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DSP Lab Model Questions

The document contains 6 questions related to digital signal processing lab experiments. The questions include: 1) Writing a MATLAB program to simulate a 2nd order IIR bandpass filter and plot the magnitude and phase response. 2) Writing a MATLAB program to simulate a 2nd order IIR bandstop filter and plot the response. 3) Writing a MATLAB program to simulate a 41-tap FIR bandpass filter and plot the impulse response and magnitude/phase response. 4) Writing a program to generate a 1kHz sine wave using a DSP board. 5) Implementing a FIR lowpass filter using a DSP board. 6) Implementing a 4th order Butterworth IIR lowpass filter using a DSP board.

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annzera
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2K views

DSP Lab Model Questions

The document contains 6 questions related to digital signal processing lab experiments. The questions include: 1) Writing a MATLAB program to simulate a 2nd order IIR bandpass filter and plot the magnitude and phase response. 2) Writing a MATLAB program to simulate a 2nd order IIR bandstop filter and plot the response. 3) Writing a MATLAB program to simulate a 41-tap FIR bandpass filter and plot the impulse response and magnitude/phase response. 4) Writing a program to generate a 1kHz sine wave using a DSP board. 5) Implementing a FIR lowpass filter using a DSP board. 6) Implementing a 4th order Butterworth IIR lowpass filter using a DSP board.

Uploaded by

annzera
Copyright
© © All Rights Reserved
Available Formats
Download as DOCX, PDF, TXT or read online on Scribd
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DSP Lab model questions

Write a Matlab program for simulation of I.I.R Digital Band Pass filter for meeting
following specification in analog domain .
Order of the filter Second order
Lower cut-off frequency=4khz
Upper cut-off frequency=5khz
Sampling frequency=16Khz.
Find H(Z) and plot the magnitude and phase response of the filter.

Write a Matlab program for simulation of I.I.R Digital Band Stop filter for meeting
following specification in analog domain .
Order of the filter Second order
Lower cut-off frequency=4khz
Upper cut-off frequency=5khz
Sampling frequency=16Khz.
Find H(Z) and plot the magnitude and phase response of the
3 Write a Matlab program for simulation of a Digital FIR band pass filter for
Order of the filter N=41
Sampling frequency = 48Khz
Lower cut-off frequency=1Khz
Upper cut-off frequency= 2Khz
Use rectangular window
Plot impulse response h(n) and Magnitude and phase response of the filter.

Using DSP board write a program for generating 1KHz Sine wave.

Using DSP board write a program for implementing a FIR low pass filter for
Cutt-off frequency 2Khz. Sampling frequency 16Khz.

Using DSP board write a program for implementing a IIR fourth order Butterworth
low pass filter for Cutt-off frequency 2Khz. Sampling frequency 16Khz

20 TOP MOST Digital Signal Processing LAB VIVA Questions


and Answers ECE
Digital Signal Processing LAB VIVA Questions and Answers :1.Classify signals?
Continuous time, continuous amplitude (Analog Signals)
Discrete time, continuous amplitude
Continuous time, discrete amplitude
Discrete time, discrete amplitude
2.What is the use of Random Signals?
Random signals are used to test dynamic response statistically for very small amplitudes
and time duration.
3. Classify Systems.
Linear, stable and time invariant.
4.What do you mean by aliasing in digital signal processing? How it can be avoided?
Aliasing refers to an effect due to which different signals become indistinguishable. It also
refers to distortion in the reconstructed signal when it is reconstructed from the original
continuous signal.
To avoid aliasing we can simply filter out the high freuency components of the signal by
using antialiasing filter like optical antialiasing filter.
5. What are the differences between a microprocessor and a DSP processor?
DSP processors are featured to support high performance and repeatitive and intensive
tasks whereas microprocessors are not application specific and they are designed to
process controloriented tasks.
6. What is the convolution?
Convolution is the technique of adding two signals in time domain. We can also do this
uite easily by changing the domain of signals from time domain to freuency domain
using Fast Fourier Transform (FFT).
7. What is FFT?
FFT is a fast way to calculate Discrete Fourier Transform (DFT). It is much more efficient
then DFT and reuire less number of coding lines. Due to FFT several kind of techniues
are feasible.

8. What is the advantage of a Direct form II FIR over from I?


Direct Form II FIR filters requires half the number of delay units as much as used by
Form I.
9. What is interpolation and decimation?
Interpolation is the process of increasing the sample rate in dsp whereas decimation is
the opposite of this that is, it is the process of decreasing the sample rate in dsp.
10. Difference between DFT and DTFT.
DFT
1.Limited number of samples of periodic signal

DTFT
1.unlimited number of samples.

2.input is always periodic

2.input may not always be periodic

3 physically realizable

3 mathematically precise

4 frequency becomes discrete

4 frequency is continuous

11.What is a Digital Signal Processor (DSP)?


Microprocessor specifically designed to perform fast DSP operations (e.g., Fast Fourier
Transforms, inner products, Multiply & Accumulate)
Good at arithmetic operations (multiplication/division)
Mostly programmed with Assembly and C through Integrated Development Environment
(IDE)
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