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Unit-1 Signals and Systems

1. Define correlation and list its types.
Correlation is a measure of the degree to which two signals are similar.
There are two types, 1.Auto correlation 2.Cross correlation.
2. What are the elementary discrete time signals?
• Unit sample sequence (unit impulse)
δ (n)= {1, n=0
0, Otherwise
• Unit step signal
U (n) ={1, n > 0
0, Otherwise
• Unit ramp signal
Ur(n)={n, for n > 0
0, Otherwise
• Exponential signal
x (n) = an where a is real
x(n)-Real signal
3. Define periodic and aperiodic signal.
A signal x (n) is periodic in period N, if x (n + N) = x (n) for all n. If a signal does not
satisfy this equation, the signal is called aperiodic signal.
4. What are energy and power signals? ∞
The energy of a discrete time signal is defined as, E = ∑ │x(n)│2
n=-∞ N
1
The average power of a discrete time signal is defined as, P = lim ∑ │x(n)│2
2N +1
N--> ∞ n=-N
5. Define symmetric and antisymmetric signal.
A real value signal x (n) is called symmetric (even) if x (- n) =x (n). On the other hand the
signal is called antisymmetric (odd) if x (- n) = - x (n).
6. Define dynamic and static system.
A discrete time system is called static or memory less if its output at any instant 'n' depends
on the input sample at the same time but not on past and future samples of the input.
y(n) =a x (n)
In any other case the system is said to be dynamic and to have memory.
y(n) =x (n) + 3 x(n-1)
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7. Define time variant and time invariant system.

A system is called time invariant if its output , input characteristics does not change with
time. y(n) = x(n)+x(n-1)
A system is called time variant if its input, output characteristics changes with time.
y(n) = x(-n).
8. Define linear and Non-linear system.
Linear system is one which satisfies superposition principle.
T [a1x1(n) + a2x2(n)]=a1T [x1(n)] + a2 T[x2(n)]
Superposition principle: The response of a system to a weighted sum of signals be equal to
the corresponding weighted sum of responses of system to each of individual input signal.
y(n)= n x(n)
A system which does not satisfy superposition principle is known as non-linear system.
y(n)=x2(n)
9. Define causal and anticausal system.
The system is said to be causal if the output of the system at any time ‘n’ depends only on
present and past inputs but does not depend on the future inputs.
y (n) =x (n) + x (n-1)
A system is said to be non-causal if a system does not satisfy the above definition.
10. What is BIBO stability? What is the condition to be satisfied for stability?
A system is said to be BIBO stable, if and only if every bounded input produces a bounded
output.
The condition for an LTI system is that the impulse response of the system should be
absolutely summable .
∞
∑ │h(k)│ < ∞
k= - ∞
11. Define sampling theorem.
A continuous time signal can be completely represented in its samples and recovered back if
the sampling frequency F ≥ 2 fmax.
Where F = Sampling frequency and fmax = The maximum frequency present in the signal.
12. What is meant by aliasing? How it can be avoided?
If the sampling frequency is less than twice of the highest frequency content of the signal,
then the aliasing occurs. In aliasing, the high frequencies of the signal mix with lower
frequencies and create distortion in frequency spectrum.
Aliasing can be avoided by two ways,
i) Sampling frequency must be higher than twice of highest frequency present in the signal.
ii) A low pass filter must be used before sampling to get a band limited signal.
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13. Define Region of convergence & List its properties.

The region of convergence (ROC) of X(Z) the set of all values of Z for which X(Z) attains
finite value.
Properties of ROC.
• The ROC does not contain any poles.
• When x(n) is two sided sequence, then ROC is entire Z-plane except Z=0 or Z=∞.
• If X(Z) is causal,then ROC includes Z=∞.
• If X(Z) is anti casual,then ROC includes Z=0.
14. State the initial value theorem and final value theorem.
If x (n) is causal sequence, then its initial value is given by,
If x (n) is causal sequence, then its final value is given by
15. Define linear (or) discrete convolution
The linear convolution of x (n) and h (n) is given as,

16. State the convolution property and time shifting property of z-transform.
Convolution property: If Z[x(n)] = X(Z) & Z[h(n)] = H(Z)
Then Z[x(n) * h(n)] = X(Z) . H(Z)
Time shifting property: If Z[x(n)] = X(Z)
then, Z[x(n-m)] = Z -m X(Z)
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UNIT – II FREQUENCY TRANSFORMATIONS

1. Define DFT and IDF or (DFT Pair).
DFT of a signal x(n) is given by
N-1
X(k)= ∑ x(n) e- j2πnk /N where K = 0 to N-1
n=0 N-1
IDFT of X(K) is defined by x(n)= 1/N ∑ X(K) e j2πnk /N where n = 0 to N-1
K=0
2. 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
3. What are the applications of FFT algorithm?
The applications of FFT algorithm includes
1) Linear filtering
2) Correlation
3) Spectrum analysis
4. 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
computation of DFT of a sequence of length N into successively smaller DFTs.
5. How many multiplications and additions are required to compute N point DFT using
redix-2 FFT?
N
The number of multiplications required to compute N point DFT is 2 log 2 N

The number of additions to compute N point DFT is N log 2 N

6. Why FFT is needed?
The direct evaluation of DFT requires N2 complex multiplications and N(N-1) 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.
7. Draw the basic butterfly diagram of DIT-FFT
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8. Distinguish between linear convolution and circular convolution of two sequences.

Linear convolution Circular convolution

If x(n) is a sequence of L samples and If x(n) is a sequence of L number of samples

h(n) with M samples, after convolution and h(n) with M samples, after convolution y(n)
y(n) will have N=L+M-1 samples. will have N = max(L,M) samples.

Zero padding is not necessary to find the Zero padding is necessary to find the response
response of a linear filter. of a filter.

Multiplication of two sequences in time Multiplication of two sequences in frequency

domain is called as Linear convolution domain is called as circular convolution.
9. Difference between overlap save and overlap add method
Overlap save method Overlap add method

In this method, L samples of the current In this method L samples from input
segment and (M-1) samples of the sequence and padding M-1 zeros forms
previous segment forms the input data data block of size N.
block.

Initial M-1 samples of output sequence There will be no aliasing in output data
are discarded which occurs due to blocks.
aliasing effect.

To avoid loss of data due to aliasing last Last M-1 samples of current output
M-1 samples of each data record are block must be added to the first M-1
saved. samples of next output block. Hence
called as overlap add method.
10. Draw the basic butterfly diagram of DIF-FFT
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11. Difference between DFT & FFT

DFT FFT

DFT requires large number of Radix-2 FFT algorithms requires less

computations as compared with FFT number of computations.
algorithms.
The number of multiplications = N2 The number of multiplications =
N
2 log 2 N

The number of additions N(N-1) The number of additions = N log 2 N

DFT does not requires Splitting Splitting operation is done on time domain
operation basis (DIT) or frequency domain basis
(DIF)
Processing time is more and more for Processing time is less
large number of N hence processor
remains busy
As the value of N in DFT increases , The As the value of N in DFT increases , The
efficiency of DFT decreases. efficiency of FFT decreases.
Used in applications that use smaller N Applied in Digital filter design , Linear
filtering.
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UNIT-3 IIR FILTER DESIGN

1. What do you mean by recursive and non recursive filters?
In recursive filter, the output y(n) is a function of past outputs, present and past inputs.
Example. IIR filter
In non-recursive filter, output y(n) is a function of present and past inputs only.
Example. FIR filter.
2. Which types of structures are used to realize IIR systems?
* Direct form -1structure (DF-1 )
* Direct form -2structure (DF-2 )
* Cascade form structure
* Parallel form structure
3. 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.
4. Distinguish direct-I and direct-II forms.
The direct-form I realization requires M+N+1 multiplications, M+N additions and M+N+1
memory locations.
The direct-form II realization requires M+N+1 multiplications, M+N additions and the
maximun of (M,N) memory locations.
5. What is warping effect or frequency warping?
The relation between the analog and digital frequencies in bilinear transformation is given
2 ω
by, Ω = T tan 2 . For smaller values of ω, there exists linear relationship between ω
and Ω .But for Larger values of ω, 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.
6. What do you understand by backward difference?
One of the simplest methods of converting analog to digital filter is to approximate the
dy(t ) y(nT) - y(nT - T)
differential equation by an equivalent difference equation. = at t=nT
dt T

7. How can you design digital filter from analog filter?

1. Map the desired digital filter specifications into those for an equivalent analog filter
2. Derive the analog transfer function for the analog protype.
3. Transform the transfer function of the analog prototype into an equivalent digital
filter transfer function.
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8. What is Bilinear transformation?

The bilinear transformation is conformal mapping that transforms the s-plane to z- plane. In
this mapping the imaginary axis of s-plane is mapped into the unit circle in z- plane. The left
half of s-plane is mapped into interior of unit circle in z-plane and the right half of s-plane is
mapped into exterior of unit circle in z-plane. The Bilinear mapping is a one-to-one mapping
and it is accomplished when
2 1  Z 
1
S= T
1+ Z 
1

9. 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 are 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.
10. Write a note on pre warping.
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.
2 ω
Prewarping frequency is given by, Ω = T tan  
2
11. Distinguish between Butterworth and Chebyshev filter.
1. The magnitude response of Butterworth filter decreases monotonically as the frequency
increases from 0 to ∞, whereas the magnitude response of the Chebyshev filter exhibits
ripple in the passband and monotonically decreasing in the stopband.
2. The transition band is more in Butterworth filter compared to Chebyshev filter.
3. The order of the Chebyshev filter is less than that of Butterworth.
12. State the advantages and disadvantages of FIR filter over IIR filter.
Advantages: i) FIR filter has linear phase characteristics. ii) FIR filters are inherently
stable. iii) The design of FIR filters is fairly simple compared to IIR filters.
Disadvantages: i) FIR filters need higher order compared to IIR filter. ii) Processing time is
more in FIR filter. iii) FIR filters need more memory. iv) FIR filters are all zero filters.
13. 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
3. Bilinear transformation
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UNIT-4 FIR FILTER DESIGN

1. What do you mean by Gibbs phenomenon?
jw
One possible way of finding an FIR filter that approximates H(e ) would be to truncate the

infinite fourier series at n = +

N  1 . Abrupt truncation of the series will lead to oscillation
2
in passband and in stopband. This phenomenon is known as Gibbs Phenomenon.
2. What is a window ?Why it is necessary ?
jw
One possible way of finding an FIR filter that approximates H(e ) would be to truncate the
infinite fourier series at n = +
N  1 . Abrupt truncation of the series will lead to oscillation
2
in passband and in stopband. This oscillation is known as Gibbs Oscillation. These
oscillations can be reduced by multiplying the infinite impulse response with a finite
weighing sequence w(n) called a window.
3. Write about the principle of frequency sampling technique?
The desired magnitude response is sampled and linear phase response is specified. The
samples of desired frequency response is identified as DFT coefficients. The filter
coefficients are then determined as the IDFT of this set of samples.
4. Distinguish between FIR and IIR filters.
FIR filter IIR filter

These filters can be easily designed to have These filters do not have linear phase.
perfectly 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
magnitude response. of filters.
Errors due to roundoff noise are less severe in The roundoff noise in IIR filters are
FIR filters, mainly because feedback is not used. more.
5. What is the condition for linear phase of a digital filter?
A FIR filter will have linear phase if, h (n) = h (N-1-n) for symmetric response.
h (n) = - h (N-1-n) for antisymmetric response.
Where ‘N’ is the length of unit sample response of the filter.
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6. What are the techniques of designing FIR filters?

There are three well-known methods for designing FIR filters with linear phase.
These are 1) windows method 2) Frequency sampling method 3) Optimal or minimax design
7. 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 .
8. What are the desirable characteristics of windows?
i) The length of the window should be as large as possible.
ii) The width of the main lobe should be as small as possible.
iii) The amplitude of side lobes should be very small.
9. Write the Different window functions.
i) Rectangular window

WR(n) = 1 for n = -
N  1 <n< +
N  1
2 2
= 0, otherwise
ii) Hanning window

WHn(n) = 0.5 + 0.5 cos [

2n
] for n = -
N  1 < n < + N  1
N 1 2 2
= 0, otherwise
iii) Hamming window

WHn(n) = 0.54 + 0.46 cos [

2n
] for n = -
N  1 < n < + N  1
N 1 2 2
= 0, otherwise
10. Why FIR filter is always stable?
In FIR filter all poles will lie at the origin. Therefore it is always stable.
11. How constant group delay & phase delay is achieved in linear phase FIR filters?
The following conditions have to be satisfied to achieve constant group delay & phase delay.
Phase delay, α = (N-1)/2
Group delay, β = π/2
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UNIT V FINITE WORD LENGTH EFFECTS IN DIGITAL FILTERS

1. What do you mean by rounding?
Rounding a number to ' b' bits is accomplished by choosing the rounded result as the 'b' bit
number closest to the original number unrounded.
2. Distinguish between fixed-point arithmetic and floating point arithmetic.
Fixed-point arithmetic Floating point arithmetic

Fast operation Slow operation

Relatively economical More expensive
Small dynamic range Increased dynamic range
Roundoff error occurs only for addition Roundoff error occurs for both addition &
multiplication
Overflow occurs in addition Overflow does not arise
Used in small computers Used in large, general purpose computers

3. What is quantization step size?

Let us assume a sinusoidal signal varying between +1 and -1 having a dynamic
range 2. If the ADC used to convert the sinusoidal signal employs b+1 bits including sign
bit, the number of levels available for quantizing x(n) is 2b+1. Thus the interval between
successive levels
R 2
q= = = 2-b
2 b +1 2 b +1
4. What is meant by zero limit cycle oscillations?
When a stable IIR filter is excited by a finite input sequence, that is constant, the
output will decay to zero. However the nonlinearities due to the finite precision arithmetic
operations often cause periodic oscillations to occur at the output. Such oscillations are
called zero limit cycle oscillations
5. Define over flow error.
An error occurs in addition of two or more binary numbers occurs when the sum
exceeds the word size available in the digital implementation of the system. Such an error is
called overflow error. When an overflow is detected the sum of the adder is set equal to the
maximum value.
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6. What is quantization error & list the errors due to finite word length registers.
The quantization of numbers is done by truncation or rounding off during storing the results,
an error occurs. That is called quantization error.
The errors are
 Input quantization error
 Coefficient quantization error
 Product quantization error
7. Explain briefly the need for scaling in the digital filter realization.
When a digital filter is implemented using fixed point arithmetic, overflow may occur at the
input to the multiplier and at the output of the adder, which may lead the oscillation with
large amplitude .This can be minimized by scaling. To prevent overflow, the signal level at
certain points in the digital filters must be scaled so that no overflow occur in the adder.
8. Define Dead band of the filter.
The limit cycle occurs as a result of quantization effects in multiplication. The amplitudes of
the output during a limit cycle are confined to a range of values called the deadband of the
1 b
filter. y(n-1) < 2
2
1-|α|
9. What is truncation?
Truncation is the process of discarding all the bits less significant than the least significant
bit that is retained. For example, truncate 0.110111 to b=4bits.
Result= 0.1101
10. Define product round off error?
Product quantization error occurs at the output of the multiplier. Multiplication of a 'b' bit
data with a 'b' bit coefficient results a product of '2b' bits. Since a 'b' bit register is used, the
multiplier output must be rounded or truncated to 'b' bits, which produces error. This is
known as product round off error.