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Communications Systems, 5 Edition: Solutions Manual For
Communications Systems, 5 Edition: Solutions Manual For
Communications Systems,
5th edition
by
Karl Wiklund, McMaster University,
Hamilton, Canada
Michael Moher, Space-Time DSP
Ottawa, Canada
and
Simon Haykin, McMaster University,
Hamilton, Canada
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
Chapter 2
2.1 (a)
g (t ) = A cos(2 f c t )
fc =
T T
t ,
2 2
1
T
1
sin( fT )
[ ( f fc ) + ( f + fc )] AT
fT
2
Writing out the convolution:
AT sin(T )
G( f ) =
[ ( ( f + f c ) + ( ( f f c ) ] d
2
T
A sin( ( f + f c )T ) sin( ( f f c )T )
+
f + fc
f fc
2
A cos( fT ) cos( fT )
=
1
2 f 1
f+
2T
2T
fc =
1
2T
A cos( fT ) cos( fT )
G( f ) =
exp( j fT )
1
2 f 1
f+
2T
2T
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(c)The half-sine pulse is identical to the half-cosine pulse except for the centre frequency
and time-shift.
fc =
1
2Ta
cos( fTa ) cos( fTa )
+j
j
4 f f c
f + fc
f fc
f + fc
G( f ) =
A
2
4
f fc
f + fc
(d) The spectrum is the same as for (b) except shifted backwards in time and multiplied
by -1.
A cos( fT ) cos( fT )
exp( j fT )
G( f ) =
1
2 f 1
f+
2T
2T
A exp( j 2 fT ) exp( j 2 fT )
=
1
4 f 1
f+
2T
2T
(e) Because the Fourier transform is a linear operation, this is simply the summation of
the results from (b) and (d)
G( f ) =
1
1
4
f
f+
2T
2T
A cos(2 fT ) cos(2 fT )
=
1
2 f 1
f+
2T
2T
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2.2
g (t ) = exp(t ) sin(2 f c t )u(t )
1
( ( f f c ) ( f + f c ) )
1 + j 2 f 2 j
1
1
1
=
2 j 1 + j 2 ( f f c ) 1 + j 2 ( f + f c )
G( f ) =
2.3 (a)
g (t ) = g e (t ) + g o (t )
1
[ g (t ) + g (t )]
2
t
g e (t ) = Arect
2T
g e (t ) =
1
[ g (t ) g (t )]
2
1
1
t 2T
t + 2T
g o (t ) = A rect
rect
T
T
g o (t ) =
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(b)
By the time-scaling property g(-t) R G(-f)
1
[G ( f ) + G ( f ) ]
2
1
= [sinc( fT ) exp( j 2 fT ) + sinc( fT ) exp( j 2 fT ) ]
2
= sinc( fT ) cos( fT )
Ge ( f ) =
1
[G ( f ) G ( f )]
2
1
= [sinc( fT ) exp( j 2 fT ) sinc( fT ) exp( j 2 fT ) ]
2
= jsinc( fT ) sin( fT )
Go ( f ) =
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1
1
j (t )
exp( j 2 Wt )
j 2 t
2
1
sin(2 Wt )
= +j
2 t
t
g (t ) =
2.5
1
+
t
g (t ) =
=
1
1
j (t )
exp( j 2 Wt )
j 2 t
2
t +T
u2
exp
2 du
t T
1
h( )d +
t T
t +T
h( )d
0
dg (t )
1
1
= h(t T ) + h(t + T )
dt
2j
H ( f ) sin(2 f )
But H ( f ) = exp( f 2 2 )
1
exp( f 2 2 ) sin(2 fT )
G( f ) =
f
sin(2 fT )
= exp( f 2 2 )
f
= 2T exp( f 2 2 )sinc(2 fT )
lim G ( f ) = 2Tsinc(2 fT )
0
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.6 (a)
1
[ g (t ) + g (t )]
2
and g (t ) = g * (t ) G ( f ) = G* ( f )
g (t ) =
1
G* ( f ) = [G* ( f ) + G * ( f )]
2
1 *
1
G ( f ) = G* ( f )
2
2
*
G ( f ) = G( f )
G ( f ) is all real
If g(t) is odd and real then
1
[ g (t ) g (t )]
2
and g (t ) = g * (t ) G ( f ) = G* ( f )
g (t ) =
1
G ( f ) = [G ( f ) G ( f )]
2
1
1
G* ( f ) = G* ( f ) G* ( f )
2
2
*
*
G ( f ) = G ( f )
G* ( f ) = G ( f )
G ( f ) must be all imaginary
(b)
( j 2 t )G (t ) R
t G (t ) R
d
g ( f ) by duality
df
j d
g ( f )
2 df
j (n)
t n G (t ) R
g ( f )
2
Replacing g with h
n
j (n)
t n h(t ) R
H (f)
2
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(c)
n
j (n)
Let h(t ) = t g (t ) and H ( f ) =
G (f)
2
n
j (n)
h(t )dt = H (0) = 2 G (0)
(d)
g1 (t ) R G1 ( f )
g 2* (t ) R G2 ( f )
g1 (t ) g 2 (t ) R
G ( )G ( f )d
1
g1 (t ) g 2* (t ) R
G ( )G (( f ))d
1
G ( )G ( f )d
1
(e)
g1 (t ) g 2* (t ) R
G ( )G ( f )d
1
g (t ) g (t )dt R G(0)
1
*
2
g (t ) g (t )dt R G ( )G ( )d
1
*
2
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.7 (a)
g (t ) R ATsinc 2 ( fT )
g (t ) dt = AT
max G ( f ) = G (0)
= ATsinc 2 (0)
= AT
The first bound holds true.
(b)
dg (t )
dt = 2 A
dt
j 2 fG ( f ) = 2 fATsinc 2 ( fT )
= 2 fAT
= 2A
But,
sin( fT ) sin( fT )
fT
fT
sin( fT )
sin( fT )
fT
sin( fT )
sin( fT ) 2 A
fT
j 2 fG ( f ) 2 A
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.7 c)
( j 2 f ) 2 G ( f ) = 4 2 f 2G ( f )
sin 2 ( fT )
= 4 f AT
( fT ) 2
2
4A 2
sin ( fT )
T
4A
T
=
d 2 g (t )
4A
dt =
2
dt
T
( j 2 f ) 2 G ( f )
d 2 g (t )
dt
dt 2
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.8. (a)
g1 (t ) g 2 (t ) R G1 ( f )G2 ( f )
= G2 ( f )G1 ( f ) by the commutative property of multiplication
b)
g1 ( f ) [ g 2 ( f ) g 3 ( f ) ] R G1 ( f ) [G2 ( f )G3 ( f ) ]
Because multiplication is commutative, the order of the multiplication
doesn't matter.
G1 ( f ) [G2 ( f )G3 ( f ) ] = [G1 ( f )G2 ( f ) ] G3 ( f )
G1 ( f ) [G2 ( f )G3 ( f ) ] R [ g1 ( f ) g 2 ( f ) ] g3 ( f )
c)
Taking the Fourier transform gives:
G1 ( f ) [G2 ( f ) + G3 ( f )]
Multiplication is distributive so:
G1 ( f )G2 ( f ) + G2 ( f )G 3 ( f ) R g1 (t ) g 2 (t ) + g1 (t ) g 2 (t )
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.9 a)
Let h(t ) = g1 (t ) g 2 (t )
dh(t )
R j 2 fH ( f )
dt
= j 2 fG1 ( f )G2 ( f )
= ( j 2 fG1 ( f ) ) G2 ( f )
( j 2 fG1 ( f ) ) G2 ( f ) R
dg1 (t )
g 2 (t )
dt
d
dg (t )
[ g1 (t ) g 2 (t )] = 1 g 2 (t )
dt
dt
b)
g (t ) g (t )dt R
1
1
G (0)G2 (0)
( f )
G1 ( f )G2 ( f ) + 1
j 2 f
2
G (0)
=
G1 ( f ) G2 ( f ) + 1 ( f ) G2 ( f )
2
j 2 f
G (0)
=
G1 ( f ) + 1 ( f ) G2 ( f )
2
j 2 f
t
t
g1 (t ) g 2 (t )dt = g1 (t ) g 2 (t )
2.10.
Y( f ) =
X ( ) X ( f )d
X ( ) 0 if W
X ( f ) 0 if f W
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
f 2W
1
t
rect , for which the area under g(t) is
T
T
1
t
rect
T
T
R sinc( fT )
g (t ) = 2Wsinc(2Wt )
f
2Wsinc(2Wt ) R rect
2W
lim 2Wsinc(2Wt ) = (t )
2
lim rect
=1
2W
2.12.
1 1
+ sgn( f )
2 2
By duality:
G( f ) =
1
1
G ( f ) R ( t )
j 2 t
2
j
1
g (t ) = (t ) +
2
2 t
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G( f ) =
1
4 2 f 2
k exp( j 2 ft )
i
A
tb t a
1 A
G( f ) = 2 2
[ exp( j 2 ftb ) exp( j 2 fta ) exp( j 2 fta ) + exp( j 2 ftb ) ]
4 f tb ta
A
= 2 2
[cos(2 ftb ) cos(2 fta )]
2 f ( tb ta )
1
But: sin( f (tb ta )) sin( f (tb + ta )) = [ cos(2 fta ) cos(2 ftb ) ] by a trig identity.
2
A
G( f ) = 2 2
[sin( f (tb ta )) sin( f (tb + ta ))]
f (tb ta )
2.14 a) let g(t) be the half cosine pulse of Fig. P2.1a, and let g(t-t0) be its time-shifted
counterpart in Fig.2.1b
= G ( f )G* ( f )
= G( f )
G( f )
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.14 b)Given that the two energy densities are equal, we only need to prove the result for
one. From before, it was shown that the Fourier transform of the half-cosine pulse was:
AT
1
[sinc(( f + fc )T ) + sinc(( f f c )T )] for fc =
2
2T
After squaring, this becomes:
sin( ( f + f c )T ) sin( ( f f c )T )
A2T 2 sin 2 ( ( f + f c )T ) sin 2 ( ( f f c )T )
+
+
2
4 ( ( f + f c )T ) 2
( ( f f c )T ) 2
2T 2 ( f + f c )( f f c )
sin 2 fT +
2
cos 2 ( fT )
2 cos ( fT )
=
=
2
2
2
2T 2 ( f + f c )
fT +
fT +
2
2
sin 2 fT
cos 2 ( fT )
2
= 2 2
2
2
T ( f fc )
fT
2T 2 f 2 2
4T
1 cos(2 fT )
=
1
2T 2 f 2 2
4T
2 cos 2 ( fT )
=
1
2T 2 f 2 2
4T
2
2
2 2
2
cos ( fT )
A T cos ( fT ) cos ( fT )
+
2
2
2
1
1
4 2T 2
1
1
f+
f
f +
2T
2T
2T
2T
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.14 b)Contd
By rearranging the previous expression, and summing over a common denominator, we
get:
2 2
2
A T cos ( fT )
2
4 2T 2 2
1
f 2
4T
A2T 2
cos 2 ( fT )
= 2 4
4 T 1 1 4T 2 f 2 1 2
(
)
16 T 4
A2T 2 cos 2 ( fT )
= 2
( 4T 2 f 2 1)2
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dg (t )
R j 2 fG ( f )
dt
dg (t )
dt
By Rayleighs theorem:
g (t ) dt =
2
W T
2
t
=
t
=
G( f )
df
g (t ) dt f 2 G ( f ) df
2
g (t ) dt
4 2
( g (t ) dt )
2
2
2
16 2 g (t ) dt
*
t dt ( g (t ) g (t ) ) dt
=
2
16 2 g (t ) g * (t )dt
d
2
2
t
g
(
t
)
dt
=
dt
g (t )
1
16 2
1
WT
4
W 2T 2
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
W 2T 2 =
2
2
2
2
t exp(2 t )dt f exp(2 f )df
exp(2 t
)dt
exp(ax 2 )dx =
1
4a a
t 2 exp(2 t 2 )dt =
1 1
4 2
f 2 exp(2 t 2 )df =
exp(2 t
)=
4
2
2
T W =
for a > 0
1
4
1
2
1
2
1
1
2
1
=
4
1
TW =
4
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.16.
Given:
However, if
h(t ) dt < .
X ( f ) df < and
2
applies to h(t).
Y( f ) = H( f )X ( f )
Y ( f ) df =
2
X ( f )H ( f ) X
( f ) H * ( f )df
X ( f ) H ( f ) df
Y ( f ) df
X ( f ) df
H ( f ) df
<
Y( f )
df <
By Rayleighs theorem:
Y( f )
df = y (t ) dt
2
y (t ) dt <
2
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.17.
The transfer function of the summing block is: H1 ( f ) = [1 exp( j 2 fT ) ] .
The transfer function of the integrator is: H 2 ( f ) =
1
j 2 f
( 2 f )
[1 exp( j 2 fT )]
[1 2 exp( j 2 fT ) + exp( j 4 fT )]
( 2 f )
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.18.a) Using the Laplace transform representation of a single stage, the transfer function
is:
1
H 0 (s) =
1 + RCs
1
=
1+ 0s
H0 ( f ) =
1
1 + j 2 f 0
These units are cascaded, so the transfer function for N stages is:
H ( f ) = ( H ( f ))
1
=
1 + j 2 f 0
T2
4 2 N
1
ln H ( f ) = N ln
1 + j 2 f 0
= N ln (1 + j 2 f 0 )
b) For N, and 02 =
jfT
= N ln 1 +
jfT
let z =
, then for very large N , z < 1
N
We can use the Taylor series expansion of ln(1 + z )
1
m +1
N ln(1 + z ) = N ( 1) z m
m =1 m
m
1
fT
m +1
= N ( 1) j
N
m =1 m
(next page)
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
1
fT
fT
f 2T 2
m +1
lim N ( 1) j
N
j
=
N
2N
N
N
m =1 m
1
= f 2T 2 j N fT
2
1
H ( f ) = exp( f 2T 2 ) exp( j N ft )
2
1
H ( f ) = exp( f 2T 2 )
2
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.19.a) y (t ) =
x( )d
t T
This is the convolution of a rectangular function with x(). The interval of the
rectangular function is [(t-T),T], and the midpoint is T/2.
T
t
rect R Tsinc( fT ), but the function is shifted by .
2
T
H ( f ) = Tsinc( fT ) exp( j fT )
b)BW =
1
1
=
RC T
H( f ) =
T
T
exp( j 2 f )
1 + j 2 RC f
2
1
T
=
1
exp( j fT )
RC
+ j 2 f
RC
T
T
T
1
h(t ) =
exp
(t ) u (t )
2
2
RC
RC
T
T
1
= exp (t ) u (t )
2
2
T
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.20. a) For the sake of convenience, let h(t) be the filter time-shifted so that it is
symmetric about the origin (t = 0).
N 1
2
k =1
k =1
H ( f ) = wk exp( j 2 fk ) +
N 1
2
exp( j 2 fk ) + w0
N 1
2
= 2 wk cos(2 fk )
k =1
G ( f ) = exp ( j f ( N 1) ) 2 wk cos(2 fk )
k =1
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.21 Given an ideal bandpass filter of the type shown in Fig P2.7, we need to find the
response of the filter for x(t ) = A cos(2 f 0t )
1
f fc 1
f + fc
rect
rect
+
2B
2B 2B
2B
1
X ( f ) = [ ( f f 0 ) + ( f f 0 ) ]
2
If f c f 0 is large compared to 2B, then the response is zero in the steady state.
H( f ) =
However:
A
A
A
A
+ ( f f0 ) +
+ ( f + f0 )
x(t )u (t ) R
j 2 ( f + f 0 ) 2
j 2 ( f f 0 ) 2
Since f c f 0 is large, assume that the portion of the amplitude spectrum lying inside the
A
.
4 ( f c f 0 )
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Taking the envelope by retaining the positive frequency components, shifting them to the
origin, and scaling by 2:
A exp j 2 j 2 ft0
if B < f < B
Y ( f )
2 ( f c f 0 )
0
otherwise
y (t ) =
AB
sinc [ 2 B(t t0 )]
j ( f c f 0 )
y (t )
AB
sinc [ 2 B(t t0 ) ] sin(2 f ct )
( fc f0 )
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2.22
H ( f ) = X ( f ) exp( j 2 fT )
A
T
[ ( f fc ) + ( f + f c )] Tsinc( fT ) exp( j 2 f )
2
2
AT
=
[sinc(T (f fc )) + sinc(T (f + fc ))] exp( j fT )
2
N
Let f c =
for N large
T
X(f ) =
Y( f ) = H( f )X ( f )
= X ( f ) exp( j 2 fT ) exp( j fT )
AT
sinc (T ( f f c ) ) + sinc (T ( f + f c ) )
2
A2T 2
sinc (T ( f f c ) ) + sinc (T ( f + f c ) ) sinc ( T ( f f c ) ) + sinc (T ( f + f c ) )
4
A2T 2
= exp( j 2 fT )
[sinc( fT N ) + sinc( fT + N )][sinc( fT N ) + sinc( fT + N )]
4
= exp( j 2 fT )
But sinc(x)=sinc(-x)
Y ( f ) = exp( j 2 fT )
A2T 2
[sinc( fT N ) + sinc( fT + N )]
2
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2.23 G(k)=G
gn =
1
N
N 1
G (k ) exp( j
k =0
2
k n)
N
2
G N 1
exp( j
k n)
N k =0
N
2
2
G N 1
cos( j
k n) + j sin( j
k n)
N k =0
N
N
G N 1
1 = G
N k =0
For n 0 , we are averaging over one full wavelength of a sine or cosine, with regularly
sampled points. These sums must always be zero.
If n = 0, g (n) =
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.24. a) By the duality and frequency-shifting properties, the impulse response of an ideal
low-pass filter is a phase-shifted sinc pulse. The resulting filter is non-causal and
therefore not realizable in practice.
Overshoot (%)
9,98
9.13
9.71
100
Ripple Period
1/5
1/10
1/20
No visible ripple
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2.24 (d)
t
T/100
T/150
T/200
Overshoot (%)
100
16.54
~0
Ripple Period
No visible ripple.
1/100
No visible ripple.
Discussion
Increasing B, which also increases the filters bandwidth, allows for more of the highfrequency components to be accounted for. These high-frequency components are
responsible for producing the sharper edges. However, this accuracy also depends on the
sampling rate being high enough to include the higher frequencies.
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2.25
BT
5
10
20
100
Overshoot (%)
8.73
8.8
9.8
100
Ripple Period
1/5
1/10
1/20
-
The overshoot figures better for the raised cosine pulse that for the square pulse. This is
likely because a somewhat greater percentage of the pulses energy is concentrated at
lower frequencies, and so a greater percentage is within the bandwidth of the filter.
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2.26.b)
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2.26 b)
If B is left fixed, at B=1, and only T is varied, the results are as follows
BT
5
2
1
0.5
0.45
Max. Amplitude
1.194
1.23
1.34
0.612
0.286
As the centre frequency of the square wave increases, so does the bandwidth of the signal
(and its own bandwidth shifts its centre as well). This means that the filter passes less of
the signals energy, since more of it will lie outside of the pass band. This results in
greater overshoot.
However, as the frequency of the pulse train continues to increase, the centre frequency is
no longer in the pass band, and the resulting output will also be attenuated.
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c)
BT
5
2
1
0.5
0.45
Max. Amplitude
1.18
1.20
1.27
0.62
0.042
Extending the length of the filters impulse response has allowed it to better approximate
the ideal filter in that there is less ripple. However, this does not extend the bandwidth of
the filter, so the reduction in overshoot is minimal. The dramatic change in the last entry
(BT=0.45) can be accounted for by the reduction in ripple.
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.27
a)At fs = 4000 and fs = 8000, there is a muffled quality to the signals. This improves
with higher sampling rates. Lower sampling rates throw away more of the signals high
frequencies, which results in a lower quality approximation.
b)Speech suffers from less muffling than do other forms of music. This is because a
greater percentage of the signal energy is concentrated at low frequencies. Musical
instruments create notes that have significant energy in frequencies beyond the human
vocal range. This is particularly true of instruments whose notes have sharp attack times.
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
2.28
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Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
Chapter 3
3.1
s(t) = Ac[1+kam(t)]cos(2fc t)
where m(t) = sin(2fs t)
ka
(sin(2 ( f c + fs)t ) + sin(2 ( f c f s )t )]
2
s(t) is the signal before transmission.
f
106
The filter bandwidth is: BW = c =
= 5714 Hz
Q 175
m(t) lies close to the 3dB bandwidth of the filter, m(t) is therefore attenuated by a factor
of a half.
s (t ) = Ac [cos(2 f c t ) +
m' (t ) = 0.5m(t )
or ka' = 0.5ka
ka' = 0.25
Copyright 2009 John Wiley & Sons, Inc. All Rights Reserved.
3.2 (a)
v
) 1]
VT
Using the Taylor series expansion of exp(x) up to the third order terms, we get:
i = I 0 [exp(
v 1 v 1 v
i = I 0 [ + ]
VT 2 VT 6 VT
(b) v(t ) = 0.01[cos(2 f mt ) + cos(2 f c t )]
Let = 2 t
fc + fm
f fm
, = 2 t c
2
2
4
4
0.023 9
3
=
[ (cos( + ) + cos( )) + (cos( + 3 ) + cos( 3 )
16 2
2
3
1
+ (cos(3 + ) + cos(3 )) + (cos(3 + 3 ) + cos(3 3 ))]
2
2
0.022 9
3
[ (cos(2 f ct ) + cos(2 f mt )) + (cos(2 (2 f c f m )t ) + cos(2 (2 f m ft )t )
16 2
2
3
1
+ (cos(2 (2 f c + f m )t ) + cos(2 (2 f m + f t )t )) + (cos(6 f c t ) + cos(6 f mt ))]
2
2
v 3 (t ) =
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The bandpass filter must be symmetric and centred around fc . It must pass components
at fc+ fm, but reject those at fc+2 fm and higher.
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(d)
Term #
1
2
3
Carrier
0.01
Message
0.0001
-6
2.25 x 10
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Taylor Coef.
-38.46
739.6
-9.48 x 103
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(b)
Let the voltage Vb-Vd be the voltage across the output resistor, with Vb and Vd being the
voltages at each node.
Using the voltage divider rule for condition (a):
Vb = V
Rb
,
R f + Rb
Vd = V
Rf
R f + Rb
Vb Vd =V
Rb R f
R f + Rb
Vb = V
Rf
R f + Rb
Vd = V
Rb
,
R f + Rb
Vb Vd =V
Rb + R f
R f + Rb
The two voltages are of the same magnitude, but are of the opposite sign.
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3.16 (a)
s (t ) =
1
1
a Am Ac cos(2 ( f m + f c )t ) + (1 a ) Am Ac cos(2 ( f m + f c )t )
2
2
Am Ac
[a(cos(2 f ct ) cos(2 f mt ) sin(2 f c t ) sin(2 f mt ))
2
+ (1 a)(cos(2 f c t ) cos(2 f mt ) + sin(2 f c t ) sin(2 f mt ))]
s (t ) =
Am Ac
[cos(2 f c t ) cos(2 f mt ) + (1 2a ) sin(2 f c t ) sin(2 f mt ))]
2
A
m1 (t ) = m cos(2 f mt )
2
A
m2 (t ) = m (1 2a ) sin(2 f mt )
2
s (t ) =
b)Let:
s (t ) =
1
1
Ac m(t ) cos(2 f c t ) + Ac ms (t ) sin(2 f c t )
2
2
2
1
ka ms (t )
1
= Ac 1 + ka m(t ) 1 + 2
1
2
1 + k m(t )
a
2
The second factor in s (t ) is the distortion term d(t). For the example in (a), this
becomes:
2
1
(1 2a ) sin(2 f mt )
d (t ) = 1 + 2
1 + 1 cos(2 f mt )
2
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c)Ideally, d(t) is equal to one. However, the distortion factor increases with decreasing a.
Therefore, the worst case exists when a = 0.
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(a)
1
Ac Ac' cos(2 f c't )[m(t ) cos(2 f c t ) m (t ) cos(2 f c t )]
2
The only lowpass components will be those that are functions of only t and f. Higher
frequency terms will be filtered out, and so can be ignored for the purposes of
determining the output of the detector.
vo (t ) =
vo (t ) =
1
Ac Ac' [m(t ) cos(2 f t ) m (t ) sin(2 f t )] by using basic trig identities.
4
When the upper side-band is transmitted, and f>0, the frequencies are shifted inwards
by f.
Vo ( f ) contains {99.98,199.98,399.98} Hz
(b) When the lower side-band is transmitted, and f>0, then the baseband frequencies are
shifted outwards by f.
Vo ( f ) contains {100.02, 200.02, 400.02} Hz
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f1 = f c f W
3.22.
f 2 = f c + f
v1 (t )v2 (t ) = A1 A2 cos(2 f1t + 1 ) cos(2 f 2t + 2 )
=
A1 A2
[cos(2 ( f1 f 2 )t + 1 2 ) + cos(2 ( f1 + f 2 )t + 1 + 2 )]
2
vo (t ) =
1
A1 A22 [cos(2ft + 1 2 2 ) cos(2 f 2t + 1 2 + 2 )]
2
n+2
n+2
1
= A1 A22 [cos(2 ( f c + 2f ) + 1 2 2 ) + cos(2 f c t + 1 2 + 2 )]
4
n+2
n+2
=
vo (t ) =
+ 2 = 0
n+2 n+2
rearranging and solving for 2 :
2 =
1
n +1
n+2
1 =
n+2
+ 1 = 0
2
n+3
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3.23. Assume that the mixer performs a multiplication of the two signals.
y1 (t ) {1, 2,3, 4,5, 6, 7,8,9} MHz
y2 (t ) {100, 200,300, 400,500, 600, 700,800,900} kHz
This system essentially produces a DSB-SC signal centred around the frequency of y1(t).
The lowest frequencies that can be produced are:
1
yo (t ) = [cos(2 ( f1 f 2 )t ) + cos(2 ( f1 + f 2 )t )]
2
f1 = 1 MHz
f1 f 2 = 0.9 MHz
f 2 = 100 kHz
f1 + f 2 = 1.1 MHz
f1 f 2 = 8.1 MHz
f 2 = 900 kHz
f1 + f 2 = 9.9 MHz
The resolution of the system is the bandwidth of the output signal. Assuming that no
branch can be zeroed, the narrowest resolution occurs with a modulation frequency of
100 kHz. The widest bandwidth occurs when there is a modulation frequency of 900
kHz.
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3.24 Given the presence of the filters, only the baseband signals need to be considered.
All of the other product components can be discarded.
(a) Given the sum of the modulated carrier waves, the individual message signals are
extracted by multiplying the signal with the required carrier.
For m1(t), this results in the conditions:
cos(1 ) + cos( 1 ) = 0
cos( 2 ) + cos( 2 ) = 0
cos( 3 ) + cos( 3 ) = 0
i = i
For the other signals:
m2 (t ) :
cos(1 ) + cos( 1 ) = 0
cos( 2 1 ) + cos( 2 1 ) = 0
cos( 3 1 ) + cos( 3 1 ) = 0
1 = 1
( 2 1 ) = ( 2 1 )
( 3 1 ) = ( 3 1 )
Similarly:
m3 (t ) :
(1 2 ) = ( 1 2 )
( 3 2 ) = ( 3 2 )
m4 (t ) :
(1 3 ) = ( 1 3 )
( 2 3 ) = ( 2 3 )
(b) Given that the maximum bandwidth of mi(t) is W, then the separation between fa and
fb must be | fa- fb|>2W in order to account for the modulated components corresponding to
fa- fb.
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3.25 c)
The high-pass filter applied to the envelope detector eliminates the DC component.
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Carrier Frequency
Modulation Frequency
modulation index
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Vc(k)=Vc(k-1)-0.023*Vc(k-1);
end
end
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Chapter 4 Problems
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Problem 4.7.
s (t ) = Ac cos( (t ))
(t ) = 2 f c t + k p m(t )
Let = 0.3 for m(t) = cos(2fmt).
s (t ) = A c cos(2 f c t + m(t ))
= Ac [cos(2 f c t ) cos( cos(2 f mt )) sin(2 f c t ) sin( cos(2 f mt ))]
for small :
cos( cos(2 f mt )) 1
sin( sin(2 f mt )) cos(2 f mt )
s (t ) = Ac cos(2 f ct ) Ac sin(2 f c t ) cos(2 fmt )
= Ac cos(2 f c t )
Ac
[sin(2 ( f c + f m )t ) + sin(2 ( f c + f m )t )
2
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Problem 4.14.
v2 = av12
s (t ) = Ac cos(2 f c t + sin(2 f mt ))
= Ac cos(2 f c t + m(t ))
v2 = a s 2 (t )
= a cos 2 (2 f c t + m(t ))
=
a
cos(4 f c t + 2 m(t ))
2
The square-law device produces a new FM signal centred at 2fc and with a frequency
deviation of 2. This doubles the frequency deviation.
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The response of |X1(f)| after the resonant peak is the same as for a single pole low-pass
filter. From a table of Bode plots, the following gain response can be obtained:
| X 1 ( f ) |=
1
f fB
1+
| X 1 ( f ) |=
d | X 1 ( f ) |
df
1
f
1+
B
=
f = kB
k
3
2 2
B(1 + k )
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Because the filters are symmetric about the central frequency, the contribution of the
second filter is identical. Adding the filter responses results in the slope at the central
frequency being:
d | X ( f ) |
df
=
f = kB
2k
3
2 2
B(1 + k )
In the original definition of the slope filter, the responses are multiplied by -1, so do this
here. This results in a total slope of:
2k
3
2 2
B(1 + k )
As can be seen from the following plot, the linear approximation is very accurate
between the two resonant peaks. For this plot B = 500, f1=-750, and f2=750.
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Problem 4. 23
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Problem 4.24
The amplitude spectrum corresponding to the Gaussian pulse
p(t ) = c exp c 2t 2 * rect[t / T ]
P ( f ) = F c exp c 2t 2 F rect ( t / T )
= c exp f 2 c 2 Tsinc [ fT ]
where we have used the convolution theorem
Problem 4.25
f 59.7
=
= 1.19 . From the
W
50
BT
= 6 , which implies BT = 6(59.7) = 358.2 kHz.
f
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Problem 4.26.
a)
Beta
1
2
5
10
# of side frequencies
1
2
8
14
b)By experimentation, a modulation index of 2.408, will force the amplitude of the
carrier to be about zero. This corresponds to the first root of J0(), as predicted by the
theory.
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Problem 4.27.
a)Using the original MATLAB script, the rms phase error is 6.15 %
b)Using the plot provided, the rms phase error is 19.83%
Problem 4.28
a)The output of the detected signal is multiplied by -1. This results from the fact that
m(t)=cos(t) is integrated twice. Once to form the transmitted signal and once by the
envelope detector.
In addition, the signal also has a DC offset, which results from the action of the envelope
detector. The change in amplitude is the result of the modulation process and filters used
in detection.
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The above signal has been multiplied by a constant gain factor in order to highlight the
differences with the original message signal.
c)The earliest signs of distortion start to appear above about fm =4.0 kHz. As the
message frequency may no longer lie wholly within the bandwidth of either the
differentiator or the low-pass filter. This results in the potential loss of high-frequency
message components.
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4.29. By tracing the individual steps of the MATLAB algorithm, it can be seen that the
resulting sequence is the same as for the 2nd order PLL.
e(t ) is the phase error e (t ) in the theoretical model.
The theoretical model of the VCO is:
t
2 (t ) = 2 kv v(t )dt
0
Integrator +input
b)For smaller kv, the lock-in time is longer, but the output amplitude is greater.
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Chapter 5 Problems
5.1.
( x x )2
exp(
)
(a) Given f ( x) =
2 x2
2 x2
1
and exp( t 2 ) R exp( f 2 ) , then by applying the time-shifting and scaling properties:
F( f ) =
1
2
2
x
2 x2 exp( ( 2 x2 ) 2 f 2 ) exp( j 2 f x )
= exp( 2 2 x2 f 2 + j x 2 f )
1
= exp( j x 2 x2 )
2
and let = 2 f
(b)The value of x does not affect the moment, as its influence is removed.
Use the Taylor series approximation of x(x), given x = 0.
1
2
x ( ) = exp( 2 x2 )
x2
exp( x) =
n=0 n !
E[ X n ] =
d nx ( )
d n v =0
2k 2k
1
x ( ) = x
k!
2
k =0
k
d nx ( )
leaves the lowest non-zero derivative as 2k-n.
d n
When this derivative is evaluated for v=0, then E[ X n ] =0.
For even values of n, only the terms in the resulting derivative that correspond to 2k-n =
0 are non-zero. In other words, only the even terms in the sum that correspond to k = n/2
are retained.
E[ X n ] =
n!
x2
(n / 2)!
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5.2. (a) All the inputs for x 0 are mapped to y = 0. However, the probability that x > 0
is unchanged. Therefore the probability density of x 0 must be concentrated at y=0.
Because fy(y) is a
f y ( y )dy = 0.5
Therefore, the integral over the delta function must be 0.5. This means that the factor k
must also be 0.5.
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5.3 (a)
p y ( y ) = p y ( y | x0 ) P( x0 ) + p y ( y | x1 ) P( x1 )
Assume: P( x0 ) = P( x1 ) = 0.5
1
p y ( y ) = [ p y ( y | x0 ) + p y ( y | y1 )
2
1
( y + 1) 2
( y 1) 2
py ( y) =
[exp(
)
exp(
)]
+
2 2
2 2
2 2 2
(b) P ( y ) = p y ( y )dy
, 2 ( y ) =
( y )2
exp(
)dy
2 2
2 2
1
1
P( y ) = [ 1, 2 ( ) + 1, 2 ( )]
2
1
y
)]
But, , 2 ( y ) = [1 + erf (
2
2
1
+ 1
1
P( y ) = [2 + erf
+ erf
]
2
2
2
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Problem 5.4
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Problem 5.5
= E [ Z *( s + ) Z ( s ) ]
= E [ Z ( s ) Z ( s + )] *
= RZ* ( )
where we have used the change of variable s = t - .
(iii) Taking an approach similar to that of Eq. (5.67)
2
0 E ( Z (t ) Z (t + ) )
= E ( Z (t ) Z (t + ) )( Z *(t ) Z *(t + ) )
2
= 2E Z (t ) 2 Re {E [ Z *(t ) Z (t + ) ]}
= 2 RZ (0) 2 Re { RZ ( )}
Thus Re { RZ ( )} RZ (0) .
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E[ Z (t1 ) Z * (t2 )]
= E[( A cos(2 f1t1 + 1 ) + jA cos(2 f 2t1 + 2 )) ( A cos(2 f1t2 + 1 ) + jA cos(2 f 2t2 + 2 ))]
Let 1=2f1 2=2f2
After distributing the terms, consider the first term:
A2 E[cos(1t1 + 1 ) cos(1t2 + 1 )]
=
A2
E[cos(1 (t1 t2 )) + cos(1 (t1 + t2 ) + 21 )]
2
The expectation over 1 goes to zero, because 1 is distributed uniformly over [-,].
This result also applies to the term A2 [cos(2t1 + 2 ) cos(2t2 + 2 )] . Both cross-terms go
to zero.
R(t1 , t2 ) =
A2
[cos(1 (t 1 t2 )) + cos(2 (t1 t2 ))]
2
A2
[cos(1 (t1 t2 )) + cos(2 (t1 t2 )) + j cos(2t1 1t2 ) j cos(1t1 2t2 )]
2
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Problem 5.7
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Problem 5.8
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Problem 5.9
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Problem 5.10
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Problem 5.11
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Problem 5.12
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Problem 5.13
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Problem 5.14
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Problem 5.15
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Problem 5.16
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Problem 5.17
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Problem 5.18
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Problem 5.19
Problem 5.20
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Problem 5.21
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Problem 5.22
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Problem 5.23
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Problem 5.24
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Problem 5.25
Problem 5.26
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Problem 5.27
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Problem 5.28
( f )
1 + (2 f ) df <
2
ln S x ( f )
1 + (2 f )2 df <
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Problem 5.29
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Problem 5.30
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Problem 5.31
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Problem 5.32
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Problem 5.33
(a) The receiver position is given by x(t) = x0+vt Thus the signal observed by the
receiver is
x
r (t , x) = A( x) cos 2 f c t
c
x + vt
= A( x) cos t 2 f c t 0
c
f v
x
= A( x) cos 2 f c c t f c 0
c
c
The Doppler shift of the frequency observed at the receiver is f D =
fc v
.
c
1
2
1
2
exp ( j 2 f cos ) d
D
exp ( j 2 f sin ) d
D
= J 0 ( 2 f D )
where the second line comes from the symmetry of cos and sin under a
-/2 translation.
Eq. (5.174) follows directly from this upon noting that, since the expectation result is
real-valued, the right-hand side of Eq.(5.173) is equal to its conjugate.
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Problem 5.34
The histogram has been plotted for 100 bins. Larger numbers of bins result in larger
errors, as the effects of averaging are reduced.
Distance
0
1
2
3
4
Relative Error
0.94%
2.6 %
4.8 %
47.4%
60.7%
The error increases further out from the centre. It is also important to note that the
random numbers generated by this MATLAB procedure can never be greater than 5.
This is very different from the Gaussian distribution, for which there is a non-zero
probability for any real number.
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y = 0.0343 + j 0.0493
y2 = 5.597
The theoretical values are: y = 0 (by inspection).
The theoretical value of y2 =5.56. See 5.35 (c) for the calculation.
5.35 (b)
From the plots, it can be seen that both the real and imaginary components are
approximately Gaussian. In addition, from statistics, the sum of tow zero-mean Gaussian
signals is also Gaussian distributed. As a result, the filter output must also be Gaussian.
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5.35 (c)
y (n) = ay (n 1) + w(n)
Y ( z ) = aY ( z ) z 1
H ( z) =
1
n
R h( n) = a u ( n)
1
1 az
1
(1 az )(1 az )
-1
Rh(z) = H(z)H(z ) =
a
z 1
1
1
+
2
1
2
1 a 1 az
1 a 1 az
w2
1 a2
an
< n <
From the plots, the measured and observed autocorrelations are almost identical.
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Chapter 6 Solutions
Problem 6.3
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Problem 6.4
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Problem 6.5
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Problem 6.6
Problem 6.7
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Problem 6.8
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Problem 6.9
Problem 6.10
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Problem 6.11
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Problem 6.12
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Problem 6.13
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Problem 6.24
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Problem 6.15
Problem 6.16
Problem 6.17
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Chapter 7 Problems
Problem 7.1
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Problem 7.2
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Problem 7.3
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Problem 7.4
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Problem 7.5
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Problem 7.6
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Problem 7.7
Problem 7.8
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Problem 7.9
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Problem 7.10
Problem 7.11
Problem 7.12
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Problem 7.13
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Problem 7.14
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Problem 7.15
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Problem 7.16
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Problem 7.17
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Problem 7.18
Problem 7.19
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Problem 7.20
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Problem 7.21
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Problem 7.22
The maximum slope of the signal s (t ) = A sin ( 2 ft ) is 2fA. Consequently, the
Problem 7.23
(a) Theoretically, the sampled spectrum is given by
Ss ( f ) =
H ( f nf )
n =
where Hs(f) is the spectrum of the signal H(f) limited to f f s / 2 . For this
example, the sample spectrum should look as below.
-5 kHz
5 kHz
(b)
The sampled spectrum is given by
2.5
x 10
Amplitude Spectrum
1.5
0.5
0
-5
-4
-3
-2
-1
0
1
Frequency (kHz)
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(ii)
x 10
Amplitude Spectrum
1.5
0.5
0
-6
-4
-2
0
Frequency (kHz)
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Problem 7.23
(a) The expanding portion of the -law compander is given by
exp log(1 + ) 1
m=
(1 + ) exp 1
(b)
(i) For the non-companded case, the rms quantization error is determined by step size.
The step size is given by the maximum range over the number of quantization steps
2A
= Q
2
For this signal the range is from +10 to -1, so A = 10 and with Q = 8, we have = 0.078.
From Eq. ( ) , the rms quantization error is then given by
1 2 2 R
2
Q2 = mmax
3
1
= (10) 2 216
3
= 0.0005086
and the rms error is Q 0.02255.
(ii) For a fair comparison, the signal must have similar amplitudes.
The rms error with companding is 0.0037 which is significantly less. The plot is shown
below. Note that the error is always positive.
0.035
0.03
0.025
0.02
0.015
0.01
0.005
0
-0.005
50
100
150
200
250
300
350
400
450
Rest TBD.
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Problem 7.24
Problem 7.25
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Chapter 8
Problem 8.1
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Problem 8.2
Problem 8.3
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Problem 8.4
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Problem 8.5
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Problem 8.6
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Problem 8.7
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Problem 8.8
Problem 8.9
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Problem 8.10
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Problem 8.11
Problem 8.12
.
Problem 8.13
Problem 8.14
Problem 8.15
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Problem 8.16
Problem 8.17
Problem 8.18
Problem 8.19
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Problem 8.20
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Problem 8.21
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Problem 8.22
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Problem 8.23
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Problem 8.24
Problem 8.25
Problem 8.26
Problem 8.29
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Chapter 9
Problem 9.1
The three waveforms are shown below for the sequence 0011011001. (b) is ASK, (c) is
PSK; and (d) is FSK.
Problem 9.2
The bandpass signal is given by
s (t ) = g (t ) cos ( 2 f c t )
The corresponding amplitude spectrum, using the multiplication theorem for Fourier
transforms, is given by
S ( f ) = G ( f ) * [ ( f f c ) + ( f + f c ) ]
= G( f fc ) + G( f + fc )
To be done
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Problem 9.4
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Problem 9.5
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Problem 9.6
**The problem here is solved as erfc here and in the old edition, but listed in the
textbook question as Q(x).
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Problem 9.7
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Problem 9.8
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Problem 9.9
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Problem 9.10
Problem 9.11
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Problem 9.12
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Problem 9.13
Problem 9.14
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Problem 9.15
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Problem 9.16
Problem 9.17
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Problem 9.18
Problem 9.19
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Problem 9.20
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Problem 9.21
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Problem 9.22
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Problem 9.23
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Chapter 10 Problems
Problem 10.1
Problem 10.2
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Problem 10.3
Problem 10.4
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Problem 10.5
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Problem 10.6
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Problem 10.7
a)
b)To be done
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Problem 10.8
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Problem 10.9
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Problem 10.10
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Problem 10.11
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Problem 10.12
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Problem 10.13
Problem 10.14
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Problem 10.15
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Problem 10.16
Problem 10.17
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Problem 10.18
Problem 10.19
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Problem 10.20
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Problem 10.21
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Problem 10.22
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Problem 10.23
Problem 10.24
Problem 10.25
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Problem 10.26
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Problem 10.27
Problem 10.28
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Problem 10.29
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Problem 10.30
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