A

Course File On

“DIGITAL SIGNAL PROCESSING”

Submitted by

V VIJAYA KUMAR

Asst. Professor, ECE.

In the department of

Electronics & Communication Engineering

NARASIMHA REDDY ENGINEERING COLLEGE

(Affiliated to J.N.T.U, HYDERABAD)

MAISAMMGUDA (V), DHULAPALLY(P),MEDCHAL(M) SECUNDERABAD-14

(2018-2019)
 COURSE FILE

Subject: DIGITAL SIGNAL PROCESSING

Year: III– B.Tech, II SEM Branch: ECE

S.NO CONTENTS
1 Department vision & mission

2 List of PEOs ,PSOs and POs

3 Course Out comes & Mapping of course out comes with POs

4 Mapping of course outcomes with PSO’s

5 Syllabus copy

6 Individual time table

7 Detailed lecture plan

8 Session execution log

9 Assignment Questions

10 Sample assignment script

11 Unit-wise course material

12 Mid exam question papers

13 Sample mid answer script

14 Material collected from Internet/Websites

15 Power point presentations

University Questions / Question Bank

16

17 References (Text books/websites/Journals)

18 Advanced topics beyond the syllabus

 19 Mid questions with CO mapping

20 Assignment questions with CO mapping

21 Course attainment value for internal marks

22 Course attainment value for External marks

1.Department Vision & Mission

Vision of the Department:

The Electronics and Communication Engineering Department strives to be recognized

globally through quality education producing well-qualified Engineers who are innovative in
research, possessing modern skills, ethically strong, empire-building with goal..
Mission of the Department:
1. To create knowledgeable and skilled graduates with strong fundamentals, through rigorous
teaching , learning and assessment process that develops the ability to solve problems individually
and work in teams
2. To create professional leaders for the society with research attitudes in the core areas of electronics
and communication with collaborations to impact quality of academics, industry and social needs.
3. To create competent technical man power with ethical values, leadership qualities serving the
professional and societal problems
2. List of PEOs, POs & PSOs
PEOs:
1. To exhibit professional competency in Electronics and Communication Engineering with good
foundation in Mathematics and basic sciences.
2. To possess lifelong learning process and augment their engineering skills for new challenges
with sustainability.
3. To have effective communication skills and work in multidisciplinary teams.
POs:
Engineering knowledge: Apply the knowledge of mathematics, science, engineering
1 fundamentals, and an engineering specialization to the solution of complex engineering
problems
Problem analysis: Identify, formulate, review research literature, and analyze complex
2 engineering problems reaching substantiated conclusions using first principles of
mathematics, natural sciences, and engineering sciences.
Design/development of solutions: Design solutions for complex engineering problems
and design system components or processes that meet the specified needs with appropriate
3
consideration for the public health and safety, and the cultural, societal, and environmental
considerations.
Conduct investigations of complex problems: Use research-based knowledge and
4 research methods including design of experiments, analysis and interpretation of data, and
synthesis of the information to provide valid conclusions.
Modern tool usage: Create, select, and apply appropriate techniques, resources, and
5 modern engineering and IT tools including prediction and modeling to complex
engineering activities with an understanding of the limitations.
The engineer and society: Apply reasoning informed by the contextual knowledge to
6
assess societal, health, safety, legal and cultural issues and the consequent responsibilities
 relevant to the professional engineering practice.
Environment and sustainability: Understand the impact of the professional engineering
7 solutions in societal and environmental contexts, and demonstrate the knowledge of, and
need for sustainable development.
Ethics: Apply ethical principles and commit to professional ethics and responsibilities and
8
norms of the engineering practice.
Individual and team work: Function effectively as an individual, and as a member or
9
leader in diverse teams, and in multidisciplinary settings.
Communication: Communicate effectively on complex engineering activities with the
engineering community and with society at large, such as, being able to comprehend and
10
write effective reports and design documentation, make effective presentations, and give
and receive clear instructions.
Project management and finance: Demonstrate knowledge and understanding of the
11 engineering and management principles and apply these to one’s own work, as a member
and leader in a team, to manage projects and in multidisciplinary environments.
Life-long learning: Recognize the need for, and have the preparation and ability to
12 engage in independent and life-long learning in the broadest context of technological
change.

PSOs:
PSO 1 (Engineering Knowledge and Analysis): Analyze specific engineering problems relevant to
Electronics & Communication Engineering by applying the knowledge of basic sciences, engineering
mathematics and engineering fundamentals.
PSO 2 (System Design): Ability to design and solve problems in the field of Electronics &
Communication Engineering by applying the knowledge acquired from Electronic Devices and Circuits,
VLSI Design, Embedded systems, Analog & Digital Communication and other allied topics.
PSO 3 (Application of the knowledge on society/environment): Apply the contextual knowledge of
Electronics and Communication Engineering to assess societal, environmental, health, safety, legal and
cultural issues with professional ethics and function effectively as an individual or a leader in a team to
manage different projects in multidisciplinary environments as the process of life-long learning.
3.COURSE OUTCOMES
Cousre name: DSP
At the end of the course student will be able to
Understand the classification od different discrete signals and systems, Z-transform
C326.1
concepts and Realization of digital filters

C326.2 Understand the Discrete Fourier Series and Discrete Fourier transform (DFT), its
applications and implementation by FFT techniques

C326.3 Apply several design techniques for IIR type digital filters: “pole-zero placement”,
the “derivative approximation”

C326.4 Applies a design technique for FIR type digital filters known as the “windowing
method”.

C326.5 Understand fundamental concepts & Theory of Multirate signal processing and it’s
applications.

Course Outcome (CO)-Program Outcome (PO) Matrix:

Course Program Outcomes (PO’s)
Outcomes
(CO’s) PO1 PO2 PO3 PO4 PO5 PO6 PO7 PO8 PO9 PO10 PO11 PO12
C326.1 2 1 3 - 2 2 - - - - 3 2

C326.2 2 2 3 - 3 2 - - - - 3 2

C326.3 2 2 3 - 3 2 - - - - 3 2

C326.4 2 1 3 - 3 2 - - - - 3 2

C326.5 2 2 3 - 3 2 - - - - 3 2

Course Program Specific Outcomes (PSO’s)

Outcomes
PSO1 PSO2 PSO3
(CO’s)
C326.1 2 2 2

C326.2 2 2 2

C326.3 2 2 2

C326.4 2 2 2

C326.5 2 2 2

4. SYLLABUS
UNIT I:
Introduction: Introduction to Digital Signal Processing:
Introduction to Digital Signal Processing: Discrete time signals & sequences, linear shift invariant
systems, stability, and causality, linear constant coefficient difference equations. Frequency domain
representation of discrete time signals and systems
Realization of Digital Filters:
Applications of Z - transforms, solution of difference equations of digital filters, System function.
Stability criterion. Frequency response of stable systems, Realization of digital filters - Direct, Canonic,
Cascade and Parallel forms
UNIT II:
Discrete Fourier series:
DFS representation of Periodic Sequences. Properties of Discrete Fourier Series., Discrete Fourier
Transforms: Properties of DPT. linear convolution of sequences using DPT. Computation of DFT: Over-
lap Add method, Over-lap Save method, Relation between DTFT, DFS. DFT and Z-Transform.
Fast Fourier Transforms:
Fast Fourier transforms (FFT) - Radix-2 decimation-in-time and decimation-in-frequency FPT
Algorithms, Inverse FFT and FFT with general Radix-N
UNIT III:
IIR Digital Filters:
Analog filter approximations - Butterworth and Chebyshev, Design of IIR Digital filters from analog
filters. Step and Impulse invariant techniques. Bilinear transformation method, Spectral transformations.
UNIT IV:
FIR Digital Filters:
Characteristics of FIR Digital Filters. Frequency response. Design of FIR Filters: Fourier Method.
Digital Filters using Window Techniques, Frequency Sampling technique, Comparison of IIR & FIR
filters
UNIT V:
 Multirate Digital Signal Processing
Introduction. Down sampling, Decimation. Up sampling, Interpolation, Sampling Rate
Conversionconversion of band pass signals. Concept of re-sampling. Applications of multi rate signal
processing
Finite Word Length Effects:
Limit cycles. Overflow oscillations. Round-off noise in IIR digital filters. Computational output round
off noise. Methods to prevent overflow. Trade off between round off and overflow noise. Measurement
of coefficient quantization effects through pole-zero movement. Dead band effects.
5. INDIVIDUAL TIME TABLE

Day I II III IV V VI VII

& 10.10 – 12.40 -

09.10 – 11.00 – 11.50 – 01.20 – 02.20 – 03.10 –
Time 10.10 11.00 11.50 12.40 01.20 02.20 03.10 04.00

MON DSP DSP LAB (III ECE-A)

TUE DSP DSP DSP LAB (III ECE-B)

WED DSP DSP LAB (III ECE-A)

THU

FRI EC LAB II EEE ASSIGMNT

SAT DSP DSP LAB (III ECE-B)

6. DETAILED LECTURE PLAN

Method
of
Suggested
Tentative Topic as per JNTUH Teaching
S.No Topic Actually Covered Book
Date Syllabus Black
Board /
PPT
UNIT-I
Introduction to Digital Signal Introduction to Digital Signal
1 24/12/2018 Processing Processing T1,R1 BB

Discrete time signals & Discrete time signals &

2 27/12/2018 sequences sequences T1,R1 BB

28/12/2018 Linear shift invariant systems Linear shift invariant systems

3 T1,R1 BB
29/12/2018 Stability and causality Stability and causality
4 T1,R1 BB
linear constant coefficient linear constant coefficient
5 31/12/2018 difference equations difference equations T1,R1 BB

Frequency domain Frequency domain

6 2/1/2019 representation of discrete time representation of discrete time T1,R1 BB
signals and systems signals and systems

3/1/2019 Introduction of Realization of Introduction of Realization of

7 Digital Filters Digital Filters T1,R1 BB

4/1/2019 Applications of Z - transforms Applications of Z - transforms

8 5/1/2019 T1,R1 BB

7/1/2019 solution of difference equations solution of difference

9 8/1/2019 of digital filters equations of digital filters T1,R1 BB

9/1/2019 System function,Stability System function,Stability

10 criterion criterion T1,R1 BB
10/1/2019 Frequency response of stable Frequency response of stable
11 11/1/2019 systems systems T1,R1 BB

12/1/2019 Realization of digital filters - Realization of digital filters -

12 16/1/2019 Direct, Canonic Direct, Canonic T1,R1 BB

17/1/2019 Realization of digital filters - Realization of digital filters -

13 18/1/2019 Cascade and Parallel forms Cascade and Parallel forms T1,R1 BB

UNIT-II
 Discrete Fourier series:DFS Discrete Fourier series:DFS
19/1/2019
representation of Periodic representation of Periodic
14 21/1/2019 T4 BB
Sequences Sequences

22/1/2019 Properties of Discrete Fourier Properties of Discrete Fourier

15 23/1/2019 Series Series T4 BB

24/1/2019 Discrete Fourier Transforms and Discrete Fourier Transforms

16 25/1/2019 Properties of DFT and Properties of DFT T4 BB

28/1/2019 linear convolution of sequences linear convolution of

17 29/1/2019 using DPT sequences using DPT T4 BB

30/1/2019 Computation of DFT: Over-lap Computation of DFT: Over-

31/1/2019 Add method, Over-lap Save lap Add method, Over-lap
18 T4 BB
method Save method

1/2/2019 Relation between DTFT, DFS Relation between DTFT, DFS

19 2/2/2019 T4 BB

4/2/2019 Relation between DFT and Z- Relation between DFT and Z-

20 5/2/2019 Transform Transform T4 BB

6/2/2019 Introduction of Fast Fourier Introduction of Fast Fourier

21 7/2/2019 transforms (FFT) transforms (FFT) T4 BB

8/2/2019 Radix-2 decimation-in-time FFT Radix-2 decimation-in-time

22 9/2/2019 Algorithm FFT Algorithm T4 BB

11/2/2019 decimation-in-frequency FFT decimation-in-frequency FFT

23 12/2/2019 Algorithm Algorithm T4 BB

13/2/2019 Inverse FFT and FFT with Inverse FFT and FFT with
24 14/2/2019 general Radix-N general Radix-N T4 BB

UNIT-III
15/2/2019 Introduction to IIR Digital
25 16/2/2019 Filters Introduction to IIR Digital T4,R3 BB
Filters
21/2/2019 Analog filter approximations -
26 22/2/2019 Butterworth Analog filter approximations - T4,R3 BB
Butterworth
23/2/2019 Analog filter approximations - Analog filter approximations -
27 Chebyshev T4,R3 BB
25/2/2019 Chebyshev
26/2/2019 Design of IIR Digital filters
28 Design of IIR Digital filters T4,R3 BB
27/2/2019 from analog filters
from analog filters
Step and Impulse invariant
29 28/2/2019 techniques Step and Impulse invariant T4,R3 BB
techniques
1/3/2019 Bilinear transformation method Bilinear transformation
30 method T4,R3 BB
2/3/2019
5/3/2019 Spectral transformations Spectral transformations
31 T4,R3 BB
6/3/2019

UNIT-IV
7/3/2019 Introduction to FIR Digital Introduction to FIR Digital
32 8/3/2019 Filters Filters T4 BB
9/3/2019
11/3/2019 Characteristics of FIR Digital Characteristics of FIR Digital
33 12/3/2019 Filters Filters T4 BB
13/3/2019
 14/3/2019 Frequency response Frequency response
34 15/3/2019 T4 BB
16/3/2019
18/3/2019 Design of FIR Filters: Fourier Design of FIR Filters: Fourier
35 19/3/2019 Method Method T4 BB
20/3/2019

21/3/2019 Digital Filters using Window Digital Filters using Window

36 22/3/2019 Techniques Techniques T4 BB
23/3/2019
25/3/2019 Frequency Sampling technique Frequency Sampling
37 26/3/2019 technique BB
27/3/2019 T4
Comparison of IIR & FIR filters Comparison of IIR & FIR
38 28/3/2019 BB
filters
UNIT-V
1/4/2019 Introduction Multirate Digital Introduction Multirate Digital
39 2/4/2019 Signal Processing Signal Processing T4 BB

3/4/2019 Down sampling Down sampling

40 4/4/2019 T4 BB

6/4/2019 Decimation Decimation

41 T4 BB
8/4/2019
Up sampling Up sampling
9/4/2019
42 T4 BB
10/4/2019

11/4/2019 Interpolation Interpolation

43 T4 BB
12/4/2019
Sampling Rate Sampling Rate
15/4/2019 Conversionconversion of band Conversionconversion of band
44 pass signals pass signals T4 BB
16/4/2019

17/4/2019 Concept of re-sampling Concept of re-sampling

45 T4 BB
18/4/2019
Applications of multi rate signal Applications of multi rate
46 20/4/2019 processing signal processing T4 BB

TEXT BOOKS:
1. Digital Signal Processing, Principles, Algorithms, and Applications: John G. Proakis, Dimitris
G. Manolakis.
Pearson Education / PHI. 2007.
2. Discrete Time Signal Processing-A. V. Oppenheim and R.W. Schaffer. PHI, 2009
3. Fundamentals of Digital Signal Processing - Loney Ludeman.John Wiley
REFERENCE BOOKS:
1. Digital Signal Processing - Fundamentals and Applications - Li Tan, Elsevier. 2008
2. Fundamentals of Digital Signal Processing using Matlab - Robert J. Schilling. Sandra L,
Harris, Thomson. 2007
3. Digital Signal Processing - S.Salivahanan. A.Vallavaraj and CGnanapriya.TMH.2009
4. Discrete Systems and Digital Signal Processing with MATLAB -Taan S.EIAli.CRC press.
2009.
7. Session execution log

S.NO UNITS COMPLETED

DATE

UNIT-I
1

UNIT:2
2

UNIT:3
3

UNIT:4
4

UNIT:5
5

8. Assignment Questions

ASSIGNMENT: 1

SET:1

SET:2

SET:3

SET:4
ASSIGNMENT: 2

SET:1

SET:2

SET:3

SET:4
9.Sample assignment scripts

10.Unit-wise course material

11. Mid exam question papers

NARASIMHAREDDY ENGINEERING COLLEGE

DEAPRTMENT OF ECE
II B.TECH II SEMESTER ECE (A,B) MID-I
Sub: DSP Time: 60Mins
Answer any two Questions each Questions carry 5 marks. 2*5=10M

***********ALL THE BEST************

NARASIMHAREDDY ENGINEERING COLLEGE

DEAPRTMENT OF ECE
II B.TECH II SEMESTER ECE (A,B) MID-II
Sub: DSP Time: 60Mins
Answer any two Questions each Questions carry 5 marks. 2*5=10M

***********ALL THE BEST************

12. Sample mid answer script( Best,Average & Low)
13. Material collected from Internet/Websites

14. Power point presentations

15. University Questions / Question Bank ( Minimum 7 Questions from each Unit)

Unit:1

Unit:2

Unit :3

Unit: 4

Unit: 5

16. References (Text books/websites/Journals)

TEXT BOOKS :

1. Digital Signal Processing, Principles, Algorithms, and Applications: John G. Proakis, Dimitris
G. Manolakis. Pearson Education / PHI. 2007.
2. Discrete Time Signal Processing-A. V. Oppenheim and R.W. Schaffer. PHI, 2009
3. Fundamentals of Digital Signal Processing - Loney Ludeman.John Wiley
REFERENCES :

1. Digital Signal Processing - Fundamentals and Applications - Li Tan, Elsevier. 2008

2. Fundamentals of Digital Signal Processing using Matlab - Robert J. Schilling. Sandra L,
Harris, Thomson. 2007
3. Digital Signal Processing - S.Salivahanan. A.Vallavaraj and CGnanapriya.TMH.2009
4. Discrete Systems and Digital Signal Processing with MATLAB -Taan S.EIAli.CRC press.
2009.
WEBSITES:
 USEFUL URL’S:

JOURNALS:

17: ADD on topics (ADVANCED TOPICS)

Objective type question bank

1.Application of Convolution:
a) FIR Filtering b) Addition c) Manipulation d) None of these
2.Condition for aliasing problem:
a) fs<fm b) fs<2fm c)fs=fm d) all of these
3.DFT stands as:
a.Discrete Fourier transform b.digital function transform
c.digital frequency transform d.none
4. FFT stands as:
a. Fast Fourier transform
b. Fourier function transform
c. Fast frequency transform
d. none
6. Calculate DFT of x (n) = {1, 0, 1, 0}.
a.x (k) = {2, 0, 2, 0} b.x (k) = {1, 0, 1, 0}
c.x (k) = {2, 0, 1, 0} d.none
7. Calculate DFT of x (n)= (n) .
a.1 b.0 c. Wn d. none
8. Calculate DFT of x (n)= (n n0)where 0<n0<N.
𝑗2
a.𝑒 𝑗𝑤𝑛0 b.𝑒𝑁 c.1 d. none
9. The FFT algorithms:
a.eliminate the redundant calculation and enable to analyze the spectral properties of a signal.

b.enable the redundant calculation and redundant to analyze the spectral properties of a signal.

c.a & b d.none

10.The relation between DFT and Fourier series coefficients of a periodic sequence is
a.X(K) = Ck/N b.X(K)= Ck c.X(K) = NCk d.X(K)=1/Ck
11.If x(n) ------N pt DFT------X(K) Then x*(-n, (mod N)) -------------N pt DFT------
a.X*(-K) b.X*(K) c.X(-k) d.X(K)
12.If the Nyquist rate for xa(t) is Ωs , what is the Nyquist rate for d xa(t)/dt
a.dΩs/df b.Ωs c.Ωs/2 d.2Ωs

13.If the Nyquist rate for xa(t) is Ωs , what is the Nyquist rate for xa(2t)
a.2Ωs b.Ωs/2 c.Ωs d.Ωs/4

14.If the Nyquist rate for xa(t) is Ωs , what is the Nyquist rate for xa2(t)
a.2Ωs b.Ωs/2 c.Ωs d.Ωs/4

15.If the Nyquist rate for xa(t) is Ωs , what is the Nyquist rate for xa(t)Cos(Ω0 t)
a.Ωs + 2Ω0 b.Ωs * 2Ω0 c.Ωs /2Ω0 d. Ωs - 2Ω0
16.The minimum sampling frequency for xa(t) is real with Xa(f) non-zero only for 18 KHz < |f|< 22 KHz is
a.8.8 KHz b.9 KHz c.11 KHz d.17.6 KHz

17.The minimum sampling frequency for xa(t) is complex with Xa(f) non-zero only for
30 KHz <|f| < 35 KHz is
a.6 KHz b.5 KHz c.15 KHz d.17.5 KHz

18.Find two different continuous-time signals that will produce the sequence
x(n) = cos( 0.15 nπ) when sampled with a sampling frequency of 8 KHz.
a.sine(1200πt) and Cos(17200πt) b.Cos(1200πt) and Sine(17200πt)
c.Cos(1200πt) and Cos(17200πt) d.Sine(1200πt) and Sine(17200πt)

19.A continuous-time signal xa(t) is known to be uniquely recoverable from its samples xa(nTs) when
Ts = 1 ms. What is the highest frequency in Xa( f )?
a.500 Hz b.1000 Hz c.700 Hz d.5 KHz

20.Suppose that xa(t) is bandlimited to 8 kHz (that is, Xa( f ) = 0 for |f| > 8000), then what is the Nyquist
rate for xa(t)?
a.16 KHz b.4 KHz c.8 KHz d.12 KHz

21.Suppose that xa(t) is bandlimited to 8 kHz (that is, Xa( f ) = 0 for |f| > 8000), then what is the Nyquist
rate for xa(t)cos(2π . 1000t)?
a.16 KHz b.4 KHz c.18 KHz d.5 KHz
22.If a continuous-time filter with an impulse response ha(t) is sampled with a sampling frequency of fs ,
what happens to the cutoff frequency wc of the discrete-time filter as fs is increased?
a.wc increases b.wc decreases
c.wc remains constant d.wc depends upon fs

23.A complex bandpass signal xa(t) with Xa(f) nonzero for 10 kHz < f < 12 kHz is sampled at asampling
rate of 2 kHz. The resulting sequence is x(n) = δ(n), then xa(t) will be
a.xa(t) = (1/2000) (Sine(2000πt)/(πt))ej2π(11000)t
b.xa(t) = (1/2000) (Sine(2000πt)/(πt))e-j2π(11000)t
c.xa(t) = (1/2000) (Cos(2000πt)/(πt))ej2π(11000)t
d.xa(t) = (1/2000) (Cos(2000πt)/(πt))e-j2π(11000)t

24.If the highest frequency in xa(t) is f = 8 kHz, then the minimum sampling frequency for the bandpass
signal ya(t) = xa(t) Cos(Ω0t) if Ω0 = 2π.20.103 will be
a.56 KHz b.64 KHz c.16 KHz d.32 KHz
25.Drawbacks of DSP is
a. Digital processing needs pre and post processing devices b. high cost
c. No memory storage d.none of above
26. Drawbacks of DSP is
a. Digital processing needs A/D and D/A converters and associated reconstruction filters
b. high cost c. No reliable d.none of above
27.Advantages of DSP are:
a. low cost b. stable c. reliable d. all of above
28.Advantages of DSP are:
a. predictable b. repeatable
c. Sharing a single processor among a number of signals by time sharing
d. all of above
29.Advantages of DSP are:
a. low cost b. repeatable
c. storage of data is very easy d. all of above
30.Application of DSP:
a. Military b. telecommunication c. consumer electronics d. all of above
31.Application of DSP:
a. medicine b. seismology c. signal filtering d. all of above
32.Fast convolution techniques:
a. overlap save b. overlap add c. a & b d. none of above
33.Correlation
a. It gives a measure of similarity between two data sequences.
b. It gives a measure of dis-similarity between two data sequences
c.a & b
d.none of above
34. Find the response of an FIR filter with impulse response h(n)= {1,2,4} to the input sequence
x(n)={1,2}.
a. y(n)={1,4,8,8} b. y(n)={1,4,6,6}
c. y(n)={1,2,8,8} d.none of above

35. IIR filters

a)Use feedback
b)Are sometimes called recursive filters
c)Can oscillate if not properly designed
d)all of the above
36. A Blackman window can eliminate ripple in FIR filters. The tradeoff is
a)larger transition bandwidth
b)smaller transition bandwidth
c)a non-linear phase response
d)possible instability
37. The output of two digital filters can be added. Or, the same effect can be achieved by
a)adding their coefficients b)subtracting their coefficients
c)convolving their coefficients
d)averaging their coefficients and then using a Blackman window
38. The letter A below indicates the filter
a) stopband
b) transition band
c) passband

d) ripple
39. A DSP convolves each discrete sample with four coefficients and they are all equal to 0.25. This
must be a
a)low-pass filter b)high-pass filter
c)band-pass filter d)band-stop filter
40. The inverse Fourier transform
a)converts from the frequency domain to the time domain
b)converts from the time domain to the frequency domain
c)converts from the phasor domain to the magnitude domain
d)is used to make real-time spectrum analyzers
41. This is the impulse response for
a)an IIR high pass filter b)an FIR band pass filter
c)an IIR low pass filter d)an FIR low pass filter
42. Coefficient symmetry is important in FIR filters because it provides
a)a smaller transition bandwidth b)less passband ripple
c)less stopband ripple d)a linear phase response
43. This time graph shows the
a)frequency response of an IIR filter b)amplitude response of an IIR filter
c)impulse response of an IIR filter d)none of the above

44. Curve A is the

a)phase response of a low pass filter b)amplitude response of a lowpass filter

c)both of the above d)none of the above

45. This windowed sinc FIR filter has ripple caused by
a)non-symmetrical coefficients b)Gibb's phenomenon
c)too few taps d)a defective accumulator
46. Two digital filters can be operated in cascade. Or, the same effect can be achieved by
a)adding their coefficients b)subtracting their coefficients
c)convolving their coefficients
d)averaging their coefficients and then using a Blackman window
47. A DSP convolves each discrete sample with four coefficients and they are all equal to 0.25. This
must be an

a)IIR filter b)FIR filter c)RRR filter d)All of the above

48. This frequency response graph is for a
a)low pass filter b)high pass filter c) band pass filter d)band stop filter
49. The letter C below indicates the filter
a)stopband b)passband c)transition band d)ripple
50. Curve B is the
a)phase response of a low pass filter b)amplitude response of a low pass filter

c)both of the above d)none of the above

51. The letter B below indicates the filter

a)stopband b)passband c)transition band d)ripple
52. If a linear phase filter has a phase response of 40 degrees at 200 Hz, what will its phase response be
at a frequency of 400 Hz (assuming that both frequencies are in the passband of the filter)?
a)35 degrees b)40 degrees c)45 degrees d)80 degrees
53. Which of the following is used to alter FIR filter coefficients so they smoothly approach zero at both
ends?
a)rectangular window b)Blackman window c)Laplace window d)Hilbert window
54. Point C is called

a)a phase reversal b)the half-power point

c)a phase discontinuity d)a phase wrap

55. A DSP convolves each discrete sample with these coefficients: -0.25, -0.25, 1.0, -0.25, and -
0.25.This must be a
a)low-pass filter b)high-pass filter c)band-pass filter d)band-stop filter
56. The basic process that's going on inside a DSP chip is
a)quantization b)MAC c)logarithmic transformation d)vector calculations
57. For the rectangular window function, the transition width of the main lobe is approximately (here M
is the length of the filter)
a)4*pi*M b)pi/4M c)pi*M/4 d)4*pi/M
58. For the rectangular window function, the first side lobe will be __________ dB down the peak of the
main lobe.
a)12 dB
b)11 dB
c)13 dB
d)14 dB
59. For the rectangular window function, the roll-off will be __________ dB/decade.
a)25 Db b)20 dB c)15 dB d)10 dB
60. For the hamming window function, the width of the main lobe is approximately (here M is the length
of the filter)
a)8*pi*M b)pi/8M c)pi*M/8 d)8*pi/M
61. For the hamming window function, the peak of the first side lobe will be at __________ dB.
a)-40 dB b)-48 dB c)-43 dB d)-45 dB
62. For the hamming window function, the side lobe roll-off will be __________ dB/decade.
a)25 dB b)20 dB c)15 dB d)10 dB
63. For the hamming window function, the width of the main lobe is approximately (here M is the length
of the filter)
a)8*pi*M b)pi/8M c)pi*M/8 d)8*pi/M
64. For the hamming window function, the peak of the first side lobe will be at __________ dB.
a)-35 dB b)-32 dB c)-40 dB d)-43 dB
65. For the Blackman window function, the width of the main lobe is approximately (here M is the
length of the filter)
a)12*pi*M b)pi/8M c)pi*M/8 d)12*pi/M
66. For the hamming window function, the peak of the first side lobe will be at __________ dB.
a)-58 dB b)-48 dB c)-45 dB d)-43 dB
67. What is FIR filter?
a.FIR filters are “finite” there is a specific limit to the number of times that any delayed sample is added
to a new input sample.
b.FIR filters are “finite” there is a specific limit to the number of times that any delayed sample is added
to a new output sample.
c.a & b d.none of above
68. FIR filters have ……., and IIR filters have ……….
a.Zeros, poles & zeros b.poles & zeros, Zeros
c.Zeros, zeros d.none of above

69. How to define IIR filters term as infinite:

a. As with any feedback device, create a loop, hence the term infinite.
b. As with any non-feedback device, create a loop, hence the term infinite.
c. As with any feedback device, create a open loop, hence the term infinite.
d. None of above
70.The filter coefficients are stored in:
a. binary registers b. digital system c. binary memory d. none
71.Truncation or rounding of the data results in
a. degradation of system performance b. increase system performance
c. grow power d. none
72.the process of quantization is introduce
a. error b. noise c. power d. none
73.Issue connected with finite word length effects:
a. quantization effects in A/D conversion
b. product quantization and coefficient quantization errors in digital filters
c. a & b d. none.
74.Issue connected with finite word length effects:
a. limit cycles in IIR filters
b. product quantization and coefficient quantization errors in digital filters
c.a & b d.none.
75.Issue connected with finite word length effects:
a. finite word length effects in FFTs
b. product quantization and coefficient quantization errors in digital filters
c. limit cycles in IIR filters
d. all of above
76.Rounding or truncation introduces an error whose magnitude depends
a. On the number of bits truncated or rounded bits. b. On the number of bits
c. On the number of bits truncated bits. d.all of above.
78.The dynamic range is
a. DR =-2*log Pe(n) b. DR =-logPe(n)
c. DR =-10*logPe(n) d. none of above
79. Coefficient quantization effects in Direct form realization of IIR filters is
a.Y’(z) = [Hideal(z) X(z) + E(z)] b. Y’(z) = [Hideal(z) + E(z)]
c.Y’(z) = [Hideal(z) X(z)] d.none of above
80. Stray filter
a. coefficient quantization error in Direct form realization of IIR filters
b. coefficient quantization error in cascaded-Direct form realization of IIR filters
c. coefficient quantization error in ladder form realization of IIR filters
d.none of above
81. Limit cycle is
a. zero input limit cycle b. over flow limit cycle c.a & b d.none of above
82.The effects of limit cycles in first and second order systems were studied by
a. Hendy using an effective value model
b. Thomson using an effective value model
c. Jackson using an effective value model
d. none of above
83.If a is positive the limit cycle will have
a. variable magnitude by alternating sign.
b. constant magnitude by alternating sign.
c. constant phase by alternating sign.
d.none of above
84. What is scaling?
a. Scaling must be done in such a way that no overflow occurs at the summing point.
b. Scaling must be done in such a way that overflow occurs at the summing point.
c. Scaling must be done in such a way that no underflow occurs at the summing point.
d.none of above
85.The band of integers is known as the "dead band".
a. true b. false c. either true or false d. none of above
86.In the second order system, under rounding, the output assumes a cyclic set of values of the dead
band.
a. limit-cycle. b. band-cycle. c. dead-cycle. d.none of above

87. With finite precision the response does not converge to the origin but assumes cyclically a set of
values:
a. the limit-cycle. b. band-cycle. c. dead-cycle. d.none of above
88.With infinite precision the response converges to the ..........
a. origin. b. center. c. mid. d. none of above
89. below figure shows:
a. Quantization error in rounding.
b. Quantization error in truncation in 2’s complement.
c. Quantization error in truncation in sign magnitude.
d.none of above
90.This is a deterministic frequency response error is referred to as…………...
a. coefficient quantization error b. product quantization error
c. a & b d. none of above
91.A digital system is characterized by the difference equation y(n)=0.9 y(n-1) + x(n) with x(n)=0 and
initial condition y(-1)=12. Determine dead band of the system.

a.[-5,5] b.[-3 , 3] c.[-1,1] d. none of above

92.FIR filters are ….. generally as sensitive to coefficient round off.
a. not b. less c. most d. none of above
93.FIR filters often require more computation, because you must do ………… for each term in the
impulse response.
a. multiply-add b. add c. multiply d.all of above
94. FIR filters can be ………. delay, IIR filters can………….
a. constant, not b. not, constant c. not, not d.none of above
95. “Linear Phase” (constant delay), If a filter has a ………….delay, the phase shift of the filter will be
t*w, where t is the time delay, and w the natural frequency (2*pi*f). a.
a. constant b. variable c. equal d.none of above

96. Non-linear delay, This is the part of the phase shift (in and around the filter’s passband) that
is not modeled by a ………...
a. straight line b. circle c. square d.none of above
97. If you don’t want a zero at pi, you can’t use a symmetric ……-length filter. You can use an anti
symmetric even length filter if you want a high pass filter, but then you’ll have a zero at DC. This means
that symmetric high pass filters are of …… length.
a. even , odd b.odd, even c .even, even d.none of above
98.Used to increase the sampling rate by an integer factor
a. Up-sampler b. down sampler c. a & b d. none of above
99.Used to decrease the sampling rate by an integer factor
a. Up-sampler b. down sampler c. a & b d. none of above
100.This block represents
a. Up-sampler
b. down sampler
c.a & b
d.none of above
101. Up-sampling operation is implemented by inserting L-1 equidistant ……..-valued samples between
two consecutive samples of x[n].
a. zero b. one c. two d.none of above

102. Input-output relation for ………..x[n]=x[n/L] , n=0,L,2L

0 , otherwise

a. Up-sampler
b. down sampler
c.a & b
d.none of above
103. In practice, the zero-valued samples inserted by the up-sampler are replaced with appropriate
nonzero values using some type of filtering process, Process is called…….
a. interpolation
b. decimation
c.a & b
d.none of above
104. ………operation is implemented by keeping every M-th sample of x[n] and removing M-1 in-
between samples to generate y[n].
a. Up-sampling
b. Down-sampling
c.a & b
d.none of above
105.Input-output relation for ……… y[n] = x[nM]
a. Up-sampler
b. down sampler
c. a & b
d. none of above
106.The up-sampler and the down-sampler are ……..but time-varying discrete-time systems:

a. linear
b. none linear
c.a & b
d.none of above
107. A factor-of-2 sampling rate expansion leads to a compression of X(ej ) by a factor of 2 and a 2-
foldrepetition in the baseband [0, 2 ]. This process is called………
a. imaging
b. sampling
c. decimation
d.none of above
108. A ……..is formed by an interconnection of the up-sampler, the down-sampler, and the components
of an LTI digital filter.
a. complex multirate system
b. complex single-rate system
c.a & b
d.none of above
109. An interchange of the positions of the branches in a cascade often can lead to a computationally
……….realization.
a. efficient
b. non-efficient
c. neither efficient nor non- efficient
d.none of above
110. To implement a ……..in the sampling rate we need to employ a cascade of an up-sampler and
adown-sampler.
a. fractional change
b. constant change
c. variable change
d.none of above
111. A cascade of a factor-of-M down-sampler and a factor-of-L up-sampler is interchangeable with no
change in the input-output relation: y1[n] y2[n] if and only if M and L are relatively
…..
a. prime
b. non prime
c. natural number
d.none of above
112. From the sampling theorem it is known that a the sampling rate of a critically sampled discrete-time
signal with a spectrum occupying the full Nyquist range cannot be reduced any further since such a
reduction will introduce………..
a. aliasing
b. quantization
c. error
d.none of above
113. The bandwidth of a critically sampled signal must be reduced by ………filtering before its
sampling rate is reduced by a down-sampler.
a. lowpass
b. highpass
c.a& b
d.none of above
114. The zero-valued samples introduced by an up-sampler must be interpolated to more appropriate
values for an effective sampling rate………...
a. decrease
b. increase
c.a & b
d.none of above
115. Since up-sampling causes periodic repetition of the basic spectrum, the unwanted images in the
spectra of the up-sampled signal xu [n] must be removed by using a lowpass filter H(z),
called…………………..
a. the interpolation filter

b. the decimation filter

c. the Low pass filter

d. all of above

ASSIGNMENT QUESTIONS:

UNIT-I

PART A:
1. Determine the energy of the discrete time sequence x(n) = (½)n, n_0 =3n, n<0
2. Define multi channel and multi dimensional signals. (2)
3. Define symmetric and anti symmetric signals. (2)
4. Differentiate recursive and non recursive difference equations. (2)
5. What is meant by impulse response? (2)
6. What is meant by LTI system? (2)
7. What are the basic steps involved in convolution? (2)
8. Define the Auto correlation and Cross correlation? (2)
9. What is the causality condition for an LTI system? (2)
10. What are the different methods of evaluating inverse z transform? (2)
11. What is meant by ROC? (2)
12. What are the properties of ROC? (2)
13. What is zero padding? What are it uses? (2)
14. What is an anti imaging and anti aliasing filter? (2)
15. State the Sampling Theorem. (2)
16. Determine the signals are periodic and find the fundamental period (2)
i) sin_2_t ii) sin 20_t+ sin5_t
17. Give the mathematical and graphical representations of a unit sample, unit step sequence. (2)
18. Sketch the discrete time signal x(n) =4_(n+4) +_(n)+ 2_(n-1) +_(n-2) -5_(n-3) (2)
19.Find the periodicity of x(n) =cost(2_n / 7) (2)
20. What is inverse system? (2)
21. Write the relationship between system function and the frequency response. (2)
22. Define commutative and associative law of convolutions. (2)
23. What is meant by Nyquist rate and Nyquist interval? (2)
24. What is an aliasing? How to overcome this effect? (2)
25. What are the disadvantages of DSP? (2)
26. Compare linear and circular convolution.(2)
27. What is meant by section convolution? (2)
28. Compare over lap add and save method. (2)
29. Define system function. (2)
30. State Parseval’s relation in z -transform. (2)

PART B:
1. Determine whether the following system are linear, time-invariant (16)
i)y(n) = Ax(n) +B. (4)
ii)y(n) =x(2n). (4)
iii)y(n) =n x2(n). (4)
iv)y(n) = ax(n)(4)
2. Check for following systems are linear, causal, time in variant, stable, static (16)
i) y(n) =x(2n). (4)
ii) y(n) = cos (x(n)). (4)
iii) y(n) = x(n) cos (x(n) (4)
iv) y(n) =x(-n+2) (4)
3. (a)For each impulse response determine the system is (a) stable i(a) causal
i) h(n)= sin(_n / 2) . (4)
ii) h(n) =_(n) + sin_n (4)
(b)Find the periodicity of the signal x(n) =sin(2_n / 3)+cos (_n / 2) (8)
4. (a)Find the periodicity of the signali) x(n) =cos(_/4)cos(_n /4). (4)ii) x(n) = cos (_n2/ 8) (4)
(b) State and proof of sampling theorem. (8)
5. Explain in detail about A to D conversion with suitable block diagram and to
reconstruct the signal. (16)
6. What are the advantages of DSP over analog signal processing? (16)

CONVOLUTION:
7.Find the output of an LTI system if the input is x(n) =(n+2) for 0_n_3and h(n) =anu(n) for all n
 (16)
8. Find the convolution sum of x(n) =1 n = -2,0,1= 2 n= -1= 0 elsewhereand h(n) =_(n) –_(n-
1)+_( n-2) -_(n-3). (16)
9. (a)Find the convolution of the following sequence x(n) = u(n) ; h(n) =u(n-3). (8)
(b)Find the convolution of the following sequence x(n) =(1,2,-1,1) , h(n) =(1, 0 ,1,1). (8 )
10.Find the output sequence y(n) if h(n) =(1,1,1) and x(n) =(1,2,3,1) using a circular
Convolution. (16)
11. Find the convolution y(n) of the signals (16)
x(n) ={_n, -3_n_5 and h(n) ={ 1, 0_n_4
0,elsewhere } 0, elsewhere

UN IT-II
PART A:
1.How many multiplication and additions are required to compute N point DFT using radix 2
FFT? (2)
2. Define DTFT pair. (2)
3. What are Twiddle factors of the DFT? (2)
4. State Periodicity Property of DFT. (2)
5. What is the difference between DFT and DTFT? (2)
6. Why need of FFT? (2)
7. Find the IDFT of Y (k) = (1, 0, 1, 0) (2)
8. Compute the Fourier transform of the signal x(n) = u(n) –u(n-1). (2)
9. Compare DIT and DIF? (2)
10. What is meant by in place in DIT and DIF algorithm? (2)
11. Is the DFT of a finite length sequence is periodic? If so, state the reason. (2)
12. Draw the butterfly operation in DIT and DIF algorithm? (2)
13. What is meant by radix 2 FFT? (2)
14. State the properties of WNk? (2)
15. What is bit reversal in FFT? (2)
16. Determine the no of bits required in computing the DFT of a 1024 point sequence with
SNR of 30dB. (2)
17. What is the use of Fourier transform? (2)
18. What are the advantages FFT over DFT? (2)
19. What is meant by section convolution? (2)
20. Differentiate overlap adds and save method? (2)
21.Distinguish between Fourier series and Fourier transform. (2)
22.What is the relation between fourier transform and z transform. (2)
23. Distinguish between DFT and DTFT. (2)

PART B:
1.The impulse response of LTI system is h(n)=(1,2,1,-1).Find the response of the system to the
input x(n)=(2,1,0,2) (16)
2. Determine the magnitude and phase response of the given equationy(n) =x(n)+x(n-2) (16)
3. Determine the response of the causal system y(n) y(n-1) =x(n) + x(n-1)to inputsx(n)=u(n) and
x(n) =2–nu(n).Test its stability (16)
4. Determine the frequency response for the system given byy(n)-y3/4y(n-1)+1/8 y(n-2) = x(n)-
x(n-1) (16)
5. Determine the pole and zero plot for the system described difference equations
y(n)=x(n)+2x(n-1)-4x(n-2)+x(n-3) (16)
6.A system has unit sample response h(n) =-1/4_(n+1)+1/2_(n)-1/4 (n-1).Is the system
BIBO stable? Is the filter is Causal? Find the frequency response? (16)
7. Find the output of the system whose input-output is related by the difference equation
y(n) -5/6 y(n-1) +1/6 y(n-2) = x(n) -1/2 x(n-1) for the step input. (16)
8. Find the output of the system whose input-output is related by the difference equationy(n) -5/6
y(n-1) +1/6 y(n-2) = x(n) -1/2 x(n-1) for the x(n) =4nu(n). (16)
1. (a) Determine the Fourier transform of x (n) =a|n|;-1<1 (8)
(b) Determine the Inverse Fourier transform H (w) = (1-ae-jw) -1(8)
2. State and proof the properties of Fourier transform (16)
3. Determine the Discrete Fourier transform x (n) = (1, 1, 1, 1) (16)
4. Derive and draw the 8 point FFT-DIT butterfly structure. (16)
5. Derive and draw the 8 point FFT-DIF butterfly structure. (16)
6.Compute the DFT for the sequence.(0.5,0.5,0.5,0.5,0,0,0,0) (16)
7.Compute the DFT for the se
quence.(1,1,1,1,1,1,0,0) (16)
8.Find the DFT of a sequence x(n)=(1,1,0,0) and find IDFT of Y(k) =(1,0,1,0) (16)
9. If x (n) = sin (n_/2), n=0, 1, 2, 3h (n) = 2n, n=0,1,2,3.Find IDFT and sketch it. (16)
10.(a) Find 4 point DFT using DIF of x(n) =(0,1,2,3) (8)
(b).Proof x(n)*h(n) =X(z) H(z) (8)
11.Discuss the properties of DFT. (16)
12.Discuss the use of FFT algorithm in linear filtering. (16)
13.Explain the application of DFT in linear filtering and spectral analysis? (16)

UNIT –III
PART A:
1. Define canonic and non canonic form realizations.(2)
2. Draw the direct form realizations of FIR systems? (2)
3. Mention advantages of direct form II and cascade structure? (2)
4. Define Bilinear Transformation. (2)
5. What is prewar ping? Why is it needed? (2)
6. Write the expression for location of poles of normalized Butterworth filter. (2)
7. Distinguish between FIR and IIR Filters. (2)
8. What is linear phase filter? (2)
9. What are the design techniques available for IIR filter? (2)
10. What is the main drawback of impulse invariant mapping? (2)
11. Compare impulse invariant and bilinear transformation. (2)
12. Why IIR filters do not have linear phase? (2)
13. Mention the properties of Butterworth filter? (2)
14. Mention the properties of Chebyshev filter? (2)
15. Why impulse invariant method is not preferred in the design of high pass IIR filter? (2)
16. Give the transform relation for converting LPF to BPF in digital domain. (2)

PART B:
1. Obtain the cascade and parallel form realizations for the following systems (16)Y (n) =-0.1(n-
1) + 0.2 y (n-2) + 3x (n) +3.6 x (n-1) +0.6 x (n-2)
2. Obtain the Direct form I and II
y (n) = -0.1(n-1) + 0.72 y(n-2) + 0.7x(n) -0.252 x(n-2) (16)
3. Obtain the (a) Direct forms i(a) cascade ii(a) parallel form realizations for the following
systems y (n) = 3/4(n-1) –1/8 y(n-2) + x(n) +1/3 x(n-1) (16)
4. Find the direct form I and IIH (z) =8z-2+5z-1+1 / 7z-3+8z-2+1 (16)
5.Find the direct form –I, cascade and parallel form for (16)H(Z) = z-1-1 / 1 –0.5 z-1+0.06 z-2
6. Explain the method of design of IIR filters using bilinear transform method. (16)
7. (a)Discuss the limitations of designing an IIR filter
using impulse invariant method. (8)
(b)Derive bilinear transformation for an analog filter with system function H(s) = b/ s + a (8)
8 (a)For the analog transfer function H(s) = 2 / (s+1) (s+3) .Determine H (z) using bilinear
transformation. With T=0.1 sec (8)
(b)Convert the analog filter H(s) = 0.5 (s+4) / (s+1)(s+2) using impulse invariant
transformation T=0.31416s (8)
9. The normalized transfer function of an analog filter is given byHa(sn) = 1/ sn2+1.414 sn+1.
Convert analog filter to digital filter with cut offfrequency of 0.4_using bilinear
 transformation.(16)
10. Design a single pole low pass digital IIR filter with -
3db bandwidth of 0.2_by using bilinear transformation. (16)
11.For the constraints0.8_|H (ejw)|_1, 0_ _ _0.2_|H (ejw)|_0.2, 0.6_ _ _ __with T= 1
sec.Determine systemfunction H(z) for a Butterworth filter using Bilinear transformation. (16)
12.Design a digital Butterworth filter satisfying the following specifications0.7_|H (ejw)|_
1, 0_ _ _0.2_|H (ejw)|_0.2, 0.6_ _ _ __with T= 1 sec .Determine system
function H(z) for a Butterworth filter using impulse invariant transformation. (16)
13. Design a digital Chebyshev low pass filter satisfying the following specifications
0.707_|H (ejw)|_1, 0_ _ _0.2_|H (ejw)|_0.1 0.5_ _ __with T= 1 sec using for bilinear
transformation. (16)
14.Design a digital Butterworth High pass filter satisfying the following specifications0.9_|H(ejw)|_1, 0_
_ _ _/2|H (ejw)|_0.2, 3_/4_ _ __with T= 1 sec. using impulse
invariant transformation. (16)
15. Design a realize a digital filter using bilinear transformation for the followingspecifications
i) Monotonic pass band and stop band
ii) -3.01 db cut off at 0.5_rad
iii) Magnitude down at least 15 db at_= 0.75_rad. (16)

UNIT –IV
FIR FILTER DESIGN
PART A :
1. What are Gibbs oscillations?(2)
2. Explain briefly Hamming window(2).
3. If the impulse response of the symmetric linear phase FIR filter of length 5 is h(n) =
{2, 3, 0, x, y), then find the values of x and y.(2)
4. What are the desirable properties of windowing technique?(2)
5. Write the equation of Bartlett window.(2)
6.Why IIR filters do not have linear phase?(2)
7.Why FIR filters are always stable?(2)
8.Why rectangular window are not used in FIR filter design using window method?(2)
9.What are the advantages of FIR filter? (2)
10.What are the advantages and disadvantages of window? (2)
11.What is the necessary condition and sufficient condition for the linear phase characteristic of a FIR
filter? (2)
12.Compare Hamming and Hanning window? (2)
13.Why triangular window is not a good choice for designing FIR Filter? (2)
14.Why Kaiser window is most used for designing FIR Filter? (2)
15.What is the advantages in linear phase realization of FIR systems? (2)
PART B:
1. Prove that an FIR filter has linear phase if the unit sample response satisfies the condition
h(n) = ± h(M-1-n), n =0,1,..... M-1.Also discuss symmetric and anti symmetric cases of FIR
filter. (16)
2. Explain the need for the use of window sequence in the design of FIR filter. Describe the
window sequence generally used and compare the properties. (16)
3. Explain the type 1 design of FIR filter using Frequency sampling technique. (16)
4. A LPF has the desired response given below (16)H (ej) =e-3 j,0_ _ _ _/20._/2_ _ _ _.Determine the
filter coefficients h(n) for M=7using frequency sampling technique.
5. Design a HPF of length 7 with cut off frequency of 2 rad/sec using Hamming window. Plot
the magnitude and phase response. (16)
6. Explain the principle and procedure for designing FIR filter using rectangular window (16)
7. Design a filter with Hd(e-j) =e -3 j,_/4_ _ _ _/40._/4_ _ _ _using a Hanning window with N=7.(16)
8.Design a FIR filter whose frequency response (16)H (ej) =1_/4_ _ _3_/40. |_|_
3_/4.Calculate the value of h(n) for N=11 and hence find H(z).
9.Design an ideal differentiator with frequency response H (ej) =jw -_ _ _ _ _using
hamming window for N=8 and find the frequency response. (16)
10.Design an ideal Hilbert transformer having frequency responseH (ej) =j -_ _ _ _0-j 0_ _ _ _
for N=11 using Blackman window. (16)

FIR structures:
12.(a) Determine the direct form of following system (8)H (z) =1+2z-1-3z-2+ 4z-3-5z-4
(b) Obtain the cascade form realizations of FIR systems (8)H (z) = 1+5/2 z-1+ 2z-2+2 z-3

UNIT -V
FINITE WORDLENGTH EFFECTS
PART A:
1. What are the three quantization errors due to finite world length registers in digital filters?(2)
2. What do you mean by limit cycle oscillations? (2)
3. Explain briefly quantization noise. (2)
4. Represent 15.75 in fixed point and in floating point representations. (2)
5.What is the need for scaling in digital filters? (2)
6.List the well known techniques for linear phase FIR filter? (2)
7.What is quantization step size? (2)
8.State the advantages of floating point over fixed point representations? (2)
9.Why rounding preferred over truncation in realizing digital filter? (2)
10.What is meant by dead band? (2).
11.What is over flow limit cycle? How overflow can be eliminated? (2)
12.Sketch the noise probability density functions for rounding? (2)
13.Sketch the noise probability density functions for truncation?. (2)
14.What is meant by finite word length effect in digital filter? (2)
15.Explain the fraction 7/8 and -7/8 in sign magnitude ,1’s, 2’s complement. (2)
16.Convert in decimal to binary 20.675 (2)
17. Convert in binary to decimal 1110.01 (2)
18. What is product quantization error? (2)
19. What is input quantization error? (2)
20.What is coefficient quantization error? (2)
PART B:
1. Explain in details about quantization in floating point realizations of IIR filter? (16)
2. 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 three bits plus a sign bit.
The input x(n) = 0.875_(n).Explain the limit cycle oscillations in the above filter, if
quantization is performed by means of rounding and signed magnitude representation is
used.(16)
3. Explain briefly(1) Effects of coefficient quantization in filter design. (6)(2) Effects of product
round off error in filter design. (6)(3) Speech recognition (4)
4. Explain briefly
(A)Define limit cycle oscillation. Explain. (8)
(B) Explain the different representation of fixed and floating point representation. (8)
5.Two first order LPF whose system function are given below connected in cascade.
Determine the over all output noise power (16H1(z) = 1/1-
0.9z-1and H2(z) =1/1-0.8z-1
6. (a) Describe the quantization error occur in rounding and truncation in twos complement.(8)
(b) Draw a sample and hold circuit and explains its operation? (8)
7. (a)Explain dead band in limit cycles? (8)
(b)Draw the stastical model of fixed point product quantization and explain (8)
8. (a)What is dead band of a filter? Derive the dead band of second order linear filter? (12)
(b)Consider all pole second order IIR filter described by equation y(n) = -0.5 y(n-1) –0.75y(n-
2) + x(n).Assuming 8 bits to represent pole, determine the dead band region governing
The limit cycle. (4)
9. Determine the variance of the round off noise at the output of two cascaded of the filter with
system function H(z) =H1(z) .H2(z) where H1(z) =1 / 1-0.5 z-1H2(z) = 1 / 1-0.25z-1(16)
10. Explain with suitable examples the truncation and rounding off errors (16)
11 .a) Explain the application of DSP in Speech processing? (8)
b) What is a vocoder? Explain with a block diagram? (8)
12. Determine the dead band of the filter of y(n) = 0.95 y(n-1) +x(n) (16)