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Telecommunication
Overview
In the previous week the we have examined the Fourier
Transform applied to energy signals.
Reading
2.5, 2.6
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Lecture Objectives
To show how the delta function allows the Fourier Transform to
be applied to power signals (i.e., periodic signals).
Z f f f 0
This results in
e j 2 f 0 t f f 0
1 1
sin 2 f 0t f f0 f f0
2j 2j
1 1
cos 2 f 0t f f 0 f f 0
2 2
2
Fourier Transform of Periodic Signals
Using this same approach we can develop the Fourier
Transform for any periodic signal.
X f F ck e j 2 kf0t ck F e j 2 kf0t ck f kf 0
k k k
TW= T0 /2
To A=1
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Example 3.1 – cont.
Writing the signal in terms of its Fourier Series coefficients:
1 k
x t
2 k
sinc e j 2 kf0t
2
Using the linearity property and the FT for a complex
exponential:
F e j 2 f 0 t f f 0
1 k 1 k
X f F sinc e j 2 kf0t sinc F e j 2 kf0t
2 2
2 k 2 k
1 k
sinc f kf 0
2 k 2
Note that since the signal is periodic the spectrum
(i.e., Fourier Transform) is discrete.
T0=0.1 sec
f0=10 Hz
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Why study LTI systems?
Several parts of a communication system can be modeled as an
LTI system including
● Filters
● Equalizers
● The channel
● Pulse shaping
y (t ) a1 x1 (t ) a2 x2 (t ) a1 x1 (t ) a2 x2 (t )
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Impulse Response
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Impulse Response – cont.
Let us approximate a general input x(t) using a series of
impulses
x(t ) x nt (t nt ) t
n 0
x nt
t
Since the system is LTI we can approximate the output as
y (t ) x nt h(t nt ) t
n0
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Impulse Response – cont.
Now, if we let ∆t go to zero we get:
y (t ) x h t d
Also,
y (t ) h x t d
This says that the output of an LTI system is the convolution of
the input and the impulse response.
Thus we can find the output for any input if we know the
impulse response.
Impulse Response
Interpretation
The current output of an LTI system is a weighted integral over
the past history of the input signal,
weighted according to the impulse response of the system.
y (t ) x h t d h x t d
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Impulse Response in the Frequency
Domain
Recall the Fourier Transform Property (convolution):
w1 (t ) * w2 (t ) W1 ( f )W2 ( f )
y (t ) x(t ) * h(t ) Y ( f ) X ( f ) H ( f )
Transfer Function
Rewriting
Y( f )
H( f )
X(f )
H(f) is termed the Transfer Function of the system.
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Transfer Function – cont.
Alternative Definition
Consider a LTI system characterized by impulse response h(t)
with input x(t)
x t e j 2 f 0 t
The output can be determined as
y t x h t d
h x t d
h e
j 2 f 0 t
d
h e
j 2 f 0 t j 2 f 0
e d
e j 2 f 0 t H f 0
Thus, when the input to a LTI system is a complex sinusoid of
frequency fo,
the output is also a complex sinusoid of frequency fo
weighted by the transfer function at that frequency, H(fo).
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Transfer Function – cont.
Alternative Definition
In general, a signal x(t) can be expressed in terms of the
inverse Fourier Transform
x(t ) X f e j 2 f t df
which can be written in the limiting form
x(t ) lim
f 0
X k f e
k
j 2 k f t
f
Thus, from the previous development we can write
y (t ) lim
f 0
k
H k f X k f e j 2 k f t f H f X f e
j 2 f t
df
Y f
Thus Y ( f ) H f X f
If h t is real, then H ( f ) H * ( f ).
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Magnitude of H(f)
Proof
Taking the magnitude
H( f ) h t e j 2 f t dt h t cos 2 f t dt j h t sin 2 f t dt
2 2
h t cos 2 f t dt h t sin 2 f t dt
h t e h t cos 2 f t dt j h t sin 2 f t dt
j 2 f t
H ( f ) dt
Angle of H(f)
Proof
Now, we can take the angle
H ( f ) h t e j 2 f t dt
h t cos 2 f t dt j h t sin 2 f t dt
h t sin 2 f t dt h t sin 2 f t dt
tan 1 tan 1
h t cos 2 f t dt h t cos 2 f t dt
where we assumed a real signal h(t) to make the first step.
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Angle of H(f) – cont.
Proof
Substituting for –f we have
H ( f ) h t e dt
j 2 f t
h t cos 2 f t dt j h t sin 2 f t dt
h t sin 2 f t dt
tan 1 H ( f )
h t cos 2 f t dt
Example 3.2
Consider a system with impulse response
h t e 5t u t
Determine the output when the input is
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Example 3.2 – cont.
Frequency domain
Let’s try this in the frequency domain instead.
1
Hf
5 j 2 f
1
X f f 100 f 100 f 10 f 10
2
Y f H f X f
1 1 1
e at u (t ) , cos 2 f 0 t f f 0 f f 0
a j 2 f 2 2
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Example 3.2 – cont.
Frequency domain
1 1 1
Yf f 100 f 100
2 5 j 200 5 j 200
1 1
f 10 f 10
5 j 20 5 j 20
1 5 j 200 5 j 200
Yf f 100 f 100
2 25 200 2
25 200
2
5 j 20 5 j 20
f 10 f 10
25 20 25 20
2 2
1 5
Yf f 100 f 100
2 25 200 2
200 1
f 100 f 100
25 200
2
j
5
f 10 f 10
25 20
2
20 1
f 10 f 10
25 20
2
j
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Example 3.2 – cont.
Time domain
In the time domain we have
5 200
y t cos 200 t sin 200 t
25 200 25 200
2 2
5 20
cos 20 t sin 20 t
25 20 25 20
2 2
Now
a b 1
cos 2 f 0t 2 sin 2 f 0t cos 2 f 0t
a b
2 2
a b 2
a b2
2
b
tan 1
a
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Example 3.2 – cont.
Transfer Function of the System
1 1 2 f
Hf , Hf , H f tan 1
5 j 2 f 25 2 f
2
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Summary
The Dirac delta function (or impulse) can be used to define the
Fourier Transform for periodic signals.
When the Fourier Series combined with the Dirac delta
function, the Fourier Series coefficients allow us to determine
the Fourier Transform.
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Introduction to
Telecommunication
Filtering
Overview
In the previous classes we have discussed using the Fourier
Transform to characterize systems.
Reading
2.6, 2.7
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Filters
A filter is a system which passes certain frequencies and rejects
other frequencies.
Types of filters
• Lowpass filter
• Highpass filter
• Bandpass filter
• Bandstop filter
Ideal filter
• An ideal filter is one which perfectly passes frequencies in a
certain range (termed the pass band) and perfectly rejects
frequencies in another range termed the stop band.
• An ideal filter doesn’t distort the signal in the pass band.
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The Ideal Lowpass Filter
Magnitude / Phase Responses
f j 2 f t0 f
H f rect e rect
2B 2B
f j 2 f t0
arg H f arg rect e
2B
arg cos 2 f t0 j sin 2 f t0 f B
0 f B
1 sin 2 f t0
tan f B 2 f t0
f B
cos 2 f t0
f B
0
0 f B
f
H f rect H f H f e j 2 f t0
2B
A f e j 2 f t0
arg H f 2 f t0
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The Ideal Lowpass Filter
Impulse response
The impulse response of the ideal LPF can be found by taking
the inverse Fourier Transform of the ideal LPF frequency
response.
h t F 1 H f
2B=5, to=2
5sinc(5(t-2))
f j 2 f t0
F 1 rect e
2B
2 Bsinc 2 B t t0
1 f
sinc 2Wt rect ,
2W 2W
x t t0 e j 2 f t0
Xf
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The Ideal Lowpass Filter
Causal LPF
We can make the filter causal by simply truncating the impulse
response before t=0.
2B=5, to=0
5sinc(5t)u(t)
2B=5, to=2
5sinc(5(t-2))u(t)
This makes the system closer to ideal, but requires a delay which
some applications may not tolerate.
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The Ideal Lowpass Filter
Causal LPF
A second option is to delay the impulse response and truncate it.
Delay = 5
2B=5, to=5
5sinc(5(t-5))u(t)
h t 5 e 5 t Hf
1
1 j f / 2.5
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The Ideal Lowpass Filter
Pulse Response of Ideal LPF
Let us apply a rectangular pulse x(t)=rect(t/T) to an ideal
lowpass filter (impulse response h(t)=2B sinc(2Bt) ).
y t x t h t
T /2
rect 2 Bsinc 2 B t d 2 Bsinc 2 B t d
T T /2
T /2
sin 2 B t d
2B d , 2 B t d
T / 2
2B t 2 B
2 B t T /2
1 sin
2 B t T /2
d
Si 2 B t T / 2 Si 2 B t T / 2 , where
1
x
sin
Si x d is the Sine integral.
0
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Time-Bandwidth Product
The product of the signal’s duration and its bandwidth is always
a constant.
B T = Bandwidth × Duration = constant
Whatever definition we use for the bandwidth of a signal, the
time-bandwidth product remains constant over certain classes
of pulse signals.
Example 3.3
Pulse Response of Ideal LPF
Consider a time-domain square pulse of width 1 second which
is passed through a filter with a bandwidth of 5Hz (BT=5).
The bandwidth restriction does not cause a substantial change
in the pulse shape.
Time Domain
Frequency Domain
B=5Hz
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Example 3.3 – cont.
Pulse Response of Ideal LPF
Now consider a filter with a bandwidth of 3Hz (BT=3).
The bandwidth restriction still does not cause a substantial
change in the pulse shape.
Ringing occurs near sharp transitions.
B=3Hz
B=2Hz
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Example 3.3 – cont.
Pulse Response of Ideal LPF
Now consider a filter with a bandwidth of 1Hz (BT=1).
The bandwidth restriction now changes the shape considerably.
B=1Hz
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The Ideal Bandpass Filter
The ideal bandpass filter can be written as
Ae j 2 f t0 fL f fH
Hf
0 else
f f0 f f 0 j 2 f t0
A rect rect e
f f
timedelay
magnitude
response
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The Ideal Bandpass Filter – cont.
Ideal BPF is also noncausal.
Realistic bandpass filters will be causal and not have a perfect
frequency response.
fL=7.5Hz, fH=12.5Hz
∆f=5Hz, fo=10Hz
Example 3.4
Noise
A major use of filters is the elimination of noise.
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Example 3.4 – cont.
Noise
Consider a square pulse with duration one second that is
received with the addition of noise.
The ratio of the received desired signal power to the noise
power, signal-to-noise ratio or SNR, is 2.
Desired signal
x(t)
Received signal
r(t)=x(t)+n(t)
Desired signal
Filter
Desired Noise
signal PSD
Received signal
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Example 3.4 – cont.
Noise
We know that increasing the bandwidth to B=3Hz will reduce
the amount of distortion to the original signal.
However, it also lets more noise in.
Desired signal
Filter
Desired Noise
signal PSD Received signal
Summary
An important function in communication systems is to filter the
input signal.
Filtering helps eliminate noise in the system but can also
distort the desired signal.
Increasing the value of the time-bandwidth product BT tends to
reduce the rise time and decay time of the filter pulse response
and helps to preserve the pulse shape.
Increasing the bandwidth of the filter also allows more noise
into the system.
Ideal filters are noncausal.
• Causal filters can approximate the ideal filter using delay.
• More delay can allow better approximation.
Filters are built in the analog domain using resistors, capacitors
and inductors.
They are often implemented digitally using simple tapped-
delay lines in DSP.
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