DSP Problem Set

1. Find out the impulse and step response of the systems described as
i) y(n)  2(0.4) n u(n)  (0.2) n u(n)
ii) y(n)  0.447(1.618) n u(n)  0.447(0.618) n u(n)

2. Verify sampling theorem.

3. Generate an amplitude modulated signal using MATLAB.

4. Generate a frequency modulated signal using MATLAB.

*** Simulation of Discrete-Time Systems

For the simulation of causal LTI discrete-time systems the command filter can be used. There are several versions
of this command. If we denote
num = [p0 p1 . . . pM],
den = [d0 d1 . . . dN],
then y = filter(num,den,x) generates an output vector y of the same length as the specified input vector x with zero
initial conditions, that is, y[-1] y[-2] = ... = y[-N] = 0. The output can also be computed using y =
filter(num,den,x,ic) where ic = [y[-1], y[-2], ..., y[-N]] is the vector of initial conditions. Access to final
conditions is obtained using [y,fc] filter(num,den,x, ic).

[ ** The Moving Average System

A causal version of the three-point smoothing filter is obtained by simply delaying the output by one sample period,
resulting in the FIR filter described by
1
y (n)  ( x[n]  x[n  1]  x[n  2])
3

Generalizing the above equation we obtain

M 1
1
y ( n) 
M
 x[n  k ]
k 0
which defines a causalM-point smoothing FIR filter. The system of the above Eq is also known as a moving average
filter . We illustrate its use in filtering high-frequency components from a signal composed of a sum of several
sinusoidal signals.]

5. Simulation of an M-point Moving Average Filter to filtering out high-frequency components

from a signal composed of a sum of several sinusoidal signals.

[ ** Cascade of LTI Systems

In practice a causal LTI discrete-time system of higher order is implemented as a cascade of
lower order causal LTI discrete-time systems. For example, the fourth-order discrete-time
system given below

can be realized as a cascade of two second-order discrete-time systems:

 ]
6. Generate a sequence x[n], and then uses it as the input of the fourth-order system, generating
the output y[n]. It then applies the same input x[n] to Stage No. 1 and finds its output sequence
y1[n]. Next, it uses y1[n] as the input of Stage No. 2 and finds its output y2[n].

7.
Write a MATLAB Program to compute the outputs of the above two systems for an
Input

8. Response of Butterworth filter to square signal

9. Frequency response of raised cosine filter

10. Impulse response of raised cosine filter

11. Recovering a sinusoidal signal buried in noise

12. Digital Modulations (ASK, PSK and FSK) of Sine Signals

13. Write a program in MATLAB to compute the frequency response of the system
(2  z )
x( z ) 
(1  0.6 z )

14. Write a program to find out the impulse response of any discrete system.

15. Write a program to find the cross-correlation of two signals.

16. Perform Decimation process using Matlab programming.

17. To perform interpolation process using Matlab programming.

18. Sampling rate conversion by a rational form using MATLAB

19. Calculate and plot the spectrum of the 1D digital signal:

20. Write a MATLAB code to illustrate the following DFT properties: time-delay,
frequency shifting, modulation and time inversion, using the digital signal:

21. Consider the following two digital signals: x = [0, 0.25, 0.5, 0.75, 1] and
h = [0, 0.5, 1, 0.5, 0] . Calculate the product of their DFTs and then come back to the
time domain by inverse DFT. Compare the obtained result to the convolution
product of the two signals and conclude.

22. Consider the sum of two sinusoids of 50 Hz and 120 Hz, corrupted by an additive, zero-
mean, white noise. Calculate and plot its power spectral density.

23. Consider the following impulse response of a digital filter:

a. Calculate its z-transform

24.
DFT calculation with a digital computer requires truncating the signal because the number of
samples has to be finite (generally a power of 2 in the case of the fast Fourier transform). The
signal is truncated by default with a rectangular window, but several other weighting windows
can also be used (Hamming, Hanning, Blackman, etc.). Its choice depends on the spectral and
dynamic resolutions required for a given application.

# Generate a sinusoid on N = 32 points, with the amplitude 1 V and a frequency of 100 Hz,
sampled at 256 Hz. Calculate its DFT on 1,024 points using successively the rectangular,
triangular, Hamming, Hanning and Blackman windows to truncate the signal.
25.
Plot its spectrum obtained for different weighting windows and comment upon this

26.
Design a digital filter which meets the following specifications using the bilinear
method:
- sampling frequency: 1 Hz,
- 0 dB attenuation of the DC component,
- maximum attenuation of 1 dB at 0.1 Hz,
- minimum attenuation of 15 dB at 0.15 Hz.
Consider a Butterworth and then a Chebychev model for the filter transfer
function.

27.
Design a 25th order FIR lowpass filter with a minimum transition band, using the window
method. Consider a sampling frequency fs = 20 kHz and the following lower and upper passband
frequency edges: f1 = 3 kHz and f2 = 7 kHz. Plot the transfer function, the impulse response and
the zeros of the designed filter.

28.
Design a 24th order FIR highpass filter using the frequency sampling method. Consider a cutoff
frequency fc = 6 kHz and a sampling frequency fs = 20 kHz. Plot the transfer function, the
impulse response and the zeros of the designed filter.

29.
Consider a sum of two sinusoids, whose frequencies are 1 kHz and 1.56 kHz, sampled at 10 kHz.
Extract the first sinusoid of this mixture using a lowpass FIR filter.

30. Write a MATLAB code to simulate a channel which is characterized by a bandwidth of 4,900
Hz and a zero-mean white Gaussian additive noise with the variance 1. Filter the transmitted
signal using an IIR 8th order Chebychev type I filter. Plot on the same figure the ideal signal, the
noisy signal and the matched filter output signal.