/3 (n -2) (6)
13. From the first principles obtain the signal flow graph for computing 8 point DFT using
radix-2 DIT - FFT algorithm. Using the above compute the DFT of sequence x(n) = 2 sin n
/ 4 for 0 n 7
(16)
14. What is circular convolution? Explain the circular convolution property of DFT and
compute the circular convolution of the sequence x(n)=(2,1,0,1,0) with
itself
(8)
15. Perform circular convolution of the sequence using DFT and IDFT technique
x1(n) = {0,1,2,3} x2 (n) = {1,0,0,1}

(8)
n

16. i) Compute the DFT of the sequence x (n) = (-1)


(4)
ii) What are the differences and similarities between DIT FFT and DIF FFT
algorithms?

(4)

17. From the first principles obtain the signal flow graph for computing 8 point DFT using
radix-2 DIT - FFT algorithm. Using the above compute the DFT of sequence x(n) = cos n / 4
(16)
for 0 n 7
18. Compute 4-point DFT of the sequence x (n) = (0, 1, 2, 3)

(6)

19. Compute 4-point DFT of the sequence x (n) = (1, 0, 0, 1)

(6)

20. Explain the procedure for finding IDFT using FFT algorithm

(6)

21. Compute the output using 8 point DIT FFT algorithm for the sequence
x(n) = {1,2,3,4,5,6,7,8}

(16)

22.Determine the 8-point DFT of the sequence x(n) = {0,0,1,1,1,0,0,0}


23. Find the circular convolution of x(n) = 1,2,3,4} and h(n) = {4,3,2,1}
24. Determine the 8 point DFT of the signal x(n) = {1,1,1,1,1,1,0,0}. Sketch its
magnitude and phase.

*********************************************************************

UNIT - IV DESIGN OF DIGITAL FILTERS


PART A
1. An analog filter has a transfer function H(s) = 1 / s+2. Using impulse invariance method,
obtain pole location for the corresponding digital filter with T = 0.1s.
2. What is frequency warping in bilinear transformation?
3. If the impulse response of the symmetric linear phase FIR filter of length 5 is h (n)
= {2,3,0,x,y}, find the values of x and y.
4. What is prewarping? Why is it needed?

5. Find the digital transfer function H (z) by using impulse invariance method for the analog
transfer function H(s) = 1 / s+2.
6. What are the different structures of realization of FIR and IIR filters?
7. What are the methods used to transform analog to digital filters?
8. State the condition for linear phase in FIR filters for symmetric and anti symmetric
response.
9. Draw a causal FIR filter structure for length M= 5.
10. What is bilinear transformation? What are its advantages?
11. Write the equation of Barlett (or) triangular and Hamming window.
12. Write the equation of Rectangular and Hanning window.
13. Write the equation of Blackman and Kaiser window.
14. Write the expression for location of poles of normalized Butter worth filter.

Pk = c ej (N+2k +1) / 2N
Where k = 0, 1, . (N-1) and for a normalized filter c = 1 rad / sec
15.
16.
17.
18.
19.
20.
21.
22.
23.
24.
25.
26.

Write the expression for location of poles of normalized Chebyshev filter.


rd
Draw the magnitude response of 3 order Chebyshev filter.
Draw the magnitude response of 4th order Chebyshev filter.
Draw the basic FIR filter structure.
Draw the direct form I structure of IIR filter.
Draw the direct form II structure of IIR filter.
Draw the cascade form realization structure of IIR filter.
Draw the parallel form realization structure of IIR filter.
When cascade form realization structure is preferred in filters?
Distinguish between FIR and IIR filters.
Compare analog and digital filters.
Why FIR filters are always stable?
Because all its poles are located at the origin.

27. State the condition for a digital filter to be causal and stable.
28. What are the desirable characteristics of windows?
29. Give the magnitude function Butterworth filter. What is the effect of varying the order of N on
magnitude and phase response?
30. List out the properties of Butterworth filter.
31. List out the properties of Chebyshev filter.
32. Give the Chebyshev filters transfer function and draw its magnitude response.
33. Give the equation for the order N and cut off frequency c of Butterworth filter
34. Why impulse invariance method is not preferred in the design of IIR filter other than low pass
filters?
35. What are the advantages and disadvantages FIR filters?
36. What are the advantages and disadvantages IIR filters?
37. What is canonic structure?
If the number of delays in the structure is equal to the order of the difference equation
or order of transfer function, then it is called canonic form of realization.
38. Compare Butterworth and Chebyshev filters.
39. What are the desirable and undesirable features of FIR filters?
40. What are the design techniques of designing FIR filters?
Fourier series method
Windowing technique Frequency
sampling method

PART B
1. With suitable examples, describe the realization of linear phase FIR filters

(8)

2. Convert the following analog transfer function H(s) = (s+0.2) / [(s+0.2) + 4] into
equivalent digital transfer function H (z) by using impulse invariance method assuming
T= 1 sec.
3. Convert the following analog transfer function H(s) = 1 / (s+2) (s+4) into

(8)

equivalent digital transfer function H (z) by using bilinear transformation with T = 0.5 sec.
2

4. Convert the following analog transfer function H(s) = (s+0.1) / [(s+0.1) + 9] into
equivalent digital transfer function H (z) by using impulse invariance method assuming
T= 1 sec.

(8)

5. Convert the following analog transfer function H(s) = 2/ (s+1) (s+3) into
equivalent digital transfer function H (z) by using bilinear transformation with T = 0.1
sec.Draw the diect form II realization of digital filter.

(8)

6. Design a high pass filter of length 7 samples with cut off frequency of 2 rad / sec
using Hamming window. Plot its magnitude and phase response.

(16)

7. For the constraints


0.8 _H()_ 1.0 , 0 0.2
_ H()_ 0.2, 0.6
With T= 1 sec determine the system function H(z) for a Butterworth filter using
bilinear transformation.

(16)

8. Describe the effects of quantization in IIR filter. Consider a first order filter with difference
equation y (n) = x (n) + 0.5 y (n-1).Assume that the data register length is 3

bits plus a sign bit. The input x (n) = 0.875 (n). Explain the limit cycle oscillations in
the above filter, if quantization is preferred by means of rounding and signed
magnitude representation is used.

(16)

9. With a neat sketch explain the architecture of TMS 320 C54 processor.

(16)

10. For the constraints


0.7 _H()_ 1.0 , 0 /2
_ H()_ 0.2, 3/4
With T= 1 sec, design a Butterworth filter.

(16)

11. Explain the quantization effects in design of digital filters.

(16)

12. Discuss about the window functions used in design of FIR filters

(8)

13. Obtain the cascade and parallel realization of system described by difference equation
y(n) = -0.1 y(n-1) + 0.2 y(n-2) + 3x(n) +3.6 x(n-1) + 0.6 x(n-2)

(10)

14. Design a digital Butterworth filter satisfying the following constraints with T= 1 sec,
using Bilinear transformation.
0.707 _H () _ 1.0, 0 /2
_ H () _ 0.2, 3/4
15. Design a digital Chebyshev filter satisfying the following constraints with T= 1 sec,
using Bilinear transformation.
0.707 _H () _ 1.0, 0 /2

(16)

_ H () _ 0.2, 3/4

(16)

th

18. Draw and explain cascade form structure for a 6 order FIR filter.

(6)

19. Explain impulse invariance method of digital filter design.

(10)

20. Derive an expression between s- domain and z- domain using bilinear transformation.
Explain frequency warping.

(10)

21. Draw the structure for IIR filter in direct form I and II for the following transfer
-1

-1

-2

-1

-1

-2

Function H (z) = (2 + 3 z ) (4+ 2 z +3 z ) / (1+0.6 z ) (1+ z +0.5 z ) (10)


22. Design a filter with
Hd() = e

-j2

- /4 /4
/4

=0

Using a Hamming window with N= 7


23. Discuss about frequency transformations in detail.

(16)
(8)

24. Design a LPF with


Hd() = e

-j3

- 3/4 3/4
3/4

=0

Using a Hamming window with N= 7

(16)

25. Using the bilinear transformation and a low pass analog Butterworth prototype,
design a low pass digital filter operating at a rate of 20 KHz and having pass band
extending to a 4 KHz with a maximum pass band attenuation of 0.5 dB and stop band
starting at 5KHzwith a minimum stop band attenuation of 10 dB.

(16)

26. Using the bilinear transformation and a low pass analog Chebyshev type I prototype, design a
low pass digital filter operating at a rate of 20 KHz and having pass band extending to a 4 KHz
with a maximum pass band attenuation of 0.5 dB and stop band

starting at 5KHzwith a minimum stop band attenuation of 10 dB.

(16)

27. Obtain the cascade realization of linear phase FIR filter having system function H(z) = (
-1

-2

-1

-2

1+1/2 z + z ) (2 + z +2z ) using minimum number of multipliers.(8)

28. Design an ideal Hilbert transformer having frequency response


j

H(e ) = j
= -j

for - 0
for 0

for N=11, using i. rectangular window


ii. Blackmann window
29. Obtain the direct form I, direct form II, cascade and parallel form of realization for
the system y(n) = -0.1 y9n-1) + 0.2 y(n-2) + 3 x(n) + 3.6 x (n-1) + 0.6 x(n-2)
30. Using Bilinear transformation and a low pass analog Butterworth prototype, design a low
pass digital filter operating at the rate of 20k Hz and having pass band extending to 4 kHz

with maximum pass band attenuation of 10 dB and stop band starting at 5 kHz with a
minimum stop band attenuation of 0.5 dB
31. Using Bilinear transformation and a low pass analog Chebyshev type I prototype, design
a low pass digital filter operating at the rate of 20k Hz and having pass band extending to
4 kHz with maximum pass band attenuation of 10 dB and stop band starting at 5 kHz
with a minimum stop band attenuation of 0.5 dB
32. Design a low pass filter using Hamming window for N=7 for the desired frequency
Response D () = e

j3

= 0 for

for -3 / 4 3 / 4
3 / 4

33. Design an ideal differentiator for N=9 using Hanning and triangular window
**************************************************************************
UNIT - V DSP HARDWARE
PART - A
1. Compare fixed point arithmetic and floating point arithmetic.
2. What is product quantization error or product round off error in DSP?
3. What are the quantization methods?
4. What is truncation and what is the error that arises due to truncation in floating point
numbers?
5.
6.
7.
8.
9.
10.
11.
12.
13.
14.
15.
16.
17.
18.
19.

What is meant by rounding? Discuss its effects?


What are the two kinds of limit cycle oscillations in DSP?
Why is rounding preferred to truncation in realizing digital filters?
What are the 3 quantization errors due to finite word length registers in digital filters?
List out the features of TMS 320 C54 processors.
What are the various interrupt types supported by TMS 320 C54?
Mention the function of program controller of DSP processor TMS 320 C54.
List the elements in program controller of TMS 320C54.
What do you mean by limit cycle oscillations?
What is pipelining? What is the pipeline depth of TMS 320 C54 processor?
What are the different buses of TMS 320 C54 processor?
What are quantization errors due to finite word length registers in digital filters?
Differentiate between Von Neumann and Harvard architecture.
Define limit cycle oscillations in recursive systems.
How to prevent overflow in digital filters?
PART B

1. Describe the function of on chip peripherals of TMS 320 C54 processor.

(12)

2. What are the different buses of TMS 320 C54 processor? Give their functions. (4)
3. Explain the function of auxiliary registers in the indirect addressing mode to point the
data memory location.
4.Explain about the MAC unit.

(8)
(8)

5.What is meant by instruction pipelining? Explain with an example how pipelining


increases through put efficiency.
6.Explain the operation of TDM serial ports in P-DSPs

(8)
(8)

7.Explain the characteristics of a limit cycle oscillation with respect to the system
described by the equation y (n) = 0.95 y (n-1) + x (n).Determine the dead band of the
filter.

(10)

8. Draw the product quantization noise model of second order IIR filter.

(6)

9. In a cascaded realization of the first order digital filter, the system function of the
-1

-1

individual section are H19z) = 1 / (1-0.9 z ) and H2(z) = 1 / (1-0.8z ). Draw the
product quantization noise model of the system and determine the output noise
power.

(16)

10. Explain the statistical characterization of quantization effects in fixed point


realization of digital filter.

(16)

11. Give a detailed note on Direct memory Access controller in TMS 320 C54x
processor.
12. Find the effect of quantization on the pole locations of the second order IIR filter Given
-1

-1)

by H(z) = 1 / (1-0.5z ) (1- 0.45 z

when it is realized in direct form I and in cascade

form. Assume a word length of 3 bits.


13. Determine the variance of the round off noise at the output of the two cascade
-1

realizations of the filters with system functions H1 (z) = 1 / 1-0.5 z


-1

H2 (z) = 1 / 1- 0.25 z

Cascade I, H (z) = H1 (Z) H2 (z)


Cascade II, H (z) = H2 (z) H1 (z)

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