0% found this document useful (0 votes)
22 views5 pages

Introduction To The Simulation of Signals and Systems: Please Turn The Page Over

The document describes a lab simulation exercise involving signals and systems. It contains 5 problems covering topics like simulating discrete-time signals using Simulink, designing digital filters, modeling A/D converters, and separating signals. The problems involve tasks like simulating sampled sine waves, finding transfer functions, plotting responses, and designing digital filters.

Uploaded by

Aman
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
0% found this document useful (0 votes)
22 views5 pages

Introduction To The Simulation of Signals and Systems: Please Turn The Page Over

The document describes a lab simulation exercise involving signals and systems. It contains 5 problems covering topics like simulating discrete-time signals using Simulink, designing digital filters, modeling A/D converters, and separating signals. The problems involve tasks like simulating sampled sine waves, finding transfer functions, plotting responses, and designing digital filters.

Uploaded by

Aman
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
You are on page 1/ 5

Introduction to the Simulation of Signals and Systems

Objective:
The objective of this lab is twofold. First, the student will get familiar with Simulink as a tool
to simulate discrete-time and continuous-time signals and systems. Second, some typical systems
will be simulated and the results will be analyzed using Matlab.

(1) Use Simulink to simulate the system shown in Figure 1. The system is composed of a sampled
sine wave of amplitude equal to 5, frequency f1 , and initial phase shift of π/3 radians. The
signal will be displayed on the scope. Assume that the sampling period is 5 µsec.

(a) If f1 = 30 kHz, simulate the system with a time equivalent to 4 periods of the sine
wave. Display the waveform on the scope and also export the data of the output to the
Matlab workspace to plot the amplitude (discrete time versus time in msec). Also, plot the
amplitude versus the time index n.

(b) Change f1 to 230 kHz and investigate the output signal. Compare to part (a) and comment.

(c) What is the apparent frequency of the output if f1 = 370 kHz? Compare to part (a) and
comment.

Figure 1: Block Diagram for Exercise 1

(d) For the above cases, give the digital frequencies and indicate if they have aliases in the
range 0 ≤ f ≤ 100 kHz.
1
please turn the page over

1
Figure 2: Finding ytran (n)

Figure 3: Generating a Pulse Input

(2) Consider the digital filter given by the

y(n) − 1.7654y(n − 1) − 0.81y(n − 2) = x(n) − 0.5x(n − 1)

(a) Find the transfer function of the above filter. Hence, determine the poles and zeros. Check
for stability.

(b) Using Simulink, assuming fsamp = 200 kHz, and a simulation time of 0.5 msec, find the
step response.

(c) Determine the steady-state value, Go . Hence, plot the transient response, ytran . (Hint:
Use the model shown in Figure 2.)

(d) If the input is a pulse equivalent to x(n) = 2u(n)u(−n + 39), find the pulse duration in
msec, and hence determine the pulse response. (Hint: Use the model shown in Figure 3).

(3) It is required to design an anti-aliasing low-pass filter to filter out frequencies in the frequency
band 0 ≤ f ≤ 5 kHz. The requirements are as follows:

ˆ The pass-band attenuation αs = 0.5 dB


1
please turn the page over

2
Figure 4: Data Acquisition System

ˆ The stop-band attenuation αs = 46 dB

ˆ The Order of filter n=8.

(a) Determine the stop band frequency edge fs and the pass-band frequency edge fp .

(b) Find the transfer function and hence plot the magnitude and phase responses of your
design.

(c) For your design, what will be the order of the filter you designed if Butterworth, Chebychev,
or Elliptic filter approximations are used? Comment.

(d) To test the anti-aliasing lowpass filter, construct the model shown in Figure 4, which
represents a part of a data acquisition system. Assume that the digital filter is given by
the following difference equation:

y(n) = 0.5335y(n − 1) − 0.7265y(n − 2) + 0.8633[x(n) + 0.618x(n − 1) + x(n − 2)]

What is the type of this digital filter? Test for stability. Plot the magnitude and phase
responses.

(e) Simulate the model using Simulink for input frequencies fi = i kHz, i = 1, 2, . . . , 7. What
is the frequency where the output is zero? Verify that there is no aliasing. Investigate the
different waveforms seen on the screen of the scopes and comment on the results.

(4) In this exercise, we would like to model an A/D converter with the following parameters:
1
please turn the page over

3
Figure 5: Block Diagram for problem 4

ˆ Number of bits is nb

ˆ Full scale-level is Vref = 5 V . (This means that the dynamic range is ±Vref )

ˆ Sampling frequency of 12 kHz.

The model is shown in Figure 5.

(a) For the following values of the parameters, model the A/D converter and plot the transfer
characteristics. Hint: Use one period of a low frequency signal (1 Hz).

ˆ nb = 4 bits

ˆ nb = 8 bits

(b) For an analog input sinewave of frequency 7 Hz and amplitude = Vref − ∆ (quantization
step), plot the output and the quantization noise signals for both cases of part (a).

(c) Calculate the quantization average power and hence the S/N ratio in dB of the output
signal. Hint: Use about 10 cycles of the sinewave to calculate the average power.

(d) Find the effective number of bits if the S/N is 45 dB.

(5) Consider the following continuous-time signal

1
please turn the page over

4
xa (t) = xa1 (t) + xa2 (t) + xa3 (t),

where,

π
xa1 (t) = 16 cos4 (2πf1 t − ) + 2 cos(2πf4 t),
3
π
xa2 (t) = 6 cos(2πf2 t) cos(2πf3 t + ),
4
π
xa3 (t) = 12 sin(2πf5 t) cos(2πf6 t + ),
3

f1 = 50 Hz, f2 = 200 Hz, f3 = 800 Hz, f4 = 1500 Hz, f5 = 400 Hz, and f6 = 800 Hz. It is
required to design a digital signal processing-based system to separate the signals xa1 (t) and
xa3 (t) from the signal xa (t). Assume that the attenuation in the passband αp = 1 dB and
αs = 48 dB in the stop-bands.

(a) Determine the minimum required sampling rate fsamp(min) . Hence, use twice this value.

(b) Draw a block diagram of the system indicating the requirements of each block (use the
minimum possible number of blocks and filters).

(c) Design and write the difference equations of the digital filters needed (show only six terms).

(d) Plot the attenuation response (in dB) for each filter to verify your design.

You might also like