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1 Min Short Questions DSP

The document discusses various concepts in signal processing, including energy and power signals, causal and static systems, and the transfer function of LTI systems. It covers properties of Discrete Fourier Series, the time reversal property of DFT, and the definition of twiddle factors, as well as comparisons between IIR and FIR filters. Additionally, it addresses multirate signal processing, quantization errors, and scaling in signal processing.

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sai vasu
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8 views10 pages

1 Min Short Questions DSP

The document discusses various concepts in signal processing, including energy and power signals, causal and static systems, and the transfer function of LTI systems. It covers properties of Discrete Fourier Series, the time reversal property of DFT, and the definition of twiddle factors, as well as comparisons between IIR and FIR filters. Additionally, it addresses multirate signal processing, quantization errors, and scaling in signal processing.

Uploaded by

sai vasu
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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1.What isan energy andpower signal?

Anenergy signalhas finite energy and zero


average power, typically localized in time. A power
signal has finite, non-zero average power and
infinite energy, usually periodic or persistent over
time.

2. Define causal and static systems.


Asystem is causal if its output depends only on
present and past inputs, not future ones. Astatic
(memoryless) system's output depends solely on
the current input value.
3. Describe the transfer function of an LTI system.
The transfer function of an LTI system is the Z
transform of its impulse response. It characterizes
the system's input-output relationship in the
frequency domain.

4. Derive the relation between DTFT and Z


Transform.
The DTFT is aspecial case of the Z-transform
evaluated on the unit circle, i.e., z = e .
Therefore, X(e)= X(2)|=ee, provided the
unit circle is within the ROC.
1. State any two properties of Discrete Fourier
Series.

Linearity: The DFS of a sum of signals is the


sum of their individual DFS.

Time Shifting: Shifting a signal in time results


in a phase shift in its DFS coefficients.

2. State and prove time reversal property of DFT.


DFT
Property: If z(n] + x[k], then z(-n
DFT
mod N X-k mod N].
Proof: This follows from substituting z-n into
the DFT formula and changing summation index,
yielding a DFT output mirrored in frequency.
3. Define Twiddle Factor.
The twiddle factor is a complex exponential
WN = e Ñ used in DFT and FFT computations.
It represents the roots of unity and facilitates
efficient computation in FFT algorithms.

4. How many multiplications andadditions are


required for N-point DFT &FFT?
An N-point DFT requires N complex
multiplications and N(N - 1)additions.
An N-point FFT requires log N complex
multiplications and N log, N additions.
1. Compare lIR and FIR Filters.
IIR (Infinite Impulse Response) filters have
feedback and potentially infinite-duration impulse
responses, offering higher efficiency but less
stability.
FIR (Finite Impulse Response) filters are inherently
stable and linear phase but often require more
coefficients for similar performance.

2. Define Chebyshev Type Iand Type Il


approximation techniques.
Chebyshev Type Ifilters have ripple in the
passband and a monotonic stopband.
Chebyshev Type Il filters have a flat passband and
ripple in the stopband.
3. What is Bilinear Transformation and Impulse
Invariant Method?

Bilinear transformation maps an analog filter to a


digital one by warping the frequency axis, avoiding
aliasing.
The impulse invariant method samples the analog
filter's impulse response, preserving time-domain
characteristics but may suffer from aliasing.

4. Differentiate Butterworth and Chebyshev


approximation techniques.
Butterworth filters have a maximally flat frequency
response in the passband with no ripples.
Chebyshev filters achieve faster roll-off by allowing
ripples in either the passband (Type I) or stopband
(Type lI).
1. Why FIR Filters have Linear Phase
Characteristics?

FIR filters can be designed with symmetric or anti


symmetric impulse responses, which ensures
linear phase.
This means allfrequency components experience
equal delay, preserving the waveform shape of
signals.

2. Define Group Delay and Phase Delay.


Group delay is the negative derivative of phase
with respect to frequency and indicates the delay
of signalenvelopes.
Phase delay is the phase shift divided by
frequency and represents the delay of individual
sinusoidal components
3. What is Wrapping Effect?
The wrapping effect, or aliasing in the frequency
domain, occurs when frequency components
exceed the Nyquist limit and fold back into the
lower frequency range.
It commonly arises during sampling or DFT
computations with limited frequency resolution.

4. What are the Characteristics of a Window?


A
window function has characteristics like
mainlobe width and sidelobe levels that affect
spectral leakage and resolution.
Good windows reduce leakage by tapering signal
edges, trading off frequency resolution for better
dynamic range.
1.What is the need for multirate signal processing
and give its classification?
Multirate signal processing is needed to efficiently
process signals at different sampling rates,
reducing computation and memory requirements.
It is classified into decimation (downsampling),
interpolation (upsampling), and sampling rate
Conversion.

2. Explain Up sampling &Down sampling.


Upsampling increases the sampling rate of a
signal by inserting zeros between samples and
then filtering. Downsampling reduces the sampling
rate by keeping every M" sample and discarding
the rest, usually after low-pass filtering.
3. List out the errors arise due to quantization of
numbers.
Quantization introduces round-off errors and
truncation errors. These errors cause signal
distortion and limit the accuracy of digital signal
representation.

4. Define Scaling.
Scaling in signal processing refers to adjusting the
amplitude of a signal or number to fit within a
specific range. This is crucial to prevent overflow
or underflow during arithmetic operations.

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