Dept.

of Electrical Engineering
University of Toronto
Communication Systems
ECE316
Problems 1 - Solutions

2 + 2j
1. Simplify the expression -------------- + 5j and put it in both polar and rectangular form.
1 – 2j

2 31 1 π
Rectangular: – --- + ------ j , Polar: --- 965e jθ where θ = --- + atan 31
------ .
5 5 5 2 2

1
--- – 2j
2
Redo for -------------- ( 3 – j ) .
–j

2. Consider the expression x ( t ) = e j4πt + cos ( 4πt ) . This is a complex function of time. That
is, for each value of time t, x ( t ) is a complex number. Now determine the real part of this function,
the imaginary part and the magnitude, as functions of time.

Real: x r ( t ) = cos ( 4πt ) + cos ( 4πt ) = 2 cos ( 4πt )

Imagenary: x i ( t ) = sin ( 4πt )

e 2πt – sin ( 8πt )

Redo for: x ( t ) = ------------------------------------ .
1 + e – 6πt

π
3. Consider the signal x ( t ) = 2 cos ( 3πt ) cos  5πt + --- . Write this signal as a summation of two
 3
sinusoidal signals.

π π
cos  8πt + --- + cos  2πt + --- .
3 3

π π
Redo for: x ( t ) = 3 cos  2πt – --- sin  3πt + --- .
4 8

π
4. Consider the signal x ( t ) = cos ( 8πt ) + 2 sin  8πt + --- . Write this signal as a single sinusoidal
6
signal. Give the amplitude, frequency, and phase for the resultant signal.

π π
– j --- – j ---
x ( t ) = A cos ( 8πt + θ ) , where A = 1 + 2e 3 andθ = arg  1 + 2e 3 .
 π
Redo for: x ( t ) = – cos  10πt – --- + 10 sin ( 10πt ) .
 2

5. Consider the following signal x ( t ) defined as

1
x ( t ) = 1 – 2t if t < --- , 2 if 1 < t < 3 , and 0 elsewhere.
2

For each of the following plot y ( t ) .

a) y ( t ) = x ( t ) .

3
2
1
x ( t) 0
−1
−2
−3
−5 −4 −3 −2 −1 0 1 2 3 4 5
t

b) y( t) = x(t + 3) .

3
2
1
x ( t+3) 0
−1
−2
−3
−5 −4 −3 −2 −1 0 1 2 3 4 5
t

c) y( t) = x(t – 1 ) .
 3
3
2
1
x ( t−1) 0
−1
−2
−3
−3
−5 −4 −3 −2 −1 0 1 2 3 4 5
−5 t 5

d) y ( t ) = x ( 2t )

3
2
1
x ( 2 ⋅t) 0
−1
−2
−3
−5 −4 −3 −2 −1 0 1 2 3 4 5
t

e) y ( t ) = x ( – t )

3
2
1
x ( − t) 0
−1
−2
−3
−5 −4 −3 −2 −1 0 1 2 3 4 5
t

f) y ( t ) = x ( 3t + 2 )
 3
2
1
x ( 3 ⋅t+2) 0
−1
−2
−3
−5 −4 −3 −2 −1 0 1 2 3 4 5
t

g) y ( t ) = x 2 ( t )

6
5
4
3
2 2
x ( t) 1
0
−1
−2
−3
−5 −4 −3 −2 −1 0 1 2 3 4 5
t

h) y ( t ) = x' ( t )

1
2 δ t + --- – 2Π ( t ) + 2 δ( t – 1 ) – 2 δ( t – 3 )
 2
t

i) y ( t ) =  x ( τ ) dτ .
–∞

t
Redo for: x ( t ) =  1 – ---- u ( t + 1 ) if t < 3 and 0 elsewhere.
2

6. The following plot shows a sinusoidal signal displayed in a scope. Each division in the hori-
zontal axis corresponds to 0.125 ms, and each division in the vertical axis corresponds to 1.2 volts.
Give an equation to specify the signal. The equation should be in terms of a cosine.

x ( t ) = 6 cos ( 2πf 0 t + θ ) , where f 0 = 640 Hz, θ = 1 rad

Redo assuming that the graph is pushed so that the origin is at the point (2,2), i.e. two divisions to
the right and two divisions up.
 ∞

7. Consider the signal x ( t ) =  Λ ( t – 2k ) , where Λ ( t ) is defined as follows:

k = –∞

t
1 1---
– ---
2 2

a) Plot the signal.

1
y ( t)
0

−1
− 10 −5 0 5 10
t

b) Find the average power of the signal.

 1
P ave = --- .
6

c) Find the d.c. component of the signal.

1
dc value = ---
4

d) What is the period of the signal?

Period = 2

e) What is the fundamental frequency of the signal?

1
f 0 = --- Hz
2

f) Find the component with fundamental frequency (also called the first harmonic) of the signal.

α cos ( 2πf 0 t ) , where , α = 0.203 .

g) What is the average power in the above fundamental component?

1--- 2
α
2

Redo: Repeat for x ( t ) =  Λ ( t – 3k )

k = –∞

8. Now define y ( t ) = x ( 2t ) . Give answers to all the questions in 7. with x ( t ) replaced by y ( t ) .

a)

1
y ( t)
0

−1
− 10 − 9 − 8 − 7 − 6 − 5 − 4 − 3 − 2 − 1 0 1 2 3 4 5 6 7 8 9 10
t
b) Average power is the same.

c) DC value is the same.

d) Period = 1.

e)

fundamental freq. = 1.

f) Amplitude of 1st harmonic is the same. But the frequency is obviously twice.

g) Average power is the same as before.

Redo the above for new x ( t ) .

9. Consider the signal x ( t ) = cos3 ( 4t ) . What is the fundamental frequency in rad/s? What is the
fundamental frequency in Hz? Find the Fourier series with complex exponentials.

2
ω 0 = 4 , f 0 = --- .
π

1
x ( t ) = ----3- ( e j 6π f 0 t + 3e j 2π f 0 t + 3e –j 2π f 0 t + e –j 6π f 0 t ) .
2

1 3 1 1
X – 3 = --- , X – 1 = --- , X 1 = --- , and X 3 = --- .
8 8 8 8

Redo for x ( t ) = cos4 ( 3t )

1
10. Consider the signal z ( t ) = Λ ( t ) – Λ  t – --- , where Λ ( t ) is defined in 7. above.
2

a) Plot the signal z ( t ) .

 1

0.5

z( t) 0

− 0.5

−1
−4 −3 −2 −1 0 1 2 3 4
t

b) Find the energy of the signal z ( t ) .

1
Energy = --- .
2

c) Find the Fourier transform of z ( t ) using a direct computation method.

Break into three integrals and evaluate the integrals.

d) Find the Fourier transform of the signal using the Fourier transform properties.

1 f 2
Z ( f ) = G ( f ) – e –j π f G ( f ) = G ( f ) ( 1 – e –j π f ) , where G ( f ) = ---  sinc  ---  .
2  2 

Redo for z ( t ) = sgn ( t ) ( u ( t + 1 )u ( 1 – t ) )

11. Consider the random signal defined as follows:

x( t) =  ak p ( t – 2k ) ,
k=0

t
where p ( t ) is a triangular pulse p ( t ) = 1 – ---- for t ≤ 2 , and 0 elsewhere, and the coefficients
2
a k are chosen at random as follows: For each a k flip a coin 2 times to get heads (H) or tails (T).
For TT set a k = – 3 , for TH a k = – 1 , for HT a k = 1 , and for HH a k = 3 . Plot the signal x ( t ) .

I used a function to generate random numbers. The function rnd(x) generates a random number
with uniform distribution from 0 to x . Let x = 4 . Considter y = floor ( rnd ( x ) ) . Then y is a ran-
dom variable with equal probability (1/4) in the set { 0, 1, 2, 3 } . Now take
a k = 2y – 3 = 2floor ( rnd ( 4 ) ) – 3 .
Here is the a k

T
a = 0 1 2 3 4 5 6 7 8 9
0 1 -1 1 1 -3 3 1 -1 3 1

x( t)
−2 0 2 4 6 8 10 12 14 16 18 20 22
−2

−4

12. Consider a rotating pointer (e.g. a magnetic needle). The length of the need is 2 cm, i.e. from
the pivot point, located at the origin, to the tip. At t = 0 the needle makes an angle of 30 degrees
with respect to the x-axis. Represent the points in the plane as complex numbers. Give an equa-
tion for the tip of the needle as a function of time if the needle rotates clockwise at a speed of 10
m/s. Then give the equation under the same conditions if the needle rotates counterclockwise.

The pointer will trace a circle of radius r = 0.02 m. For an angle θ the tip of the pointer is at
c = ( r cos θ, r sin θ ) . The length of the arc traced by the pointer is l = rθ . The speed of rotation
dl – dθ – 10 – 10
is s = ----- = r --------- = 10 . θ ( t ) = --------- t = ---------- t = – 500 t + C . Taking the initial condition we
dt dt r 0.02
π
have θ ( t ) = – 500 t + --- .
6

Hence the equation is

π π π π
( x ( t ), y ( t ) ) =  r cos  – 500t + --- , r sin  – 500t + ---  = r  cos  500t – --- , – sin  500t – ---  .
  6  6    6  6 

π
– j  500 t – ---
6
Written as a complex number we have z ( t ) = re

π
j  500 t – ---
 6
For counter-clockwise the result is z ( t ) = re

13. For the counterclockwise case of 12. give an expression for the length of the projection of the
needle on the x-axis, and give a second expression for the projection of the needle on the y-axis.
 π π
The projections are r cos  500t – --- and r sin  500t – --- .
6 6
 Dept. of Electrical Engineering
University of Toronto
Communication Systems
ECE316
Problems 2

1. Find the Fourier transform of the following signal. Simplify the expression as much as
possible.

1
t
3 5 10
–1

We view the above as three shifted rectangular pulses.

Pulse 1 - width = 3, height = 1, shift left by 1.5

Pulse 2 - width = 2, height = 2, shift left by 4

Pulse 3 - width = 5, height = -1, shift left by 7.5

Let W = width, A = height.

sin ( πfW )
AW -----------------------
πfW

X ( f ) = 3e –j3πf sinc ( 3f ) + 4e –j8πf sinc ( 2f ) – 5e – j15πf sinc ( 5f )

2. Find the energy spectral density for the above signal.

ψ ( f ) = X ( f ) 2 = 9sinc 2 ( 3f ) + 16sinc 2 ( 2f ) + 25sinc 2 ( 5f ) + 24 cos ( 5πf ) sinc ( 3f )sinc ( 2f )

– 30 cos ( 12πf ) sinc ( 3f )sinc ( 5f ) – 40 cos ( 7πf ) sinc ( 2f )sinc ( 5f )

80

60

( X( f ) ) 240
20

0
0 0.5 1 1.5 2
f
3. Consider the output of a half-wave rectified circuit when the input is x ( t ) = cos ( 120πt ) .
Find the Fourier series for this signal. Give an expression for the Fourier transform of the
signal.

This is a sinusoid with frequency = 60 Hz and zero phase.

The following is the half-wave rectified signal, with the period normalized to 1.

y ( t ) 0.5

0
−2 −1 0 1 2
t

Use the real Fourier series. Let the output be y ( t )

∞ ∞

y ( t ) = a0 +  ak cos ( 2πkf0 t ) +  bk sin ( 2πkf0 t )

k=0 k=0

1
f 0 = 60 T 0 = ----
f0

b k = 0 because y ( t ) is even

T T
----0- ----0- T0
π
sin  ---
2 4 -----
4
1 1 2 sin ( 2πf 0 t ) 2  2 1
a 0 = -----
T0  y ( t ) dt = -----
T0  cos ( 2πf 0 t ) dt = ----- -------------------------
T 0 2πf 0
= ----- ---------------- = ---
T 0 2πf 0 π
T T 0
– ----0- – ----0-
2 4

T0 T0 T0
------ ------ ------
2 4 4
2 2 2
ak = ------
T0  y ( t ) cos ( 2πkf 0 t ) dt = ------
T0  cos ( 2πf 0 t ) cos ( 2πkf 0 t ) dt = ------
T0  ( cos ( 2π( k + 1 )f0t ) + cos ( 2π( k – 1 )f0t ) ) dt
T0 T0 0
– ------ – ------
2 4

Now treat the case k = 1 seperately

 T0
------
4
2 1
a ------
1 = T
0
 ( cos ( 4πf0 t ) + 1 )dt = --2-
0

T0
------
 sin  2π (k + 1 )) 2π ( k – 1 )
sin ( 2π ( k + 1 )f t ) sin ( 2π ( k – 1 )f t )
4 -------------------------- sin  ----------------------- 
2 0 0   4   4 
a k = ------  -------------------------------------------- + ------------------------------------------- = 2  ---------------------------------------- + ------------------------------------- for k ≥ 2
T  2π ( k + 1 )f 2π ( k – 1 )f  2π ( k + 1 ) 2π ( k – 1 ) 
0 0 0 
0  

 sin  π (k + 1 ))
----------------------- sin  π ( k – 1 )- 
-------------------  cos  kπ ------ cos  ------ 
kπ
  2   2  1   2   2 
ak = --1-  ------------------------------------- + --------------------------------- = ---  -------------------- – -------------------- for k ≥ 2 .
π k+1 k–1  π k + 1 k–1 
   

cos  ------
kπ
1 1 1 2 2
a k = --- cos  ------  ------------ – ----------- = – --- --------------------
kπ
for k ≥ 2 .
π  2   k + 1 k – 1 π k2 – 1

Note that for k odd cos  ------ is zero, and for k even cos  ------ = ( – 1 ) k / 2 .
kπ kπ
 2  2
Hence

cos  ------
∞
kπ
1 1 2 2
y ( t ) = --- + --- cos ( 2πt ) – ---
π 2 π  -------------------- cos ( 2πkf 0 t ) with sum being over k even.
2
k –1
k=2
Now change the index of summation to go over all k, i.e. replace k by 2k.
2kπ
∞ cos  ---------
1 2  2 
y ( t ) = --- – ---
π π  ------------------------ cos ( 2π ( 2k )f 0 t )
( 2k ) 2 – 1
k=1
∞
1 1 2 ( –1 ) k
y ( t ) = --- + --- cos ( 2πf 0 t ) – ---
π 2 π  ----------------- cos ( 4πkf 0 t )
4k 2 – 1
k=1

The Fourier transform is

∞
1 1 2 ( –1 ) k
Y ( f ) = --- δ( f ) + --- ( δ( f – f 0 ) + δ( f + f 0 ) ) – ---
π 4 π  ----------------- ( δ( f – 2kf 0 ) + δ( f + 2kf 0 ) )
4k 2 – 1
k=1
4. Consider the square wave with 50% duty cycle as follows:

1
... ...

1--- t
2

You can compute the average power of this signal using two methods: 1) the easy method (by in-
spection) and the hard method 2) Parseval’s theorem. Use he 2nd approach to find an infinite se-
ries to compute π .

T
----0-
2 ∞

Parseval theorem is  x ( t ) 2 dt =  X k 2 . In the time domain we get 1/2, by inspection.

T k = –∞
– ----0-
2

The above is for the complex Fourier series. For the real Fourier series, since x ( t ) is even, we
T
----0-
∞ 2
2
have x ( t ) = a 0 +  ak cos ( 2πkf0 t ) , where ak = -----
T0  x ( t ) cos ( 2πkf 0 t ) for k > 0 and the
k=1 T
– ----0-
2
∞
1
average power is P = a 02 + ---
2  ak2 . Note that f 0 = 1 .
k=1

1
Finding the Fourier series we have a 0 = --- , and
2

sin  ------
kπ
2 2
a k = --- ------------------- for k > 0 .
π k

sin  ------
kπ
2  2
a k = --- ------------------- for k odd, 0 for k even.
π k

2 2
 sin  -----
∞
kπ-  ∞  sin  kπ ------ 
1 2 1  2  2  1 2   2 
x ( t ) =  --- + ---   --- ------------------- = --- + -----2   ------------------- (sum over k odd)
 2 2 π k  4 π  k 
k = 1  k = 1 
Summing the above over k odd is the same as summing over all k after replacing k with 2k – 1 .

Hence we have

2 2
 sin  ( 2k – 1 ) π
∞ ---  ∞ sin  kπ – π ---  ∞
1 2   2  1 2   2  1 2 cos ( kπ )- 2
P = --- + -----2   ------------------------------------- = --- + -----2   -----------------------------  –----------------------
4 π  2k – 1  4 π  2k – 1 
= --- + -----2
4 π   2k – 1 
k = 1  k = 1  k=1

∞
1 2 1 ( –1 )  k 2
 –-------------------
= --- + -----2
4 π   2k – 1 
k=1

∞
1 2 1
= --- + -----2
4 π  ---------------------
( 2k – 1 ) 2
-
k=1

∞
1 2 1 1
Now equating the above to --- (computed in the time domain), we have -----2
2 π  ---------------------
( 2k – 1 ) 2
- = --- .
4
k=1
Hence solving for π we have

∞ ∞
1 - , or π = 1
π2 = 8  ---------------------
( 2k – 1 ) 2  ---------------------
( 2k – 1 ) 2
-
k=1 k=1

∞
sin4 ( x )
5. Use Parseval’s theorem to evaluate the following integral:  ----------------
x4
- dx .
–∞
sin4 ( x ) sin2 ( x ) 2
- =  -----------------
We write ----------------
x4  x2 

consider x to be time domain, t . Hence we can use the Rayleigh energy theory.
∞ ∞

 x ( t ) 2 dt =  X ( f ) 2 df . Find the Fourier transform of x ( t ) .

–∞ –∞
sin ( t )
First find the F.T. of -------------- and then convolve. The F.T. of this sinc is
t
π

f
1-
-----
2π
Convolve the above to get

1 f
---
π

Now square this function and integrate.

--1- --1-
π π
f 3 2π
2  ( π 2 f ) 2 dt = 2π 4 ---- = ------
3 0
3
0

6. The signal x ( t ) = rect  --- is input to a linear system with impulse response h ( t ) = 2e –2t .
t
 T
a) Find the output of the system, as a time function, using a direct time domain computa-
tion approach.

The impulse response should be h ( t ) = 2e – 2t u ( t ) . Flip h ( t ) , slide it back and do the rest ...

1
Note that rect ( t ) = 1 for t < --- and 0 elsewhere.
2

T
x ( t )*h ( t ) = 0 for t < – ---
2

T T
= 1 – e –( 2t + T ) for – --- < t < ---
2 2

T
= e –2t ( e T – e – T ) for t > ---
2

b) Find the output of the system, as a time function, using a frequency domain approach.

The Fourier transform of the input is X ( f ) = T sin c ( fT ) and the transfer function is
2 2T sin c ( fT )
H ( f ) = -------------------- . Hence the Fourier transform of the output is Y ( f ) = ---------------------------- .
2 + j2πf 2 + j2πf

Now we have to find the inverse Fourier transform of Y ( t ) . We can write it as

sin ( πfT ) 1 1 1 1 1
Y ( f ) = 2T ---------------------------------- = ---  ----------------------- ----- ( e jπfT – e –jπfT ) = -------- ----------------------- ( e jπfT – e –jπfT )
πfT ( 2 + j2πf ) 
π f ( 1 + jπf ) 2j  j2π f ( 1 + jπf )
 1 1 jπ
Now, we can use a partial fraction expansion to write ----------------------- = --- – --------------------- , hence the
f ( 1 + jπf ) f ( 1 + jπf )
above becomes
1 1 e jπfT e jπfT e – jπfT e – jπfT
Y ( f ) =  ---------- – ------------------------ ( e jπfT – e –jπfT ) = ----------- – ------------------------ – ------------- + ------------------------
 j2πf 2 ( 1 + jπf ) j2πf ( 1 + jπf )2 j2πf ( 1 + jπf )2

Now we invert each of the above terms using the shifting property

1
note that for the step function we have u ( t ) ⇔ ---------- .
j2πf

1 e jπfT
Hence ---------- ⇔ u ( t ) and ----------- ⇔ u  t + --- (using the shifting property)
T
j2πf j2πf  2

– 2  t + --- T
– e jπfT 2 
u t + ---
T
Similarly we have ------------------------ ⇔ – e
( 1 + jπf )2  2

– e –jπfT
and ---------------- ⇔ – u  t – ---
T
j2πf  2

 T
– 2  t – ---
e –jπfT 2 
u  t – ---
T
and ---------------- ⇔ e
1 + jπf 2

Combining these we get the final result

y ( t ) = u  t + --- – u  t – --- + e – 2t  e T u  t – --- – e –T u  t + ---  . This is thge same as the direct

T T T T
2 2 2 2
computation although it is written differently using branches.

c) Find the energy spectral density of the input signal.

ψ x ( f ) = X ( f ) 2 = T 2 ( sinc ( fT ) ) 2 .

d) Find the energy spectral density of the output signal.

4T 2 ( sinc ( fT ) ) 2 2 ( sinc ( fT ) ) 2
- = T
ψ y ( f ) = Y ( f ) 2 = H ( f )X ( f ) 2 = ------------------------------------ ---------------------------------
4 + ( 2πf ) 2 1 + π2 f2

7. Consider the following second order filter:

 R
C
L

where R = 10 K Ω , C = 10 pF, and L = 1 mH. Assume that a signal source driving the filter
has an output impedance equal to zero. This means that when we connect the filter it does not
affect the amplitude of the input signal, so we can treat the filter as a linear system represented by
a black box.

a) Find the transfer function for the filter. Plot the amplitude response.

If the input source has zero output impedance then we can just use the voltage divider equation.

L-
---
C
-------------------------------
1
( jωL )  ---------- 1 jωL
( jωL ) + ---------- -----------------------
 jωC jωC 1 – ω 2 LC jωL
H ( ω ) = --------------------------------------------- = ------------------------------------------ = ---------------------------------- = -------------------------------------------
2 RLC
-
 1  L
----
jωL
R + ---------------------- - R + jωL – ω
R + ( jωL )  ---------- 2 LC
jωC R+
C
------------------------------- 1 – ω
1
( jωL ) + ----------
jωC

jω ( 10 – 3 ) jω ( 10 – 3 )
= ---------------------------------------------------------------------------------------
- = -------------------------------------------------------------------
-
10 4 + jω ( 10 –3 ) – ω 2 ( 10 4 10 –3 10 – 11 ) 10 4 + jω ( 10 – 3 ) – ω 2 ( 10 –10 )

jω
= -------------------------------------------------
10 + jω – ω 2 ( 10 – 7 )
7

Now convert to frequency in Hertz

j2πf
H ( f ) = -----------------------------------------------2
( 2πf ) -
10 7 + j2πf – ---------------
10 7

The amplitude response is then

 2πf
A ( f ) = -----------------------------------------------------------------
2πf ) 2- 2
 10 7 – (--------------- + ( 2πf ) 2
 10 7 

The plot is shown below with the frequency axis being in log scale.

0.8
0.6
A( f )
0.4
0.2
0
3 4 5 6 7 8 9
1×10 1×10 1×10 1×10 1×10 1×10 1×10
f

b) Is this a low-pass, band-pass, or high-pass filter?

This is a bandpass filter. The center frequency is 1.59 MHz.

c) This filter is used to separate two narrow-band signals at the frequencies f 1 = 10 6 Hz

and f 2 = 10 3 Hz, where the signal at f 2 is considered to be an intereference signal. The
two signals have the same average power. What is the SNR for the signal at f 1 at the filter
output.

SNR is the Signal to Noise Ratio. It is the ratio of the powers. In this case it is signal to interfer-
ence ratio. The output signal signal power is H ( f 1 ) 2 S i and the Output interference power is
H ( f1 ) 2 Si H ( f1 ) 2 A ( f 1 ) 2
H ( f 2 ) 2 I i . Hence the SNR is ----------------------- - =  ------------
- = ------------------ - = 1.3 × 10 6 . Usually we spec-
H ( f2 ) 2 Ii H ( f2 ) 2  A ( f 2 )

ify this in decibels 10 log ( 1.3 × 10 6 ) = 61 dB.

8) For the above filter consider the bandwidth to be defined at points where the amplitude
response drops by 6 dB. What is the bandwidth and center frequency of the filter?

We take the maximum value of H ( f ) 2 , and let f c be the frequency where this value is attained,.
The find the frequencies f 1 and f 2 on either side of f c where the value drops by a factor of 2.

The center frequency is f c = 1.592 MHz, f 1 = 0.984 MHz, and f 2 = 2.575 .

Hence the bandwidth of the filter is B = f 2 – f 1 = 2.575 – 0.984 = 1.592 MHz.

9) In 8) change the value of the resistor to 20 K Ω . What is the bandwidth and center fre-
quency of the filter?

The center frequency remains the same. In fact it is determined by the resonance frequency
1
f c = ------------------ . f c = 1.592 MHz. f 1 = 1.243 MHz, f 2 = 2.038 MHz.
2π LC

The bandwidth is B = f 2 – f 1 = 0.8 MHz. Note that the bandwidth decreases as R increases.

In fact for R = 500 KΩ the bandwidth is approximatey 32 KHz.

10) Repeat 8) with the capacitance of the capacitor changed to 1 pF.

In this case the center frequency will change to

f c = 5.03 MHz. The bandwidth is B = 15.9 MHz.

11) For the above circuit swap the inductor and resistor and find the transfer function. Plot
the amplitude response. What type of filter is it?

The transfer function, using the voltage divider equation is now

1
R  ----------
jωC
--------------------
1
R  ---------- R + ----------
1 R
-----------------------
jωC jωC 1 + jωRC R
H ( ω ) = ---------------------------------------- = ------------------------------------- = ---------------------------------------- = -------------------------------------------
2 RLC
-
1 1 R + – ω
jωL + R  ---------- R  ---------- jωL + ----------------------- R jωL
 jωC  jωC 1 + jωRC
jωL + --------------------
1
R + ----------
jωC

The amplitude response is

R 1
A ( ω ) = ----------------------------------------------------------- = ------------------------------------------------------------
( R – RLCω 2 ) 2 + ω 2 L 2 L 2
( 1 – LCω 2 ) 2 + ω 2  ---
 R

In terms of Hz we have

1
A ( f ) = ------------------------------------------------------------------------------- . Note that in doing this I am re-defining the function
L 2
( 1 – LC ( 2πf ) 2 ) 2 + ( 2πf ) 2  ---
 R
A( .) .
Also, instead of the amplitude we may plot the amplitude squared P ( f ) = A 2 ( f ) . The 3 dB point is
then the point at which P ( f ) drops by a factor of 2 relative to the main passband value of 1, i.e.
1
the frequency f at which P ( f ) = --- . This is ploted for these values R = 10 KΩ , L = 1 mH,
2
C = 10 pF.

1.5

P( f ) 1

0.5

0
3 4 5 6 7 8 9
1×10 1×10 1×10 1×10 1×10 1×10 1×10
f

We can see that now we have a low pass filter. Indeed we expected it because for low frequencies
the inductor acts like a short and clearly the ouput voltage is equal to the input, whereas for very
high frequencies the inductor acts like an open circuit and the output voltage then goes to zero. If
we define the bandwidth as the 3 dB bandwidth (with the 3 dB point taken with respect to the
value at 0) then the bandwidth is 1.77 MHz.

And now, even though the question did not ask, since I am having fun, and since this can easily be
programmed in Matlab or in my case Mathcad, I am going to do the remaining case where I take
the original circuit and exchange the resistor with the capacitor. The transfer function is now

jωRL
-------------------
R jωL R + jωL jωRL – ω 2 RLC
H ( ω ) = ----------------------------------- = ------------------------------------- = -------------------------------------------------------- = -------------------------------------------
-
1- jωRL 1- 1- R + jωL – ω 2 RLC
R jωL + --------- ------------------- + --------- jωRL + --------- ( R + jωL )
jωC R + jωL jωC jωC

And the power response is then

ω 4 ( LC ) 2
P ( ω ) = -------------------------------------------------------2-
( 1 – ω 2 LC ) 2 + ω 2  ---
L
 R

Redifining P ( . ) in terms of frequency in Hertz, we obtain

( 2πf ) 4 ( LC ) 2
P ( f ) = --------------------------------------------------------------------------2-
( 1 – ( 2πf ) 2 LC ) 2 + ( 2πf ) 2  ---
L
 R
 1.5

1
P( f )
0.5

0
3 4 5 6 7 8 9
1×10 1×10 1×10 1×10 1×10 1×10 1×10
f

In this case we get a high pass filter (HPF). The 3 dB point (with reference to the constant value

of 1 for large frequencies) is 1.25 MHz.

 Dept. of Electrical Engineering
University of Toronto
Communication Systems
ECE316
Problems 3 - solutions

1.

a) We plot the signal over a span of 5 periods.

Here is the plot for the sinusoid

m( t ) 0

−1
−3 −3 −3 −3
− 2×10 − 1×10 0 1×10 2×10
t

Here is the plot for the square wave

1.1 1

sw ( t , To , τ ) 0.5

− .1 0
−3 −3 −3 −3
− 2×10 − 1×10 0 1×10 2×10
− Tm t Tm

Here is the plot for the product

y ( t) 0

−1
−3 −3 −3 −3
− 2×10 − 1×10 0 1×10 2×10
t
b)

f m be the frequency of the sinusoid, f 0 the frequency of the square wave.

We find the Fourier transforms of both signals and then convolve.

1
For the sinusoid we have --- ( δ( f – f m ) + δ( f + f m ) ) , and for the square wave we have harmonics at
2
1
kf 0 = k10f m . Find the Fourier series, X k = --- sinc  --- .
k
2  2

The Fourier transform of the square wave is the following where the lines are delta functions (sorry
the program does not plot arrows).

The Fourier transform for the square wave is  X k δ( f – kf 0 ) . The plot is shown below where
k = –∞
for kf 0 we only show k , and use use a circle instead of arrow for the delta functions.
∞
Xk
The convolution is  ----- ( δ( f – ( kf 0 + f m ) ) + δ( f – ( kf 0 – f m ) ) ) . For each delta function at kf 0
2
k = –∞
we replace it with two delta functions. One at kf 0 + f m and the other at kf 0 – f m . And we multiply
1
the X k by --- .
2

0.4

Xk 0.2

− 0.2
− 10 − 9 − 8 − 7 − 6 − 5 − 4 − 3 − 2 − 1 0 1 2 3 4 5 6 7 8 9 10
k
2.
1
The Fourier transform of x ( t ) is X ( f ) = --------------------
a + j2πf
1
ψ x ( f ) = X ( f ) 2 = ------------------------
-
a + 4π 2 f 2
2

a)
a 2000
Note that setting a = 2πf and solving for f = ------ = ------------ = 318 Hz is the 3 dB bandwidth.
2π 2π

b)

∞ ∞
1 1
Total energy is  X ( f ) 2 df =  a------------------------
2 + 4π 2 f 2
- df = ------ = 2.5 × 10 – 4
2a
–∞ –∞

For the bandwidth we need to determine B such that

 X ( f ) 2 df
–B
--------------------------
- = 0.99
∞

 X ( f ) 2 df
–∞

We compute numerically. B = 20.2 × 10 3 . Hz

c) The bandwidth in the strict sense is infinite.

3.

1
α 1 = 1 τ 1 = 1 , α 2 = --- , τ 2 = 3 .
2

1 1
The transfer function is H ( f ) = e –j2πf + --- e –i2πf3 = e – j2πf + --- e – i6πf
2 2
We have used the units of μ s for time, hence the units for frequency will be MHz.
 2

1.5

H( ν ) 1

0.5

0
9.6 9.8 10 10.2 10.4
ν

The attenuation is the lowest at the band edges, 9.5 and 10.5 MHz, and it is the greatest at 9.75
MHz and 10.25 MHz.

4.

y ( t ) = 2x ( t ) + ax 2 ( t ) .

Y ( f ) = 2X ( f ) + a ( X ( f )*X ( f ) )

Now, generally when we convolve two functions the “width” of the result is the sum of the two
widths of the functions being convolved, and if we are convolving the function with itself then the
bandwidth doubles. E.g. convolve a square function with itself and we can see that the width dou-
bles.

Hence in this case the first term will be limited to 10 KHz, but the second term has a bandwidth of
20 KHz, hence the bandwidth of y ( t ) should be approximately 20 KHz.
 Dept. of Electrical Engineering
University of Toronto
Communication Systems
ECE316
Problems 4 - Solutions

1. The signal x ( t ) =  δ( t – kT ) is input to an ideal band-pass filter with center frequency 2

k = –∞
MHz and bandwidth 500 KHz. If T = 10 μ s, give the output of the filter.

First we find the Fourier transform. Since x ( t ) is a periodic signal we can find the Fourier series
and then find the FT term by term.

T
---
2
1 1
FS: X n = ---  x ( t )e – j2πnf 0 t dt = --- . Note that in this case we should not take the interval as 0 to
T T
T
– ---
2
T because there is a delta function at 0 and we have no way to split the delta function at 0.

n n
1 1
 where f 0 = --- . x ( t ) = --- 
j2πnf 0 t
The Fourier series is then Xn e e j2πnf0 t .
T T
n = –∞ n = –∞

n
1 1 1 -
The FT is then ---
T  δ( f – nf 0 ) . From the above f 0 = --- = ---------------------
T 10 ( 10 – 6 )
= 10 5 = 100 KHz.
n = –∞

So x ( t ) has components at all the harmonics n × 100 KHz. The components that pass through the
filter are for n = 18 , i.e. 1800 KHz. to n = 22 , i.e. 2200 KHz.

1
y ( t ) = --- ( e j2π ( –18 )f0 t + e j2π ( 18 )f 0 t + e j2π ( – 19 )f0 t + e j2π ( 19 )f 0 t + e j2π ( – 20 )f0 t + e j2π ( 20 )f 0 t )
T
+ e j2π ( –21 )f0 t + e j2π ( 21 )f0 t + e j2π ( –22 )f 0 t + e j2π ( 22 )f0 t )

2
y ( t ) = --- ( cos 36πf 0 t + cos ( 38πf 0 t ) + cos ( 40πf 0 t ) + cos ( 42πf 0 t ) + cos ( 44πf 0 t ) ) .
T

Note: Another way to to see that the delta train can be written as a summation of cosine signals.
Define the truncated sum
 N

xN ( t ) = 1 + 2  cos ( 2πnf 0 t ) . Plot this for values values of N making them larger and larger
n=1
and you see that it appears to approach a delta train.

2. The 60 Hz sinusoid is passed through a full wave rectifier. What is the fundamental frequency
of the ouput?

1
Before rectification the fundamental period is ------ s, but after full wave rectification the fundamen-
60
1 1
tal period is -------------- = --------- s. Hence the fundamental frequency of the output is equal to 120 Hz.
2 ( 60 ) 120

3. A square wave with 0 average value and amplitude equal to 1 has a frequency of 100 KHz. It is
passed through an ideal high pass filter with cut-off frequency equal to 150 KHz. Find the average
power of the output signal.

We can find the Fourier series of the output and then use Parseval’s theorem. The fundamental
frequency is 100 KHz. The filter passes all components except the first harmonic. The average

power of the input signal is 1. The Fourier series coefficients are X 0 = 0 . X n = sinc  --- .
n
 2

π
sin  ---
1  2 2
The blocked components are at n = 1 and n = – 1 , or sinc  --- = ---------------- = --- . The blocked
 2 π π
---
2
2 2 4
component is then --- e – j2πf 0 t + --- e j2πf 0 t = --- cos ( 2πf 0 t ) . Note that we can write a FS for x ( t )
π π π

∞ ∞
1
as x ( t ) =  a n cos ( 2πnf 0 t ) and the average power is P = ---
2  an2 = 1 . Hence the power of
n=1 n=1
1 1 4 2
the output is 1 – --- a 12 = 1 – ---  --- = 0.189 .
2 2  π

4. Find the complex Fourier series (in terms of complex exponentials) for the signal
x ( t ) = sin4 ( 20πt ) .

We can use the standard procedure as follows: 1) identify the fundamental period T 0 and then
T
----0-
2
1
compute the Fourier coefficients as X n = -----
T0  x ( t )e – j2πnf 0 t dt . The Fourier series is then
T
– ----0-
2
 ∞

 X n e j2πnf0 t . Now in this case the fundamental perios is the value t sunch that 20πt = π , or
n = –∞
1
T 0 = ------ . We can compute the above integral to determine the X n or take a short cut and set
20
1
sin ( 20πt ) = ----- ( e j20πf – e –j20πt ) and then expand
2j
1- j20πt –j20πt  4 1
 ---- (e –e ) = ------ ( e 80πt – 4e 40πt + 6 – 4e – 40πt + e –80πt ) .
 2j  16

Note that we used the Pascal triangle.

11
121
1331
14641

3 –1 1
Now since the fundamental is f 0 = 20 , whe have X 0 = --- , X 1 = X – 1 = ------ , X 2 = X – 2 = ------ ,
8 4 16
and X n = 0 for all the other n .

5. Find the real Fourier series (in terms of cosine and sine functions) for the above signal.

3 1 1
The real FS is x ( t ) = --- – --- cos ( 2πf 0 t ) + --- cos ( 4πf 0 t ) .
8 2 8

π
6. Find the envelope of the signal 10 cos ( 100πt ) + 20 sin  100πt + --- .
4

In this solution I am going way being the problem to explain the concept of signal envelope.

The concept of envelope applies to narrow band signals, or modulated signals. The simplest case
is a sinusoid. We write the sinusoid in standard form A cos ( 2πf c t + θ ) , whe A > 0 . Then the enve-
lope is the amplitude A . Note that if we have α cos ( 2πf c t ) , where α can be negative then we can
always say that the envelope is α . For example if we have – 3 cos ( 2πf c t ) then the envelope is
– 3 = 3 . Or we could write – 3 cos ( 2πf c t ) = 3 cos ( 2πf c t + π ) , and then just pick off the ampli-
tude 3. Now what about the summation of 2 sinusoids with the same frequency? We can use a trig
identity and write it as a single sinusoid and then pick off the ampliture. Alternatively represent
the first sinusoid by the phaser A 1 e jθ1 and the second by the phasor A 2 e jθ2 . Then add
these A 1 e jθ1 + A 2 e jθ2 = A 3 e jθ3 and the envelope of the sum of the sinusoids is A 3 . Note that in
this case you would convert each complex number into rectangular form, then add them, and then
convert back to polar form.

Solution to the above problem: Note that the two signals have the same frequency.

π
The phasor for 10 cos ( 100πt ) is 10. The phasor for 20 sin  100πt + --- is obtained by first con-
 4
verting to the standard sinusoid.
π
π π π π – j ---
20 sin  100πt + --- = 20 cos  100πt + --- – --- = 20 cos  100πt – --- . The phasor is 20e 4 .
4 4 2 4

π
– j --- π π 20 20
Add the two phasors: 10 + 20 4 = 10 + 20 cos  --- – j20 sin  --- = 10 + ------- – ------- j .
 4  4 2 2

20 2 20 2 400
The envelope is   10 + ------- +  ------- = 100 + --------- + 400 = 10 5 + 2 2 ≈ 28 . Note that we
2 2 2
are talking about a function of time that is constant. The envelope is always a function of time, i.e.
E ( t ) = 28 .

What about other cases where we add signals with different frequencies? E.g.
x ( t ) = A cos ( 2πf 1 t + θ 1 ) + A cos ( 2πf 2 t + θ 2 ) . We assume that the difference of these frequen-
cies is much smaller than either one, i.e. f 2 – f 1 « min ( f 1, f 2 ) . We can write
2πf 2 t – 2πf 1 t 2πf 2 t + 2πf 1 t 2π ( f 2 – f 1 )t
x ( t ) = 2A cos  -------------------------------- cos  -------------------------------- . The envelope is then 2 A cos  ---------------------------- .
2 2 2
In general we find the envelope for signals that are modulated signals. There is a carrier of high
frequency that is modulated by a slowly varying signal. From one cyycle of the carrier to the next
there is little change in the amplitude. We think of connecting the peaks of the sinusoid to get the
envelope. Consider the modulated signal m ( t ) cos ( 2πf c t ) where the bandwidth of the signal
m ( t ) is much smaller than the carrier frequency f c . Then the envelope is E ( t ) = m ( t ) . Now
suppose that you have the signal x ( t ) = m 1 ( t ) cos ( 2πf c t ) + m 2 ( t ) sin ( 2πf c t ) , where m 1 ( t ) and
m 2 ( t ) vary much slower with time vs. cos ( 2πf c t ) . Then we can write this signal as
m2 ( t )
x( t) = m 12 ( t ) + m 22 ( t ) cos  2πf c t – atan  -------------  . The envelope is then
m1 ( t )

E( t) = m 12 ( t ) + m 22 ( t ) . Example: Take m 1 ( t ) = sinc ( t ) and m 2 ( t ) = sinc ( t – 1 ) . The band-

1
width of these signals is B = --- (show it by finding the Fourier transform). Now consider
2
f c = 10 . Now plot the signal x ( t ) = sinc ( t ) cos ( 2πf c t ) + sinc ( t – 1 ) sin ( 2πf c t ) . The envelope

of this signal is E x = sinc 2 ( t ) + sinc 2 ( t – 1 ) . Below we plot the signal x ( t ) .

 1

x( t )
−4 −2 0 2 4

−1

The envelope is plotted below

E( t )
−4 −2 0 2 4

−1

Below we plot the two in the same plot

x( t )

E( t ) −4 −2 0 2 4

−1

t
 π
7. Find the envelope of the signal 10 cos ( 100πt ) + 20 sin  101πt + --- .
 4

In cases like this we identify a carrier frequency and write each in terms of the carrier frequency
and an extra frequency component. It is not important which frequency we choose as long as it is
close to the two frequencies. For example we can choose f c = 50 . Then we have

1 π
10 cos ( 2πf c t ) + 20 sin  2π  f c + --- t + --- . We expand so that we get terms involving cos ( 2πf c t )
  2  4
and sin ( 2πf c t ) and the collect terms so as to get A ( t ) cos ( 2πf c t ) + B ( t ) sin ( 2πf c t ) . The enve-

lope is then A 2 ( t ) + B 2 ( t ) . In the above we expand and get

π θ
10 cos ( 2πf c t ) + 20 sin ( 2πf c t ) cos  πt + --- + 20 cos ( 2πf c t ) sin  πt + --- .
4 4

Now collect terms

 10 + 20 sin  πt + π
---  cos ( 2πf c t ) + 20 cos  πt + π
--- sin ( 2πf c t ) .
  4   4

The envelope is then

2 2
E( t) =  10 + 20 sin  πt + π
---  +  20 cos  πt + π
---  = π
100 + 400 sin  πt + --- + 400
  4    4  4

π
E ( t ) = 10 5 + 4 sin  πt + --- .
 4

=================

101
Now let us do it using a different f c . Let f c = --------- . Then we have
2
π
10 cos ( 2πf c t – πt ) + 20 sin  2πf c t + ---
4

expand

π π
10 cos ( 2πf c t ) cos ( πt ) + 10 sin ( 2πf c t ) sin ( πt ) + 20 sin ( 2πf c t ) cos  --- + 20 cos ( 2πf c t ) sin  ---
4 4

collect terms

 10 cos ( πt ) + 20 sin  π
---   π--- 
  4  cos ( 2πf c t ) +  10 sin ( πt ) + 20 cos  4  sin ( 2πf c t )
 2 2
E( t) =  10 cos ( πt ) + 20 sin  π
---  +  10 sin ( πt ) + 20 cos  π
--- 
  4    4 

π π π
E( t) = 10 2 + 20 2 + 400 cos ( πt ) sin  --- + 400 sin ( πt ) cos  --- = 500 + 400 sin  πt + ---
 4  4  4

π
E ( t ) = 10 5 + 4 sin  πt + --- . This is the same as above!
 4

8. Find the average power of the signal m ( t ) cos ( 2πf 0 t ) if f 0 = 1 MHz and m ( t ) is a 1 KHz
square wave that varies between 0 and 2 and has a 50% duty cycle.

1 1
The average power for a sinusoid A cos ( 2πf 0 t ) is --- A 2 . For variable amplitude, it is --- m 2 ( t ) ,
2 2
1 1 1
where the bar denotes time average. Hence we have ---  --- × 0 + --- 2 2 = 1 .
2 2 2

9. Consider the message signal m ( t ) = 2 cos ( 4πt ) and the carrier c ( t ) = 10 cos ( 40πt ) . For
each of the following types of modulation plot the spectrum of the signal and give an expression
for the signal in the time domain. Also give the bandwidth and determine the average power of the
modulated signal.

a) DSB modulation.

modulated signal is x ( t ) = 20 cos ( 4πt ) cos ( 40πt ) .

x ( t ) = 10 ( cos ( 44πt ) + cos 36πt ) )

X ( f ) = 5 ( δ( f – 22 ) + δ( f + 22 ) + δ( f – 18 ) + δ( f + 18 ) ) .

The above can be plotted as 4 delta functions at the appropriate points.

b) LSB-SSB modulation.

After filtering the upper sideband we get

X ( f ) = 5 ( δ( f – 18 ) + δ( f + 18 ) ) .

c) USB-SSB modulation.

Here we filter the lower sideband

X ( f ) = 5 ( δ( f – 22 ) + δ( f + 22 ) )

d) AM modulation with modulation index 1.

Here we normalize m ( t ) so that it has peak value equal to 1. Let m n ( t ) be the normalized value.
Then x ( t ) = A ( 1 + μm n ( t ) ) cos ( 2πf c t ) . For μ = 1 we have
x ( t ) = 10 ( 1 + cos ( 4πt ) ) cos ( 2πf c t ) .

e) AM modulation with modulation index 1/2.

1
In this case x ( t ) = 10  1 + --- cos ( 4πt ) cos ( 2πf c t )
 2 

10. A signal m ( t ) has a complex Fourier series where all the components have amplitude equal
to 1 for k ≤ 10 , and 0 otherwise. Determine the average power of the signal.

10

The average power is  1 2 = 21 .

n = – 10

11. Consider a definition of signal bandwidth where we include enough frequency components in
the signal to account for 99% of the total power of the signal. Find the bandwidth of the following
signals:

a) A sinusoidal signal with frequency equal to 1 KHz.

There is only one component. The bandwidth is B = 1 KHz.

b) A square wave signal with frequency equal to 1 KHz.

We assume a square wave with 50% duty cycle and zero mean. X n = sinc  --- for n ≠ 0 .
n
2

 sinc 2  --2- ≥ 0.99

n
We need to find the smallest N so that 2 . We get N = 41 . Since the
n=1
fundamental is 1 KHz, the bandwidth is then 41 KHz.

c) A triangular wave signal with frequency equal to 1 KHz.

We assume a zero mean triangular wave with peak value equal 1. The Fourier coefficients can be
1
obtained as X n = sinc 2  --- , for n ≥ 1 , and 0 for n = 0 . The total power is --- .
n
2 3

N
1
Now find N such that 2  Xn2 ≥ 0.99 × --3- . The result is N = 3 . Hence the bandwidth is B = 3
n=1
KHz.

d) A 1 KHz sinusoidal signal after full-wave rectification.

Assume that the peak is equal to 1.

2 ( –1 ) n 2
X n =  – --- ----------------- for n ≥ 1 , X 0 = --- .
 π 4n 2 – 1 π

N
1 1
Power is --- . Find N such that
2  X n2 ≥ 0.99 × --- . N = 1 . Hence bandwidth B = 1 KHz.
2
n = –N

12. For a square wave as in b) above and a carrier with frequency 100 KHz, plot the spectrum for
DSB and USB-SSB using the square wave as the message signal. Plot the modulated signals in
the time domain.

1
Take the line spectrum for the square wave multipy each component by --- . Then shift to the right
2
by 100 KHz, and shift to the left by 100 KHz.
 Department of Electrical & Computer Engineering
University of Toronto
Communication Systems - ECE 316F

Solutions to Problems 5

4.2-10

The ring modulator taken an input signal which can be a message or a modulated signal and is driven by
a local sinusoidal signal that results in the input signal effectively being multiplied by a zero mean
square wave with the two levels -1 and 1.
Let f 1 = 1 MHz and f 2 = 400 KHz.

The input signal is then m ( t ) cos ( 2πf 1 t ) and the local oscillator signal has frequency f LO . The square
wave must have a fundamental frequency lying in the range 150 to 210 KHz. We must decide which har-
monic to use from the square wave and then adjust the fundamental frequency so that we achieve the
desired frequency shift. In going from 1 MHz to 400 KHz, we multiply by a local oscillator signal that
has a frequency such that either 10 6 – f LO = 400 × 10 3 , or 10 6 + f LO = 400 × 10 3 . Clearly the only
possibilty is the first case, or f LO = 10 6 – 400 × 10 3 = 600 × 10 3 .

So we need a local oscillator with frequency equal to 600 KHz. But the oscillator that we have to drive
the ring modulator has a constraint to operate in the range 150 to 210 KHz. Hence we can not work with
the fundamental. Now the 2nd harmonic is zero for a square wave. What about the 3rd? The third har-
monic is constrained to be betwwen 3 × 150 = 450 and 3 × 210 = 630 KHz. This works. We can set
the frequency equal to 200 KHz. Then the third harmonic will be 600 KHz. Now we need to determine
the coefficient for the third harmonic in the Fourier series for the square wave. To make it easier I will
add 1 to the square wave, hence it becomes a square wave with the two values being 0 and 2. The third
harmonic is clearly the same.

T
----0- T0 6πf 0 T 0
sin  -----------------
4 -----
4
2 2 sin ( 6πf 0 t ) 4 4 2 3π 2
a 3 = -----  2 cos ( 6πf 0 t ) dt = ----- ------------------------- = ----- ------------------------------ = ------ sin  ------ = – ------ .
T0 T 0 6πf 0 T0 6πf 0 3π  2  3π
T
T
– ----0- – ----0-
4 4

2
Hence in ring modulator will multiply the DSB signal by – ------ cos ( 2πf LO t ) . If the input DSB has the
3π
1 2
form Am ( t ) cos ( 2πf 1 t ) then the constant c is c = ---  A × – ------ = – ------ .
A
2 3π 3π

The moral of the story: Frequency shifting is obtained by multiplication by a sinusoid from a local
oscillator. For a given desired output frequency (i.e. the shifted frequency) there are two possibilties for
frequency of the local oscillator. Now if we implement multiplication by a local oscillator signal using a
ring modulator then we are multiplying by a square wave, i.e. we are multiplying by many sinusoids at
the same time (i.e. many harmonics). We then have the freedom of considering all the possible harmon-
ics and for each harmonic there are two possibilties. We pick the one of all these possibilities that meets
the constraints imposed on the problem. Then we use a filter to block all the other harmonics.

4.3 - 1

1 + cos ( 2ω c t )
The output of the multiplier is ( A + m ( t ) ) cos ( ω c t ) cos ( ω c t ) = ( A + m ( t ) )  ----------------------------------
 2 
A + m(t) 1
After the low-pass filter we have --------------------- , and after removing the D.C. component we have --- m ( t ) .
2 2

4.3-2

Consider the standard form for the AM signal. a ( 1 + μm n ( t ) ) cos ( 2πf c t ) . Now we know that for AM the
modulation index μ must be in the range 0 ≤ μ ≤ 1 . The question is asking to consider cases of μ > 1 . This
means that for these cases we will not have AM modulation, as we will see.

m(t)
Consider A to be some positive constant. Then ( A + m ( t ) ) cos ( ω c t ) = A  1 + ----------- cos ( ω c t ) . To put it in
 A 
m(t)
standard form we have μm n ( t ) = ----------- . Hence, since m ( t ) is given to us as having an amplitude of 10, then we
A
m(t) 10m ( t )
n 10
have m ( t ) = 10m n ( t ) and ----------- = -------------------- . Hence μ = ------ . This we can select different modulation
A A A
10
indecies by choosing different A ‘s. Or A = ------ .
μ

For μ = 0.5 , A = 20 , for μ = 1 A = 10 , for μ = 2 A = 5 , and for μ = ∞ , A = 0 .

We plot all these cases below. Note that for μ = ∞ we essential get DSB modulation. For μ = 2 we have what
is called “over-modulation” and we can not demodulate using an envolope detector. We would get distortion in the
demodulated signal.

I will plot with Mathcad. First here is a method to generate a triangular wave. You can use this with any
programming language. First we define the “mod” function. mod(x,a) = remainder after dividing x by a.
E.g. mod(8,3) = 2. mod(8,3.5) = 1. mod(-8,3.5) = -1, etc.
Also Mathcad has the function if(logical expression, a, b). If the logical expression is true then the value is a.
Otherwise the value is b. E.g. if(3 > 1, -4, 9) = -4. if(3 < 1, -4, 9) = 9.
In Math can we type “if(3 < 1, -4, 9) = “ and the answer on the right side of the “=” sign pops out. That is, we can
the “if” function with the three arguments and the returned value is the value on the right side of the equal sign.
Now define the constant A to be the amplitude of the triangular wave. In Mathcad we write A := 10. Define T to
be the period of the triangular wave, or T := 10 –3 in our case. Then the triangular wave is defined as follows:

 4 if mod( t , T) < , mod( t , T) , T − mod( t , T)  

T

m( t) := A ⋅  1 −
 2 

 T 
We plot it below
 10

m( t )
−3 −4 −4 −3
− 1×10 − 5×10 0 5×10 1×10
−5

− 10

Now the fundamental frequency here is 1 KHz. We will choose a carrier that is 30 KHz. That is, there are 30 cycles
of the carrier for each period of the message signal m ( t ) .

1
Now here is the answer for μ = --- .
2

20

x( t )
−3 −4 −4 −3
− 1×10 − 5×10 0 5×10 1×10

− 20

Here is the answer for μ = 1 .

20

10

x( t )
−3 −4 −4 −3
− 1×10 − 5×10 0 5×10 1×10

− 10

− 20

Page 3 of 6
Here is the answer for μ = 2

10

x( t )
−3 −4 −4 −3
− 1×10 − 5×10 0 5×10 1×10

− 10

Now I will choose a very large value to approximate infinity μ = 10000 .

Here is the result.

10

x( t )
−3 −4 −4 −3
− 1×10 − 5×10 0 5×10 1×10

− 10

Note that this is the same as DSB.

4.3-3

10
For μ = 0.8 . We have A = ------ = 12.5 . Hence the AM signal is A ( 1 + μm n ( t ) ) cos ( 2πf c t ) . Now
μ

1 A2 A2 A2μ2m 2( t )
n
the average power of this signal is --- ( A ( 1 + μm n ( t ) ) ) 2 = ------ ( 1 + μ 2 m n2 ( t ) ) = ------ + --------------------------
2 2 2 2

The amplitude of the carrier is A = 12.5 .

 2
 25
------
A 2 12.5 2  2 625
a) The power in the carrier is P c = ------ = ------------- = -------------- = --------- .
2 2 2 8

n A2μ2m 2( t ) 1
b) The sideband power is P s = -------------------------- . Note that for a normalized square wave the average power is --- (do
2 3
2 2
 25
------  4---  1---
 2   5  3 25 2 4 2 25 ( 4 ) 50
it). Hence P s = ---------------------------------- = ----------------------- = -------------- = ------ .
2 22523( 2) 3( 2) 3
The efficiency is
50
------
Power in sidebands- 3 8 ( 50 ) 8 ( 10 ) 80
η = ----------------------------------------------- = --------------------- = ------------------------------------- = ------------------------------------- = --------- ≈ 0.176 .
Total Power 625 3 ( 625 ) + 8 ( 50 ) 3 ( 125 ) + 8 ( 10 )
--------- + 50 455
------
8 3

4.3-6

A DSB signal has the form m ( t ) cos ( 2πf c t ) . An AM signal has the form A ( 1 + μm n ( t ) ) cos ( 2πf c t ) .
We can write the AM signal as m' ( t ) cos ( 2πf c t ) , where m' ( t ) = A ( 1 + μm n ( t ) ) . Hence the different between
DSB and AM is that the DSB signal has a modulating signal that is zero mean and the AM signal has a modulating
signal that is not zero mean, in fact it is positive for all t. The converse is not true because the envelope detector can
not be used to demodulate a DSB signal. E.g. consider the case of the message equal to a square wave.

4.3-8

Consider the AM signal φ AM ( t ) = A ( 1 + μm n ( t ) ) cos ( 2πf c t ) . Then, after squaring we get

A 2 ( 1 + μm n ( t ) ) 2
x ( t ) = A 2 ( 1 + μm n ( t ) ) 2 cos2 2πf c t = ---------------------------------------- ( 1 + cos ( 4πf c t ) ) . Expand to get
2
A 2 ( 1 + μm n ( t ) ) 2 A 2 ( 1 + μm n ( t ) ) 2
---------------------------------------- + ---------------------------------------- cos ( 4πf c t ) . And expand again to get
2 2

A 2-
 ----- A 2 μ 2 m n2 ( t )  A 2 A 2 μ 2 m n2 ( t )
2
+ A μm n ( t ) + -------------------------- + -----
- 2
+ A μm n ( t ) + -------------------------- cos ( 4πf c t ) . (**)
2 2  2 2 

Now we need to extract the message from this message. Let us look at the spectrum. Note that the first term in the
above is basically a baseband signal. The spectrum of this baseband signal is
A 2- A2μ2
----- δ( f ) + A 2 μM n ( f ) + ------------ ( M n ( f )*M n ( f ) ) . (***)
2 2

If the bandwidth of m n ( t ) is B , then the bandwidth of this baseband signal is 2B because of the signal being
convolved with itself. Now the second term in the above (**) is a modulated signal with carrier 2f c and it is easily
removed by a low-pass filter. So we are left with the first term in (**) which has the frequency domain expression

Page 5 of 6
(***). Now if we use a low-pass filter with bandwidth B then the first two terms in (***) will pass and part of the
3rd term will be blocked because it has a bandwidth of 2B . As a result we can not completely filter out the third
term and it will cause distortion in the output signal. To determine the severity of this distortion we could
determine the power in this distortion signal vs. the main signal. If we use a d.c. blocker we will remove the first
term and also a d.c. component in the third term (not that it has a non-zero d.c. component because the message
has been squared). The power in the wanted signal (2nd term in (***)) is P s = ( A 2 μm n ( t ) ) 2 . The power in the

n A2μ2m 2( t )
component that is causing distortion is the a.c. component of the signal -------------------------- , which is
2
A 2 μ 2- 2
----------- ( m n ( t ) – m n2 ( t ) ) . Note that if A >> m ( t ) , which is the same as μ << 1 , then the power in this
2
a.c. component is much less than the power in the wanted signal. The a.c. component contains a factor μ 2 , where
as the wanted signal contains the factor μ .

4.4-1

Note that we did something similar to this in the notes in page 98 of Notes 2, where we assume a phase error of θ .
If instead of θ we have put Δωt + δ then we would get the result shown in this problem.
 ECE316
Communication Systems
E. S. Sousa

Review problems

1. Consider a message signal m ( t ) = 2 cos ( 2πf m t ) and a carrier signal equal to 10 cos ( 2πf c t ) .
Give a time domain and frequency domain expression for a DSB signal. Plot the signal in the
time and frequency domain.

Time domain: x ( t ) = 20 cos ( 2πf m t ) cos ( 2πf c t ) .

10
Frequency domain: X ( f ) = ------ ( M ( f – f c ) + M ( f + f c ) ) . Now M ( f ) = δ( f – f m ) + δ( f + f m ) .
2
Hence X ( f ) = 5 ( δ( f – f c – f m ) + δ( f – f c + f m ) + δ( f + f c – f m ) + δ( f + f c + f m ) ) .

Time domain plot:

25
20

x_DSB ( t ) 0

20
25
0 5 10 15 20 25 30 35 40
0 t 40

Frequency domain: Plot 4 delta functions at the locations – f c – f m, – f c + f m, f c – f m, f c + f m . All

the delta functions have value (area) equal to 5. Note that for the above plot I have normal-
ized the time axis to the reciprocal of the carrier frequency, i.e. I chose the carrier frequency
to be equal to 1. Then I chose the frequency f m = 0.05 .
 5.005

5
5
4.995

4.995 4.99
2 1.5 1 0.5 0 0.5 1 1.5 2
2 f 2
j

2. Repeat the above for AM modulation with a modulation index of μ = 0.5 and a modulation
index of μ = 1 .

Now the first thing here is to normalize the message so that the maximum amplitude is equal to 1.
The signal is then x ( t ) = A ( 1 + μ cos ( 2πf m t ) ) cos ( 2πf c t ) . The value of A will ultimately deter-
mine the power of the signal. Here we could put A = 10 , but the way that the question is
posed other value can be justified for A . One way for the question to force a specific value for
A would be to put a value on the average power for the signal.

Time domain plot for μ = 0.5

40

20

x_AM( t ) 0

20

40
0 5 10 15 20 25 30 35 40
t

Here is the spectrum for μ = 0.5

 5.5

a
j
2

0 0
2 1.5 1 0.5 0 0.5 1 1.5 2
2 f 2
j

Now for μ = 1 . Time domain:

40
40

20

x_AM( t ) 0

20

39.459 40
0 5 10 15 20 25 30 35 40
0 t 40

Frequency domain

5.5

a
j
2

0 0
2 1.5 1 0.5 0 0.5 1 1.5 2
2 f 2
j
3. Draw a block diagram for a DSB demodulator for a system where the bandwidth of the mes-
sage signal is 10 KHz, and the carrier frequency is 1 MHz. Show all the parameters in the dia-
gram.

The input signal is multiplied with a local oscilator signal of frequency equal to 1 MHz and syn-
chronized to the carrier of the incoming signal. The multiplier is followed by an ideal low-pass
filter with bandwidth equal to 10 KHz.

4. Draw a block diagram for an ideal envelope detector for AM.

This is a circuit consisting of an ideal rectifier such as implemented by an ideal diode, followed by
an ideal low pass filter. The bandwidth of the filter is equal to the bandwidth of the message.

5. Draw a block diagram for a simple (practical) envelope detector demodulator for AM.

It is the same as the above except that the ideal low pass filter is replaced by a capacity in parallel
with a resistor. The time constant for this circuit is τ = RC . This time constant should be cho-
1 1
sen so that --- « f « --- , where B is the bandwidth of the message signal.
fc B

6. Draw a block diagram for a coherent demodulator for the above AM signal.

This is the same as a coherent demodulator for DSB but then followed by a circuit that removes the
d.c. value.

7. What is the average power of the modulated signal in 1.?

10 2
The average power is -------- m 2 ( t ) = 50 ( 2 ) = 100 .
2

8. A QAM modulation scheme uses the two modulation signals m 1 ( t ) = cos ( 2πf 1 t ) and
m 2 ( t ) = cos ( 2πf 2 t ) where f 1 = 1 KHz and f 2 = 1.2 KHz. Plot the amplitude spectrum for
the QAM signal. Give an expression for the envelope of the signal.

The time domain signal is m 1 ( t ) cos ( 2πf c t ) + m 2 ( t ) sin ( 2πf c t ) .

1 1
The spectrum is X ( f ) = --- ( M 1 ( f – f c ) + M ( f + f c ) ) + ----- ( M 2 ( f – f c ) – M 2 ( f + f c ) ) . or
2 2j
1 1 1 1
X ( f ) = --- ( M 1 ( f – f c ) ) + ----- M 2 ( f – f c ) + --- M ( f + f c ) – ----- M 2 ( f + f c ) .
2 2j 2 2j
This consists of 4 delta functions for positive frequencies and four delta functions for negative fre-
quencies. Some of these delta functions have coefficients that are imagenary but you then take
the magnitude of the complex number in order to plot the amplitude spectrum.

The amplitude spectrum is X ( f ) .

9. An FDM system is used to multiplex 12 message signals. Each of the 12 message signals has
a bandwidth of 3 KHz. In order to facilitate filtering we need a guard band between signals.
The FDM signal is to be centered at 100 MHz. Give the frequency band required for the sys-
tem, i.e. lower and upper frequencies.

For this problem I left out the specification of the guard band. If the guard band is equal to .5
KHz. then the total bandwidth required is 11 × 0.5 + 12 × 3 = 41.5 KHz. The lower fre-
quency is 100 MHz - 41.5/2 KHz and the higher frequency is 100 MHz + 41.5/2 KHz.

10. Give the block diagram of a system that uses an adder and a squarer to implement a DSB
modulator. Specify any filters needed and the corresponding parameters. If the bandwidth of
the message signal is 5 KHz, what is the smallest carrier frequency that we may have?

The circuit adds the message to the carrier to get m ( t ) + cos ( ω c t ) . Then we square this to get
m 2 ( t ) + 2m ( t ) cos ( ω c t ) + cos2 ( ω c t ) . Now we need a bandpass filter to block the low fre-
quency term m 2 ( t ) and the carrier component cos2 ( ω c t ) . Since the bandwidth of the signal
m 2 ( t ) is twice the bandwidth of the signal m ( t ) . Then the carrier frequency must be at least
2B + B = 3B where B is the bandwidth of m ( t ) .

11. Show how to use a switching modulator with an oscillator with frequency 100 KHz to
achieve DSB modulation with a carrier frequency equal to 500 KHz.

The switching circuit basically acts like a multiplication by a square wave. The square wave has
all the odd harmonics of decreasing amplitude, but non-zero amplitudes. 500 KHz is the 5th
harmonic. All that we need to do is to put at bandpass filter centered at 500 KHz and with
bandwidth equal to twice the bandwidth of the message.

12. We wish to conver an AM modulated signal with a carrier frequency of 1 MHz to a signal with
a carrier frequency of 500 KHz. Give two possible circuits for this conversion, utilizing dif-
ferent local oscillator frequencies.

The circuit consists with a local oscillator multiplying the signal, followed by a filter centered at
500 KHz and with bandwidth equal to the bandwidth of the original signal. The two possible
frequencies for the local oscillator are 1.5 MHz and 0.5 MHz.

13. A sinusoidal signal with 1 KHz frequency is passed through a hard limiter circuit with charac-
teristics as follows: y = sgn ( x ) , where sgn ( . ) is the signum function. Plot the output of the
circuit. Now assume that the output is filtered with an ideal low-pass filter of bandwidth
equal to 8 KHz. This signal is then input as the message to a modulator. For the following
types of modulator give the bandwidth of the modulated signal: AM, DSB, LSB-SSB, USB-
SSB.

The output is a square wave with frequency equal to 1 KHz and zero mean. Since the bandwidth of
the filter is equal to 8 KHz, the filter will pass all the harmonics up to the 7th (note that the
 8th is zero). The bandwidths are therefore:

AM - 14 KHz
DSB - 14 KHz
SSB - 7 KHz (both types)

14. Suppose that for the QAM system in problem 8, the demodulator has a phase error of 10
degrees. Give the output of the in-phase channel after demodulation (i.e. the channel on the
cosine carrier).

The output is m 1 ( t ) cos θ + m 2 ( t ) sin θ . See the notes and make you you derive this yourself!

15. In a frequency conversion scheme suppose the intitial frequency is 10 MHz and the target fre-
quency is 500 KHz. Give two choices for the local oscillator frequency and for each choice
specify the image frequency.

We can have upper side tuning or lower side tuning. For upper side tuning the local oscillator fre-
quency is 10.5 MHz. The image is at 11 MHz.

For lower side tuning the oscillator frequency is 9.5 MHz and the image frequency is 9 MHz.

16. In 15, if the modulating signal (message) has a bandwidth 20 KHz, specify a filter to be used
before the conversion circuit so that any signals at the image frequency will not cause interfer-
ence.

This filter should have a passband in the range (9 MHz + 10 KHz) to (11 MHz - 10 KHz). It should
have a stop band outside this range.

17. A message signal has a spectrum with shape given as follows: M ( f ) = K f for f < B and
zero elsewhere. We wish to scramble this signal in such a way that the high frequency compo-
nents became low frequency components and vise-versa. Give a circuit involving a multiuplier
and filter to achieve the required scrambling.

The circuits consists of a multiplier where we multiply the signal by cos ( 2πf LO t ) where f LO = B .
Then followed by a low-pass filter with bandwidth equal to B .

18. A wireless communication channel is modelled with two propagation paths: 1 has a delay of 1
μ s and the other has a delay of 3 μ s. For the second path the signal is attenuated relative to
the first path by a factor of 0.5. Assume that the channel covers the band 1 - 2 MHz. Plot the
transfer function of the channel.

The impulse response is δ( t – τ 1 ) + 0.5 δ( t – τ 2 ) , where τ 1 = 1 μ s, and τ 2 = 3 μ s. The transfer

1
function is then H ( ω ) = e –jωτ 1 + --- e –jωτ 2 . You should plot either the real and imaginary parts
2
of this spectrum, or the magnitude and phase in the range 1-2 MHz.
 1
19. Find the Fourier transform for the pulse given by x ( t ) = cos ( 2πt ) for t < --- and zero else-
4
where.

1
---
∞ 4

You have to compute ∫ cos ( 2πt ) e –jωt dt = ∫ cos ( 2πt ) cos ( ωt ) dt . Evaluate the last integral by
–∞ 1
– ---
4
first writing the integrand and a cos of sum and cos of difference terms.

20. Show that a lower sideband and an upper sideband SSB signal for the same message signal
have the same signal envelope.

A general signal of the form a ( t ) cos ( ω c t ) + b ( t ) sin ( ω c t ) has envelope given by a2 ( t ) + b2( t ) .

For SSB the envelope would be m 2 ( t ) + ( ± m̂ ) 2 which is the same for both cases of the sign.
“plus” is for lower sideband, and “minus” is for upper sideband.
 ECE316
Communication Systems
E. S. Sousa

Problems 6 - FM: Solutions

1
1. Consider a message signal that is a pulse Π ( t ) = 2 if 0 ≤ t ≤ ------ ms, and 0 elsewhere. Con-
10
sider a carrier 4 cos ( 2πf c t ) , where f c = 100 KHz. If k f = 5 KHz/V plot the modulated sig-
nal assuming FM modulation. (show the plot over the time interval [0, 1/5] ms).

We consider the signal over the whole real line. From – ∞ to 0 the signal has a frequency equal to
1
the carrier, or 100 KHz. From 0 to ------ ms it has a frequency equal to 100 KHz + 10 KHz =
10
1
110 KHz. From ------ to ∞ the frequency is equal to 100 KHz.
10

5
5
4

x( t ) 0

4
5 5
5 5 5 5 4 4 4 4 4 4
0 2 .10 4 .10 6 .10 8 .10 1 .10 1.2 .10 1.4 .10 1.6 .10 1.8 .10 2 .10
0 t Tmax

2. In the above, plot the modulated signal, for the same time interval as above, assuming PM
π
modulation if k p = --- rad/V.
2
1 π
At t = ------ ms, there is a phase change by --- × 2 = π rad.
10 2
 5
5
4

x( t ) 0

4
5 5
5 5 5 5 4 4 4 4 4 4
0 2 .10 4 .10 6 .10 8 .10 1 .10 1.2 .10 1.4 .10 1.6 .10 1.8 .10 2 .10
0 t Tmax

1
Note that in the above there is a phase change of π at 0 and another phase change at t = ------
10
ms.

3. Find the average power of the FM and PM modulated signals in the above?

1
Both signals have constant envelope equal to 4. The average power is --- × 4 2 = 8 .
2

4. In 1) determine the peak frequency deviation Δf .

The peak frequency deviation is Δf = 2 × 5 = 10 KHz.

5. In 1) determine the exact spectrum of the modulated signal.

The signal can be written as 4 cos ( 2πf c t ) – 2Π ( t ) cos ( 2πf c t ) + 2Π ( t ) cos ( 2π ( f c + Δf )t ) .

The spectrum is then
2 ( δ( f – f c ) + δ( f + f c ) ) – ( P ( f – f c ) + P ( f + f c ) ) + P ( f – ( f c + Δf ) ) + P ( f + f c + Δf ) , where
P ( f ) is the Fourier transform of Π ( t ) .

6. What is the bandwidth of the message signal in 1) if we use a strict definition of bandwidth?

With a strict definition the bandwidth is infinite.

7. Assume that the bandwidth of the message signal in 1) is determined by the location of the first
null in the spectrum. What is the bandwidth of the signal?

There is a sinc centered at f c with a first null at f c – B and a sinc centered at f c + Δf with a first
1
null at f c + Δf + B , where B = ----------- = 10KHz . The bandwith is therefore
1-
 -----
 10
 f c + Δf + B – ( f c – B ) = Δf + 2B = 10 + 2 ( 10 ) = 30 KHz . Note that we don’t have 2 Δf
as we expect from Carson’s rule because the signal is not zero mean.

8. Using the bandwidth determined in 7) what is the modulation index β assuming the FM
modulation scheme of 1).

Δf 10 × 10 3
β = ----- = --------------------3 = 1 .
B 10 × 10

9. Estimate the bandwidth of the FM scheme in 1) using Carson’s rule assuming the definition
of bandwidth in 7).

Using Carson’s rule we would have 2 Δf + 2B = 40 KHz . The difference here (instead of 30
KHz) is because the message is not zero mean. In the case of zero mean there is a frequency
deviation of Δf on the positive side of the carrier and an equal deviation in the negative side
of the carrier for a total of 2 Δf .

10. Determine the exact spectrum of the FM scheme in 1) assuming that the FM signal is defined
over the whole real line (i.e. over all time from – ∞ to ∞ .

This is the same as in question 5.

11. An FM signal has a message bandwidth of 10 KHz and a carrier frequency of 10 MHz. The
peak frequency deviation is 20 KHz. Determine the frequency band occupied by the signal.

We use Carson’s rule: Δf = 20 KHz and B = 10 KHz . Hence the bandwidth of the FM sig-
60
nal is 2 × 20 + 2 × 10 = 60 KHz . The band occupied is from 10 × 10 6 – ------ × 10 3
2
60
to10 × 10 6 + ------ × 10 3 , or from 9,970,000 Hz to 10,030,000 Hz.
2

12. Suppose we take the signal in 11) and pass it through a frequency doubler device. What is the
bandwidth of the output signal?

In passing through a frequency doubler we change the Δf by a factor of 2. Hence the new Δf is 40
KHz. The new bandwidth is therefore 2 × 40 + 2 × 10 = 100 KHz .

13. Sketch the block diagram of a system that implements frequency doubling. You may use ideal
filters and also common electronic components.

Techniques used in practice would usually make use of phase-lock loops (PLL). However we can
also use a full wave rectifier followed by a bandpass filter.
14. We wish to take the FM signal in 11) and double its bandwidth while keeping the same carrier
frequency. Sketch the block diagram of a system to achieve this bandwidth doubling.

The original bandwidth is 2 Δf + 2B where Δf and B are given in 11). Now we need a new peak
frequency deviation that we denote as Δf 1 . We have 2 Δf 1 + 2B = 2 ( 2 Δf + 2B ) . Now solving
Δf 1 we get Δf 1 = 2 Δf + B = 2 × 20 + 10 = 50 KHz .

Now to change Δf from 20 KHz to 50 KHz we need a circuit that multiplies the frequency by a fac-
tor of 2.5. The carrier would also be changed by a factor of 2.5 and become 25 MHz. We then
need to do frequency conversion from a carrier at 25 MHz to a carrier at 10 MHz. Use a local
oscillator at frequency 35 MHz and then a bandpassfilter at 10 MHz, or a local oscillator at 15
MHz and a BPF at 10 MHz.

15. An FM signal is passed through a half-wave rectifier. Is is possible to take the output signal
and recover the original FM signal completely? If so, sketch the system block diagram.

It is possible since the information in the FM signal is in the zero crossings. We can shift the half-
wave rectified signal slightly negative and then use a band-pass limiter as we did in the notes.

16. What is the advantage of using narrow bandwidth FM as opposed to wide-band FM?

The advantage is that the bandwidth is lower. Signals with lower bandwidth require channels with
lower bandwidth for transmission.

17. What is the advantage of using wideband FM as opposed to narrow band FM?

The advantage of using wideband FM is that the modulated signal is more robust to interference
and noise.

18. In an analog voice transmission system with quality equivalent to that of a telephone system
FM modulation is used. What is the smallest possible channel bandwidth required to transmit
the signal?

Telephone quality speech signals have a bandwidth of 3.5 KHz. Hence if we use narrowband FM,
the required bandwidth is 2 × 3.5 = 7 KHz.

19. In a first generation cellular system FM modulation with bandwidth equal to 30 KHz is used to
transmit telephone quality speech signals. What is the peak frequency deviation in this system?

This system transmits a message signal of band with equal to 3.5 KHz. Hence we have
30 – 7
30 KHz = 2 Δf + 2 × 3.5 . And Δf = --------------- = 11.5 KHz .
2

20. In the analysis of the spectrum of an FM signal subject to a sinusoidal modulating signal we
find that we have to including terms in the Fourier series expansion up to the 5th harmonic. If
the modulation sinusoid has a bandwidth equal to 5 KHz, what is the bandwidth of the FM sig-
nal?
Recall that the bandwidth is 2n 0 f m , where f m is the fundamental frequency and n 0 is the highest
harmonic that we take. Hence the bandwidth is 2 × 5 × 5 = 50 KHz. .

21. What is a voltage controlled oscillator (VCO)? Give the parameters that describe a given
VCO.

A voltage controlled oscillator is an device that has one input and one output. The output is a
sinusoid with constant amplitude and frequency equal to f c + k v v i , where k v is a constant
and v i is the input voltage.

22. An FM modulator is modelled as a black box. There are two inputs, one for the message and
one for the carrier, and one output for the modulated signal - the FM signal. When the input
message signal is set to zero the output is a sinusoidal signal with frequency equal to 100
MHz. When the message input is d.c. with 100 millivolts the output is a sinusoid with fre-
quency equal to 102 MHz. What is the modulator constant k f ? What is the peak frequency
deviation Δf ?

102 – 100
The modulator constant is k f = ------------------------ = 20 MHz/V . The peak frequency deviation depends
0.1
on the peak input voltage for the message signal. For example if the peak voltage is 500 mV,
then the peak frequency deviation would be 0.5 × 20 = 10 MHz.

23. In 22) if we apply a speech signal m ( t ) as the message, where the maximum value of m ( t )
is 0.5 volts, what is the bandwidth of the FM signal?

The bandwidth would be (Carson’s rule)

( 2 Δf + 2B = 2 × 10 MHz + 2 × 3.5 KHz ) ≅ 2 Δf = 20 MHz .

24. In 22) we are going to apply a square wave as the message signal. The amplitude is A , the
mean value is zero, and the frequency is f 0 . Now, fix the amplitude A and let the frequency
f 0 approach zero, what is the bandwidth of the FM signal in terms of the amplitude.

The bandwidth would be 2 Δf . Now Δf = k f A . Hence the bandwidth is 2k f A .

25. Repeat 24, but now we fix the frequency f 0 and let the amplitude approach zero, what is the
bandwidth if the signal in terms of the frequency of the square wave?

In this case the bandwith is 2B , where B is the bandwidth of the square wave. This depends on
how many harmonics we include in the bandwidth definition for the square wave. If we take 5
harmonics then the bandwidth would be 2 × 5 × f 0 = 10f 0 .

26. Now, repeat 24, but now we fix the amplitude and let the frequency become very large, what
 is the approximate bandwidth of the FM signal in terms of the frequency of the square wave?

Again in this case we would have Δf « B and the bandwidth would be 2B , and depending on the
number of harmonics that we include we would get the bandwidth value, e.g. 10f 0 for five har-
monics.

27. Repeat 24, but now we fix the frequency and let the amplitude become very large. What is the
approximate bandwidth of the signal in terms of the amplitude of the square wave?

In this case we would have Δf » B and the bandwidth would be 2 Δf = 2k f A , where A is the
amplitude.

28. A linear system (filter) has a transfer function with amplitude response given as in the follow-
ing plot:

f (MHz)
10 30 45 60 70
This filter is to be used along with an ideal envelope detector to demodulate FM signals with differ-
ent carriers frequencies. All the FM signals have bandwidths equal to 500 KHz. What are the
possible carriers frequencies for which the demodulator would work?

The possible carrier frequencies are frequencies in ranges where the above filter characeristic is a
straight line, i.e. a carrier anywhere in the range 10 MHz + 250 KHz to
30 MHz – 250 KHz , or a carrier anywhere in the range from 45 MHz + 250 KHz to
60 MHz – 250 KHz .

29. A non-linear device is a squarer; i.e., if the input is x ( t ) then the output is y ( t ) = x 2 ( t ) . Show
that if an FM signal is input to this device then we can essential double the carrier frequency of
the FM signal, while maintaining the capability to demodulate the signal exactly. What else do
we need to do to the signal (output of the squarer) to produce an FM signal? What happens to
the bandwidth of the original FM signal? I.e. what is the bandwidth of the new FM signal?

1
The FM signal has the form cos ( θ ( t ) ) . Squaring we get cos2 ( θ ( t ) ) = --- ( 1 + cos ( 2θ ( t ) ) ) . Since
2
t

θ ( t ) = 2πf c t + 2πk f  m ( τ ) dτ . Hence we need to filter the d.c. value 1/2. The FM signal will
–∞
have a peak frequency deviation that is doubled. Hence in terms of the original parameters the
new bandwidth is 2 ( 2 Δf ) + 2B = 4 Δf + 2B .
30. Satellite systems, used for the transmission of video signals, are based on RF channels with a
bandwidth equal to approximately 29 MHz. This bandwidth was selected in order to transmit
analog TV signals with baseband bandwidth of approximately 4.2 MHz using FM modula-
tion. Note that terrestrial and cable transmission of NTSC TV signals utilizes VSB modula-
tion with a bandwidth for the modulated signal equal to 6 MHz. What is the modulation index
β for the FM satellite transmission system? What would the bandwidth of the satellite sig-
nals if narrowband FM was used instead?

29 – 2 × 4.2 20.6
We have 29 = 2 Δf + 2B . Hence Δf = ---------------------------- = ---------- = 10.3 MHz . Hence the modula-
2 2
Δf 10.3
tion index is β = ----- = ---------- = 2.45 . In the case of narrowband FM the bandwidth would
B 4.3
be 2 × 4.2 = 8.4 MHz .
 Dept. of Electrical Engineering
University of Toronto
Communication Systems
ECE316
Problems 7 - Solutions

1. A signal g ( t ) ) is bandlimited to 10kHz. Find the Nyquist sampling rate for g ( t ) and g ( 2t ) ) .

The Nyquist sampling rate for g ( t ) is 2 × 10 = 20 KHz.

For g ( 2t ) , this signal is compressed in time by a factor of 2. Hence it is expanded in frequency by

a factor of 2, hence the bandwidth double, or 20 KHz. The Nyquist sampling rate is 40 KHz.

2. A “sampled signal” is given by g ( t ) =  g ( nT s ) δ( t – nT s ) = g ( t )p ( t ) , where

n = –∞
∞

p( t) =  δ( t – nT s ) . g ( t ) is bandlimited to B Hz. Assuming the Nyquist sampling rate is

n = –∞
fs
met, the original signal g ( t ) is recovered by inputing g ( t ) to a LPF with cutoff frequency --- ,
2
1
where f s = ----- . Consider g ( t ) = g ( t )p ( t ) , but now p ( t ) is now a periodic pulse train with
Ts
Ts
period T s and each individual pulse is of width ----- .
4

(a) Choose a convenient shape for G ( f ) and sketch G ( f ) , the Fourier transform of g ( t ) .

The question should have been clear that the pulses are square pulses with unit amplitude. First
πk
sin  ------
1  4 1 k
find the complex Fourier series for p ( t ) . A k = --- ------------------- = --- sinc  --- . The Fourier transform
4 πk 4  4
------
4
∞ ∞
1 k
of g ( t ) is then  A k G ( f – kf s ) = ---  sinc  --- G ( f – kf s ) .
4  4
k = –∞ k = –∞

1
For plotting assume that the bandwidth of g ( t ) is B = --- .
2
Then assume the following for G ( f )
 1

0.5

G( f ) 0

− 0.5

−1
−1 − 0.5 0 0.5 1
f

Then the spectrum of g ( t ) , assuming Nyquist rate sampling, is then

0.5
X( f )
0

− 0.5
−5 0 5
f

Note that assuming a sampling rate equal to 1.2 times the Nyquist rate, the spectrum would be

1
0.8
0.6
X( f ) 0.4
0.2
0
− 0.2
−5 0 5
f

(b) How would you recover g ( t ) from g ( t ) ?

1
We recover the original signal by using a LPF with bandwidth B . In this case B = --- .
2

3. In wideband speech processing, a speech signal is assumed to be bandlimited to 7 kHz.

(a) Assuming that the signal is sampled at the Nyquist rate and quantized using a 512-level
quantizer, what is the bit rate of the corresponding pulse code modulation (PCM) system?

Sampling rate is 14 K samples/s. There are 9 bits per sample, because 2 9 = 512 . Hence the bit
rate is 14 × 9 = 126 Kb/s

(b) The above system is found to achieve a signal-to-quantization noise ratio (SQNR) of 60 dB.
How many levels would the quantizer need to have in order to achieve an SQNR of 72 dB?

We need to reduce the quantization noise power by 12 dB. Since the noise power is reduced by 6
dB per bit, we need 2 extra bits per sample, hence 11 bits/sample. The bit rate is then
14 × 11 = 154 Kb/s.

4. (a) The message signal, m ( t ) , has bandwidth 4 kHz, and is scaled so that – 2 ≤ m ( t ) ≤ 2 . In
the initial design, m ( t ) is sampled at the minimum rate required to meet Nyquist’s sampling crite-
rion. What is this minimum sampling rate?

The minimum rate is twice the bandwidth, or 8 K sample/s.

(b) Next, the samples are quantized. What is the minimum number of bits required per sample if
we require the quantization error ε to satisfy – 0.25 ≤ ε ≤ 0.25 ?

The range for signal variation is 2 - (-2) = 4. The quantization interval spacing is Δ = 0.5 . The
4
number of levels is then ------- = 8 . The minimum number of bits per sample is then 3, because
0.5
23 = 8 .

(c) Using the sampling rate found in (b) and 8 bits per sample, what bit rate is required to
communicate this signal?

The question should have referred to the sampling rate found in (a).
Bit rate = 8 × 8 = 64 Kb/s.

5. A 20 kHz audio signal is sampled at its Nyquist rate.

(a) Using a 10 bit quantizer, it is found that the signal-to-quantization noise ratio (SQNR) is 55
dB. However, the system specification calls for a minimum SQNR of 60 dB. How many bits
should the quantizer have to achieve this specification?

For each bit we get an extra 6 dB. Hence we need one extra bit, or 11 bits. The SNR would be
55 + 6 = 61 dB.

(b) Suppose that it is decided to use 14 bits per sample. What is the resulting bit rate?

The bit rate would be 2 × 20 × 14 = 560 Kb/s.

6.
A zero-order hold reconstructs g ( t ) from its samples. The system is shown in the figure below:
 DELAY - OUTPUT
INPUT  ( . ) dt Ts
+

The input in this figure is the signal g ( t ) =  g ( kTs ) δ( t – kTs ) , the ideally sampled signal. The
k
delay T s is the sampling period.

(a) Find the impulse response, h ( t ) , of the overall filter in the figure. Now find H ( f ) , the
frequency response.

The inpulse response for the integrator block (the first block) is u ( t ) (i.e. the step function). The
output is then h ( t ) = u ( t ) – u ( t – T s ) . The transfer function is
T
( – j 2π f ) -----s
H(f) = e 2 sinc ( fT
s )T s = T s e –j π fTs sinc ( fT s ) .

(b) Show that the output is a staircase approximation to g ( t ) .

Basically we take samples of g ( t ) with a sampling period equal to T s . For each sample we start
a pulse of height equal to the value of the sample and width equal to T s . This is a staircase
approximation of g ( t ) .

7) Suppose we have a message signal that is a sinusoid and we quantize it using a quantizer of L
levels. Assuming that the quantization error for each sample is uniformly distributed in the
Δ Δ
interval – ---, --- , determine the signal to quantization noise ratio (SNR) in terms of L . Determine
2 2
the SNR in terms of L if the message is changed to a triangular wave, again assuming that the
errors are uniformly distributed.

Δ2
The power of the quantization noise signal is ------ (see notes and do it by yourself without looking
12
at the notes). Now, if the message is a sinusoid of amplitude A , then the signal varies over a range
2
 2A
-------
2A  L A2
of A – ( – A ) = 2A . Then Δ = ------- . Hence the quantization noise power is --------------- = --------2- . Now,
L 12 3L
1--- 2
A
1--- 2 2 3
the signal noise power is A . Hence the SNR is 2 = --- L 2 .
---------
2 A- 2
--------
3L 2
 1 2
--- A
1 2 3
For a triangular wave of amplitude A , the signal power is --- A . Hence the SNR is --------- = L2 .
3 A 2
---------
3L 2

Note that this was done in the lecture.

Now repeated this if the message signal consists of the sum of two sinusoids of unit amplitude, one
with frequency equal to f 0 and the other with frequency equal to 1.1 × f 0 . Assume that the two
sinusoids have the same phase.

Note that in this case the signal varies from -2 to 2. We plot it below for f 0 = 1 :

x( t ) 0

−1

−2
− 10 −5 0 5 10
t

1 1
The average power of the signal is --- 1 2 + --- 1 2 = 1 . (Do this rigurously). The quantization level
2 2
4 1 1 3
spacing is Δ = --- . The SNR is -----2- = -----------2 = --- L 2 .
L Δ  --4- 4
------
12  L 
-----------
12
 ECE316
Communication Systems
E. S. Sousa

Problem Set 8 - Solutions

Review problems - Sampling, Quantization, and Digital

1. A signal has the spectrum given by X ( f ) = 1 – ---f- for f < B and 0 elsewhere. If B = 1 KHz
B
give the spectrum of the sampled signal for sampling frequency f s for the following cases: 1)
f s = 1.5 KHz, f s = 2 KHz, and f s = 2.5 KHz.

f s = 1.5 KHz

1.2
1

Xs ( f )

0
.2
0 2 4 6 8 10
0 f 10

f s = 2 KHz

1.2
1

Xs ( f )

0
.2
0 2 4 6 8 10
0 f 10

f s = 2.5 KHz
 1.2
1

Xs ( f )

0
.2
0 2 4 6 8 10
0 f 10

2. The following filter is used as the reconstruction filter for a sampled signal, where the message
of the original (unsampled signal) has a bandwidth B .

f
9 KHz 11 KHz

What is the maximum bandwidth that we can have for the message signal and still be able to
reconstruct the original signal perfectly?

We need a “gap” between two repeated spectra equal to 2 KHz.

fs

B = 9 KHz
The sampling frequency is 11 + 9 = 20 KHz.

3. Suppose we sample a message signal with bandwidth B at the rate of f s = 2B and then trans-
mit the signal (i.e. input the signal) into a bandpass channel with bandwidth W , where the fre-
quency response of the channel is constant and equal to 1. What is the minimum bandwidth W
that we can have and still be able to reconstruct the original signal?

The minimum bandwidth is B.

4. In 3) can we recover the signal for a channel with any center frequency, or do we need a spe-
cific center frequency in order to reconstruct the signal?

It can be any center frequency because there is enough information in the sampled signal over a
frequency span of B Hz.
5. A sampled signal has a rate of 10,000 sampes per second. Each sample is transmitted as a
square pulse with width given by 10 μ s, and height equal to the sample value. Plot the
power spectral density of the transmitted signal.

1
1

S( f ) 0.5

0 0
4 3 2 1 0 1 2 3 4
4 f 4

1
The frequency is normalized by ------ MHz = 100 KHz.
10

6. If instead of transmitting the square pulses we transmit sinc pulses, give an expression for a
sinc pulse such that there is no interference between adjacent samples, i.e. the heights of the
pulses can be detected at the receiver without and the bandwidth is minimum? What is the
power spectral density for the resulting signal?

πt
sin  -----
1  T
The symbol period is T = ------------------ = 10 –4 . The signaling pulse is ------------------ . The power spectral
10, 000  πt
-----
 T
is the following:

1.2
1

Π ( f)
0.5

0 0
2 1 0 1 2
2 f 2

where the bandwidth is 5 KHz.

7. Using the square pulses in 5) how many signals can we multiplex using TDM?

The symbol rate from each user is 10 4 . The period is 100 μ s. Hence we can support 10 users.

8. If we use sinc pulses as in 6) where the zero crossing are spaced 10 μ s apart, what is the
 power spectral density of the resulting transmitted signal?

The PSD is a rectangular function with bandwidth equal to 50 KHz.

9. In 8) how many users could we multiplex in the channel?

We can multiplex 10 users.

10. A PCM system is used to transmit a speech signal. If the bandwidth of the speech signal is 8
KHz, what should be the sampling frequency?

The minimum sampling frequency is twice the bandwidth or 16 KHz.

11. In 10. suppose we quantize the samples such that the maximum quantization error is 0.1 %,
how many levels will there be?

It should have said 0.1% of the peak.

The maximum quantization error is equal to 1/2 of the quantization step size. Let the peak be M.
Then peak to peak is 2M. Step size is equal to 2M/Q. M/Q = 0.1% of M or M/1000.
Hence Q = 1000.

12. In 11) how many bits do we need to represent each sample?

We need 10 bits because 2 10 = 1024 .

13. In 12) what is the resulting bit rate of the PCM signal?

Bit rate is 16000 × 10 = 160 Kbps.

14. In 13. Suppose that the signal is transmitted using binary transmission where the signaling
pulse is a cosine roll-off pulse with excess bandwidth equal to 20%. What is the bandwidth of
the channel?

1+α 1+α 1.2

The bandwidth is ------------- = ------------- R = ------- × 160, 000 = 96 KHz.
2T 2 2

15. In 14, now suppose that we use 8-ary transmission with the same pulse shape. What is the
bandwidth of the channel?

For 8-ary transmission there are 8 possible transmitted pulses, instead of 2. Hence we can transmit
3 bits per pulse. Hence the pulse rate is 160/3 = 53.33. The bandwidth is then 96/3 = 32 KHz.

16. The eye digram for a digital transmission system is similar to those plotted in the following dia-
gram where the scale accross the bottom of the plot is 50 μs . What is the bit rate for the trans-
mission scheme?

The “eye” diagram covers a span of two symbol periods. Hence the symbol period is 50/2 = 25 μ s.
Now this is an 8-ary scheme (alphabet size = 8). The symbol rate is 1/25 Msps = 40 Ksps. Hence
 the bit rate is 40 × 3 = 120 Kbps.

X
j
0
0

3
0.2 0.4 0.6 0.8 1 1.2 1.4 1.6
.2 mod tt , 2 1.8
j

17. In a digital transmission system the pulse used is sinc 2  --- , where T = 1 μs . Does this pulse
t
 T
satisfy the Nyquist one criteria? If so, for which bit rate?

Yes. The bit rate, or more precisely the symbol rate, is 1/T = 1 Msps.

18. In 17), what is the excess bandwidth?

The excess bandwidth is 100%. I.e. α = 1 .

19. There is an audio signal of bandwidth equal to 20 KHz that we wish to transmit over a digital
channel. The bandwidth of the channel is 1 KHz. We wish to have a received signal with a
quality given by SNR = 50 dB. What should be the bit rate for the digitized signal in order to
achieve this SNR? To transmit over the channel what level of modulation (i.e. size of trans-
mission alphabet) do we need if the pulse used is a sinc pulse?

We can assume a uniform distribution of signal and quantization error to obtain

( 10 5 -) = 16.6 , or n = 8.3 . Hence
SNR = Q 2 = 2 2n = 10 50 / 10 = 10 5 . Hence 2n = log--------------------
log ( 2 )
to guarantee the above SNR we can use 9 bits per sample. The sampling rate is 40 Ksps.
Hence the bit rate is 40 × 9 = 360 Kbps.

Now to transmit over a channel with bandwidth equal to 1 KHz the rate is 2 Ksps. Hence we need
180 bits per symbol in a transmission scheme. The alphabet size would be 2 180 . This is huge
and not achieveable in practice because the levels would be too close together.

20. In a binary transmission system the pulse used is a triangular pulse, where the base of the tri-
angular pulse has a width equal to the symbol period. Draw the “eye” diagram.
The eye diagram (opening) has a diamond shape. There are no overshoots in the signal.

21. In a digital transmission system the pulse shape used has an excess bandwidth of 50%. The sig-
nal is generated as a baseband signal and then modulated by a carrier using DSB modulation.
What is the spectral efficiency of the scheme in terms of bps/Hz?

1 + α-
+ α- = ------------
We assume binary transmission. The bandwidth for the baseband signal is 1------------ R . For
2T 2
the modulated signal the bandwidth is ( 1 + α )R . The spectral efficiency is then
1 - = ----------------
1 = 2--- = 0.66 bps/Hz
R - = ------------
---------------------
( 1 + α )R 1+α 1 + 0.5 3

22. In 21) suppose that we use quadrature modulation (QAM) what is the spectral efficiency of the
scheme in terms of bps/Hz?

In this case the spectral effiency increases by a factor of 2 because we double the bit rate for the
4
same bandwidth, or --- , or 1.33 bps/Hz.
3
πt
sin  -----
T
23. In a digital transmission system sinc pulses are to be used, i.e. p ( t ) = ------------------ . For the pur-
πt
-----
T
posed of analysis of the transmitted signal we will truncate the sinc pulses to 0 for t > 5T . We
N

will use binary transmission where the transmitted signal is x ( t ) =  dk p ( t – kT ) , with

k=0
d k = ± 1 . For a large value of N and a random set of bits, what is the largest possible value of
the signal (i.e. the peak value), x ( t ) ?

The signal is x ( t ) =  dk p ( t – kT ) . Now we need to pick a sequence of data values (i.e. ±1 ) so

k=0
that each term in the sum is positive. This is the same as converting each term in the sum to
N

p ( t – kT ) . The sum is then m ( t ) =  p ( t – kT ) . We have to pick the value t so that

k=0

m(t) =  p ( t – kT ) . The maximum is 2.334

k = –5
24. For the system in problem 23, assume that there is noise in the channel, where the maximum
value of the noise is 0.01. Assume a baseband channel with bandwidth equal to 2 MHz. What
is the maximum bit rate that we can achieve in the channel assuming that we will never incur
errors in the received data.

We need to decide on the alphabet size and determine the minimum noise margin. If the error can
have a maximum of 0.01 then the difference between the amplitudes of the pulses should be
at least 0.02. Let α = 0.01 . The pulses used will be αp ( t ) , 3αp ( t ) , 5αp ( t ) , ..., Mαp ( t ) ,
and their negatives for some M such that M is the maximum value with Mα < 1 .
1
M < ---------- = 100 . So for M = 99 , the number of symbols would be 50 plus the negatives or 100.
0.01
Now if we want this to be a power of 2 then we can go with a maximum of 64.
=======

Let the (positive) amplitudes be a , 3a , 5a , ..., La = 1 . Then the spacing is 2a and the number
L+1
of positive symbols is ------------ . The total number of symbols is Q = L + 1 = 1--- + 1 .
2 a
1 - . Now since the minimum value for a is
Alternatively we can write the spacing as a = ------------
Q–1
1
a = 0.01 , we need to find the largest Q , a power of 2, so that ------------- > 0.01 . I.e. Q < 101 .
Q–1
The solution is Q = 64 .

-------------
Now the maximum symbol rate (assuming sinc pulse) is R = 4 Mbps. The number of bits per
symbol is 6 because 2 6 = 64 . Hence the maximum bit rate is 4 × 6 = 24 Mbps.