Digital Signal Processing All VSAQs

(Note: For additional VSAQs, please refer the questions at the end of chapters 1 to 6 in TB)

Unit – 1:
1. Test the linearity of y(n) = x2(n). (MP)

2. Relate Fourier transform and Z-transform. (MP) (Prev)

There is a close relationship between the Z transform and the Fourier transform. If we replace the
complex variable z by e^(-jω), then the Z transform is reduced to the Fourier transform. The complex
variable z can be expressed in polar form as Z = r * e^(jω), where r = |z| and ω is ∟z.
3. Determine Z-transform of x(n) = {2, -1, 3, 2, 1, 0, 2, 3, -1}. (MP)

4. Mention the difference between energy and power signals. (Prev)

An energy signal has finite energy and zero average power. Non-periodic signals are examples of energy
signals. On the other hand, a power signal has finite average power and infinite energy. Periodic signals
are examples of power signals.
5. Classify the Discrete time signals. (Prev)
Classification of systems are static or dynamic, linear or nonlinear, time variant or invariant, causal or
noncausal, stable or unstable.
6. Define Region of Convergence. (Prev)
The ROC for a given signal, is defined as the range of for which the z-transform converges.

Unit – 2:
1. Define twiddle factor in Fast Fourier Transform. (Ses-1)

2. What is the difference between DIT-FFT and DIF-FFT?

3. Distinguish between DFT and FFT. (Ses-1)

The Discrete Fourier Transform (DFT) is a mathematical algorithm used to transform discrete-time
signals from the time domain to the frequency domain, while the Fast Fourier Transform (FFT) is an
efficient algorithm for computing the DFT.
4. How many multiplications and additions are involved in the DFT and in radix-2 DIT-FFT?
(Ses-1) (MP) (Prev)
The DFT involves N2 complex multiplications and N(N-1) complex additions, where N is the number
of samples in the input sequence.
The radix-2 decimation-in-time Fast Fourier Transform (DIT-FFT) algorithm involves (N/2)log2(N)
complex multiplications and Nlog2(N) complex additions.
5. What is zero-padding? Why is it needed? (Ses-1)
Zero-padding is the process of adding zeros to the end of a signal to increase its length. It is often used
in signal processing and spectral analysis to improve the resolution of frequency-domain representations
of signals.
6. Develop basic butterfly diagram for DIF-FFT. (MP) (Prev)

7. List the applications of FFT. (MP) (Prev)

1. Audio and Video Compression: FFT is used in audio and video compression algorithms to transform
signals from the time domain to the frequency domain for more efficient encoding.
2. Digital Signal Processing: FFT is used in various applications of digital signal processing such as
filtering, convolution, and correlation.
3. Spectral Analysis: FFT is used in spectral analysis to identify the frequency components of a signal
and analyze its frequency-domain characteristics, such as frequency content, bandwidth, and harmonic
distortion.
8. Compare linear and circular convolution of two sequences. (Prev)

Property Linear Convolution Circular Convolution

Length of Length of input sequence 1 + Length Length of input sequence 1
Output of input sequence 2 - 1

Time O(N^2) where N is the length of input O(N^2) where N is the length of input
Complexity sequences sequences

Output Unbounded, can be complex-valued Bounded, always real-valued

Range

Application Used for general signal processing Used in digital filtering, cyclic convolution,
applications where input signals are and frequency-domain analysis of periodic
not periodic or circularly symmetric signals or signals with circular symmetry.

Unit – 3:
1. What is pre-warping? (MP) (Prev)
2. What are the structures for IIR filter realization? (MP)
Direct Form-I, Direct Form-II, Cascade, and Parallel.
3. What is frequency warping effect? (Prev)
Warping is the effect that the frequencies and bandwidth of higher frequency passband will tend to
reduce disproportionately.
4. Mention the difference between Chebyshev Type-1 and Type-2 filters. (Prev)
Chebyshev Type-1 filters have ripples in the passband and monotonic roll-off in the stopband, while
Chebyshev Type-2 filters have ripples in the stopband and monotonic roll-off in the passband.
5. List the methods for digitizing the analog filter into digital IIR filter. (Prev)
Butterworth and Chebyshev analog filters can be converted to digital IIR filters using Bilinear
transformation, and impulse invariance method.

Unit – 4:
1. What are desirable characteristics of frequency and impulse response of window? (Ses-2) (Prev)
2. Differentiate between FIR and IIR filters. (Ses-2)

3. What are the necessary and sufficient conditions for linear phase characteristics? (Ses-2)
4. What is windowing technique? (Ses-2)
To reduce the oscillations caused by Gibbs phenomenon, the Fourier coefficients of the filter are
modified by multiplying the infinoite impulse response with a finite sequence w(n) is called a
windowing technique.
5. What is Gibbs phenomenon? (MP) (Prev)
Gibbs phenomenon is a phenomenon that occurs when a non-periodic signal is approximated by a
truncated series of its Fourier transform coefficients. It causes oscillations or overshoots near
discontinuities or sharp transitions in the signal, even as the number of terms in the series is increased.
6. List the methods to design FIR filters. (MP)
Windows like Rectangular, Hamming, Hanning, Kaiser etc.
7. List the advantages of FIR and IIR filters. (Prev)
Advantages of FIR filters: FIR filters have linear phase response, FIR filters can achieve sharp cutoffs
and high stopband attenuation
Advantages of IIR filters: IIR filters can achieve the same level of frequency selectivity as FIR filters
with fewer coefficients, IIR filters have a causal response
8. Define Group delay and Phase delay. (Prev)
In FIR filters, the group delay is the rate of change of the filter's phase response with respect to
frequency, while the phase delay is the time delay experienced by each frequency component of a signal
passing through the filter, measured in units of time (usually seconds).
The group delay determines the amount of distortion introduced by the filter to different frequency
components of the input signal, while the phase delay determines the time alignment of the output signal
with respect to the input signal.
9. Classify the filters based on frequency response. (Prev)
Low Pass, High Pass, Band Pass, Band Stop.

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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
Innon-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-Ilstructure 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 Irealization 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
by, 2T tan For smaller values of o, there exists linear relationship between o

and 2.But for Larger values of o, the relationship is nonlinear. This introduces distortion in
the frequency axis. This effect compresses the magnitude and phase response. This Efect 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
differentialequation by an equivalent difference equation. (0- y(nl) -y(nT -) at t-nT
dt T

7. How can you designdigital filter from analogfilter?

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

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 tan
Prewarping frequency is given by, 2= T

11. Distinguish between Butterworth and Chebyshev fiter.

1. The magnitude response of Butterworth filter decreases monotonically as the frequency
increases from 0 to o, 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. i) FIR filters are inherently
stable. ii) The design of FIR filters is fairly simple compared to IIR filters.
Disadvantages: i) FIR filters need higher order compared to IR filter. i) 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 digitalfilter 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?

One possible way of finding an FIR filter that approximates H(ejw )would be to truncate the
infinite fourier series at n =+ (N-), 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
?
One possible way of finding an FIR filter that approximates H(e jw would be to truncate the
infinite fourier series at n =+ (N-).
2
Abrupt truncation of the series will lead to oscillation
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 theirLess 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 becausefeedback 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 antisymmetricresponse.
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) windowsmethod 2)Frequency sampling method 3) Optimal or minimax design
7. What isthe 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 Nis 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.

i) 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) = 1for n =
2 2

=0, otherwise
ii) Hanning window

WHa(n) =0.5 +0.5 cos [ N-1 for n N-<ns+N-)

2 2

=0, otherwise
i) Hamming window

WHa(n) = 0.54 + 0.46 cos for n= N=sns

2
+W-)
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 toachieve constant group delay &phase delay.
Phase delay, a = (N-1)/2
Group delay, B=2

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