3 Frequency Modulation

The other common form of analog data transmission that is familiar to most people is called
frequency modulation (FM). In the previous section, the message signal was multiplied with
the carrier, modulating its amplitude in order to be transmitted (this amplitude modulation).
With FM, the frequency of the carrier is modulated (varied) was the message changes. FM
is what is used to transmit the majority of radio broadcasts. It is generally preferred over
AM because it is less sensitive to noise, and it is in fact possible to trade bandwidth for
noise performance. The advantages of FM come at the expense of increased complexity in
the transmitter and in the receiver. This having been said, we will see a simple FM receiver
is actually no more complicated (to build) than its AM counterpart.
Conceptually, FM is pretty simple. If me are sending a message m(t), we send a higher
frequency wave when the amplitude of m(t) is high, and a lower frequency wave when the
amplitude is low. The example below shows a square wave message, and the corresponding
FM signal. The message here is called m
b
(t). The b stands for binary - the example signal
happens to only have two values - and dierentiates this example from the case of a general
message to which is discussed shortly.
t
1
-1
) (t m
b
t
1
-1
) (t s
FM
High amplitude High frequency
Low amplitdue Low frequency
Figure 70:
We can see that when the message m
b
(t) has a high (1) amplitude, the frequency of the
modulated signal s
FM
(t) is high. When m(t) goes low (-1), the modulated signal frequency
is reduced. Mathematically, we could describe this as
s
FM
(t) = Acos(2 [f
c
+ k
f
m
b
(t)] t). (38)
The A in front of this equation is just a constant which tells us the signal amplitude. The
constant k
f
is called the frequency sensitivity. It tells us how much the signal frequency
changes as as the message changes. From both the equation and Fig. 70, we can see that the
62
eective frequency of the cosine wave s
FM
(t) is f
w
(t) = f
c
+ k
f
m
b
(t). This translates into
f
w
= f
c
+ k
f
, m
b
= 1 (Higher Frequency)
= f
c
k
f
, m
b
= 1 (Lower Frequency).
What about for an arbitrary message m(t) ? We want the frequency of our cosine wave to
be
f
w
(t) = f
c
..
+ k
f
m(t)
. .
.
Carrier
frequency
Change due
to message
(39)
The next page or so of math culminates in Eq. 43 which tells us how to get the FM signal.
Intuitively, we might think that the FM signal would be determined using Eq. 38. This is
in fact incorrect. To get this right, we need to step back and think about the meaning of
frequency and phase. Lets start with an analogy from linear motion. If we dene position
x(t) and velocity v(t), we know the relationship between them is
v(t) =
d
dt
x(t). (40)
That is to say that velocity is the time rate of change of position. To obtain position, from
velocity, we integrate:
x(t) =
_
t
0
v(t)dt. (41)
The angular counterparts of position and velocity are phase and angular frequency =
2f. These are related in the same way. Using f, which is relevant here, the equation
corresponding to Eq. 41 is
(t) = 2
_
t
0
f(t)dt (42)
Now a cosine wave can be dened generally as c(t) = cos((t)). If we want a wave with a
xed frequency f
c
, we can calculate
(t) = 2
_
t
0
f
c
dt
= 2f
c
t
which gives the usual result c(t) = cos(2f
c
t). If we want a frequency that varies with time,
as in Eq. 39, we integrate

FM
(t) = 2
_
t
0
f
w
(t)dt
= 2
_
t
0
f
c
dt + 2
_
t
0
k
f
m(t)dt
= 2f
c
t + 2k
f
_
t
0
m(t)dt.
63
The FM signal is s
FM
(t) = Acos(
FM
(t)) and substituting the equation above gives
s
FM
(t) = Acos
_
2f
c
t + 2k
f
_
t
0
m(t)dt
_
. (43)
Equation 43 correctly indicates that the frequency (derivative of phase) of the signal is
proportional to the message m(t).
3.1 Generating an FM signal
With mathematical description of the FM signal now in hand, we must determine how to
build a circuit that will take m(t) as in input and give us s
FM
(t). This is of course a
modulator. The way these modulators work is interesting enough that it is worth discussing
a couple:
One option is to bypass altogether the previous discussion about integration and directly
control the resonant frequency of a circuit using m(t). This can be accomplished using a LC
resonant circuit incorporating a voltage variable capacitance diode, also called a varicap.
Cosine waves are typically generated by an oscillator with a circuit such as that in Fig. 71
governing the oscillation frequency. Look up Hartley or Armstrong oscillators to get a sense
of how this work. The capacitor, which would be more familiar in such as circuit, is replaced
with a variable capacitance diode. The diode capacitance changes depending on the voltage
across it. The message signal is presented across the capacitor, and the circuit is set up
(biased) so that the total capacitance C
T
is equal to
C
T
= C
0
+ k
c
m(t).
The resonance frequency of the LC circuit is f
0
= 1/(2

LC
T
) which will be the output
frequency of the oscillator.
In this example, the output frequency changes in a nonlinear fashion with the message;
however, for small changes in C
T
, we could show (using a series expansion) that the change
in output frequency is roughly proportional to m(t), that is
f
0
=
1

2LC
0
+ k
f
m(t).
This is of course Eq. 39, with the carrier frequency equal to 1/(2

LC
0
) and the frequency
sensitivity k
f
determined by the factor k
c
and the capacitances involved.
The second, less direct method of modulation uses an integrator followed by a phase
modulator. In the previous example, the frequency of oscillation was determined by m(t)
which implicitly took care of the integration in the signal equation 43. The integrated
64
T
C
) (t m
+
-
L
in
Z
Figure 71: Tank circuit of a voltage controlled oscillator. The capacitance C
T
depends on the
message signal m(t). This capacitance in turn governs the oscillation frequency of the circuit.
This LC circuit has a parallel impedance Z
in
m and would be connected to an appropriate
oscillator circuit.
message represents the phase of S
FM
, so we must come up with a circuit that can modify
the phase of a wave based on
_
m(t)dt. As you may recall, integration can be accomplished
with an operational amplier (OP-AMP) as shown in Fig. 72. Briey, we know that the
+
_
R
C
) (t m
+
-
) (t v
o
+
-
) (t i
x
Figure 72:
input terminals of the OP-AMP are at ground potential, and we know the currents owing
through the resistor and capacitor must me one and the same (equal). This gives us
i
x
(t) =
m(t)
R
= C
d
dt
v
0
(t).
Solving for the output voltage gives the expected integral
v
0
(t) =
1
RC
_
t
0
m(t)dt (44)
and puts us one step closer to frequency modulation. The next step is to use this integrated
signal to control the phase of a cosine wave.
A phase modulator circuit appears in Fig. 73. Similar to modulators we have seen
previously, this circuit has two inputs: the carrier, c(t) = Acos(2f
c
t), and the integrated
65
message signal, provided by the integrator discussed above. Phase shifting a cosine wave
means the addition of a phase angle:
Shifted Cosine = Acos(2f
c
t +
s
(t)).
Out goal is to make
s
(t) =
_
t
0
m(t)dt so that we wind up with Eq. 39.
T
C
L
R

t
dt t m
0
) (
) 2 cos( t f A
c
p
+
+
- -
) (t s
FM
Figure 73:
The circuit in Fig. 73 is a bandpass lter with a center frequency of f
0
= 1/(2

LC
T
).
The transfer function of this lter is shown in Fig. 74. Assume that the lter is designed so
f
1
f
o
90
o
90 -
0
f 0
f
) ( f H ) ( f H
Figure 74:
that when no voltage is applied to the varicap (when
_
m(t)dt = 0), the resonant frequency
is equal to the carrier frequency, f
0
= f
c
. When the integrated message signal is applied to
the capacitance C
T
will increase or decrease, causing a shift in f
0
. We can express this as
f = f
0
f
c
= k
1
_
t
0
m(t)dt.
For small deviations from f
0
, the lter has a response that can be approximated by the
dashed lines in Fig. 74. The gain is close to 1, and the phase varies linearly. When f = 0,
the carrier passes through the lter without any change in amplitude or phase since the phase
66
shift is zero at the center of the lter. A non-zero frequency dierence though, means that
the carrier is no longer passing through the center of the lter. For small f, this has little
eect on the carrier amplitude since we approximate the lter gain as 1. However, the phase
of the carrier will be shifted by k
2
f, with k
2
being the slope of the linear approximation
of the phase response.
The output signal will therefore be
S
FM
(t) = Acos(2f
c
t k
2
f)
= Acos(2f
c
t + k
1
k
2
_
t
0
m(t)dt).
Again, we have arrived at the correct frequency modulated signal in Eq. 43.
This second method of modulation is often preferred. A precision oscillator, such as a
crystal. can be employed to generate the carrier. This contrasts with the direct modulation
method wherein a tuned circuit is used to control the oscillator frequency. Such a tuned
circuit leaves the modulator more susceptible to frequency drift, due to temperature changes
in the circuit.
3.2 Demodulating FM
The goal of course is to extract m(t) back from s(t), a FM signal, picked up for example by
an antenna. Assuming the received signal has been ltered, it will contain only s(t). The
FM wave was generated by integrating the message signal so it is natural that dierentiation
is used in demodulation. The way that this works can be interpreted in both the time and
frequency domains and we will discuss both.
Start with the formula for an FM signal, a repeat of Eq. 43:
s(t) = Acos(2f
c
t + 2k
f
_
t
0
m(t)dt).
The time derivative of this waveform is
d
dt
s(t) = 2A(f
c
+ k
f
m(t)) sin
_
2f
c
t + 2k
f
_
t
0
m(t)dt
_
= 2Af
c
_
1 +
k
f
f
c
m(t)
_
sin
_
2f
c
t + 2k
f
_
t
0
m(t)dt
_
. (45)
In taking the derivative, the chain rule, and the fact that
d
dt
_
m(t)dt = m(t) have been
employed. Look at Eq. 36, which denes an amplitude modulated signal. You should be
able to see that Eq. 45 above looks very similar: the message signal, plus a constant, is
multiplied by a sine wave. The only dierences are some scaling constants, the choice of a
67
sine instead of a cosine carrier, and the frequency variation of the sine wave. Like with AM,
the envelope of this derivative signal now contains the original message and can be recovered
with an envelope detector, leaving the recovered signal
m(t) = 2Af
c
_
1 +
k
f
f
c
m(t)
_
. (46)
Recovery only depends on the envelope
k
f
fc
m(t) never falling below one, as in the AM case.
Figure 75 shown as block diagram representation of the demodulation procedure. The
envelope detector, consists of a diode rectier, followed by a RC lowpass lter, as in Fig. 50.
With this block taken care of, it is the dierentiator that presents a diculty.
) (
~
t s
) (
~
t m
dt
d Env.
Detect
) (t s
Figure 75:
A dierentiator can be made using an inverting op-amp as in Fig. 72 with the resistor and
capacitor interchanged. This is not usually done in practice, as the high frequency operation
of such a circuit is not reliable. To implement a dierentiator, we switch to analyzing the
demodulation process in the frequency domain.
The idea behind FM is that a larger m(t) results in a higher frequency, and a smaller
message a lower frequency. This was rst indicated in Fig. 70. What we are eectively
doing in converting voltage to frequency. Consequently, we will recover the voltage (m(t))
by nding a way to convert higher frequency back to a higher voltage and lower frequency
to lower voltage.
Think about the lter whose transfer function is shown in Fig. 76. The magnitude of the
transfer function varies linearly with frequency over the range f
c
W to f
c
+ W and has a
slope of p. Imagine now putting a wave with a frequency f

< f
c
into the lter, as in the
upper part of Fig. 77. The input to the lter has a lower frequency, so it will scaled by a value
somewhere towards the left hand side of the transfer function. The result is a comparatively
lower output amplitude. On the other hand, it we put a higher frequency wave, frequency f
h
into the lter, the transfer function shown that the output will be comparatively higher. This
lter does exactly what we have set out to do: the amplitude of the output is proportional to
the frequency of the input. This may be more easy to see with the third example in Fig. 78.
The wave going in to the lter has a frequency that varies with time. When the frequency
is higher, the lter lets more of the wave through, and when it is lower, less gets through.
The output of the lter now has an amplitude which follows the frequency variations in the
68
f
) ( f H
W f
c
- W f
c
+
c
f
slope, p
pW
l
f
h
f
Figure 76:
) ( f H
) ( f H
) 2 cos( t f
l
p
) 2 cos( t f
h
p
INPUT
OUTPUT
Figure 77:
input wave. We have just created a lter which acts as a frequency to voltage converter
which will be able to undo frequency modulation and recover m(t). Of course, the output is
still a sine wave. Only its amplitude is proportional to the input frequency. We recover the
message signal now by running the lter output through an amplitude detector. Looking
) ( f H
INPUT
OUTPUT
Figure 78:
back at the block diagram for FM demodulation (Fig. 75), you can safely conclude that the
lter we have just described acts as the dierentiator.
Now that we know the transfer function that will dierentiate our signal, we need to look
at how to implement such a lter. The answer is actually in the bandpass lter response in
69
Fig. 73. We will reproduce it here. Instead of focusing on the center of the lter response, we
shift our attention to the transition band. As highlighted in Fig. 79, you can see that there is
a region where the rollo of the bandpass lter varies almost linearly with frequency. In this
region at least, the response looks like the ideal in Fig. 76. We know that out FM signal will
f
) ( f H
c
f
Figure 79:
have a frequency that varies around the carrier f
c
. If we design a bandpass lter, with the
linear region of the transition band centered at f
c
, we have a good chance of dierentiating
the FM wave. Again, the operation of dierentiation is realized by creating a lter which
converts frequency variations back to amplitude variations. The only condition is that the
frequency sensitivity k
f
is chosen so that the frequency of the FM signal does not go outside
the linear region of the lter. As you may recall, k
f
tells us how the frequency of the FM
signal changes in proportion to the message - look back at Eq. 43.
3.3 FM Example
Before embarking on a more mathematical description of how the dierentiator works, lets
pause and use what we already know to do a complete example of the FM modulation /
demodulation process. The message m(t) will be the same as in Fig. 51: a sound recording
of a male speaker saying hello.
Fig. 80 (a) shows a short section of the speech waveform. This is the same section that
was examined in the AM case of Fig. 51. The dierence here is that the signal has not been
scaled and shifted. Although the scale is dierent, you can look back at Fig. 51 (c) and see
that the envelope there is the same shape as the signal here.
To transmit the signal with FM, we begin by calculating k
f
_
t
0
m(t)dt. This is plotted,
over the same small section, in plot (b) of Fig. 80. The amplitude here is just over 200
Volts. This has occurred for two reasons: k
f
was made large (300000/2) to accentuate
the modulation and make the FM waveform that we will see shortly easier to visualize; the
integrated signal has a large oset, the cumulative eect of integrating m(t) for the rst (not
70
310 315 320
-100
-50
0
50
100
Time (ms)
A
m
p
l
i
t
u
d
e

(
m
V
)
310 315 320
190
200
210
220
230
Time (ms)
A
m
p
l
i
t
u
d
e

(
V
)
(a)
(b)
1
2
3
Figure 80:
shown) 310 ms of the waveform. You can look at plot (b) and see that it decreases when
m(t) is below zero and increases in the opposite case, hopefully convincing yourself that the
second plot does represent the integral of the rst.
Having obtained the integral of the signal, we employ Eq. 43 to generate an appropriate
FM wave s(t). This is displayed in Fig. 81 (a). For clarity, the plot has been zoomed in
slightly, to span a total of 6 ms. We have used a carrier frequency f
c
=7000 Hz, and an
amplitude of 1. The points labelled (1,2,3) in the gure correspond to the signal minima
310 312 314 316
-1
-0.5
0
0.5
1
Time (ms)
A
m
p
l
i
t
u
d
e

(
V
)
Frequency (KHz)
(a)
(b)
1 2 3
0 5 10 15
0
20
40
60
A
m
p
l
i
t
u
d
e

(
m
V
)
Figure 81:
(1,2,3) in m(t) identied in Fig. 80 (a). For the lower amplitude regions of m(t), the FM
wave has a lower frequency while it has a higher frequency for the higher amplitude points
in between. The amplitude of s(t) is xed at 1, with all the message information contained
in the frequency variation.
Plot (b) in Fig. 81 shows the frequency domain representation S(f) of the FM signal.
Only the upper half of the frequency spectrum is shown. We see a peak at 7 KHz which can
be expected. The carrier frequency is 7 KHz and the frequency of the transmitted signal
varies around this value according to the message. There are a number of dierences between
71
this frequency spectrum and that of the AM signal seen earlier in Fig. 51. In that gure,
M(f) and S(f) are both shown for the AM example, with S(f) essentially as shifted copy
of M(f). The spectrum of the FM signal is not composed of shifted copies of the message
spectrum. As we will see in section 3.5, the relationship between M(f) and S(f) for FM is
actually much more complicated. Looking at M(f), there is no easy way of telling what the
message spectrum might look like. Another important dierence is that S(f) spans a greater
frequency range than it did for AM. The way to think about this is that AM simply shifts
the message over in the frequency domain so its bandwidth must remain the same. In FM,
the amount that the frequency varies really depends on the magnitude of k
f
. The choice of
this parameter is therefore what determines the width of the signal spectrum, rather that
the message itself.
0 5 10 15
0
10
20
30
Frequency (KHz)
G
a
i
n
0 5 10 15
0
0.2
0.4
0.6
0.8
1
Frequency (KHz)
G
a
i
n
(a)
(b)
310 315 320
-1.5
-1
-0.5
0
0.5
1
1.5
Time (ms)
A
m
p
l
i
t
u
d
e

(
V
)
(c)
Figure 82:
At the receiver, s(t) is demodulated with a dierentiator and envelope detector. The
dierentiator here has been realized as a series RLC bandpass lter. The lter is designed
to have a resonant frequency of 13 KHz and a bandwidth of 10 KHz, giving a response as
indicated in Fig. 82 (a). The resonant frequency and bandwidth were chosen so that the
lter response would increase almost linearly with frequency in the range around the carrier
at 7 KHz. You can see that the response shown in the plot is not entirely linear, but it will
in fact be good enough to recover our signal. The dierentiated signal s(t) is obtained by
passing s(t) though this lter. The corresponding spectrum

S(f) is shown in Fig. 82 (b).
The spectrum has now been emphasized at higher frequencies compared to S(f), but the
resulting spectrum still does not look like M(f).
72
The eect of what we have done becomes clear in the time domain, s(t) shown in plot
(c). As in the previous discussion, the amplitude of the higher frequency sections of s(t)
becomes higher, and that of the lower frequency sections lower. Looking at the envelope
s(t), you can now see that it follows the shape of m(t) from Fig. 80 (a).
310 315 320
-1.5
-1
-0.5
0
0.5
1
1.5
Time (s)
A
m
p
l
i
t
u
d
e

(
V
o
l
t
s
)
-20 -10 0 10 20
0
5
10
15
Frequency (kHz)
A
m
p
l
i
t
u
d
e

(
m
V
)
-5 0 5
0
1
2
3
4
5
Frequency (KHz)
A
m
p
l
i
t
u
d
e

(
V
o
l
t
s
)
310 315 320
0
100
200
300
Time (s)
A
m
p
l
i
t
u
d
e

(
m
V
)
(a)
(b)
(c) (d)
Figure 83:
We can now rectify and lter s(t) to get back m(t), just as was done for the AM signal.
The steps are summarized in Fig. 83. In plot (a), the dierentiator output has been rectied
by a diode. The eect in the frequency domain is shown in plot (b). The spectrum is a mess.
We can see that there is something that looks like S(f) at around 7000 KHz, and some
high frequency interference has been added as well. The dierence now though is at the
center of the spectrum. Looking back to Fig. 52, this central portion may appear familiar:
it is the spectrum of the original hello waveform we have been studying. The only major
dierence is the spike in the center due to the DC oset. We knew that this had to be the
case - we could see m(t) in the envelope of s(t) - but it should be gratifying nonetheless.
After ltering, we get the recovered M(f) by itself (plot (c)), and can see that it looks nearly
identical to that in Fig. 51.
Returning to the time domain, plot (d) shows the demodulated signal over the short
time interval. Compared with the original in Fig. 80, we can see that the demodulated
signal is close but not quite identical. There are a number of reasons that this is the case;
FM is nonlinear and by nature allows may allow corrupting inuences to modify slightly
73
the signal being transmitted. The dierentiator that we used to recover the message was
not perfectly linear; this may explain why the recovered signal is most corrupted at low and
high amplitudes (corresponding to frequencies in s(t) furthest from the central linear region
of the dierentiator). The larger bandwidth of s(t) may have encroached into that of m(t)
during demodulation (see Fig. 83 (b)). Regardless, the signal has been recovered, and when
played over speakers in fact sounds clearer than the original, due to the ltering away of
high frequency noise.
3.4 The Dierentiator
We have seen that a dierentiator can be implemented over a narrow frequency range using
the transition band of a bandpass lter. This is sucient to understand how an FM demod-
ulator works; however, a more mathematical discussion of how the dierentiator works gives
some additional insight into the process. Specically, we will show why the lter works, and
develop an equation which tells us the amplitude of s(t).
We will start by determining the frequency response of an ideal dierentiator. Suppose
we have a signal g(t) with a frequency spectrum G(f). Since the former is the inverse Fourier
transform of the latter, we can write
g(t) =
_

G(f)e
j2ft
df (47)
which is Eq. 21. We take the time derivative of both sides of the equation by putting a d/dt
in front of each:
d
dt
g(t) =
d
dt
_

G(f)e
j2ft
df. (48)
The integral is with respect to time so the derivative can be moved inside without changing
the meaning. The term G(f) also appears as a constant to the dierentiation so we get
dg(t)
dt
=
_

G(f)
d
dt
_
e
j2ft

df
=
_

[j2fG(f)] e
j2ft
df.
By moving the dierentiation inside of the equation, we ended up just taking the derivative
of exp j2ft. Grouping terms as in the second line of the equation, it is apparent that it
is now the expression in the square brackets which is being inverse Fourier transformed. In
words, this says the inverse Fourier transform of j2fG(f) is
d
dt
g(t). This can be expressed
as the time dierentiation property of Fourier Transforms:
dg(t)
dt
j2fG(f); (49)
74
taking the derivative in the time domain is the same as multiplying by f in the frequency
domain. Thus, a lter with a transfer function of H(f) = j2f will be a dierentiator. The
transfer function of such a lter is plotted in Fig. 84. The j2 is just a scaling constant so
f
slope,
) ( f H
p 2 j
Figure 84:
that any linear function of f, like the lter we implemented in FM demodulation, will serve
the same purpose.
A simple way to see why this works is to look at the case of a sine wave with a frequency
of f
0
. The derivative of this wave is
d
dt
sin 2f
0
t = 2f
0
..
cos(2f
0
t)
. .
.
= Derivative
proportional
to frequency
Phase shift of
90 deg.
The factor of 2f is apparent. The j that shows up in Eq. 49 is reected in the change from
a sine to a cosine. Since any signal can be split into sines and cosines (what a FT does), we
can see how the time dierentiation property comes about.
We can now use the time dierentiation property to examine how the bandpass lter
demodulates the FM signal. Assume that the signal bandwidth is 2W, and that the lter
response looks like the plot on the left in Fig. 85. It is linear from f
c
W to f
c
+W, and has
a slope of j2. This lter diers from the ideal dierentiator as it is shifted from f = 0: the
ideal dierentiator response crosses zero at f = 0 but this has been shifted over by f
c
W.
It will actually be easier if we assume the lter has been shifted over by f
c
and shifted up
as well, so we break the lter response down into a at part H
1
(f) and sloped part H
2
(f)
as in the gure.
We assume that the FM signal going into the lter is dened as usual by
s(t) = Acos
_
2f
c
t + 2k
f
_
t
0
m(t)dt
_
. (50)
75
f
W f
c
- W f
c
+ c
f
W j p 2
f
W f
c
- W f
c
+ c
f
slope,
W j p 2
p 2 j
f
W f
c
- W f
c
+ c
f
+
=
slope, p 2 j
) (
1
f H
) (
2
f H
) ( f H
Figure 85:
When s(t) goes through H
1
(f), it is simply scaled by a factor of j2W. The j eectively
turns the cosine into a negative sine so we can write the output of this lter as
s
1
(t) = 2AW sin
_
2f
c
t + 2k
f
_
t
0
m(t)dt
_
.
The situation is trickier for H
2
(f). We know that if we shift H
2
(f) down by f
c
, we would
have an ideal dierentiator. We can use this to determine the output of this lter by shifting
s(t) down in frequency by f
c
, dierentiating it, and shifting it back up. Shifting the signal
down is essentially accomplished by dropping the 2f
c
t term inside of the cosine function
- we are eectively only dierentiating the baseband part of the signal. Think of this as
applying the chain rule to dierentiate but not including the 2f
c
t term when taking the
derivative of the inside part. The output of the lter is
s
2
(t) = 2k
f
m(t) sin
_
2f
c
t + 2k
f
_
t
0
m(t)dt
_
.
Putting this together, we determine the combined dierentiator output to be
s(t) = 2AW
_
1 +
k
f
W
m(t)
_
sin
_
2f
c
t + 2k
f
_
t
0
m(t)dt
_
. (51)
This is important because the term in the square brackets must always be larger than zero
for diode detection to work, just like in AM. As you can see, this actually depends on the
width of the modulated signal (or the slope of the dierentiator response), giving us another
thing to think about when designing an FM system.
76
3.5 FM Mathematics
Although it required more thought in the derivation, we can see that in their simplest form,
FM modulation and demodulation are barely more complex than for AM. We can modulate
with the addition of a lter network, and demodulate with a similar addition. Despite
this simple description, the nonlinearity of the modulation Equation 43 makes FM more
challenging to analyze mathematically.
The goal of our mathematical analysis of FM is to determine the bandwidth of the
modulated signal. As with AM, it will be necessary to understand this bandwidth and the
factors that aect it if we wish to share a channel amongst messages modulated at dierent
carrier frequencies. With AM, the signal bandwidth was simply that of the message (or half
that for SSB). We have already observed that this is not the case for FM. The bandwidth
will depend on some way of k
f
, with the modulated signal spectrum bearing no resemblance
to that of the message.
We know that the general form of the FM signal is
s(t) = Acos
_
2f
c
t + 2k
f
_
t
0
m(t)dt
_
. (52)
To keep things compact, we dene the excess phase
(t) = 2k
f
_
t
0
m(t) (53)
so that
s(t) = Acos(2f
c
t + (t)). (54)
Now, we use the trigonometric identity
cos(a + b) = cos a cos b sin a sin b (55)
to rewrite Eq. 52 as
s(t) = A[cos((t)) cos(2f
c
t) sin((t)) sin(2f
c
t)] . (56)
We rst assume that the modulation is narrow band, meaning that at all times, |(t)| 1.
Remember, this angle is proportional to the frequency sensitivity k
f
which tells us how much
the signal frequency changes as the message changes. The narrow band condition essentially
means that the frequency of s(t) does not change vary much from f
c
. In practice, this
approximation is valid for about max |(t)| < 0.3.
77
Under this narrow band condition, the small value of (t) lets us approximate cos((t))
1 and sin((t)) (t) to give
s(t) = A

:
1
cos((t)) cos(2f
c
t) A

:
(t)
sin((t)) sin(2f
c
t)
= Acos(2f
c
t) A(t) sin(2f
c
t). (57)
Looking at this equation, we can see that what has happened is actually a kind of
amplitude modulation. Compare Eq. 57 to Eq. 36. The rst term in Eq. 57 is the carrier, a
cosine at frequency f
c
. The second term consists of (t), modulated with a sine wave at the
carrier frequency. Remember that (t) = 2k
f
_
m(t)dt. We can think of (t) as the new
message which is just being amplitude modulated. The situation in the frequency domain
c
f
c
f - f
) ( f S
2
1
) ( f Q
) 0 ( Q
) (
2
1
c
f f
j
+ Q
2
1
) (
2
1
c
f f
j
- Q -
Figure 86:
is explained in Fig. 86. The angle (t) has a frequency representation (f), given by its
Fourier transform. This is related to the message by
(f) = F{2k
f
_
t
0
m(t)dt}
so it can be calculated given a known message. Upon modulation, the sine term in Eq. 57
splits the spectrum and shifts it left and right. The cosine term, the carrier, results in a
spikes at f
c
. Notice that the copies of the spectrum (f) have opposite signs and are
multiplied by a factor of 1/(2j) because the decomposition of a sine function into complex
exponentials (responsible for the frequency shifting)
sin(2f
c
t) =
e
j2fct
e
j2fct
2j
.
Comparing what has happened to our previous studies of AM, we conclude that in the
narrow band case, FM is really akin to AM, with the integrated message modulated with a
sine instead of a cosine.
78
The above analysis was possible because we linearized the equation for s(t). That is, we
used the assumption that (t) was small to represent the nonlinear functions cos((t)) and
sin((t)) with linear functions 1 and respectively. If (t) is large, this linearization is not
possible.
What we want to do is come up with a mathematical way of relating m(t) to the nal
FM bandwidth. Lets start again by looking at the FM signal represented Eq. 56. What we
really have is two functions cos((t)) and sin((t)) amplitude modulated with a cosine and
sine carrier respectively. Each of these functions can be interpreted as a message being
amplitude modulated. We can make this clear by rewriting
s(t) = A[cos((t)) cos(2f
c
t) sin((t)) sin(2f
c
t)]
= Am
1
(t) cos(2f
c
t) Am
2
(t) sin(2f
c
t). (58)
We have now dened two new message functions
m
1
(t) = cos((t)) = cos(2k
f
_
t
0
m(t)dt) (59)
m
2
(t) = sin((t)) = sin(2k
f
_
t
0
m(t)dt) (60)
and we can see from Eq. 58 that these new messages are being amplitude modulated, the
rst with a cosine carrier and the second with as sine carrier. We already know how to
analyze the bandwidth of this amplitude modulation and so have all the steps necessary to
go from the original message m(t) to the FM waveform. Specically, we start with m(t)
to determine the new functions m
1
(t) and m
2
(t). These will have frequency spectra M
1
(f)
and M
2
(f) with a dened bandwidth. We then analyze these as if they are being amplitude
modulated - each is split and shifted up and down by f
c
to arrive at the frequency spectrum
of our FM signal.
Lets look at a specic example of this procedure, and use it to develop a general way of
estimating the bandwidth of an FM signal. Consider a message m(t) that consists of a lone
cosine wave. This is called a single tone signal since a cosine wave represents the waveform
of a pure sound tone (like the whistle in Fig. 5). This message message is
m(t) = a cos(2f
0
t). (61)
We want the instantaneous frequency to be f
i
= f
c
+k
f
m(t) and so corresponding phase
79
is
(t) = 2k
f
_
t
0
m(t)dt
= 2k
f
_
t
0
a cos(2f
0
t)dt
= 2k
f
a
2f
0
sin(2f
0
t)
=
k
f
a
f
0
sin(2f
0
t). (62)
For single tone FM, we know that the largest value of m(t) will be the amplitude a so the
largest instantaneous frequency change will be f = k
f
a which is called the frequency devi-
ation. We will see later that the impact of this frequency deviation on the signal bandwidth
depends on the tone frequency f
0
. For this reason, we dene the modulation index which
is
=
k
f
a
f
0
=
f
f
0
=
frequency deviation of tone
frequency of tone
(63)
so the FM signal is
s(t) = Acos(2f
c
t + sin(2f
0
t)). (64)
t
a
0
1
f
) (t m
) (t s
t
t
0
1
f
) (t q
b
Figure 87:
One cycle of m(t) is shown in Fig. 87. The length of one period is T = 1/f
0
as indicated.
Also shown is the nal FM signal s(t). You can see that it is a wave whose frequency is lower
in the middle of the plot (where m(t) has a lower amplitude) and higher at the sides. In this
case, the carrier frequency is f
c
= 20f
0
and = 5. The excess phase (t) is also shown. As
we calculated, it is a sine wave with an amplitude equal to the modulation index.
80
t
0
1
f
t
0
1
f
) (
1
t m ) (
2
t m
Figure 88:
To calculate the frequency spectrum of s(t), we determine rst use Eq.s 59 and 60 to
determine m
1
(t) = cos( sin(2f
0
t)) and m
2
(t) = sin( sin(2f
0
t)). Each of these message
functions is plotted in Fig. 88, over one period of m(t). Remember that m(t) is assumed to
be a cosine wave going on for ever so m
1
(t) and m
2
(t) are also innite periodic functions.
The new messages m
1
(t) and m
2
(t) are now amplitude modulated according to Eq. 58 to
get s(t). This is best analyzed in the frequency domain. We start by determining the spectra
M
1
(f) and M
2
(f) which will be split and shifted up and down during the modulation. The
time domain functions are nonlinear - calculating their fourier transforms is possible but
mathematically complex. Instead, we will evaluate these spectra using a computer (FFT of
a ten period long signal) which gives the approximate answer. We will use this result for our
ultimate goal of deriving an expression for S(f). The spectra M
1
(f) and M
2
(f) which are
the Fourier transforms of m
1
(t) and m
2
(t) are shown in Fig. 89. Only the magnitudes are
shown. Each spectrum is composed of a series of peaks. This makes sense because we know
that m
1
(t) and m
2
(t) are periodic waveforms. The peak separation in each case is 2f
0
, and
we can see that M
1
(f) has peaks at frequencies which are even multiples of f
0
while M
2
(f)
only has peaks at the odd multiples.
To get the spectrum of s(t), we must take each of the spectra in Fig. 88 and split it an
shift it up and down by f
c
. Again, this comes from the interpretation of FM as amplitude
modulation of the messages m
1
(t) and m
2
(t) as in Eq. 58. To verify this, we use a computer
to determine the FT of s(t) which was plotted in Fig. 87. The resulting spectrum S(f) is
plotted in Fig. 90. As we expect, the spectrum is centered around f
c
. If we look closely
at one side of the spectrum, as in the plot in the right hand side, we can see that it consists
of a combination of the peaks present in M
1
(f) and M
2
(f). Since these peaks occurred at
alternating frequencies in the baseband signals, they do not interfere with each other after
modulation. The peaks in S(f) are now each separated by f
0
Hz. From this plot, we can
determine the bandwidth of the modulated spectrum. If we consider the few smaller peaks
81
f
f
0
0
4 f
0
4 f -
0
8 f -
0
8 f
0
f
0
f -
0
5 f
0
5 f -
| ) ( |
1
f M
| ) ( |
2
f M
Figure 89:
f
f
0
c
f
c
f -
) ( f S
c
f
0
6 f f
c
+
0
6 f f
c
-
) ( f S
Figure 90:
at the sides to be insignicant, this bandwidth is about 12f
0
.
We now know the FM spectrum and approximate bandwidth for a single tone message.
However, we needed to use a computer to calculate the spacing and magnitude of the peaks
in the spectrum. Understanding how we got this spectrum allows us to proceed to the usual
mathematical analysis of single tone FM. Rather than calculate and analyze m
1
(t) and m
2
(t),
we can actually start with
s(t) = Acos(2f
c
t + sin(2f
0
t))
and use a variation of a mathematical technique called the Jacobi Anger Expansion to
rewrite it as
s(t) = A

n=
J
n
() cos(2(f
c
+ nf
0
)t). (65)
This equation may look dicult to understand but we can break it down as follows: we
know that cos(2f
i
t) will have a frequency domain representation consisting of peaks at
f = f
i
, each with an amplitude of 1/2. Since this is symmetrical for positive and negative
frequencies, lets only worry about the positive frequencies.
82
The cos(2(f
c
+ nf
0
)) term in Eq. 65 has f
i
= f
c
+ nf
0
, so in the positive frequencies,
this means peaks as
f
c
+ nf
0
, n = , 2, 1, 0, 1, 2,
As shown in Fig. 91, this means a peak at the carrier f
c
, and another peak every f
0
Hz in
both directions. The J
n
() term in Eq. 65 tells us how high each of these peaks will be in
f
...
...
c
f
0
f f
c
+
0
2 f f
c
+ 0
f f
c
-
0
2 f f
c
-
2
1
Figure 91:
the spectrum of s(t). The function J
n
() is called a Bessel function. Since is xed for
a given modulator and message, this is really just a function of n. If you look back at the
Eq. 15 which describes a Fourier cosine series, you can see that J
n
() acts in the same way
that a
a
does there. In fact, we can make Eq. 65 easier to look at by re-writing it as
s(t) = A

n=
c
n
cos(2(f
c
+ nf
0
)t). (66)
Bessel functions do not have a closed form solution, and typically need to be looked up or
evaluated on a computer. To give a sense of what these coecients look like, Fig. 92 shows
the coecients c
n
= J
n
() for various values of . A property of Bessel functions is that
|J
n
()| = |J
n
()|. If we are only interested in the magnitude of the coecients, these plots
can simply be reected about n = 0 to give us the coecients for negative n.
You can see in the plots that for smaller , the coecients quickly go to zero; they take
longer to do so as increases. It is these coecients that tell us the amplitudes of the peaks
shown in Fig. 91 which come from the cosine term in Eq. 65. Fig. 91 is thus modied -
examples of this modication are shown in Fig. 93. The actual height of the coecients will
depend of and A according to c
n
= AJ
n
()/2. Figure. 93 plots the spectrum S(f) for
positive frequencies only and for values of 1, 2, and 5. The increasing values correspond
to an increase in k
f
, which increases f and thus . You can see that the peaks are spaced
apart by f
0
as in Fig. 91 and that the amplitudes are given by the coecients plotted in
Fig. 92. If you look at the plot for = 5, you will see that it is exactly the plot we found
by treating FM as AM of m
1
(t) and m
2
(t). The advantage of the bessel function approach
is that it formalizes the procedure and gives us values for the amplitudes of the peaks in the
frequency spectrum.
83
0 2 4 6 8 10
-1
-0.5
0
0.5
1
n
J
n
(
0
.
1
)
0 2 4 6 8 10
-1
-0.5
0
0.5
1
n
J
n
(
0
.
5
)
0 2 4 6 8 10
-1
-0.5
0
0.5
1
n
J
n
(
2
)
0 2 4 6 8 10
-1
-0.5
0
0.5
1
n
J
n
(
0
.
3
)
0 2 4 6 8 10
-1
-0.5
0
0.5
1
n
J
n
(
1
)
0 2 4 6 8 10
-1
-0.5
0
0.5
1
n
J
n
(
5
)
Figure 92:
84
) 5 ( 2
0
f
) 1 ( 2
0
f
) 2 ( 2
0
f
c
f
c
f
c
f
1 = b
2 = b
5 = b
0
f
Figure 93:
After all this work, it is best to summarize single tone FM as follows: if you are given
m(t) = a cos(2f
0
t), rst use the frequency sensitivity k
f
to calculate = k
f
a/f
0
. Look at a
plot like the one in Fig. 92 to determine the shape of the spectrum for the you found. The
spectrum S(f) will consist of the peaks from Fig. 92, centered at f
c
and spaced apart by
a frequency f
0
. The peaks from Fig. 92 will of course be mirrored around f
c
as well. Look
back at Fig. 90 and make sure you understand how it can be computed using the procedure
above.
Our goal in all of this is to determine the eective bandwidth of the signal. This band-
width could be dened as the width over which the spectrum has a signicant amplitude -
it never quite gets to zero but we can neglect values of c
n
which are very small. In Fig. 93, a
width equal to 2f
0
is indicated in each plot. You can see that most of the signicant (non
zero) part of the spectrum is within this range. If we extend the range by an extra f
0
on
either side, we come very close to capturing all the signicant peaks in the spectrum. This
has led to an empirical equation, known as Carsons rule, which states that
Bandwidth 2(f
0
+ f
0
). (67)
This says that the eective signal bandwidth is given by the span specied in the gure, plus
an extra two peaks. You can see that it is not exact - but it is simple to use and is a good
rule of thumb for bandwidth calculations. Using the denition = f/f
0
, Carsons rule is
often written as 2(f + f
0
).
Remember that this derivation was for a purely sinusoidal m(t) = a cos(2f
0
t). For an
85
arbitrary message, there will be no analytical expression for the signal bandwidth. However,
if we know that the message has a bandwidth of 2W and is centered at f = 0 (see eg.
Fig. 41), then the highest frequency wave present in the message has a frequency f
0
= W.
The maximum possible frequency deviation for an arbitrary m(t) will be f = k
f
max |m(t)|.
Inserting these worst case values into Carsons rule gives
Bandwidth 2(k
f
max |m(t)| + W). (68)
This equation gives the worst case scenario: it is not likely that the maximum signal ampli-
tude will actually be associated with the highest frequency component in the signal. However,
it is again the quickest way of calculating the approximate bandwidth. If a more accurate
measure is needed, it analyzing the spectrum of s(t) either using hardware measurements or
computer experiments would be necessary.
86