Qbank DSP

1 How do you provide time scaling for a signal?
Or Show that time scaling of a signal is

provided by interpolation and decimation
Illustrate the decimation operation by decimating the above signal by a factor 3.
3 Define and explain the periodicity of a discrete time signal, which is a sinusoidal. Example
,cos(ωn)
4 If 𝑥𝑎 (𝑡) represents an analog signal and its sampled version is 𝑥(𝑛), where sampling period
T=1/fs, where t=NT=N/ fs, establish the relation between 𝛺 𝑎𝑛𝑑 𝜔.
5 Consider an analog signal, 𝑥𝑎 (𝑡) = 3 cos 100 𝜋t

a. Determine the minimum sampling rate
b. Suppose that the signal is sampled at the rate Fs=200 Hz. What is the discrete –
time signal obtained after sampling?
c. Suppose that signal is sampled at 75 Hz, what is the signal obtained after
sampling?
d. What is the frequency 0 ≤ 𝐹 ≤ 𝐹𝑠 /2 of a sinusoid that yields samples identical
to the one obtained in part c
6 Let us consider two analog signals, 𝑥1 (𝑡) = cos 2𝜋(10)t and 𝑥2 (𝑡) = cos 2𝜋(50)t which are
sampled at 40 Hz. Obtain the corresponding discrete time signals and show that the discrete
signals thus obtained are same , i.e 𝑥1 (𝑛) = 𝑥2 (𝑛)
7 𝑥𝑎 (𝑡) = 3 cos 50 𝜋t+10 cos 300 𝜋t- cos 100 𝜋t, what is the Nyquist rate for this signal?
8 Consider the analog signal 𝑥𝑎 (𝑡) = 3 cos 2000 𝜋t+5 sin 6000 𝜋t +10 cos 12000 𝜋t .
a) What is the Nyquist rate for this signal?
b) Assume now that we sample this signal using sampling rate 𝐹𝑠 =
5000 𝑠𝑎𝑚𝑝𝑙𝑒𝑠/𝑠 . What is the discrete –time signal obtained after sampling?
c) What is the analog signal 𝑦𝑎 (𝑡) that we can reconstruct from samples if we use
ideal interpolation?
9 If 𝑥(𝑛)is an aperiodic finite energy signal, having its frequency spectrum of 𝑋(𝜔), show that
the signal 𝑥(𝑛) can be obtained from 𝑥𝑝 (𝑛), without any aliasing, provided 𝑥𝑝 (𝑛)]where N ≥
𝐿, L being the length of the signal 𝑥(𝑛).
10 State Linearity, periodicity and Symmetry properties of DFT
11 Let X(k) be a 14 point DFT of length 14 sequence x(n). The eight samples are given by, X(0)=12,
X(1)=-1+3j X(2)=3+j4, X(3)=1-j5, X(4)=-2+j2 , X(5)= 6+j3, X(6)=-2-j3 , X(7)=10
Determine the remaining samples of X(k). Evaluate the following functions of x(n),
without computing IDFT
13 13
x(7) b) ∑𝑛=0 x(n) c) ∑𝑛=0 𝑒 𝑗4𝜋𝑛/7 x(n) d) ∑13 𝑛=0 |x(n)|
2
12 𝑥(𝑛) = 𝛿(𝑛) + 2𝛿(𝑛 − 5),Find the 10 point DFT of the signal.
13 Comment on the characteristics of DFT of the signal x(n), when (a) Signal is Real and
circularly even (b)Signal is Real and circularly odd
14 Let x(n)={1, 2, 0,3}, Find the circularly folded signal x1(n)=x((-n))4and hence
determine the circularly even and odd part of x(n)
15 State and interpret Conjugate Symmetry Property of DFT
16 Show that correlation between two discrete signals lead to Perseval’s theorem
17 Let 𝑥(𝑛), 0≤ 𝑛 ≤ 𝑁 − 1, be a length N sequence with PN point DFT X(k), 0≤ 𝑘 ≤ 𝑃𝑁 − 1,

define 𝑦(𝑛) = 𝑥((𝑛)𝑁 ); 0≤ 𝑛 ≤ 𝑃𝑁 − 1.
How would you compute MN point DFT Y(k), knowing only X(k)?
18 Show that DFT of individual signals x(n) and g(n) for N Point , leading to 2 N Point DFT
computation , can be performed by using one N point DFT by considering x(n) as real part
and g(n) as an imaginary part of a complex signal. Note: take an arbitray signal of length 4 ,
x(n) and g(n).
19 Give Examples on DFT of time shifted signal , shift is circular
20 Give Examples on IDFT of frequency shifted DFTs , using circular frequency shift properties
21 Calculating energy of the signal using Parseval’s theorem
22 Consider a bandlimited continuous signal 𝑥𝑎 (𝑡) with 𝑋𝑎 (𝛺) = 0 for |𝛺| ≥ 8000𝜋. Assume
that anti-aliasing filter is ideal and sampling rate of the ADC is, fs=16000 samples/sec. If the
DFT samples X(k) are to be equivalent to the samples of 𝑋𝑎 (𝛺) that are utmost 15 Hz apart,
what is the minimum value for DFT size. Calculate the equivalent continuous time, frequency
spacing if we use FFT
23 State the property of multiplication of two DFTs and circular convolution
24 By means of DFT and IDFT determine the response of FIR filter with impulse response, ℎ(𝑛) =
{1,2,3} , 𝑥(𝑛) = {1,2,2,1}
25 Overlap save and Overlap add method , long data sequence convolution examples
26 2𝜋
𝑥𝑝 (𝑛) = cos 10 n, −∞ ≤ 𝑛 ≤ ∞ with frequency 𝑓0 = 1/10, and fundamental period N=10.
Determine the 10 point sequence 𝑥(𝑛)=𝑥𝑝 (𝑛), 0 ≤ 𝑛 ≤ 𝑁 − 1
27 Compute the 8 point DFT of the sequence, 𝑥(𝑛) = {1 , 1 , 1 , 1 , 0,0,0,0} using the in-place radix-2
2 2 2 2
decimation in time and decimation in frequency algorithm
28 X(k)={17< -1.12-j7.12, j3, 3.12+j2.87, 3, 3.12-j2.87, -j3, -1.12+j7.12}

Find x(n) using decimation in time IFFT algorithm
29 Let x(t) be an analog signal with bandwidth B=3kHz. We wish to use an N=2Lpoint DFT to
compute the spectrum of signal with resolution less than or equal to 50Hz. Determine
(i)The minimum sampling rate
(ii) The minimum required number of samples
(iii)Minimum length of the analog signal record
29 Given |𝐻𝑎 (𝑗𝛺)|2 = 1
7
, determine the analog filter system function 𝐻𝑎 (𝑠)
1+128𝛺
30 Convert the analog filter with system function

𝑠+0.1𝑠
𝐻𝑎 (𝑠) = (𝑠+1)2+9 into a digital filter by means of impulse invariance
method
31 Convert the analog filter with system function,

𝑠+0.1𝑠
𝐻𝑎 (𝑠) = (𝑠+0.1)2 +6
into digital IIR filter by means of BLT. The digital filter is to have a resonant frequency of
𝜋
𝜔𝑟 = 2
32 Design a single –pole lowpass digital filter with a -3dB gain at 𝜔𝑐 = 0.2𝜋 using BLT applied to
the analog filter
33 Design a low pass Butterworth filter using impulse invariant method for satisfying following
specificatoions
33 Design an IIR low pass Butterworth filter using Bilinear Transformation Technique for the
following specifications.
Pass Band: 0.8 ≤ |𝐻(𝑒 𝑗𝜔 )| ≤ 1 ; |𝜔| ≤ 0.22𝜋
Stop Band: |𝐻(𝑒 𝑗𝜔 )| ≤ 0.2 ; 0.64𝜋 ≤ |𝜔| ≤ 𝜋
Assume T= 1 second
34 Determine the order and poles of a lowpass Chebyshev filter that has a 1-dB ripple in the pass
band , a cutoff frequency 𝛺𝑐 = 1000 𝜋, a stopband frequency of 2000 𝜋, and attenuation of
40dB or more for 𝛺>=𝛺𝑠
35 A digital low-pas filter is required to meet the following specifications

Passband ripple :<= 1dB
Passband edge = 4 KHz
Stopband Attenuation :>=40 dB
Stopband edge = 6 kHz
Sample rate :24 KHz
The filter is to be designed by performing a bilinear transformation on an analog system function.
Determine what order Butterworth, Chebyshev must be used to meet the specifications in digital
implementation
36 A Digital Low pass filter needs to be designed with following response , Order M=7
Determine the frequency response of this FIR filter with rectangular window used for
truncation.
37 Realize the system whose impulse response ℎ(𝑛)is given by,

Using linear phase form.
Design a high pass FIR filter which is expected to have symmetric magnitude response of
order as 9 and cut off frequency as 0.25π. Plot the Magnitude response using Hamming
window.
38 With block schematic and relavant equations , explain the application of an adaptive filter for
interference removal and system identificating in a digital signal, 𝑥(𝑛)
39 Show that the structure of the following figure is a LTI system, determine its transfer function
G L
L
x(n) yn)
40 Given a DSP upsampling system with the following specifications , sampling rate = 6000Hz,
input audio frequency range =0-800 Hz, pass band ripple =0.02 dB, stop band attenuation =
50 dB, upsampling factor L= 3. Determine the FIR filter length , cut off frequency and
window type , if windowing method is used
41 With a neat block diagram and relevant equations explain the working principle of Adaptive
channel equalization

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1 How do you provide time scaling for a signal?

Or Show that time scaling of a signal is

Illustrate the decimation operation by decimating the above signal by a factor 3.

5 Consider an analog signal, 𝑥𝑎 (𝑡) = 3 cos 100 𝜋t

12 𝑥(𝑛) = 𝛿(𝑛) + 2𝛿(𝑛 − 5),Find the 10 point DFT of the signal.

determine the circularly even and odd part of x(n)

15 State and interpret Conjugate Symmetry Property of DFT

17 Let 𝑥(𝑛), 0≤ 𝑛 ≤ 𝑁 − 1, be a length N sequence with PN point DFT X(k), 0≤ 𝑘 ≤ 𝑃𝑁 − 1,

19 Give Examples on DFT of time shifted signal , shift is circular

21 Calculating energy of the signal using Parseval’s theorem

23 State the property of multiplication of two DFTs and circular convolution

28 X(k)={17< -1.12-j7.12, j3, 3.12+j2.87, 3, 3.12-j2.87, -j3, -1.12+j7.12}

30 Convert the analog filter with system function

31 Convert the analog filter with system function,

35 A digital low-pas filter is required to meet the following specifications

37 Realize the system whose impulse response ℎ(𝑛)is given by,

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