You are on page 1of 4


Determine the 4-point Discrete Fourier Transform (DFT) of the discrete input sequence
x( n) 0,1,2,3.

2. The difference equation of Linear Time Invariant (LTI) discrete time system is given as
y ( n) 0.4 y ( n 2) x (n) 0.9 x(n 2) . Determine the frequency response H(ej).

Then evaluate and plot the magnitude and phase response.


Consider a causal linear shift invariant system with system function

1 a 1 z 1
H ( z)
where a is real. Plot the pole-zero diagram for 0 < a < 1 and
1 az 1
determine, the range of values of a for which the system is stable and Show analytically
that this system is an all pass system.
4. Consider the discrete time sequence x( n) u ( n) u ( n 8) . Determine the ztransform of the sequence and sample it at 6 points on the unit
circle using the
relation X (k ) X ( z )
and also find the inverse DFT of
k 0,1,2,...,5.
z e
j 2 k / 6

X (k ) . Then discuss the result of inverse of X(K) by comparing with input sequence.

5. Using Decimation In Time-Fast Fourier Transform (DIT-FFT) algorithm, compute 8point Discrete Fourier Transform (DFT) of the discrete input sequence x(n) = {1, 1, 1, 1,
0, 0, 0, 0}.
6. Using Decimation In Frequency - Fast Fourier Transform (DIF-FFT) algorithm,
compute 8-point Discrete Fourier Transform (DFT) of the discrete input sequence x(n) =
{1, 2, 3, 4, 4, 3, 2, 1}.
7. In the Discrete Time (DT) system, system transfer function is given as,
H ( z)

(1 0.875 z 1 )
(1 0.7 z 1 )(1 0.2 z 1 0.9 z 2 )

Determine Direct Form I and II implementation and then obtain at least two Cascade
Form implementations with elements of order 2.
Z 2 0.36
z 2 0.1z 0.72
Obtain the following system realization Direct form I, II and cascade form.
8. The system transfer function of a discrete system is given as X ( z )

9. The system transfer function of the discrete time Finite Impulse Response (FIR)
system is given as,
H ( z)

(1 0.875 z 1 )
. Obtain Dirct form I, II system realization.
(1 0.7 z 1 )(1 0.2 z 1 0.9 z 2 )

10. Develop the parallel form structure for the second order discrete time system whose
1 2 z 1 z 2
H ( z)
system transfer function is given as,
1 z 1 z 2

11. The system transfer function of the discrete time Finite Impulse Response (FIR)
system is given as, H ( z ) (1 2 z 1 z 2 )(1 z 1 z 2 ) . Obtain the cascade realization
of the given system.
12. Compare the important performance characteristics of Butterworth and Chebyshev
13. Enumerate the computational advantages of Fast Fourier Transform (FFT) over the
Discrete Fourier Transform (DFT).
14. Design a digital Butterworth IIR filter by meeting the specifications in Table.

0 to 20 kHz (-3dB at 20 kHz)


30 kHz

Passband Ripple
Stopband Attenuation
Sampling Frequency

0 dB
-10 dB (at 30 kHz)
100 kHz

The filter is to be designed by performing an Impulse Invariance transformation on an

analogue filter. For this, determine the order of the analogue filter that must be used to
meet the above specifications. Then obtain the transfer function, H(z) of the digital filter.
15. Design a digital Chebyshev filter to meet the following constraints:
H ( e j ) 1


0 0.2 .

0 H (e j ) 0.1

for 0.5 .
Use Bilinear Transformation technique and assume sampling period T = 1 sec.
16. Design and obtain the coefficients of an FIR (Finite Impulse Response) low pass filter
to meet the specifications given below using the Hamming window method. Compute
only the middle five impulse response of the filter.
Passband edge frequency
= 1.5 kHz
Stopband attenuation > 50dB
Stopband edge frequency

= 2.0 kHz

Sampling frequency = 8 kHz

17. Determine the filter coefficients in order to design a low pass Finite Impulse
Response (FIR) filter using a rectangular window technique with the following
Unity passband gain ( p )
Cutoff frequency ( f c ) =1000Hz
Sampling frequency (F ) = 5 kHz
Length of the impulse response = 7.
18. An ideal High Pass Finite Impulse Response (FIR) filter has frequency response
H ( e j ) 1
for .

H (e

) 0

for .

Where = angular frequency. Determine the impulse response h(n) and its coefficients
for N = 11, where N = number of samples using Hanning window.
19. A signal is sampled at a frequency of 44000 Hz, has a useful content from 0 to
11000 Hz and is corrupted with noise from 12000 to 22000 Hz. Design a digital FIR lowpass filter to attenuate the noise by at least 60 dB without affecting the useful content
by more than 1 dB using a fixed window (Blackman).
20. Determine the filter coefficients of a linear all pass (ideal Hilbert transformer filter)
filter using a rectangular window technique. The desired frequency response of this filter
is given by

H ( e j ) j
H (e




21. A multirate discrete system is shown in Figure. Develop an expression for the output
y(n) as a function of x(n).

22. For the discrete-time system illustrated in Figure, derive the difference equation of the

23. The block diagram of a three stage decimator which is used to reduce the sampling
rate from 3072 kHz to 48 kHz is given in the below figure.

1. Indicate the sampling rate at the output of each of decimator in the three stages.
2. Determine the band-edge frequencies for the decimating filter at each stage.
Input sampling frequency, Fs
3072 kHz
Decimation factor, M
Passband ripple
0.01 dB
Stopband ripple
60 dB
Frequency band of interest
0-20 kHz
3. Assuming that the input and output sampling rates for the decimator are 3072 kHz and
48 kHz, respectively. Determine the overall decimation factor; determine all the possible
sets of integer decimation factors, assuming two stages of decimation. Repeat with three
stages and four stages of decimation.
4. Calculate the total number of multiplications per second (MPS) and the total storage
requirements (TSR) in terms of filter lengths N1, N2 and N3.
24. Design, at a block diagram level, a two stage decimator that down samples an audio
signal by a factor of 30 and satisfies the given in table.
Input sampling frequency, Fs
240 kHz
Highest frequency of interest in the 3.4 kHZ
Passband ripple
0.05 dB
Stopband ripple
0.01 dB
10 log( p s ) 13
Filter length


f normalised



Specify the sampling frequencies at the input and output of each stage of decimation,
then analysis the computational and storage complexities, and possible set of decimations
and band edge frequencies.
(refer page no 599 DSP by Emmanuel Ifeachor)