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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).

3.

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,

.

3

1

1 z 1 z 2

2

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

filter.

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.

Passband

Stopband

30 kHz

Passband Ripple

Stopband Attenuation

Sampling Frequency

0 dB

-10 dB (at 30 kHz)

100 kHz

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:

1

H ( e j ) 1

2

for

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

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

specifications:

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 .

4

H (e

) 0

for .

4

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

)j

for

for

0

0

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

system.

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

64

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

data

Passband ripple

0.05 dB

Stopband ripple

0.01 dB

10 log( p s ) 13

Filter length

N

14.6f

where

f normalised

transistion

width

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)

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