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GEORGIA INSTITUTE OF TECHNOLOGY

SCHOOL of ELECTRICAL and COMPUTER ENGINEERING

ECE 3084
Summer 2014
Problem Set #4
Assigned: 13-June-14
Due Date: 20-June-14

Your homework is due at 3:00 PM on Friday, June 20, in his Van Leer W431 office. If he is
not there, you may slip it under his office door. (Of course, you are also welcome to turn it in on
Thursday in class if you get it done early.)
A sheet of Laplace transform pairs and properties has been posted to the Resources
section of the T-square website. You will want to print this out and have it handy while you are
working on the homework, along with the sheet of Fourier transform pairs and properties.

Refrain from looking at backfiles of homework and


exam solutions i.e., word in Georgia Tech parlance
from previous versions of ECE2025, ECE2026, or ECE3084,
beyond your own materials assembled while taking those
classes and any old material we explicitly provide to you.

From http://www.phdcomics.com/comics/archive.php?comicid=976

PROBLEM 4.1:
Consider the following amplitude modulation system:
x(t)

v(t)

- LTI System

H(j)

y(t)
-

cos(1000t)
Assume that the input signal x(t) has a bandlimited Fourier transform as depicted below
X(j)
6

1
@
@
@
@
@

100

100

and the linear system has the frequency response of an ideal bandpass filter:

1
900 < || < 1000
H(j) =
0
otherwise.
(a) Plot the frequency response H(j) of the ideal BPF specified above. Be sure to plot for both
negative and positive frequencies.
(b) Plot the Fourier transform V (j) of the signal v(t) at the output of the multiplier.
(c) Plot the Fourier transform Y (j) of the output signal y(t) from the filter. Do not try to find
y(t).
(d) The output signal is called a single sideband signal. Can you see why?
(e) Draw a block diagram of a system that will recover the original input signal x(t) from
y(t).
Note that the negative frequency portion of the Fourier transform X(j) is shaded. Mark the
corresponding region or regions in your plots of V (j) and Y (j).

PROBLEM 4.2:
Although it may seem counterintuitive, the field of digital communication is mostly about analyzing analog waveforms. For instance, the first modem Prof. Lanterman ever used was a 300 baud
modem that encoded a mark as a sinusoid of one frequency and a space as a sinusoid of a
different frequency. To distinguish between two frequencies, a first thought might be to build two
bandpass filters and compare the energy at the output of both filters. The problem considers an
alternative decoder shown below, which employs baseband demodulation, filtering, and slicing,
which is the s(t) = y(t)[y(t td )] operation.
x(t)

w(t)- Analog
LPF

y(t)
-

ejc t

Imag
s(t)
- part
Im{ }
6

d(t)
-

complex

- Delay

- conj

td

{ }

(a) Suppose that the input signal to the above system is x(t) = x1 (t) = cos(1 t) where 1 =
c 0 with 0 > 0. Using the frequency-shift property of Fourier transforms and the Fourier
transform of the cosine signal, determine and plot the Fourier transform of the signal
w(t).
(b) Now suppose that the lowpass filter (LPF) in the above diagram has the frequency response
depicted below:
Hlp (j)
6

1
@
@
@
@
@

Determine the smallest value for p and the largest value for s such that the output of the
filter is
y(t) = 21 ej0 t .
This will give the largest transition region between passband and stopband.
(c) Show that for the passband and stopband frequencies found in part (b), the overall output
is a constant; i.e., d(t) = d1 (t) = 14 sin(0 td ).
(d) Now suppose that x(t) = x2 (t) = cos(2 t) where 2 = c + 0 and the cutoff frequencies of
the filter are the same as found in part (b). Show that the overall output is again a constant,
but in this case, d(t) = d2 (t) = 41 sin(0 td ).
(e) Assume that the input signal can be either x1 (t) or x2 (t), and assume that td < /0 .
Propose a simple algorithm for determining which signal was used.

PROBLEM 4.3:

x(t)

xs (t)

xr (t)

- LTI System

Hr (j)

p(t) =

(t nTs )

n=

The input signal for the above sampling/reconstruction system is


x(t) = 2 cos(20t) + cos(50t + /4)

<t<

and the frequency response of the lowpass reconstruction filter is



Ts
|| < /Ts
Hr (j) =
0
|| > /Ts
where Ts is the sampling period.
(a) Determine the Fourier transform X(j) and plot the Fourier transform Xs (j) for
2/Ts < < 2/Ts for the case where 2/Ts = 200. Carefully label your sketch to receive
full credit. What is the output xr (t) in this case?
(b) Now assume that s = 2/Ts = 80. Plot the Fourier transform Xs (j) for 2/Ts <
< 2/Ts for the case where 2/Ts = 80. Carefully label your sketch to receive full credit.
What is the output xr (t) in this case?
(c) Is it possible to choose the sampling rate so that xr (t) = A + 2 cos(20t), where A is a
constant? If so, what is the value of Ts and what is the numerical value of A?

PROBLEM 4.4:
In this problem, we will find some Laplace transforms.
(a) Find Xa (s), the Laplace transform of xa (t) = (t 5)u(t 2) by first rewriting it as xc (t) =
(t 2 3)u(t 2) and meditating upon that for a while.
(b) Use the Laplace transform table entry for sin(0 t)u(t), along with the time-shift property, to
easily find Xb (s), the Laplace transform of xb (t) = sin(t /3)u(t /3).
(c) Curiously, finding the Laplace transform of xc (t) = sin(t /3)u(t) requires more work
than doing the transform in part (d). The shift property will not help us here. There
are several ways to tackle this problem. One approach is to use the trigonometric identity
sin(A + B) = sin(A) cos(B) + cos(A) sin(B), and let A = t and B = /3. Use this approach
and the Laplace transform table entries for cos(0 t)u(t) and sin(0 t)u(t) to find Xc (s), the
Laplace transform of xe (t).

PROBLEM 4.5:
A common sine wave generation technique is to simply send samples of a sine wave stored in a table
through an analog-to-digital converter. To save memory, usually only one quarter of the sinusoidal
waveform is stored, with the other three quarters generated by reading the samples out in reverse
order and/or negating (basically putting a minus sign in front) the value as needed to create one
full period of a sine wave.
The Yamaha OPL series of sound synthesis chips was frequently used in PC soundcards in the
1990s. The designers of these chips employed the cheap trick of using alternative readout orders
and signs (or not outputting anything at all) for each quarter-period to generate other waveforms.
For instance, you can see the four waveforms available from the OPL2 here:
http://en.wikipedia.org/wiki/Yamaha YM3812
You can see the eight waveforms1 available from the OPL3 from page 53 of the applications
manual:
http://www.msxarchive.nl/pub/msx/docs/datasheets/opl4.pdf
In this problem, we will look at waveform 3 of the OPL3, which consists of the first half of the
positive-going hump of the sine wave, with the second half zeroed out. Note that the drawing in
the applications manual shows two periods of the waveform.
Suppose one period of the periodic signal x(t) with fundamental period T0 is specified by the
equation2
 
(
sin T0 t
for 0 t T0 /2,
x(t) =
0
for T0 /2 < t < T0 .
The nonperiodic signal x1 (t) specified by the equation


sin C t for 0 t C/2,
x1 (t) =
0
otherwise
has Fourier transform given by
X1 (j) =


j exp j C2
,

2
2
C

In class, we saw that in general, a periodic signal that can be written as f (t) =
has Fourier series coefficients given by


2
1
ak =
F1 j k ,
T0
T0

`= f1 (t`T0 )

where F1 (j) is the Fourier transform of f1 (t). Use this trick, along with the knowledge of X1 (j)
given above, to easily compute the Fourier series coefficients for x(t). This is much easier
than computing them from scratch using the Fourier analysis integral!

1
Waveform number 8 is not based on a sine table; its actually based on an exponentiation table. Its there
since fast hardware multipliers take up a lot of real estate on an integrated circuit, so the OPL series handles
multiplication implicitly by adding logarithms and then taking an exponential, which incidentally is how a slide rule
works. The designers decided to let the exponentiation table do double-duty as an alternative waveform.
2
A real soundcard employs samples of this waveform passed through a lowpass filter. Since x(t) is not bandlimited,
the the output waveform is corrupted by slight aliasing effects. In this problem, we will pretend that x(t) is pure,
i.e., it has no aliasing issues.

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