D. S. P.

2022

PYQ
SOLUTIONS
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
(a) Two properties of discrete-time systems are linearity and time invariance.

(b) The relationship between the Discrete Fourier Transform (DFT) and Z-
transform is given by substituting in the Z-transform equation.

(c) A system is causal and stable if its Region of Convergence (ROC) satisfies
the following conditions:
• Causality: The ROC must lie outside the outermost pole.
• Stability: The ROC must include the unit circle |z| = 1.

(d) One-sided Z-transform is applicable to signals that are causal (nonzero

only for nonnegative time indices), while two-sided Z-transform considers
signals that extend in both positive and negative time directions.

Chebyshev approximation allows for a steeper roll-off in the stopband but

introduces ripples in the passband, providing a trade-off between
passband ripple and stopband attenuation.

Butterworth approximation, on the other hand, has a smoother passband

response with no ripples but a slower roll-off in the stopband compared to
Chebyshev. Butterworth filters sacrifice steepness for a more uniform
frequency response.
(G) The transposed form structure in digital signal processing refers to a
different arrangement of computations in the implementation of a filter. It
involves reorganizing the operations to achieve computational efficiency,
often resulting in a more compact or resource-friendly design.

(H) A twiddle factor is a complex constant used in the context of Fourier

transforms, such as the Discrete Fourier Transform (DFT) or Fast Fourier
Transform (FFT). It simplifies the calculation of sinusoidal components and
is commonly represented as e^(-j*2*pi*k/N), where k is the index and N is
the size of the transform.

(I) The salient features of windowing techniques in signal processing

include mitigating spectral leakage, reducing the impact of finite-duration
signals, and improving the accuracy of frequency domain analysis. Windows
shape the signal to minimize artifacts in the frequency domain.

(J) Cross-correlation is a measure of similarity between two signals as a

function of the time lag applied to one of them. It is significant for tasks
like detecting similarities, aligning signals in time, and estimating time
delays. In communication systems, cross-correlation is used for
synchronization and channel estimation.
3. A.
Explain in detail how a band-limited signal can be reconstructed from
its samples in time and frequency domains without any loss of signal
information.

To reconstruct a band-limited signal without loss of information, we can

use the Nyquist-Shannon sampling theorem. Here’s a detailed explanation
in both time and frequency domains:

Time Domain Reconstruction:

1.Sampling Theorem:
• According to the Nyquist-Shannon sampling theorem, a band-limited
signal can be perfectly reconstructed if it is sampled at a rate greater
than twice its bandwidth.
• If the bandwidth of the signal is (B), then the minimum sampling
rate (fs) is given by (fs > 2B).

2.Sampling Process:
• The continuous band-limited signal (x(t)) is sampled at regular
intervals with a sampling rate (fs).
• The samples are obtained as (Xn = x(nTs)), where (n) is an integer
and (Ts) is the sampling period ((Ts = \frac{1}{f_s})).

3. Reconstruction:
• The original signal (x(t)) can be reconstructed from its samples using
an interpolation formula. A common method is the sinc interpolation.
• The reconstructed signal is given by (x_r(t) = \sum_{n=-\infty}
^{\infty} x_n \cdot \text{sinc}\left(\frac{t - nT_s}{T_s}\right)), where
sinc is the sinc function.
Frequency Domain Reconstruction:

1.Frequency Replication:
• Due to sampling, the frequency spectrum of the original signal is
replicated around multiples of the sampling frequency ((f_s)).
• The replicas are centered at (kf_s) and (-kf_s), where (k) is an
integer.

2.Low-pass Filtering:
• To eliminate the replicated spectra and obtain the original signal, a
low-pass filter is applied.
• The cut-off frequency of the filter should be set at half the sampling
rate ((\frac{f_s}{2})) to retain the original signal.

3. Ideal Reconstruction:
• If the low-pass filter perfectly removes the replicas without affecting
the original spectrum, the reconstructed signal in the frequency domain is
identical to the original.

In summary, by satisfying the Nyquist-Shannon sampling theorem,

carefully sampling the signal, and employing proper reconstruction
techniques, a band-limited signal can be accurately reconstructed in both
time and frequency domains without any loss of information.
3. B.
Determine the terms sampling theorem, Nyquist rate, Nyquist interval and
aliasing.

1.Sampling Theorem:
• The Sampling Theorem, also known as the Nyquist-Shannon Sampling
Theorem, states that in order to accurately reconstruct a continuous
signal from its samples, the sampling rate must be at least twice the
highest frequency present in the signal. Mathematically, if the maximum
frequency in a signal is (B), then the sampling rate (fs) should be greater
than (2B).
2.Nyquist Rate:
• The Nyquist rate is the minimum sampling rate required to avoid
aliasing and accurately represent a signal. It is precisely twice the
maximum frequency component in the signal. Nyquist rate (fs) is given by
(fs = 2B), where (B) is the bandwidth of the signal.
3. Nyquist Interval:
• The Nyquist interval is the time period between successive samples
and is the reciprocal of the Nyquist rate. It is denoted by (Ts) and is
given by (Ts = 1/fs), where (fs) is the Nyquist rate.
4.Aliasing:
• Aliasing occurs when a signal is improperly sampled, leading to false
representation of frequencies in the sampled signal. If the sampling rate
is less than twice the highest frequency component in the signal, high-
frequency components fold back into the lower frequencies, causing
distortion and misinterpretation. Aliasing can be avoided by ensuring the
sampling rate exceeds the Nyquist rate.

In summary, the Sampling Theorem sets the foundation for proper signal
sampling, defining the Nyquist rate as the minimum sampling rate, the
Nyquist interval as the time between samples, and highlighting the
importance of avoiding aliasing to ensure accurate signal representation.
5. A. State and prove time reversal property of DFT
 https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https:/
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
/chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https:/ /chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https:/
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
/chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https:/
https:/ /chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
/chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https:/ /chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https:/
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
/chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https:/ /chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https:/
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
/chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
https://chat.whatsapp.com/DMq2l0JQagKDfrdTNam0st
7.(a). Drive the expressions for order and cut-off frequency of a Butterworth filter.

The magnitude squared frequency

response of a butterworth filter is
given by

A common set of conditions for

the frequency response of a LPF
is given by
The value of N is rounded off to next nearest integer.
 8. A.
What are the effects of finite register length in the implementation of
digital filters?

The effects of finite register length in the implementation of digital filters,

often referred to as quantization effects, can lead to several challenges
and compromises:

1.Quantization Noise:
• Limited register length results in quantization noise, which is an
inevitable consequence of representing continuous values with a finite
number of bits.
• Quantization noise can introduce spurious signals in the output,
impacting the filter’s signal-to-noise ratio.
2.Overflow and Clipping:
• If the dynamic range of the input signal exceeds the representable
range of the finite register, overflow or clipping may occur.
• Overflow can cause loss of information and distortion in the output
signal.
3. Frequency Response Distortion:
• Finite register length affects the accuracy of coefficient
representation, leading to deviations from the ideal frequency response.
• The frequency response may exhibit distortions, especially in the
stopband and transition regions.
4.Coefficient Quantization:
• The coefficients of digital filters are often quantized, impacting the
precision of the filter design.
• Coefficient quantization can result in deviations from the desired filter
characteristics.
5.Limitations on Filter Order:
• The finite register length imposes constraints on the maximum
achievable filter order. Higher-order filters may require longer word
lengths, increasing computational complexity.
 6.Non-Linearities:
• Non-linear effects may arise due to quantization, causing nonlinear
distortions in the output signal.
7. Limitations on Filter Design Flexibility:
• The precision limitations imposed by finite register length can restrict
the flexibility in designing filters with specific characteristics.
• Some filter structures or design methods may be more susceptible to
quantization effects.

To mitigate these effects, designers often employ techniques such as

increased word lengths, dynamic range scaling, dithering, and carefully
selecting filter structures that are less sensitive to quantization.
Additionally, advanced digital signal processing hardware and software
techniques are used to minimize the impact of finite register length in
practical implementations.
8. B.
Explain how the DFT and FFT are helpful in power spectral estimation.
The Discrete Fourier Transform (DFT) and its fast implementation, the Fast
Fourier Transform (FFT), play a crucial role in power spectral estimation,
providing efficient methods for analyzing the frequency content of a signal.
Here’s how they are helpful:

1.Power Spectral Density (PSD) Estimation:

• DFT and FFT enable the computation of the Power Spectral Density
(PSD) of a signal, which describes the distribution of power across different
frequency components.
• PSD estimation helps in identifying dominant frequency components and
understanding the frequency characteristics of a signal.
2.Frequency Resolution:
• DFT allows the decomposition of a signal into its frequency components.
The resulting frequency resolution is determined by the length of the signal
and the sampling rate.
• FFT accelerates this process, making it feasible to compute the DFT
efficiently, especially for long signals.
 3. Windowing Techniques:
• Windowing is often applied before computing the DFT or FFT to reduce
spectral leakage and improve the accuracy of frequency analysis.
• Different window functions can be employed to trade off main lobe
width and sidelobe levels, optimizing spectral estimation for specific
applications.
4.Peak Detection:
• By examining the magnitudes or squared magnitudes of the DFT or FFT
coefficients, one can identify peaks corresponding to dominant frequencies
in the signal.
• This is particularly useful in applications such as signal processing,
communication, and vibration analysis.
5.Spectral Averaging:
• Power spectral estimation often involves averaging multiple segments
of a signal’s spectrum to improve the reliability of the results.
• The efficient computation of FFTs facilitates spectral averaging,
providing a more accurate representation of the underlying power
distribution.
6.Real-time Analysis:
• FFT algorithms, being computationally efficient, are essential for real-
time power spectral estimation in applications like audio processing,
communication systems, and control systems.
7. Spectral Leakage Understanding:
• DFT and FFT help in understanding and mitigating the effects of
spectral leakage, which occurs when a signal is not integer periodic within
the observation window.

In summary, DFT and FFT are powerful tools for power spectral
estimation, offering efficient algorithms for analyzing and characterizing
the frequency content of signals. Their applications range from identifying
dominant frequencies to performing real-time spectral analysis in various
fields of signal processing and communication.
8. C.
What is the need for multirate signal processing? Explain the
process of interpolation and decimation with suitable examples.

Need for Multirate Signal Processing:

Multirate signal processing is employed to efficiently manipulate signals

at different rates, addressing various requirements such as bandwidth
reduction, data compression, and resource optimization. Some key
reasons for using multirate signal processing include:

1.Bandwidth Compression:
• Multirate techniques are utilized to reduce the bandwidth of a
signal, especially when certain frequency components are not of
interest.
• Bandwidth compression is crucial in applications like communication
systems where efficient use of available bandwidth is essential.
2.Data Rate Conversion:
• Multirate processing allows for the conversion of signals between
different sampling rates, accommodating compatibility between systems
with diverse data rates.
3. Filter Bank Structures:
• Multirate systems are integral in the design of filter banks, which
are used in applications like audio and image compression, providing
efficient frequency domain analysis.
4.Digital Signal Processing Efficiency:
• Multirate signal processing can lead to computational efficiency by
processing signals at lower rates where detailed information is not
necessary and only key features need to be retained.
Interpolation and Decimation:

1.Interpolation:
• Interpolation involves increasing the sampling rate of a signal by
introducing additional samples between existing ones. The process is
typically achieved using interpolation filters.
• Let (x[n]) be the original signal sampled at rate (f_s), and (x_i[n]) be
the interpolated signal sampled at a higher rate (f_{si} > f_s).
• The interpolated signal (x_i[n]) can be expressed as (x_i[n] = \sum_{k=-
\infty}^{\infty} x[k] \cdot h[n - kM]), where (M) is the interpolation factor,
and (h[n]) is the interpolation filter.
Example:
• Original signal (x[n]): [1, 2, 3, 4]
• Interpolation factor (M = 2)
• Interpolation filter (h[n]): [0.5, 1, 0.5]
• Interpolated signal (x_i[n]): [1, 1.5, 2, 2.5, 3, 3.5, 4]
2.Decimation:
• Decimation involves reducing the sampling rate of a signal by selecting
only a subset of samples. It is often accompanied by anti-aliasing filtering
to prevent aliasing.
• Let (y[n]) be the original signal sampled at rate (f_s), and (y_d[n]) be
the decimated signal sampled at a lower rate (f_{sd} < f_s).
• The decimated signal (y_d[n]) can be expressed as (y_d[n] = y[nD]),
where (D) is the decimation factor.
Example:
• Original signal (y[n]): [1, 2, 3, 4, 5, 6, 7, 8]
• Decimation factor (D = 2)
• Decimated signal (y_d[n]): [1, 3, 5, 7]

In summary, interpolation and decimation are fundamental processes in

multirate signal processing, enabling manipulation of signal rates for various
applications, including data rate conversion and efficient use of resources.