1.

a Explain the Classifications of Discrete Time Systems

1.b Determine whether the following signals are energy or power signals.
𝟏 n 𝒏𝝅
(i) x[n]= ( ) u[n] (ii) x[n]=sin ( )
𝟒 𝟑
2.Determine whether the system y(n) = x(-n-2) is i) Causal ii) Linear iii) Dynamic iv) Time
invariant v) Stable
3.a What are the necessary and sufficient conditions for stability of a system? Prove it
3. b) Prove that convolution in time domain leads to multiplication in frequency domain for
discrete time signals
4. Determine the frequency response of an LTI system described by
𝟑 𝟏
y(n) - 𝟒 y(n-1) + 𝟖
y(n-2) = x(n)
5.Determine the convolution sum of two sequences using graphical method: x(n) = {3, 2, 1, 2};
h(n) = {1, 2, 1, 2}
6. a) Determine the convolution of the following signal sequences
 𝟏 n
x(n)= 2n u(n) & h(n)= ( ) u(n)
𝟐
6.b.Determine the pole-zero plot for the system described by the difference equation
y(n)=(5/6)y(n-1)-(1/6)y(n-2)+x(n)-x(n-1)
 Unit 2

1. Determine the DFT of a sequence x(n) = {0, 1, 2, 3} and check the validity of the
answer by calculating IDFT
2.What is DFT? Explain the properties of DFT
3.Draw the magnitude and phase spectrum of the 8 – point DFT of
𝟎, 𝟎≤𝒏≤𝟏
𝒙(𝒏) = { 𝟏, 𝟐≤𝒏≤𝟓
𝟎, 𝟔≤𝒏≤𝟕
4.Calculate the circular convolution of the following sequences
x(n) = {2,1,2,1}
h(n) = {1,2,3,4}
using DFT and IDFT method
5.Find the DFT of a sequence x(n)= {1,2,3,4,4,3,2,1} using DIF algorithm
6.Find the DFT of a sequence x(n)= {1,2,3,4,4,3,2,1} using DIT algorithm
1.Design a digital Butterworth filter satisfying the following constraints
0.8≤ǀH(𝒆𝒋𝒘 )ǀ ≤ 𝟏 for 0 ≤ω ≤ 0.2π
𝒋𝒘
ǀH(𝒆 )ǀ ≤ 0.2 for 0.6π≤ ω ≤ π with T=1 sec using Impulse Invariance method
(10 M)
2. Develop the following IIR system using direct form – I, direct form – II, cascade
structure and parallel structure.
y(n) = – 0.1 y(n – 1) + 0.2 y(n – 2) + 3 x(n) + 3.6 x(n – 1) + 0.6 x(n – 2) (10M)
3.Design a digital Chebyshev filter to meet the following constraints
0.8≤ǀH(𝒆𝒋𝒘 )ǀ ≤ 𝟏 for 0 ≤ ω ≤ 0.2π
𝒋𝒘
ǀH(𝒆 )ǀ ≤ 0.2 for 0.6π ≤ ω ≤ π
with T= 1 sec using bilinear transformation
4.a Explain the IIR filter design by impulse invariant method with a neat mapping
Diagram
 𝟏
4.b Determine 𝑯(z) given H(s) = , using (i)impulse invariant method(ii)
𝒔𝟐 +𝟕𝐬+𝟏𝟎
Bilinear
Transformation
 UNIT -4
1a)Realize the following system function using minimum number of multipliers
(i)H(z)= 1+1/3 z-1+ 1/4 z-2+1/4 z-3+1/3 z-4+z-5
(ii)H(z)= (1+z-1) (1+1/2z-1+1/2 z-2+z-3)
1b) Compare IIR and FIR filters and give their applications
2. Design a low pass filter with the desired frequency Response
H(𝒆𝒋𝒘 ) = 𝒆−𝟓𝒋𝒘 for - π/2 ≤ω ≤ π/2
0 for π/2 ≤ ω ≤ π.
Use BlackmanWindow with N=11 (10 M)
3. Design a filter with the desired frequency Response
H(𝒆𝒋𝒘 ) = 𝒆−𝟑𝒋𝒘 for - π/4 ≤ω ≤ π/4
0 for π/4 ≤ ω ≤ π.
using Hamming Window with N=7. (10 M)
4. Design an ideal lowpass filter with a frequency response
H(𝒆𝒋𝒘 ) = 𝟏 for - π/2 ≤ω ≤ π/2
0 for π/2 ≤ ω ≤ π .Find the values of h(n) for N=11. Find H(z).
 UNIT-5
1a) Represent the following numbers in floating point format with five bits for mantissa
and three bits for exponent
(i)710 (ii) 0.2510 (iii) - 710 (iv) - 0.2510 (5M)
b)Distinguish fixed point and floating point arithmetic (5M)
2. Consider the discrete time signal x(n)= {1,2,3,4}. Determine the Upsampled and
Downsampled version of the signals for the sampling rate multiplication factor (a)I=3
(b)D=2
3. Describe the process of Decimation. With necessary equations explain the spectrum of
the Decimated signal (10M)
4. Explain Sampling rate conversion by a rational factor I/D (10M)
 AY: 2024 – 25
III B. TECH – II SEM – (R-21) DIGITAL SIGNAL PROCESSING (21A040422)

Two Mark Questions & Answers

1. Define ROC and state its properties

Region of convergence of X(z) is the set of all value of z for which z transform attains
a finite value.
1.The Region of convergence is a ring or disk in the Z plain centred at the origin.
2. The Region of convergence cannot contain any poles.
3. The Region of convergence of an LTI stable system contains the unit circle.
4. The Region of convergence must be a connected region.

2. Mention any some applications of DSP

1. Speech Processing – Speech Compression & decompression for voice storage
system and for transmission and reception of voice signals.

2. Communication – Elimination of noise by filtering and echo cancellation by

adaptive Filtering in transmission channels.

3. Biomedical – Spectrum analysis of ECG, EEG, etc., signals to identify various

disorders in heart, brain, Etc

3. Give the similarities and differences between DIT and DIF-FFT algorithms

Difference:

For DIT, the input is bit reversed while the output is in natural order, where as for DIF,
the input is in natural order while the bit is 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.
Similarities:

Both algorithms require same number of operations to compute the DFT, both
algorithms can be done in place and both need to perform but reversal at some place
during the computation

4. What is Twiddle factor? State its properties

It is a rotating vector Quantity. Twiddle factors are a set of values that are used in FFT
to speed up the operation of DFT & IDFT.
𝟐𝜫
WN = ⅇ−𝒋 𝑵

Properties
1. Periodicity property
WNk+N = WNk
2. Symmetry property
WNk+N/2 = - WNk

5. What is meant by zero padding?

Appending zeros to a sequence in order to increase the size or length of the sequence is
called zero padding. In circular convolution, when the two input sequence are of
different size, then they are converted to equal to size by zero padding.

6. What is the condition for system stability?

The necessary and sufficient condition guaranteeing the stability of a linear time –
invariant system is that its impulse response is absolutely summable.

7. What are the properties of convolution?

i)Commutative law:
x (n) * h (n) = h (n) * x (n)
ii) Associative law:
[x (n)*h1 (n) * h2 (n) = x (n) * [h1 (n) * h2 (n)]

iii) Distributive law:

x (n) * [h1 (n) + h2 (n)] = x (n) * h1 (n) + x (n) * h2 (n)
8. What are the advantages of FFT algorithm over direct computation DFT

FFT requires less multiplications & additions compared to direct computation

DFT
FFT algorithm can be implemented fast on the DSP processor
The calculation of DFT and IDFT both are possible by proper combination of
FFT algorithm

9. What are the applications of FFT algorithm?

The applications of FFT algorithm includes,

i) Linear Filtering
ii) Correlation
iii) Spectrum analysis

10. If DFT of x(n) is X(k) What is the DFT of x(n-1)

𝟐𝝅𝒌𝒏𝟎
DFT of a time delayed sequence x(n-n0) = X(k) ⅇ−𝒋 𝑵
𝟐𝝅𝒌
So x(n-1) = X(k) ⅇ−𝒋 𝑵
here n0 value is 1

11. What are the advantages of DSP?

The advantages of DSP are

i) The programs can be modified easily for better performance.
ii) Better accuracy can be achieved by using adaptive algorithm.
iii) The digital signals are easily stored and transported.
iv) The digital systems are cheaper than analog equivalent.
12. Define energy & power signals

A signal x(n) is called as energy signal if the energy satisfies 0<E<∞. For an energy signal
P=0.

A signal x(n) is called Power signal if power satisfies 0< P< ∞. For a power signal energy is
infinite
13. Compute the DFT of unit impulse Sequence.

14. Draw the basic butterfly diagram of radix -2 DIF-FFT

15. Differentiate DIF & DIT FFT

1. What is frequency warping
In bilinear transformation the relation between analog and digital frequencies is non
linear. When the s plane is mapped into z plane using bilinear transformation , this non
linear relationship introduces distortion in frequency axis , which is called frequency
warping or warping effect.

2. 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
2(1+𝑧 −1 )
S= 𝑇(1−𝑧 −1 )

3. What are the desirable characteristics of the window function?

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 ω tends
to π

4. What are the advantages and disadvantages of FIR filters

Advantages
Linear phase FIR filter can be easily deigned.
Efficient realization of FIR filter exists as both recursive and non – recursive structures.
FIR filter realize non recursive realization is stable.
The round off noise can be made small in non recursive realization of FIR filter.
Disadvantages
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.
5. What is finite word length effects
The effects due to finite precision representation of number in a digital system are
commonly referred to as Finite word length effects

6. State the advantage of direct form II structure over direct form I structure.
In direct form II structure, the number of memory locations required is less than that of
direct form I structure

7. Mention the various methods for digitizing the transfer function of an analog filter.
Impulse invariance method
Bilinear transformation method
Approximation of derivatives
The matched z transformation technique

8. What is Gibb’s phenomenon?

One possible way of finding an FIR filter that approximates H (e jw) would be to truncate
the infinite Fourier series at n = ± (N -1)/2. Abrupt truncation of the series will lead to
Oscillation both in passband and stopband.This phenomenon is called Gibbs
phenomenon.

9. What is Truncation
The truncation is the process of reducing the size of binary number (or reducing the
number of bits in a binary number) by discarding all bits less significant than the least
significant bit that is retained.

10. What is limit cycle.

In recursive systems, when the input is zero or some nonzero constant value, the
nonlinearities due to finite precision arithmetic operations may cause periodic
oscillations in the output. Such oscillations are called limit cycles.

11. Write short notes 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
12. Define IIR filter?
The filter designed by considering all the infinite samples of impulse response are called
IIR filter. The IIR filters are of recursive type, where by the present output sample
depends on the present input, past input samples and output samples

13. What is the necessary and sufficient condition for the linear phase characteristic of FIR
filter?
The phase function should be a linear function of ω, which in turn requires constant
group delay and phase delay.

14. List the well-known design technique for linear phase characteristic of a FIR filter
design?
1. Fourier series method and window method.
2. Frequency sampling method.
3. Optimal filter design method

15. What is Quantization error

Quantization error is due to representation of the sampled signal by a fixed number of digital
levels.