DSP - Question Bank

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DSP - Question Bank

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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 }

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

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

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)]

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

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= .

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)

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)

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)

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.

3.

linear filter.

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?

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

Where

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

* 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 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.

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.

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?

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 )

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

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

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)

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.

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.

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?

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

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

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)

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

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.

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.

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.

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

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.

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

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}

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)

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

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.

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

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).

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.

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}

(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)

(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

(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)

(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)

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.

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

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.

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)

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)

0.7 _H()_ 1.0 , 0 /2

_ H()_ 0.2, 3/4

With T= 1 sec, design a Butterworth filter.

(16)

(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)

(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

22. Design a filter with

Hd() = e

-j2

- /4 /4

/4

=0

23. Discuss about frequency transformations in detail.

(16)

(8)

Hd() = e

-j3

- 3/4 3/4

3/4

=0

(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

(16)

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

-1

-2

-1

-2

j

H(e ) = j

= -j

for - 0

for 0

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 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

(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)

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)

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)

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

-1

-1

H2 (z) = 1 / 1- 0.25 z

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

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

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