1

Angle Modulation
Prof. Bikash Kumar Dey
IIT Bombay

Motivation:
Angle modulation keeps the amplitude of the modulated signal constant, and it allows achieving higher
SNR at the receiver output at the cost of more transmit bandwidth. This trade-off is not available in
amplitude modulation.
Types of angle modulation:
1. Frequency modulation
2. Phase modulation
3. Some form of combination, as will be discussed later. Actually practical so called FM modulation uses
a combination of FM and PM.
Angle modulation:
Consider a sinusoidal signal A cos θ(t).
Instanteneous phase := θ(t)
Instanteneous (angular) frequency wi (t) := θ̇(t)
For example, the phase of A cos(wc t + θ0 ) is (wc t + θ0 ), and the instanteneous frequency is wc .
In angular modulation, the phase of the carrier signal A cos wc t is changed according to some function
of m(t).
Phase modulation (PM):

θ(t) = wc t + θ0 + kp m(t)
where kp is a constant, wc is the carrier frequency. Wlog, we assume θ0 = 0. Then
θ(t) = wc t + kp m(t)
So the modulated signal is
sP M (t) = A cos(wc t + kp m(t)).
The instanteneous frequency of this signal is
wi (t) = wc + kp ṁ(t) (1)

Frequency modulation (FM):

wi (t) = wc + kf m(t)
 2

where kf is a constant, wc is the carrier frequency. The instanteneous phase is

Z t
θ(t) = (wc + kf m(τ ))dτ
0
Z t
= wc t + kf m(τ )dτ (2)
0
assuming that θ0 = 0. The modulated signal is
 Z t 
sF M (t) = A cos wc t + kf m(τ )dτ
0

Relation between PM and FM:

Eq. (1) and (2) clearly means that each of FM and PM can be done using the other as shown in the
figures below.
Phase modulation can be done as
m(t)
.
m(t) s PM (t)
d
FM
dt

Frequency modulation can be done as

m(t) a(t) s FM (t)
PM

In both PM and FM, the phase/angle of the carrier is varried according to some function of the message
signal.
Note that, in both the figures, the differentiator and the integrator are linear time-invariant (LTI) filters.
For instance, in the first figure, the filter has the transfer function H(s) = s.
In general, any processing as shown below
m(t) Filter a(t) s(t)
PM
H(s)

where H(s) is an invertible filter, is an angle modulation, which produces the modulated signal
 Z t 
s(t) = A cos wc t + kf m(τ )h(t − τ )dτ
0
Similarly, if FM can be used:
m(t) Filter a(t) s(t)
FM
H(s)

Then the modulated signal will be

 Z t 
s(t) = A cos wc t + kf a(τ )dτ
0
 Z tZ τ 
= A cos wc t + kf m(τ1 )h(τ − τ1 )dτ1 dτ
0 0
 3

In commercial FM audio broadcast, a “pre-emphasis filter” is used together with an FM modulator. The
details will be discussed later.
Power of an angle modulated wave:
A2
Since the amplitude A remains constant, the power is always 2
.
Example 1 (Lathi; Example 5.1)

Sometimes, when the derivative does not exist, or the signal is not even continuous, one needs to take
the direct approach.
 4

Bandwidth of an angle modulated signal

Theoretical insight:
We restrict to FM and mention the PM case later. Let
Z t
a(t) := m(τ )dτ
0
and
s̃F M (t) = Aej(wc t+kf a(t)) (3)
so that
sF M (t) = A cos(wc t + kf a(t))
= Re [s̃F M (t)]
Expanding (3), we get
s̃F M (t) = Aejwc t · ejkf a(t)
kf2 2 kn
 
n f n
= A 1 + jkf a(t) − a (t) + · · · + j a (t) + · · · · (cos wc t + j sin wc t)
2! n!
Here we used ex = 1 + x + x2 /2! + · · · . So
sF M (t) = Re [s̃F M (t)]
kf2 2 kf3 3
 
= A cos wc t − kf a(t) sin wc t − a (t) cos wc t + a (t) sin wc t + · · · (4)
2! 3!
 5

Now note that

1) The first term is the carrier term at wc .
2) The second term has spectrum A(f ) shifted to ±wc . If m(t) has bandwidth B, i.e., M (f ) = 0 for
|f | > B, then A(f ) also has bandwidth B, as a(t) is obtained by filtering m(t) (impulse response
1
jw
.) So the bandwidth of the second term is 2B.
2
3) a (t) has FT A(f ) ∗ A(f ), and so the third term has bandwidth 4B around ±wc .
4) In general, an (t) has FT A(f ) ∗ A(f ) ∗ · · · ∗ A(f ) (n times). So, the (n + 1)-th term has bandwidth
2nB around ±wc .
Since there are infinitely many terms, and the terms have increasing bandwidth around wc , the ideal
bandwidth of the FM signal is ∞.
Practical BW: Note that kfn an (t) is exponentially growing, and n! grows much faster (like en log n ). Thus
kfn an (t)
n!
→ 0 as n → ∞. Thus most of the modulated power is in a finite bandwidth. For practical purposes,
we can filter the signal in a finite bandwidth and transmit the resulting signal.
Narrow-band FM (NBFM): Suppose |kf a(t)| << 1. Then (4) can be approximated as
sF M (t) = A [cos wc t − kf a(t) sin wc t] .
This has bandwidth 2B.
Similarly, for NBPM,
sF M (t) = A [cos wc t − kp m(t) sin wc t] .
This also has bandwidth 2B.
Remarks:
1) Both AM and narrow-band angle modulation have a bandwidth 2B.
2) The sideband spectrum in NBFM/NBPM has a phase shift of π/2 with respet to the carrier, whereas
in AM, it is in phase with the carrier.
3) In spite of the similarity of the expressions for AM and NB-angle modulation, the modulated signals
look very different due to the relative phase difference of the two terms.
Practical bandwidth estimation of WBFM:
When |kf a(t)| ̸≪ 1, it is called WBFM. Now the higher order terms with higher bandwidth cannot be
neglected.
A message signal m(t) with bandwidth B can be approximated by a staircase signal as shown below.
 6

∞
X
m̂(t) = m(tk )rect(2B(t − tk ))
k=−∞

The FM signal for m̂(t) is given by

X
sF M (t) = cos [wc t + kf m(tk )t + θk ] rect(2B(t − tk )) (5)
k

over all k. For a sinusoidal pulse rect(2Bt) cos(wc t + kf m(tk )t), the FT is
   
w + wc + kf m(tk ) w − wc − kf m(tk )
(1/2)sinc + (1/2)sinc
4πB 4πB
as shown in the figure below.

This has the zero-crossing BW = 4B (angular BW = 8πB).

Each term in (5) has the same magnitude in its FT with different scaling m(tk ). Only the phase is different
for different terms.
The zero-crossing one-sided bandwidth of the signal is 4B. If mp := max |m(t)|, then the center (angular)
frequency of the positive frequency sinc() functions varry between wc − kf mp and wc + kf mp .
 7

So, the bandwidth of the frequency modulation of m̂(t) is

1
BF M = (2kf mp + 8πB) Hz
2π 
kf mp
=2 + 2B Hz
2π

Remarks:
1. Just considering the variation in the instanteneous frequency neglects the spread of 4B for the sinc
pulses.
2. This is specially erroneous when kf mp is small, as then the 2B term is the significant term. This
happens because the faster variation ⇒ highter B ⇒ narrower pulses in m̂(t) in m(t) contributes more
in the FM bandwidth.
We now define the peak deviation (one-sided) of the instateneous frequency (from fc )
kf mp
∆f = .
2π
So, we can express FM bandwidth as
BF M = 2(∆f + 2B) (6)

Carson’s rule:
While the total instanteneous frequency deviation 2∆f underestimates BF M (specially for NBFM), the
(6) overestimates the true required BW, as m(t) is smoother than m̂.
1. The discontinuities in m̂(t) (so the rect(·) function in the summation) result in a higher spread (by
sinc(·)) than actually happens. This becomes significant when kf mp is small, as then this spread is
relatively more significant.
2. For instance, for NBFM, when kf mp ≪ 1, (6) says that the BW is 4B, whereas we showed that it is
≈ 2B.
This suggests that a better BW estimate is
BF M ≈ 2(∆f + B) (Carson’s rule)

For very wideband FM, this gives

BF M ≈ 2∆f when ∆f ≫ B.
 8

∆f
We now define β := B
. Then Carson’s rule is expressed as
BF M ≈ 2B(β + 1).

Phase modulation:
wi = wc + kp ṁ(t)
[ṁ(t)]max − [ṁ(t)]min
∆f = kp ·
4π
ṁp
Assuming ṁp := [ṁ(t)]max = −[ṁ(t)]min , we have ∆f = kp · 2π
.
max
So,
 
kp ṁP
BP M = 2(∆ + B) = 2 +B .
2π

Special case of tone modulation

m(t) = α cos wm t
Thus
α
a(t) = sin wm t
wm
Thus
k α
 
j wc t+ wf sin wm t
s̃(t) = Ae m

Moreover, ∆w = kf mp = αkf , and angular BW of m(t) is 2πB = wm . Then deviation ratio is given by
∆f ∆w αkf
β= = =
B 2πB wm
Hence
s̃(t) = Aejwc t ejβ sin wm t
The last term is periodic
P with fundamental frequency wm /2π. Suppose its Fourier series expansion is
given by ejβ sin wm t = ∞
n=−∞ D n ejwm nt
, where
Z π
wm wm jβ sin wm t −jnwm t
Dn = e e
2π wπ
Z πm
1
= ej(β sin x−nx) dx
2π −π
= Jn (β),
where Jn (β) is the Bessel function of the first kind of n-th order. These can be computed numerically.
Thus
X∞
jβ sin wm t
e = Jn (β)ejwm nt .
n=−∞
 9

Hence we get
∞
X
s̃(t) = A Jn (β)ej(wc t+wm nt) ,
n=−∞

and
∞
X
s(t) = A Jn (β) cos(wc + wm n)t.
n=−∞

So, S(w) has infinite number of sidebands of frequencies wc ± wm , wc ± 2wm , · · · .

The strength of the sideband at wc ±nwm is Jn (β) (we know that J−n (β) = (−1)n Jn (β)). Jn (β) decreases
with n as shown below.

For β ≥ 1,
Jn (β) contains (as a function of n) most significant components in n ≤ β + 1. Thus
BF M ≈ 2(β + 1)fm = 2(∆f + B) Carson’s formula

For β ≪ 1,
Jn (β) can be neglected for n > 1. Thus
BF M ≈ 2fm = 2B NBFM approximation

Note: This method of finding the spectrum of FM works for any periodic m(t).
Generation of FM (and PM) signal
A. Narrow-band FM:

sN BF M (t) ≈ A [cos wc t − kf a(t) sin wc t]

sN BP M (t) ≈ A [cos wc t − kp m(t) sin wc t]
 10

Because of the approximation involved in reaching the expressions above, there is an amplitude variation
in the generated signals. This is corrected by a bandpass limitter as shown below.
Bandpass limitter:
The bandpass limitter consists of a hard limitter followed by a bandpass filter. The response of the hard
limitter is given by
vo = sign(vi )

A signal of the form

vi (t) = A(t) cos θ(t),
 11

Rt
is given as input. Here A(t) ≥ 0 is the envelope, and θ(t) = wc t + kf ∞
m(τ )dτ is the phase for an FM
signal.

Since A(t) ≥ 0, the output depends only on θ(t):

v0 (θ(t)) = sign(cos θ(t))

As a function of θ(t), the signal vo (θ(t)) is a periodic pulse. So it has the Fourier series expansion:
 
4 1 1
v0 (θ(t)) = cos θ(t) − cos 3θ(t) + cos 5θ(t) + · · ·
π 3 5
  Z   Z 
4 1
= cos wc t + kf m(τ )dτ − cos 3 wc t + kf m(τ )dτ
π 3
 Z  
1
+ cos 5 wc t + kf m(τ )dτ + · · ·
5
The output of a suitably designed BPF is then π4 cos wc t + kf m(τ )dτ , which is the constant amplitude
 R 
FM signal.
B. WBFM generator
Direct method
A voltage controlled oscillator can be used to produce the FM signal. The VCO should produce the carrier
signal cos wc t under zero input. Then the input message signal deviates the instanteneous frequency. VCOs
are typically imprelemted by using a voltage controlled capacitor in a resonant circuit of an oscillator.
For instance, a reverse biased diode acts as a capacitor whose capacitance varies with the bias voltage.
Specialized diodes, known as varactor/vericap diodes, are used whose capacitance is proportional to the
 12

bias voltage. A major disadvantage of this approach is the instability of VCOs, which is partially controlled
by feedback.
Indirect method
Suppose we want a WBFM with frequency deviation n∆f and center frequency nfc . This is obtained
by first doing NBFM with frequency deviation ∆f at center frequency fc and then passing it through a
frequency multiplier by n.

Frequency multiplier by n is implemented by a non-linear device of order ≥ n and a BPF.

If the non-linear device has the response

y(t) = a0 + a1 x(t) + a2 x2 (t) + · · · + an xk (t),
h Rt i
and x(t) = A cos wc t + kf 0 m(τ )dτ , then by trigonometric identities, y(t) has the form
 Z t 
y(t) = c0 + c1 cos wc t + kf m(τ )dτ
0
 Z t 
+ c2 cos 2wc t + 2kf m(τ )dτ
0
..
.
 Z t 
+ ck cos nwc t + nkf m(τ )dτ
0

The n-th term has frequency deviation n∆f . If the BPF is taken with center frequency nfc and suitable
bandwidth, then we get the n-th term, which is WBFM with n∆f deviation and center frequency nfc .
Remarks:
1. Sometimes we need to multiply by large n to increase ∆f significantly. So much (nfc ) carrier frequency
may not be needed, then nfc is reduced by frequency mixing. Multiplication by cos f1 t and then BPF at
center frequency fc − f1 reduces the center frequency to fc − f1 .
2. Frequency multiplication by large n can be done by many stages of small multiplication, and frequency
shifting can be done in any/multiple stages also.
3. The scheme apparently has good frequency stability, but suffers from noise due to excessive multiplica-
tion and distortion at lower frequency where ∆f /fm is not small and as a result the NBFM approximation
is poor.
 13

Example of commercial FM transmitter Need fc = 91.2M Hz, ∆f = 75kHz for audio signals with
spectrum in 50Hz to 15kHz. Choose ∆f = 24.41Hz so that required n = 7500024.41
= 3072 = 64 × 48.
The minimum β is for the tone of frequency fm = 50Hz, β = 0.5. Choose fc = 200kHz.
Frequency multiplication by 64 × 48 can be done in stages of frequency doublers and triplers. The overall
implimentation is as shown below.

Robustness of angle modulation against non-linear distortion

If the non-linear device has the response
y(t) = a0 + a1 x(t) + a2 x2 (t) + · · · + an xk (t),
h Rt i
and x(t) = A cos wc t + kf 0 m(τ )dτ , then by trigonometric identities, y(t) has the form
 Z t 
y(t) = c0 + c1 cos wc t + kf m(τ )dτ
0
 Z t 
+ c2 cos 2wc t + 2kf m(τ )dτ
0
..
.
 Z t 
+ ck cos nwc t + nkf m(τ )dτ .
0

Clearly, the output can be filtered with a BPF to get only the desired term.
FM demodulation
Let us consider an FM signal received at the receiver:
s(t) = cos θ(t)
The signal is passed through a differentiator to get
d
(s(t)) = −θ̇(t) sin θ(t)
dt
= −wi (t) sin θ(t),
where wi (t) is the instaneneous frequency. Now an envelope detector produces wi (t), from which the
message signal can be obtained by DC removal.
 14

The combination of a differentiator and an envelope detector is called a slope detector or frequency
discriminator.
Practical demodulators:
In practice, an ideal diferentiator is difficult to implement. Instead, it is more easier to design a circuit
that has almost linear response in the transmission band. Suppose, a linear filter has frequency response
H(f ) ≃ (a|ω| + b)ejϕ(ω) .
Since the instanteneous frequency ωi in s(t) = A cos θ(t) = 0.5(ejθ(t) + e−jθ(t) ) varies slowly, the output
of the filter is approximately (note that ωi > 0)
0.5(aωi + b) ej(θ(t)+ϕ(ωi )) + e−j(θ(t)+ϕ(ωi ))

=(aωi + b) cos(θ(t) + ϕ(ωi ))

=(kf am(t) + aωc + b) cos(θ(t) + ϕ(ωi )).
An envelope detector will give kf am(t) from the above signal.
Balanced discriminator: Any tuned circuit can be used as a filter either below or above the resonant
frequency, as its response is approximately linear. (A simple RC circuit can also be used at low frequency
where the response is jwRC/(1 + jwRC), which can be approximated as jwRC when wRC ≪ 1.)
However, such circuits have linear response in a small band, and so it causes distortion. It is partly
avoided by using a balanced discriminator, where two filters with opposite slopes are used to cancel
the distortion of each other.

Let us consider any wi > 0. In a balanced discriminator shown in the figure, H1 (w) and H2 (w) are related
as H2 (wc + ∆w) = H1 (wc − ∆w). For wi = wc + ∆wi , suppose
H1 (wi ) = a∆wi + b + e(wc + ∆wi )
H2 (wi ) = −a∆wi + b + e(wc − ∆wi )
in the valid range of wi (awc is absorved in the constant b). If the error e(wi ) is approximately symmetric
about wc , then the outputs of the envelope detectors are given by (we use ∆wi = kf m(t))
kf am(t) + b + e(wc + kf m(t))
and − kf am(t) + b + e(wc − kf m(t))
 15

Now we use the symmetry that e(wc + ∆wi (t)) ≈ e(wc − ∆wi (t)). This is roughly true in the tuned circuit
filter that were used (LC filter, obtained using a center tapped transformer together with a capacitor in
each branch). The difference of the envelops is given by 2kf am(t), which is the desired message signal.

The overall advantage of a balanced discriminator stems from the fact that the deviation of the filter
response from the desired linear response is cancelled when we take the difference of the responses of the
two filters. The overall characteristic is more linear and also has a linear range covering a wider frequency
band.
Using PLL In the following PLL circuit, when the circuit is locked, the output is approximately propor-
tional to m(t). The exact analysis is skipped.

Zero crossing detectors: Using modern digital ICs, frequency counters using zero crossing detector can
be implemented.
Noise performance of FM and PM
SNR and transmission bandwidth:
Note that the output of the demodulator is proportional to kf , which directly influences the frequency
deviation ∆f and hence the transmission bandwidth (Carson’s rule: BT ≃ 2(∆f + B). The output power
is proportional to kf2 , hence the overall SNR increases as BT increases. This allows a trade-off between
the transmission bandwidth and the output SNR. This is one of the main reasons for using FM.
Power spectral density of the output noise:
Under additive Gaussian noise, the receiver output can be shown to have a noise with power spectral
density

(
N0 f 2
A2
, |f | ≤ B
Sn (f ) =
0, otherwise
On the other hand, in a PM receiver, the noise can be shown to have a constant power spectral density.
 16

Receiver
S n (f)
in dB

s(t) Band−pass Baseband

Limiter Discriminator
filter LPF
Message
Band−pass limiter bandwidth −B B
log(f)
Noise

Pre-emphasis and de-emphasis

FM receiver output has more noise in high frequencies, whereas a PM receiver output has constant
noise power spectral density. Pre-emphasis and de-emphasis filters in FM system effectively make the
modulation FM in low frequencies and PM in high frequencies at the same time. This is achieved by
using a pre-emphasis filter at the transmitter and a de-emphasis filter at the receiver as shown in the figure.

Transmitter Receiver
m(t) s(t)
Pre−emphasis De−emphasis
FM modulator FM receiver
filter filter

Noise psd
Pre−emphasis filter De−emphasis filter
S n (f)
|H(f)|
6 dB/Octave in dB

dB w1 2π B
|H(f)|
dB w
log scale
w1 2π B w ω1 2π B
log scale −6 dB/Octave log(f)

Pre Emphasis filtering is also done in other systems like audio tape recording where some frequencies are
more susceptible to noise. Those frequencies are amplified before recording to compensate for the high
noise, and the played back signal is attenuated in turn at those frequencies.
Stereo FM radio broadcast
Two audio signals, the left and right channels, are transmitted using FM such that it is compatible with
earlier monophonic FM receivers (as per FCC rule).
The signals are recombined to get mL (t) + mR (t) and mL (t) − mR (t). A monophonic receiver only
recovers mL (t) + mR (t). The difference signal is DSB-SC modulated at 38kHz and added with the sum
signal. The output is then frequency modulated with ∆f = 75kHz. The transmission bandwidth is taken
as 200 kHz.
 17

• A pilot is added at 19kHz instead of 38 kHz since it is easier to separate it since there is no signal
at 19kHz.
• Monophonic receiver low-pass filters the demodulator output and gets only the mL (t) + mR (t).
• Pre-emphasis is done on the left and the right signals before multiplexing. It is shown equivalently
on the recombined signals in the figure.
• Ideally, pre-emphasis should be done just before frequency modulation. However that would result in
a very high amplification of the difference signal. This would result in a large transmission bandwidth
requirement. (high amplitude high frequency signals require high transmission bandwidth in FM).
• In fact, for the same reason, even moniphonic FM sometimes did not use (or uses little) pre-emphasis
for music channels where there is usually large high frequency power.
• Under a highly noisy channel condition, the difference signal is received with large noise due to FM
demodulator’s high noise psd. The sum signal is still received with good quality. Some receivers used
to come with a switch to fall back to monophonic reception to output only the sum signal, resulting
in an improvement on quality.

Supplementary notes
Phase locked loops (PLL)
• Some applications: in frequency synthesizer, carrier and symbol clock recovery at receivers, FM and
other demodulators.
• Three main components: Phase detector (PD), loop filter (LPF), voltage controlled oscillator (VCO)
• Both analog and digital designs implemented
 18

Km

Kv

vin (t) = Ai sin[w0 t + θi (t)]

vo (t) = Ao cos[w0 t + θo (t)]
Z t
θo (t) = Kv v2 (τ )dτ
−∞

In the locked state, the vo (t) and vi (t) are expected to be in quadrature, i.e., in 90 degree phase shift. For
this reason, they are expressed as cos and sin respectively.
w0 is assumed to be the free running frequency of the VCO. A frequency difference can always be
captured in the phases θi (t) and θo (t).
A very diverse types of designs are used for the Phase detector in different applications. It can be a
simple circuit, a small IC, or a complete box by itself. A simple product modulator (e.g. ring modulator)
followed by a LPF (this LPF can be the same that follows the PD already in PLL) can also work as an
approximate phase detector with some limitations. For such an analog PLL,
v1 (t) = Km Ai A0 sin[w0 t + θi (t)] cos[w0 t + θ0 (t)]
Km Ai Ao Km Ai Ao
= sin[θi (t) − θo (t)] + sin[2wo t + θi (t) + θo (t)]
2 2
The LPF gives
Km Ai Ao
= sin[θi (t) − θo (t)]
2
= Kd sin[θe (t)]
≈ Kd θe (t),
 19

assuming that the PLL is in near lock (phase error is small). Here Kd is the gain of the phase detector.
Transfer functions:
Θo (f )
(a) Open loop transfer function: G(f ) :=
Θe (f )
Θo (f )
(b) Closed loop transfer function: H(f ) :=
Θi (f )
Θe (f )
(c) Phase error transfer function: He (f ) :=
Θi (f )

(a) Open loop transfer function:

Z t
θo (t) = Kv v2 (τ )dτ
−∞
Kv Kd F (f )
⇒Θo (f ) = Θe (f )
jw
Θo (f ) Kv Kd F (f )
⇒G(f ) = =
Θe (f ) jw

(b) Closed loop transfer function: Now,

Θi (f ) − Θo (t) = Θe (f )
 
1
⇒Θi (f ) = Θo (f ) 1 +
G(f )
G(f ) + 1
⇒Θi (f ) = Θo (f ) ·
G(f )
Θo (f ) G(f ) Kv Kd F (f )
⇒H(f ) := = =
Θi (f ) G(f ) + 1 jw + Kv Kd F (f )

(c) Phase error transfer function:

Θe (f ) Θi (f ) − Θo (f )
He (f ) = = = 1 − H(f )
Θi (f ) Θi (f )
jw
⇒He (f ) =
jw + Kd Kv F (f )

and
Θe (f ) = He (f ) · Θi (f )

Steady state error: Using final value theorem of the Laplace transform,
Θe (∞) = lim sΘe (s); s = jw
s→0
s2
= lim Θi (s) · ; s = jw
s→0 s + Kd Kv F (s)
This can be used to understand the steady-state behavious of the PLL for different kinds of input.
 20

We consider two cases:

First order PLL: all-pass filter: F (f ) = 1: This is achieved in the analogu PLL design if the frequency
response is flat in the pass-band (message band).

G(f ) K d Kv
H(f ) = =
1 + G(f ) Kd Kv + jw

Example 1: Suppose the loop is locked and we have a phase step change.
θi (t) = ∆θ · u(t)
⇒Θi (s) = ∆θ/s
So using final value theorem,
s · ∆θ
Θe (∞) = lim =0
s→0 s + Kd · Kv

So the PLL regains lock.

Example 2: Suppose the loop is locked and we have a frequency step change.

wi (t) = wc + ∆w · u(t)
⇒θi (t) = ∆w · tu(t)
⇒Θi (s) = ∆w/s2
Using final value theorem,
s2 ∆w
Θe (∞) = lim · Θi (s) =
s→0 s + Kd · Kv Kd · Kv
Thus there will be a constant phase error, which is small if Kd · Kv is large.
More precisely,
θi (t) = ∆w · tu(t)
⇒Θi (f ) = ∆w/(jw)2
So
V1 (f ) = Kd · Θe (f ) = Kd · He (f ) · Θi (f )
jw
= Kd · Θi (f ) ·
jw + Kd Kv F (f )
Kd · jw
= 2
(jw) (jw + Kd Kv F (f ))
Kd
=
jw(jw + Kd Kv F (f ))
 
∆w 1 1
= · −
Kv jw jw + Kd · Kv
∆w
⇒ v1 (t) = (1 − e−kt )u(t)
Kv
∆w
⇒ θe (t) = (1 − e−kt )u(t)
k
 21

where k = Kv · Kd . So,
∆w
θe (t) → as t → ∞
k
Note that the speed of convergence depends on k (time constant is proportional to 1/k).

Additional note:
If we want to see what happens if the input frequency is w0 for t < 0 (the free-running frequency of the
VCO, for simplicity of analysis), and then the frequency changes in steps, then
wi (t) = w0 + ∆w1 · u(t − t1 ) + ∆w2 · u(t − t2 ) + · · ·
⇒θi (t) = ∆w1 · (t − t1 )u(t − t1 ) + ∆w2 · (t − t2 )u(t − t2 ) + · · ·
∆w1 ∆w2
⇒Θi (f ) = 2
· e−jt1 w + · e−jt2 w + · · ·
(jw) (jw)2
Following the same steps as above, we get
∆w1 ∆w2
θe (t) = (1 − e−k(t−t1 ) )u(t − t1 ) + (1 − e−k(t−t2 ) )u(t − t2 ) + · · ·
k k
So, if the gaps between the changes are much larger than 1/k, then the signals θe (t), v1 (t) “settle” to the
stable value in that time interval.

FM demodulation: Now, using a quasi-static assumption (since the message signal is slowly changing),
we can think of an FM signal as a signal of constant instantaneous frequency which stays constant for a
long time, and then changes to a new frequency value. The above example then demonstrates that for a
large value of Kd · Kv , for an FM signal as input to the PLL, the PLL will output a signal proportional
to the instantaneous frequency deviation, which is proportional to the message signal.

s(t) = Ac cos(wc t + θi (t))

Z Z
⇒θi (t) = Df m(τ )dτ = 2πKf m(τ )dτ
Df
⇒Θi (s) = M (s)
s

Now,
Kv
Θo (s) = V2 (s) ·
s
Θo (s)
⇒ V2 (s) = s ·
Kv
Θi (s)H(s)
=s·
Kv
Df sH(s)
= · M (s) ·
s Kv
Df Kd Kv
= · · M (s)
Kv s + Kd Kv
 22

If Kd Kv ≫ 2πf in the message bandwidth, i.e., if Kd Kv ≫ 2πB (B is the bandwidth of m(t)), then
Df 2πKf
V2 (s) = · M (s) = · M (s)
Kv Kv
2πKf
⇒v2 (t) = · m(t)
Kv
Note: Kd Kv ≫ 2πB can be quite a demanding requirement if B is large, but this can be achieved by
using a high gain amplifier after the phase detector.