UNIT – I

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1. Consider the analog signal x(t) = 3cos50πt + 10 sin 300πt – cos 100πt. What is the nyquist
rate for this signal

Here ωmax=300π
So, 2πfm=300π
Hence, Nyquist rate Fs=2fm=300
2. State Shannon’s sampling theorem.
A band limited continuous time signal with highest frequency (band width) fm hertz ,
can be uniquely recovered from its samples provided that the sampling rate fs is greater
than or equal to 2fm samples per second
Fs≥2fm
3. Given a continuous time signal x(t)= 2cos500πt. What is the Nyquist rate and
fundamental frequency of the signal?
ω =500π
2πf= 250π
f= 250Hz
Nyquist rate Fs=2fm= 2x250= 500Hz
4. What is the Nyquist rate for the signal xa(t)=3cos 600πt+2cos1800πt?

Solution: ω1=600π ω2=1800π

2πf1= 600π 2πf2= 1800π
f1= 300Hz f2= 900Hz
Nyquist rate Fs=2fm= 2x900= 1800Hz.
5. What are the steps involved in digital signal processing?
 Converting the analog signal to digital signal, this is performed by
A/D converter 

 Processing Digital signal by digital system. 

 Converting the digital signal to analog signal, this is performed by
D/A converter. 

6. Give some applications of DSP?
  Speech processing – Speech compression & decompression for
voice storage system 

 Communication – Elimination of noise by filtering and echo cancellation. 

 Bio-Medical – Spectrum analysis of ECG, EEG etc. 
7. What is an Energy and Power signal?
Energy signal:
A finite energy signal is periodic sequence, which has a finite energy but
zero average power.
Power signal:
An Infinite energy signal with finite average power is called a power signal.
8. What is Discrete Time Systems?
The function of discrete time systems is to process a given input sequence to generate
output sequence. In practical discrete time systems, all signals are digital signals, and
operations on such signals also lead to digital signals. Such discrete time systems are
called digital filter.
9. Write the Various classifications of Discrete-Time systems.
 Linear & Non linear system
 Causal & Non Causal system
 Stable & Un stable system
10. Define Linear system
A system is said to be linear system if it satisfies Super position principle. Let
us consider x1(n) & x2(n) be the two input sequences & y1(n) & y2(n) are the
responses respectively,
T[ax1(n) + bx2(n)] = a y1(n) + by2(n)
11. Define Static & Dynamic systems
When the output of the system depends only upon the present input sample, then it is
called static system, otherwise if the system depends pa t values of input then it is called
dynamic system
12. Define causal system.
When the output of the system depends only upon the present and past input
sample, then it is called causal system, otherwise if the system depends on future
values of input then it is called non-causal system
13. What are the basic elements used to construct the block diagram of discrete
time system?
The basic elements used to construct the block diagram of discrete time
Systems are Adder, Constant multiplier &Unit delay element.
14. What is ROC in Z-Transform?
The values of z for which z – transform converges is called region ofconvergence (ROC).
The z-transform has an infinite power series; hence it is necessary to mention the ROC along with
z-transform.

15. What are the properties of convolution?

1. Commutative property x(n) * h(n) = h(n) * x(n)
2. Associative property [x(n) * h1(n)]*h2(n) = x(n)*[h1(n) * h2(n)]
3. Distributive property x(n) *[ h1(n)+h2(n)] = [x(n)*h1(n)]+[x(n) * h2(n)
 UNIT-II
DISCRETE TIME SYSTEM ANALYSIS

1. Define DTFT.
Let us consider the discrete time signal x(n).
Its DTFT is denoted as X(w).It isgiven as X(w)= x(n)e-jwn
2. State the condition for existence of DTFT?
The conditions are
• If x(n)is absolutely summable then |x(n)|<
• If x(n) is not absolutely summable then it should have finite energy for DTFT to exit.

3. List the properties of DTFT.

Periodicity
Linearity
Time shift
Frequency shift
Scaling
Differentiation in frequency domain
Time reversal
Convolution
Multiplication in time domain
Parseval’s theorem

4. Define circularly even sequence.

A Sequence is said to be circularly even if it is symmetric about the point zero on the circle.
x(N-n)=x(n),1<=n<=N-1.

5. Define circularly odd sequence.

A Sequence is said to be circularly odd if it is anti symmetric about point x(0) on the circle

6. Define circularly folded sequences.

A circularly folded sequence is represented as x((-n))N. It is obtained by plotting x(n) in
clockwise direction along the circle.
7 . State circular convolution.
This property states that multiplication of two DFT is equal to circular convolution of their
sequence in time domain.

8. State parseval’s theorem.

Consider the complex valued sequences x(n) andy(n).
If x(n)y*(n)=1/N X(k)Y*(k)
9. Define Z transform.

The Z transform of a discrete time signal x(n) is denotedby X(z) and is givenby
 X(z)= x(n)Z-n.
10. Define ROC.
The value of Z for which the Z transform converged is called region of convergence.

11.Define time shifting property

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

12. List the methods of obtaining inverse Z transform.

Inverse z transform can be obtained by using

1. Partial fraction expansion.
2. Contour integration
3. Power series expansion
4. Convolution.

UNIT-III

DISCRETE FOURIER TRANSFORM AND COMPUTATION

1. 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 w=0 to 2л.

2. Define N point DFT.

The DFT of discrete sequence x(n) is denoted by X(K). It is given by,
Here k=0,1,2…N-1
Since this summation is taken for N points, it is called s N-point

3. 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)

5. What is the disadvantage of direct computation of DFT?

For the computation of N-point DFT, N2 complex multiplications and N[N-1]
Complex additions are required. If the value of N is large than the number of into lakhs. This proves
inefficiency of direct DFT computation.

6. What is the way to reduce number of arithmetic operations during D

computation?
 Number of arithmetic operations involved in the computation of DFT is greatly reducing using
different FFT algorithms as follows.
1. Radix-2 FFT algorithms.
2. Radix-2 Decimation in Time (DIT) algorithm.
3. Radix-2 Decimation in Frequency (DIF) algorithm.
4. Radix-4 FFT algorithm

7. What is the computational complexity using FFT algorithm?

1. Complex multiplications = N/2 log2N
2. Complex additions = N log2N

7. How many multiplications and additions are required to compute N-point DFT
using redix-2 FFT?
The number of multiplications and additions required to compute N-point DFT using
redix-2 FFT are N log2N and N/2 log 2N respectively.

8. What are the applications of FFT algorithms?

1. Linear filtering
2. Correlation
3. Spectrum analysis

9. Why FFT is needed?

The direct evaluation of the DFT using the formula requires N 2 complex multiplications
and N (N-1) complex additions. Thus for reasonably large values of N (in order of 1000) direct evaluation
of the DFT requires an inordinate amount of computation. By using FFT algorithms the number of
computations can be reduced.

10. What is a decimation-in-frequency algorithm?

In decimation-in-frequency algorithm the output sequence X (K) is divided into two N/2 point
sequences and each N/2 point sequences are in turn divided into two N/4 point sequences.

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

.12. Distinguish between FIR and IIR filters.

FIR filter IIR filter

 These filters can be easily designed to have perfectly These filters do not have linear phase.
linear phase.
FIR filters can be realized recursively and non- IIR filters can be realized recursively.
recursively.
Greater flexibility to control the shape of their Less flexibility,usually limited to kind of filters.
magnitude response.
Errors due to roundoff noise are less severe in FIR The roundoff noise in IIR filters are more.
filters, mainly because feedback is not used.

UNIT IV

DESIGN OF DIGITAL FILTERS

PART A
1. How phase distortion and delay distortion are introduced?

The phase distortion is introduced when the phase characteristics of a filter is nonlinear within
the desired frequency band. The delay distortion is introduced when the delay is not constant within
the desired frequency band.
2. What is mean by FIR filter?

The filter designed by selecting finite number of samples of impulse response h (n) obtained from
inverse Fourier transform of desired frequency response H(w) are called FIR filters

3. Write the steps involved in FIR filter design

 Choose the desired frequency response Hd(w)
 Take the inverse Fourier transform and obtain Hd(n)
 Convert the infinite duration sequence Hd(n) to h(n)
 Take Z transform of h(n) to get H(Z)

4. What are advantages of FIR filter?

 Linear phase FIR filter can be easily designed
 Efficient realization of FIR filter exists as both recursive and non-recursive structures.
 FIR filter realized non-recursively stable.
 The round off noise can be made small in non recursive realization of FIR filter.

5. What are the disadvantages of FIR FILTER

The duration of impulse response should be large to realize sharp cutoff filters. The non
integral delay can lead to problems in some signal processing applications.

6. List the well known design technique for linear phase FIR filter design?
 Fourier series method and window method
 Frequency sampling method
  Optimal filter design method

7. For what kind of application, the symmetrical impulse response can be used?

The impulse response, which is symmetric having odd number of samples, can be used to
design all types of filters, i.e., lowpass, highpass, bandpass and band reject. The symmetric impulse
response having even number of samples can be used to design lowpass and bandpass filter.
.
8. Under what conditions a finite duration sequence h(n) will yield constant group delay in its
frequency response characteristics and not the phase delay?

If the impulse response is anti symmetrical, satisfying the condition H(n)=-h(N-1-n) The
frequency response of FIR filter will have constant group delay and not the phase delay.

9. What are the properties of FIR filter?

1. FIR filter is always stable.
2. A realizable filter can always be obtained.
3. FIR filter has a linear phase response.

10. What are the disadvantages of Fourier series method?

In designing FIR filter using Fourier series method the infinite duration impulse
response is truncated at n= ± (N-1/2).Direct truncation of the series will lead to fixed
percentage overshoots and undershoots before and after an approximated discontinuity in the
frequency response .
11. What is Gibbs phenomenon? OR What are Gibbs oscillations?
One possible way of finding an FIR filter that approximates H(ejw)would be to truncate the
infinite Fourier series at n= ± (N-1/2).Abrupt truncation of the series will lead to oscillation
both in pass band and is stop band .This phenomenon is known as Gibbs phenomenon.
12. What are the desirable characteristics of the windows?
The desirable characteristics of the window are

1. The central lobe of the frequency response of the window should contain most of the energy
and should be narrow.
2. The highest side lobe level of the frequency response should be small.
3. The side lobes of the frequency response should decrease in energy rapidly as w tends to p

13. What is the necessary and sufficient condition for linear phase characteristics in FIR filter?
The necessary and sufficient condition for linear phase characteristics in FIR filter is the
impulse response h(n) of the system should have the symmetry property, i.e, H(n) = h(N-1-n)
Where N is the duration of the sequence
14. What are the advantages of Kaiser Widow?
1. It provides flexibility for the designer to select the side lobe level and N.
2. It has the attractive property that the side lobe level can be varied continuously from the low
value in the Blackman window to the high value in the rectangle window.

15. What are the different types of filters based on impulse response?
 Based on impulse response the filters are of two types 1. IIR filter 2. FIR filter The IIR filters
are of recursive type, whereby the present output sample depends on the present input, past input
samples and output samples. The FIR filters are of non recursive type, whereby the present output
sample depends on the present input sample and previous input samples.

16 . What are the different types of filters based on frequency response?

Based on frequency response the filters can be classified as
1. Lowpass filter
2. Highpass filter
3. Bandpass filter
4. Bandreject filter
17 . What are the advantages and disadvantages of FIR filters?

Advantages:
1. FIR filters have exact linear phase.
2. FIR filters are always stable.
3. FIR filters can be realized in both recursive and non recursive structure.
4. Filters with any arbitrary magnitude response can be tackled using FIR sequence.
Disadvantages:
1. For the same filter specifications the order of FIR filter design can be as high as 5 to 10 times
that in an IIR design.
2. Large storage requirement is requirement
3. Powerful computational facilities required for the implementation.
18. How one can design digital filters from analog filters? ·
1.Map the desired digital filter specifications into those for an equivalent analog filter. ·
2.Derive the analog transfer function for the analog prototype. ·
3. Transform the transfer function of the analog prototype into an equivalent digital filter
transfer function
.

19. What is meant by impulse invariant method of designing IIR filter?

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. If the transfer function is
of the form, 1/s-p, then H (z) =1/1-e-pTz-1

20. What is bilinear transformation?

The bilinear transformation is a mapping that transforms the left half of S-plane into the
unit circle in the Z-plane only once, thus avoiding aliasing of frequency components. The
mapping from the S-plane to the Z-plane is in bilinear transformation is S=2/T(1-Z-1/1+Z-1

21. Write a short note on pre-warping.

The effect of the non-linear compression at high frequencies can be compensated. When
the desired magnitude response is piece-wise constant over frequency, this compression can be
compensated by introducing a suitable pre-scaling, or pre-warping the critical frequencies by
using the formula.
 22. 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

23. What are the advantages & disadvantages of bilinear transformation?

Advantages: ·
1.The bilinear transformation provides one-to-one mapping. ·
2. Stable continuous systems can be mapped into realizable, stable digital systems. ·
3. There is no aliasing.
Disadvantage: ·
1. The mapping is highly non-linear producing frequency, compression at high frequencies.
·2. Neither the impulse response nor the phase response of the analog filter is preserved in a
digital filter obtained by bilinear transformation.

UNIT V
DIGITAL SIGNAL PROCESSOR

1. Write short notes on general purpose DSP processors

General-purpose digital signal processors are basically high speed microprocessors with
hard ware architecture and instruction set optimized for DSP operations. These processors
make extensive use of parallelism, Harvard architecture, pipelining and dedicated
hardware whenever possible to perform time consuming operations

2. Write notes on special purpose DSP processors.

There are two types of special; purpose hardware.
(i) Hardware designed for efficient execution of specific DSP algorithms such as digital
filter, FFT.
(ii) Hardware designed for specific applications, for example telecommunication, digital
audio.

3.What about of Harvard architecture?

The principal feature of Harvard architecture is that the program and the data memories lie
into separate spaces, permitting full overlap of instruction fetch and execution.
Typically these types of instructions would involve their distinct type.
1. Instruction fetch
2. Instruction decode
3. Instruction execute

4. What are the types of MAC is available?

There are two types MAC’S available
1. Dedicated & integrated
2. Separate multiplier and integrated unit
5. What is meant by pipeline technique?
The pipeline technique is used to allow overall instruction executions to overlap. That is
where all four phases operate in parallel. By adapting this technique, executi n speed is
increased.

6. What are four phases available in pipeline technique?

The four phases are
(i) Fetch
(ii) Decode
(iii)Read
(iv) Execution

7.Write down the name of the addressing modes.

Direct addressing.
Indirect addressing.
Bit-reversed addressing.
Immediate addressing.
Short immediate addressing.
Long immediate addressing.
Circular addressing

8.What are the instructions used for block transfer in C5X Processors?

The BLDD, BLDP and BLPD instructions use the BMAR to point at the source or
destination space of a block move. The MADD and MADS also use the BMAR to address an
operand in program memory for a multiply accumulator operation

9. What is meant by auxiliary register file?

The auxiliary register file contains eight memory-mapped auxiliary registers (AR0-AR7),
which can be used for indirect addressing of the data memory or for temporary data storage.

10. Write the name of various part of C5X hardware.

1. Central arithmetic logic unit (CALU)

2. Parallel logic unit (PLU)
3. Auxiliary register arithmetic unit (ARAU)
4. Memory-mapped registers.
5. Program controller.

11. Write short notes about parallel logic unit.

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

12. What is meant by auxiliary register file?

The auxiliary register file contains eight memory-mapped auxiliary registers (AR0-AR7),
which can be used for indirect addressing of the data memory or for temporary data storage. Indirect
auxiliary register addressing allows placement of the data memory address of an instruction operand
into one of the AR.