1. Distinguish between DTFT and DFT.

DTFT (Discrete-Time Fourier Transform) and DFT (Discrete Fourier Transform) are two related
mathematical tools that are used to analyze the frequency content of discrete-time signals.
However, they differ in some fundamental ways:
Definition: The DTFT is a mathematical transform that converts a discrete-time signal from the
time domain to the frequency domain, whereas the DFT is a numerical algorithm that computes the
DTFT of a finite-length sequence of samples.
Domain: The DTFT is defined over the entire frequency axis (-∞ to +∞), while the DFT is only
defined at discrete frequencies (0 to N-1), where N is the length of the input sequence.
Continuity: The DTFT is continuous, which means that it can take on any value over its entire
frequency range, while the DFT is discrete and can only take on a finite number of values at
specific frequencies.
Computation: The DTFT is typically computed using an analytical formula, while the DFT is
computed using an efficient algorithm, such as the Fast Fourier Transform (FFT).s
Time Complexity: The time complexity of computing the DTFT is O(N^2), where N is the length
of the input sequence, while the time complexity of computing the DFT using the FFT is O(N log
N), which is much faster for large N.
In summary, the DTFT is a continuous-time Fourier transform that is defined over the entire
frequency axis, while the DFT is a discrete Fourier transform that computes the DTFT at a finite
number of discrete frequencies. The DFT is computed numerically using an efficient algorithm
such as the FFT, which makes it much faster than the DTFT for large input sequences.
2. Parseval’s relation for power Signal
3. Parseval’s relation for Energy Signal.
4. What is Cepstrum
Cepstrum is a mathematical technique used in signal processing to analyze the spectral content of
a signal. It is obtained by taking the inverse Fourier transform of the logarithm of the Fourier
transform of a signal. The name "cepstrum" comes from the reversed spelling of the word
"spectrum."
Cx(z)=ln(X(z))
The cepstrum of a signal can be used to extract information about the time delay or echo
characteristics of the signal, as well as the pitch or fundamental frequency of a periodic signal. It
can also be used for signal enhancement, such as removing the effects of room reverberation from
a recorded speech signal.
One common application of the cepstrum is in speech analysis, where it can be used to identify
the formants or resonant frequencies of the vocal tract, which are important for speech recognition
and synthesis.
The cepstrum has many variants, including the real cepstrum, complex cepstrum, and power
cepstrum, which are used for different types of signal analysis.

5. What is Gibb’s phenomenon? How do you reduce the effects

Gibb’s phenomena?
Gibb's phenomenon is a phenomenon that occurs in signal processing when a signal that contains
a discontinuity or sharp edge is reconstructed from its Fourier series or Fourier transform. The
phenomenon manifests as a ringing or overshoot near the discontinuity or edge in the
reconstructed signal, and can result in a loss of fidelity or accuracy in the signal.
The effect of Gibb's phenomenon can be reduced by using a smoothing or regularization technique
to smooth out the sharp edges or discontinuities in the signal before applying the Fourier
transform or Fourier series.

6. Briefly compare the Pros and cons of FIR and IIR system.
FIR (Finite Impulse Response) and IIR (Infinite Impulse Response) are two types of digital signal
processing filters. Here are some pros and cons of FIR and IIR systems:
Pros of FIR filters:
 FIR filters are always stable, which means they do not have the potential to oscillate or become
unstable.
 They have linear phase response, which is desirable in many applications such as audio
processing.
 FIR filters can have a very high stopband attenuation without affecting the passband ripple.
 They can be designed to have a zero phase response, which means that the filter output signal
will be time-aligned with the input signal.
Cons of FIR filters:
 They require more filter coefficients compared to IIR filters to achieve similar performance,
which can lead to increased computation complexity and memory usage.
 The group delay of FIR filters is longer than that of IIR filters, which can lead to a delayed
output.
Pros of IIR filters:
 IIR filters can achieve a higher filter order with fewer coefficients compared to FIR filters,
which means that they can provide comparable performance with less computational
complexity and memory usage.
 They have a shorter group delay compared to FIR filters, which means that the output
signal is not delayed as much.
 They can be designed to have a more selective frequency response with sharper transitions
between passband and stopband.
Cons of IIR filters:
 They can become unstable for certain parameter values, which can lead to oscillations or
unstable behavior.
 They have a nonlinear phase response, which can cause distortion and other undesirable effects.
 They can suffer from quantization noise due to feedback, which can limit their performance
in certain applications.
7. Write the advantages and disadvantages of digital filter.
Advantages of digital filters:
Flexibility: Digital filters can be easily adjusted and reprogrammed, allowing for greater flexibility
in signal processing applications.
Accuracy: Digital filters offer high accuracy in filtering out unwanted signals and noise, resulting
in clearer and more reliable signal output.
Cost: Digital filters are often more cost-effective than analog filters since they don't require the
use of expensive components like capacitors and inductors.
Implementation: Digital filters can be implemented using software, making them easier to
incorporate into existing systems without the need for additional hardware.
Disadvantages of digital filters:
Sampling: Digital filters require signal sampling, which can introduce errors due to the finite
sampling rate and aliasing.
Processing time: Digital filters can require significant processing time, especially for complex
filter designs, which can lead to delays in real-time signal processing applications.
Complexity: Designing digital filters can be complex, requiring knowledge of digital signal
processing theory and programming skills.
Nonlinear distortion: Digital filters can introduce nonlinear distortion due to the use of nonlinear
functions in the filter design, which can impact the accuracy of the filtered signal.

8. Why ideal low pass filter is non causal?

An ideal low-pass filter is a theoretical filter that completely eliminates all frequencies above a
certain cutoff frequency while allowing all frequencies below that cutoff to pass through. The
transfer function of an ideal low-pass filter is given by:
H(jω) = 1, for
|ω| ≤ ωc H(jω)
= 0, for |ω| > ωc
where ωc is the cutoff frequency.
An ideal low-pass filter is non-causal because its output depends on future inputs. In other words,
the output of the filter at any given time depends not only on the current input but also on future
inputs. This violates the causality principle, which states that the output of a system should only
depend on past and present inputs, not on future inputs.
9. Explain the filter type selection procedure of digital filter in brief.
The selection of a filter type for a digital filter depends on the requirements of the signal processing
application. Here are the basic steps for selecting a digital filter type:
Determine the filter specifications: The first step is to determine the specifications of the desired
filter, such as the cutoff frequency, passband and stopband ripple, transition bandwidth, and order.
Choose a suitable filter topology: Based on the filter specifications, choose a suitable filter
topology that meets the requirements of the application. There are several types of digital filter
topologies, including FIR (finite impulse response) and IIR (infinite impulse response) filters.
Determine the filter order: The filter order is determined by the number of filter coefficients
required to meet the filter specifications. The higher the order, the more complex the filter will be.
Choose a filter design method: The filter design method determines how the filter coefficients are
computed. There are several methods, including windowing, frequency sampling, and
optimization-based methods.
Implement the filter: Once the filter coefficients have been computed, they can be implemented
in hardware or software. The implementation method depends on the application requirements and
constraints.
Test and refine the filter: Finally, the filter should be tested to ensure that it meets the desired
specifications. If necessary, the filter can be refined by adjusting the filter parameters or selecting
a different filter topology or design method.
10. Write short notes on the application of DSP in ECG.
Digital signal processing (DSP) is widely used in electrocardiography (ECG) for various
applications. Here are some examples of how DSP is applied in ECG:
ECG Signal Processing: DSP techniques are used to process and analyze ECG signals. ECG signals
are often corrupted by noise and artifacts, which can affect the accuracy of diagnosis. DSP
techniques such as filtering, segmentation, feature extraction, and classification are used to remove
noise and extract meaningful information from ECG signals.
ECG Monitoring: DSP is used in ECG monitoring devices to acquire, process, and analyze ECG
signals in real-time. Portable and wearable ECG monitoring devices use DSP algorithms to filter
out noise and artifacts and provide accurate ECG recordings, which can be used for diagnosing
and monitoring cardiac conditions.
Arrhythmia Detection: DSP algorithms are used to detect and classify different types of cardiac
arrhythmias. These algorithms analyze the ECG signal for irregularities in the heart rate and
rhythm and classify the arrhythmia type based on predefined criteria.
Heart Rate Variability (HRV) Analysis: DSP techniques are used to analyze HRV, which is a
measure of the variation in time intervals between consecutive heartbeats. HRV analysis is used
to evaluate autonomic nervous system function, which can provide insights into the risk of
cardiovascular disease and other conditions.
Telemedicine: DSP is used in telemedicine applications to transmit and process ECG signals
remotely. Real- time ECG signals can be transmitted over the internet and processed using DSP
algorithms to provide accurate diagnoses, enabling remote patient monitoring and telemedicine
consultations.

11. Differentiate between Fourier series and Fourier transform.

12. Write down the Dirichlet Condition.
13. Write short notes on FIR filter.