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ECE 461/561

COMMUNICATIONS I

Dr. Mario Edgardo Magaña


EECS
Oregon State University
OBJECTIVE: The objective of a communication system is to efficiently transmit
an information-bearing signal (message) from one location to another through a
communication channel (transmission medium).

WISH LIST:

• High reliability

• Minimum signal power

• Minimum transmission bandwidth

• Low system complexity (low cost)

EFFICIENT TRANSMISSION: Can be achieved by processing the signal through


a technique called modulation.

MODULATION TYPES:

• Analog modulation

• Digital modulation 2
Amplitude modulation
In this type of modulation the amplitude of a sinusoidal carrier is varied
according to the desired message signal. Let m(t) be the message signal we
would like to transmit, ka be the amplitude sensitivity (modulation index), and
c(t) = Accos(2πfct) be the sinusoidal carrier signal, where Ac is the amplitude of the
carrier and fc is the carrier frequency.
The transmitted AM signal waveform is described by
s AM (t ) = Ac [1 + k a m(t )] cos(2πf c t )

Requirements:

k a m(t ) < 1, ∀t

f c >> Wm
• , where Wm is the message bandwidth

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Taking the Fourier transform of the modulated waveform, we get

S AM ( f ) = F { s AM (t )}
= F { Ac cos(2π f c t ) + Ac k a m(t ) cos(2π f c t )}
Ac k a Ac
= [δ ( f − f c ) + δ ( f + f c )] + [ M ( f − f c ) + M ( f + f c )]
2 2
Let |M(f)| be described by

Magnitude spectrum of m(t) 4


Then the magnitude spectrum of sAM(t) is

Magnitude spectrum of sAM(t)

Observations:
11. B AM = 2Wm
12. Carrier is transmitted explicitly

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Let m(t ) = cos(0.2π t ) , then
s AM (t ) = Ac [1 + k a cos(0.2π t )] cos(2π f c t )

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7
To conserve transmitted power, let us suppress the carrier, i.e., let the
transmitted waveform be described by

s DSB (t ) = Ac m(t ) cos(2π f c t ).

This is called double side-band suppressed carrier (DSB-SC) modulation.

8
In the frequency domain,

S DSB ( f ) = F { s (t )}
= F { Ac m(t ) cos( 2π f c t )}
Ac
= [M ( f − fc ) + M ( f + fc )]
2

The modulated waveform now has the following spectral characteristics:

Magnitude spectrum of sDSB(t)

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Observations:

2. B DSB = 2Wm (same as standard AM)

3. Transmitted power is less than standard AM

Example: Let a sinc pulse be transmitted using DSB-SC modulation.

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11
Receivers for AM and DSB
Receivers can be classified into coherent and non-coherent categories.
Definition: If a receiver requires knowledge of the carrier frequency and phase to
extract the message signal, then it is called coherent.
Definition: If a receiver does not require knowledge of the phase (only rough
knowledge of the carrier frequency) to extract the message signal, then it is
called non-coherent.
Non-coherent demodulator (receiver) for standard AM

+ +
S AM (t ) D ˆ (t )
m
R C
- -

Peak envelope detector (demodulator) 12


Observations:
• The net effect of the diode is the multiplication (mixing) of the signals applied
to its input. Therefore, its output will contain the original input frequencies, their
harmonics and their cross products.
• The network is a lowpass filter with a single pole (lossy integrator) that
removes most of the high frequencies. R and C have to be judiciously picked
so that τ = RC is neither too short (rectifier distortion ) nor too long (diagonal
clipping ). Rule of thumb is τ ≅ period of the carrier.
• A rule of thumb for choosing τ is based on the highest modulating signal
(message) frequency that can be demodulated by a peak envelope detector
without attenuation, that is,

τ=
(1 / k ) − 1 ,
2
a
ka < 1
2π f m (max)

where f m (max) is the maximum modulating signal (message) frequency in Hz.


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Let the message signal be described by the sinusoidal tone m(t ) = cos( 2π f m t ) .
Then, the following figures show the types of distortion that can occur when τ is
improperly chosen.

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Illustration of rectifier distortion and diagonal clipping 15
Coherent demodulator for DSB-SC

At the output of the mixer,

v(t ) = Aˆ c cos( 2π f c t + φ ) s DSB (t )


= Aˆ c cos( 2π f c t + φ ) ⋅ Ac m(t ) cos( 2π f c t )
Aˆ c Ac
= m(t )[ cos( 4π f c t + φ ) + cos φ ]
and 2

V ( f ) = F { v(t )}
Ac Aˆ c Ac Aˆ c
= [ M ( f − 2 fc ) + M ( f + 2 fc )] + M ( f ) cos φ
4 2
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Let the message signal have the following magnitude spectrum

Then, if fc>Wm,

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Suppose H lpf ( f ) is such that

Ac Aˆ c
Then, if W m ≤ B < 2 f c − W m , M( f ) =
ˆ M ( f ) cos φ
2

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Coherent Costas loop receiver for DSB-SC :

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I-channel:

After downconverwsion,

v I (t ) = Ac m(t ) cos( ω c t ) ⋅ cos( ω c t + φ )


Ac
= m(t )[ cos( 2ω c t + φ ) + cos φ ]
2

At the output of the lowpass filter, with |H(0)| = 1,


Ac
m I (t ) = cos φ ⋅ m(t )
2

Q-channel:
Ac
vQ (t ) = m(t )[ sin ( 2ω c t + φ ) + sin φ ]
2
A
mQ (t ) = c sin φ ⋅ m(t )
2
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Feedback path:
At the output of the multiplier,
Ac2 2
me (t ) = m (t ) sin φ cos φ
4
Ac2 2
= m (t ) sin 2φ
8

At the output of the lowpass filter,



mef (t ) = ∫ me (τ )h f (t − τ )dτ
−∞

The purpose of hf(t) is to smooth out fast time variations of me(t)

The output of the VCO is described by

xVCO (t )  cos  c t   (t ) 

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t
Where ωc is the VCO’s reference frequency and φ (t ) = k v ∫m
0
ef (τ )dτ , is the
residual phase angle due to the tracking error. The constant kv is the frequency
sensitivity of the VCO in rad/s/volt

The instantaneous frequency in radians/sec of the VCO’s output is given by

d [ω c t + φ (t )]
= ω c + k v mef (t ),
dt

Clearly, if φ(t) were small, then the instantaneous frequency would be close to ωc
and the output of the I-path would also be proportional to m(t).

Single-Sideband Modulation

Motivation: Bandwidth occupancy can be minimized by transmitting either the


lower or the upper sideband of the message signal.

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Consider the following transmitter

Suppose we want to transmit the upper sideband, then the following choice of
bandpass filter yields the correct result

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At the output of the bandpass filter,

Observation: A practical bandpass filter will cause severe distortion.


Suppose the original message signal contains a gap in its magnitude spectrum
around the origin, i.e.,

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Then, SSB demodulation can be obtained by

Synthesis and analysis of SSB:

Let the spectrum of a DSB-SC signal be given by

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Let this DSB-SC signal be the input of a lowpass filter with magnitude response

Then the resulting (output) signal would have the spectrum

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The filter H L ( f ) can be represented by the algebraic sum

1
HL( f ) = [ sgn ( f + f c ) − sgn ( f − f c ) ]
2
Also,
L
S SSB (t ) = H L ( f ) ⋅ S DSB ( f )
Ac
= HL( f ) ⋅ [M ( f + fc ) + M ( f − fc )]
2
Ac
= [ M ( f + f c ) sgn ( f + f c ) + M ( f − f c ) sgn ( f + f c ) ]
4
A
− c [ M ( f + f c ) sgn ( f − f c ) + M ( f − f c ) sgn ( f − f c ) ]
4
A
= c [ M ( f + fc ) + M ( f − fc )]
4
Ac
+ [ M ( f + f c ) sgn ( f + f c ) − M ( f − f c ) sgn ( f − f c ) ]
4
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To shed more light on the last equation, let us introduce the concept of the
Hilbert transform.
Definition
A Hilbert transform filter is a filter that simply phase shifts all frequency
components by π/2 radians, i.e., its transfer function is described by
H ( f ) = − j sgn ( f )

where the signum function is defined by


1, f >0

sgn ( f ) = 0, f =0
 − 1, f <0
Observations:

• H( f ) =1

− π / 2, f >0
• arg( H ( f ) ) = 
π / 2, f <0
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1
In the time domain, the impulse response of the filter is given by h(t ) =
πt
Therefore, the Hilbert transform of a function g(t) can be interpreted as the
convolution of the impulse response of the filter h(t) and g(t), i.e.,
g~ (t ) = h(t ) ∗ g (t )
1 ∞ g (τ )
= ∫ dτ
π − ∞ t −τ
In the frequency domain,
~
G ( f ) = − j sgn ( f ) G ( f )
Therefore, F
g~ (t ) ⇔ − j sgn ( f ) G ( f )

Using the exponential modulation theorem,

F
g~ (t )e ± j 2π f c t ⇔ − jG ( f  f c ) sgn ( f  f c )

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In the time domain, we can synthesize lower SSB mathematically as follows:
L
s SSB (t ) = F −1 { S SSB
L
( f )}
1 
= F −1  Ac [ M ( f + f c ) + M ( f − f c ) ] 
4 
1 
+ F −1  Ac [ M ( f + f c ) sgn ( f + f c ) − M ( f − f c ) sgn ( f − f c ) ] 
4 
1 1 ~ (t )e − j 2π f ct − 1 A m
= Ac m(t ) cos( 2π f c t ) + Ac m ~
c (t )e
j 2π f c t

2 − j4 − j4
1 1 ~
= Ac m(t ) cos( 2π f c t ) + Ac m (t ) sin ( 2π f c t )
2 2

~
where m(t ) is the Hilbert transform of m(t)
Likewise, the upper SSB signal is described by
1 1 ~
s USSB (t ) = Ac m(t ) cos( 2π f c t ) − Ac m (t ) sin ( 2π f c t )
2 2
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A widely used form of SSB is vestigial sideband, which is used in TV broadcasts.

SSB is a special case of VSB, since the filter that is used to obtain SSB is a
limiting case of the filter used to generate VSB.

Generation of VSB

Instead of the ideal filter used in SSB, let us use the following filter:

L
H VSB ( f ) can be analytically described by
L
H VSB [
( f ) = − j HQ ( f + fc ) − HQ ( f − fc ) ]
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The transfer function H Q ( f ) has the following characteristics:

Observations about H Q ( f ) :
• Odd symmetry about the origin
• Linear phase
Mathematically,
1 1
L
sVSB (t ) = Ac m(t ) cos( 2π f c t ) + Ac m′(t ) sin ( 2π f c t )
2 2
and 1 1
U
sVSB (t ) = Ac m(t ) cos( 2π f c t ) − Ac m′(t ) sin ( 2π f c t )
2 2
The ‘+’ sign means that a vestige of the upper sideband is transmitted and a ‘-’
sign means that a vestige of the lower sideband is transmitted. 32
Example: Now, if the magnitude spectrum of the DSB signal is as described
L
before, then the magnitude spectrum of sVSB (t ) is

Vestigial sideband magnitude spectrum (upper sideband vestige).

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To help in the demodulation process, let us insert a pilot signal to the VSB signal

1 1 
P
sVSB (t ) = k a  Ac m(t ) cos( 2π f c t ) ± Ac m′(t ) sin ( 2π f c t )  + Ac cos( 2πf c t )
2 2 
or
 1  1
P
sVSB (t ) = Ac 1 + k a m(t ) cos( 2π f c t ) ± k a Ac m′(t ) sin ( 2π f c t ) ,
 2  2
P
sVSB (t ) = m I (t ) cos( 2π f c t ) − mQ (t ) sin ( 2π f c t )
where
 1 
m I (t ) = Ac 1 + k a m(t )
 2 
1
mQ (t ) =  k a Ac m′(t )
2
Therefore, the complex envelope of this pilot aided VSB signal is
~ P
sVSB (t ) = m I (t ) + jmQ (t ) = a (t )e jφ (t ) ,

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where the real envelope a(t) is described by

a(t ) = m I2 (t ) + mQ2 (t )
1/ 2
 2  1  1
2

2

=  Ac 1 + k a m(t ) + Ac2  k a m′(t ) 
  2  2  
and its phase is
 mQ (t ) 
φ (t ) = tan  −1

m
 I (t )
If we use an envelope detector to demodulate the pilot-aided VSB, its output
would be
1/ 2
  1 
2
 1
2
 
a (t )  Ac   1  ka m(t )    ka m(t )  
  2   2  
1/ 2
  1 
2
 ka m(t )  
 1   2 
 Ac  1  ka m(t )   1    
 2    1
1  ka m(t )  
  2  
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Clearly, if m´(t) is small, then we can extract the message signal m(t) by using
a DC blocking capacitor.

In practice, m′(t ) is minimized by increasing the width of the vestigial


sideband.

Angle Modulation

Let θi(t) be the instantaneous angle of a modulated sinusoidal carrier, i.e.,


s (t ) = Ac cosθ i (t ),

where Ac is the constant amplitude.

The instantaneous frequency is


dθ i (t )
ω i (t ) = .
dt
Observation: The signal s(t) can be thought of as a rotating phasor of length Ac
and angle θi(t).
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If s(t) were an unmodulated carrier signal, then the instantaneous angle would be
θ i (t ) = ω c t + φc

where ωc ≡ Constant angular velocity in rad/s

φc ≡ Constant but arbitrary phase angle in radians

Let θi(t) be varied linearly with the message signal m(t) and φ = 0, then
θ i (t ) = ω c t + k p m(t ),

where kp ≡ Phase sensitivity of the modulator in rad/volt

In this case we say that the carrier has been phase modulated.

The phase modulated waveform is given by


s PM (t ) = Ac cos(ω c t + k p m(t ))

Let the instantaneous frequency ωi(t) be varied linearly with the message signal
m(t), i.e., ω i (t ) = ω c + k ω m(t )
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where kω ≡ Frequency sensitivity of the modulator in rad/s/volt

In this case we say that the carrier has been frequency modulated and the
instantaneous angle is obtained by integrating the instantaneous frequency, i.e.,
t t t
θ i (t ) = ∫ ω i (τ )dτ = ∫ [ω c + k ω m(τ )] dτ = ω c t + k ω ∫ m(τ )dτ
0 0 0

The modulated waveform is therefore described by


 t

s FM (t ) = Ac cos ω c t + kω ∫ m(τ )dτ 

 0 

Observation: Both phase and frequency modulation are related to each other and
one can be obtained from the other. Hence, we could deduce the properties of one
of the two modulation schemes once we know the properties of the other.

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FM modulation

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PM modulation
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The figure below shows a comparison between AM, FM and PM modulation of the
same message waveform:

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Frequency Modulation
Consider the frequency modulation of a message signal (frequency tone)

m(t ) = Am cos(2π f m t ),
The instantaneous frequency (in Hz) of the FM signal is
f i (t ) = f c + k f Am cos(2π f m t ).
Define the maximum frequency deviation as
∆f = k f Am .
The instantaneous phase angle of the FM signal is
t
θ i (t ) = 2π ∫ f i (τ )dτ
0

Am
= 2πf c t + k f sin(2π f m t )
fm
= 2πf c t + β sin( 2π f m t )

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where β = k f Am / f m , is known as the FM modulation index (for a tone) or the
maximum phase deviation (in rad) produced by the tone in question

The FM modulated tone is therefore given by:

sFM (t )  Ac cos  2 f c t   sin(2 f m t ) 


 Ac  cos(2 f c t ) cos(  sin(2 f m t ))  sin(2 f c t ) sin(  sin(2 f m t ))  .

The signal sFM(t) is nonperiodic unless fc = nfm, where n is a positive integer.

For the general case,


{
s FM (t ) = Re Ac e j [ 2π f ct + β sin( 2π f mt ) ] }
{
= Re s e (t )e j 2π f ct , }
where the complex envelope of the FM signal is described by
s e (t ) = Ac e jβ sin( 2πf mt )

Observation: Unlike sFM(t), se(t) is periodic with period 1/fm.

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Since se(t) meets the Dirichlet conditions, we can compute its Fourier series, i.e.,

s e (t ) = ∑n
c e j 2π n f m t

n = −∞
,

where the Fourier series coefficients are given by

T/2
1
T −T∫/ 2
− j 2π n f m t
cn = s e ( t )e dt , T = 1 / fm

1/ 2 fm

∫e
− j 2π n f m t
= fm s ( t )e dt
−1 / 2 f m
1/ 2 fm

∫ c
jβ sin( 2π f m t ) − j 2π n f m t
= fm A e e dt
−1 / 2 f m
1/ 2 fm
j [ β sin( 2π f m t )− 2π n f m t ]
= Ac f m ∫ e
−1 / 2 f m
dt .

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Let x = 2π f m t .
1
Then, dt = dx
2π f m
π
A j [ β sin x − nx ]
and cn = c
2π ∫
−π
e dx

= Ac J n ( β ),

π
1 j [ β sin x − nx ]
where J n ( β ) ≡
2π ∫
−π
e dx is the Bessel function of the first kind of order n.

Therefore,


s e (t ) = Ac ∑J
n = −∞
n ( β ) e j 2π nf m t

and the FM tone waveform is described by



s FM (t ) = Ac ∑J n ( β ) cos[ 2π ( f c + nf m )t ].
n = −∞
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In the frequency domain,

S FM ( f )   sFM (t )
  
   Ac 
J n (  ) cos  2 ( f c  nf m )t  
 n  

 Ac J
n 
n ( ) F  cos  2 ( f c  nf m )t  


J n ( )
 Ac 
n 
2
  ( f  f c  nf m )   ( f  f c  nf m )

Average power of the FM waveform:

The power delivered to a 1 ohm resistor load by the FM waveform is P = Ac


2
/2

But,
Ac2 ∞
P=
2
∑J
n = −∞
2
n (β )

is also the power of the FM waveform.


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47
Let us now take a look at the properties of the Bessel function.

2. J n ( β ) = (−1) n J −n ( β )

3. ∑ n (β ) = 1
J 2

n = −∞

2
Hence, the average power of an FM tone is Ac / 2.

Suppose β is small, i.e., 0 < β ≤ 0.3, then

• J 0 (β ) ≅ 1

• J1 (β ) ≅ β / 2

• J n ( β ) ≅ 0, n≥2

Under the assumption that β is small, the Fourier series representation of the FM
waveform can be simplified to three terms.

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Thus, for β small, the FM tone may be described by

 β β 
s FM ( t ) ≅ Ac cos( 2π f c t ) + cos( 2π ( f c + f m )t ) − cos( 2π ( f c − f m )t )
 2 2 

β β
= Ac cos( 2π f c t ) + Ac cos( 2π ( f c + f m )t ) − Ac cos( 2π ( f c − f m )t )
2 2

In the frequency domain,

Ac Ac β
S FM ( f ) ≅ [δ ( f + f c ) + δ ( f − f c ) ] − [δ ( f + f c − f m ) + δ ( f − f c + f m ) ]
2 4

+ c [δ ( f + f c + f m ) + δ ( f − f c − f m ) ]
4

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A plot of the magnitude spectrum of the FM tone with β small is shown below

The time domain FM waveform can be represented in phasor form as follows:


 1 1
S FM = Ac ∠0  + Ac β∠2πf m t + Ac β∠ − 2πf m t + π
2 2

For arbitrary t = t0, and small β, we can illustrate graphically the phasor
representation and arrive at some conclusion.

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The following figure shows an example of the phasor representation

Observation:
 
The resultant phasor S FM , has magnitude S FM ≅ Ac , and is out of phase with
respect to the carrier phasor Ac ∠0 .

Analytically,
 1
S FM = Ac + Ac β [ cos(2π f m t ) + j sin( 2π f m t ) + cos(−2π f m t + π ) + j sin( −2π f m t + π )]
2 51
But, cos(−2π f m t + π ) = cos(2π f m t ) cos π + sin
π sin( 2π f m t )
0

= − cos(2π f m t )

and sin( −2π f m t + π ) = − sin( 2π f m t ) cos π + sin


π cos(2π f m t )
0

= sin( 2π f m t )
Consequently, the resultant phasor is given by

S FM = Ac + jAc β sin( 2π f m t )
The magnitude of the resultant may be approximated by

S FM = Ac2 + Ac2 β 2 sin 2 (2π f m t )
 1 
≅ Ac 1 + β 2 sin 2 (2π f m t ),
 2 

since (1 + x ) ≅ 1 + nx, nx < 1


n

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Finally, the magnitude and phase of the resultant are found to be
  β2 β2 
S FM (t ) = Ac 1 + − cos(4π f m t )
 4 4 

∠S FM = φ S (t ) = tan −1 [ β sin( 2π f m t )]

Observation:

• For an FM tone, the spectral lines sufficiently away from the carrier may be
ignored because their contribution (amplitude) is very small.

FM Transmission Bandwidth:

For an FM tone, as β becomes large Jn(β) has significant lines only for
n ≤ β = k f Am / f m = ∆f / f m .

∴All significant lines are contained in the frequency range


f c ± β f m = f c ± ∆f ,

where ∆f is the peak frequency deviation. 53


Let β be small, i.e., 0 < β ≤ 0.3, then
J 0 ( β ) >> J n ( β ), n ≠ 0

which means that only the first pair of spectral lines is significant, i.e., the significant
lines are contained in the range fc ± fm

Observation: The previous analysis of an FM tone suggests that

• For large β the FM bandwidth is BFM = 2∆ f

• For small the FM bandwidth is B FM = 2 f m .

In general, the FM transmission bandwidth may be approximated by


BT ≅ 2∆f + 2 f m
= 2∆f (1 + f m / ∆f )
= 2∆f (1 + 1 / β )
This is known as the Carson’s rule.

Observation: Carson’s rule underestimates the transmission bandwidth by about


10%. 54
Alternative definition of FM tone transmission bandwidth:

A band of frequencies that keeps all spectral lines whose magnitudes are greater
than 1% of the unmodulated carrier amplitude Ac, i.e.,
BT = 2nmax f m ,
{ }
where n max = max n : J n ( β ) > 0.01 .

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General Case:
Let an arbitrary message signal m(t) have bandwidth Wm.
Define the peak frequency deviation and the deviation ratio by
∆f =ˆ k f max{ m(t ) }
t

and D =ˆ ∆f / Wm .

Then Carson’s rule can be used to define the transmission bandwidth of an


arbitrary FM signal, i.e., when m(t) is arbitrary.

Specifically, the FM transmission bandwidth can be defined by


BT ≅ 2∆f + 2Wm
= 2∆f (1 + Wm / ∆f )
= 2∆f (1 + 1 / D)

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Example: In commercial FM in the US, ∆f = 75 kHz, Wm = 15 kHz.
Therefore, the deviation ratio is D = 75 kHz/15 kHz = 5.
Using Carson’s rule, the transmission bandwidth is
BT = 2∆f (1 + 1 / D) = 180kHz,

Using the Universal curve, the transmission bandwidth is


BT = 3.2∆f = 240kHz .

In practice, FM radio in the US uses a transmission bandwidth of BT = 200 kHz.

Generation of FM

The frequency of the carrier can be varied by the modulating signal m(t) directly or
indirectly.

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Direct generation of FM
If a very high degree of stability of the carrier frequency is not a concern, then we
can generate FM directly using circuits without external crystal oscillators. Examples
of this method are VCO’s, varactor diode modulators, reactance modulators, Crosby
modulators (modulators that use automatic frequency control), etc..

+V CC

RFC 1 R5
R2 RFC 2
C1 C2

R1 C5 +
+
L1 L2 s FM (t )
R6
m(t) R3 R2 C3 R7 C4 C6 -
-
C7

Reactance FM modulator
58
Indirect generation of FM
Commercial applications of FM (as established by the FCC and other spectrum
governing bodies) require a high degree of stability of the carrier frequency. Such
restrictions can be satisfied by using external crystal oscillators, a narrowband
phase modulator, several stages of frequency multiplication and mixers.

Let us begin with the synthesis of narrow-band FM.

Narrowband frequency modulator


The narrow band FM signal is given by

 t

s NB (t ) = Ac cos 2π f c t + 2π k f ∫ m(τ )dτ 
 0 
59
with kf (and thus ∆fNB) small
Let us now consider a technique to increase the FM signal bandwidth.

Let sNB(t) be input to a nonlinear device with transfer characteristic y(t) = axn(t),
where x(t) is its input, namely.

Nonlinear device.

t
θ (t ) = 2π f c t + 2π k f ∫ m(τ )dτ
Let i , then at the output of the nonlinear device, we
0
observe
y (t ) = aAcn cos n θ i (t )

Let us expand this last equation to infer the effect of this nonlinear device.

60
cosnθi(t) can be expanded as follows:

cos n θ i (t ) = cos 2 θ i (t ) cos n−2 θ i (t )


1
= [1 + cos 2θ i (t )] cos n−2 θ i (t )
2
1 1
= cos n−2 θ i (t ) + cos 2θ i (t ) cos n−2 θ i (t )
2 2

Likewise, 1 1
cos n − 2 θ i (t ) = cos n −4 θ i (t ) + cos 2θ i (t ) cos n − 4 θ i (t )
2 2
Thus,
1 1 1
cos n θ i (t ) = cos n − 2 θ i (t ) + cos 2θ i (t ) cos n − 4 θ i (t ) + cos 2 2θ i (t ) cos n − 4 θ i (t )
2 4 4

Expanding the last term of the previous equality, we get


1
cos 2 2θ i (t ) cos n − 4 θ i (t ) = [1 + cos 4θ i (t )] cos n−4 θ i (t ).
4
61
Rewriting the equation before the last one, we get

1 1
cos θ i (t ) = cos θ i (t ) + cos 2θ i (t ) cos n − 4 θ i (t )
n n−2

2 4
1 1
+ cos n − 4 θ i (t ) + cos 4θ i (t ) cos n − 4 θ i (t )
8 8
1 1 1
= cos n − 2 θ i (t ) + cos n − 4 θ i (t ) + cos 2θ i (t ) cos n − 4 θ i (t )
2 8 4
1 1
+ cos 4θ i (t ) cos θ i (t ) + cos 2θ i (t ) cos n −6 θ i (t )
n −6

16 32
1
+ cos 6θ i (t ) cos n −6 θ i (t )
32

The last term in the expansion of cosnθi(t) is given by

1
k −1
cos kθ i (t ) cos n −k
θ i (t ).
2

62
Let n be an even number, then, when k = n, the last term is
1
n −1
cos nθ i (t )
2
If, on the other hand, n is an odd number, then when k = n-1, the last term is

1 1 1
n −2
[ cos(n − 1)θ i (t ) cosθ i (t )] = n −1
cos(n − 2)θ i (t ) + n −1
cos nθ i (t )
2 2 2
Therefore, the last term in the expansion of cosnθi(t) is
1
n −1
cos nθ i (t )
2

So, can be expanded as

Acn
y (t ) = c0 + c1 cosθ i (t ) + c 2 cos 2θ i (t ) +  + a n −1 cos nθ i (t )
2

63
Example: Consider the cases when n = 2 and n = 3.
Let n = 2, then
y (t ) = aAc2 cos 2 θ i (t )

1 + cos 2θ (t )  aA 2
aA 2

or y (t ) = aAc2  i
 = c
+ c
cos 2θ i (t )
 2  2 2

Let n = 3, then
1 1 
y (t ) = aAc3  cos 2θ i (t ) + cos 2θ i (t ) cos θ i (t )
2 2 
1 1 1 1 
= aAc3  cos θ i (t ) +  cos 3θ i (t ) + cos θ i (t )
2 2 2 2 
3aAc3 aAc3
= cos θ i (t ) + cos 3θ i (t )
4 4

64
Finally,
 t
  t

y (t ) = c0 + c1 cos 2π f c t + 2π k f ∫ m(τ )dτ  + c 2 cos 4π f c t + 4π k f ∫ m(τ )dτ 
 0   0 
Acn  t

+  + a n −1 cos 2π n f c t + 2π n k f ∫ m(τ )dτ 
2  0 
Let y(t) be input to an ideal bandpass filter with unity gain, bandwidth wide enough to
accommodate spectrum of wide band signal and center frequency nfc, i.e.,

Ideal bandpass filter


Then, aAcn  t

sWB (t ) = n −1 cos 2π n f c t + 2π n k f ∫ m(τ )dτ 

2  0 
65
The instantaneous frequency of sWB(t) is
f i (t ) = nf c + nk f m(t )
Observations about sWB(t) :
1. The carrier frequency is nfc
2. The peak frequency deviation is n∆fNB
These are the desired properties of the WB FM signal.
The overall frequency multiplier device is shown below:

Complete frequency multiplier

66
Example: Noncommercial FM broadcast in the US uses the 88-90 MHz band and
commercial FM broadcast uses the 90-108 MHz band (divided into 200 kHz
channels). In either case ∆f = 75 kHz. Suppose we target a station with fc = 90.1
MHz. Then the indirect FM generation method suggested by Armstrong enables us
to achieve our goals.

67
Armstrong indirect method of FM generation

68
Demodulation of FM signals
Consider the following receiver architecture

Frequency discriminator implementation of an FM demodulator

The slope circuit is characterized by a purely imaginary transfer function


H i ( s ), i = 1,2.

Let H1(f ) be described by


 j 2πa( f − f c + BT / 2), f c − BT / 2 ≤ f ≤ f c + BT / 2

H 1 ( f ) =  j 2πa( f + f c − BT / 2), − f c − BT / 2 ≤ f ≤ − f c + BT / 2
 0, elsewhere
 69
where a > 0 is a constant that determines the slope of H1(s ).

Define G1(f ) ≡ H1(f )/j, then g1(t ) is the impulse response of a real bandpass
system described by G1(f ).

In the time domain,


g1 (t ) = g1, I (t ) cos(2πf c t ) − g1,Q (t ) cos(2πf c t )

where g1,I(t) and g1,Q(t) are the in-phase and quadrature components of g1(t).
Therefore, the complex envelope of g1(t) is described by
g~1 (t ) = g1, I (t ) + jg 1,Q (t )

which implies that


{
g1 (t ) = Re g~1 (t )e j 2π f ct }
Using this jinformation, we get
g~1 (t )e
2π f t
c ∗
+ g~1 (t )e
− j 2π f t c
= ( g1, I (t ) + jg 1,Q (t ) ) e j 2π f ct + ( g1, I (t ) − jg 1,Q (t ) ) e − j 2π f ct
= 2 g1, I (t ) cos(2π f c t ) − 2 g1,Q (t ) sin( 2π f c t )
= 2 g1 (t )
70
But,
{
F { 2 g 1 (t )} = F g~1 (t )e j 2π f ct + g~1∗ (t )e −2 jπ f ct }
or
~ ~
2G1 ( f ) = G1 ( f − f c ) + G1∗ (−( f + f c ))
~
which implies that G1 ( f ) has a lowpass frequency response limited to f ≤ BT / 2
and that
~
G1 ( f − f c ) = G1 ( f ), f > 0.
~
Observations: G1 ( f ) can be obtained by taking the part of G1(f) that corresponds
to positive frequencies, shifting it to the origin and then scaling it by a factor of 2.

G1 ( f ) H1 ( f ) / j ~
In the next figure is replaced by and G1( f ) replaced by
~
H 1 ( f ) / j.

71
~
Frequency responses of H1(f ), 1 ( f ), and H2(f )
H
72
From the previous derivation,

~  j 4π a ( f + BT / 2), f ≤ BT / 2
H1 ( f ) = 
 0, elsewhere

But,  t

sFM ( t ) = Ac cos 2π f c t + 2π k f ∫ m( τ )dτ 
 0 
{
= Re Ac e
j 2π k f ∫0t m ( τ )dτ
⋅ e j 2π f c t }
{
= Re ~sFM ( t ) ⋅ e j 2π f c t . }

If s1(t) is the output of the slope filter H1(f ) when the input is sFM(t) then the complex
envelope of the output is
~ 1~
s1 (t ) = h1 (t ) ∗ ~
s FM (t )
2
73
In the frequency domain,
~ 1 ~ ~
S1 ( f ) = H 1 ( f ) S FM ( f )
2
~
 j 2πa( f + BT / 2) S FM ( f ), f ≤ BT / 2
=
 0, elsewhere

But,
 dx(t ) 
F  = j 2π fX ( f )
 dt 

which implies that


~ d~
s (t )
s1 (t ) = a FM + jaπBT ~
s FM (t )
dt
j 2π k f ∫0t m (τ ) dτ j 2π k f ∫0t m (τ ) dτ
= j 2πaAc k f m(t )e + 2πaAc BT e
 2k f  j 2π k f ∫0t m (τ ) dτ
= jπaAc BT 1 + m(t ) e
 BT 
74
Therefore,

s1 (t ) = Re ~ {
s1 (t )e j 2π f ct }
  2k f  j 2π f ct + j 2π k f ∫0t m (τ ) dτ 
= Re  jπ aAc BT 1 + m(t ) e 
  BT  
 2k f   t

= −πaAc BT 1 + m(t ) sin  2π f c t + 2π k f ∫ m(τ )dτ 

 BT   0 
 2k f   t
π
= πaAc BT 1 + m(t ) cos 2π f c t + 2π k f ∫ m(τ )dτ + 
 BT   0
2

Observation: s1(t ) contains both AM and FM.


However, if
kf
2 m(t ) < 1, ∀t
BT

75
Then a distortionless envelope detector can extract m(t) plus a bias, i.e.,

 2k f 
s1e (t ) = πaAc BT 1 + m(t ).
 BT 
Finally, if
~ ~
H 2 ( f ) = H 1 (− f )

 2k f 
then s 2 e (t ) = π aAc BT 1 − m(t ).
 BT 

Moreover,
mˆ (t ) = s1e (t ) − s 2 e (t )
= 4π aAc m(t )
The cascade of a slope circuit and an envelope detector is known as a frequency
discriminator.

76
Frequency discriminator

FM demodulation via phase-locked loops


We consider the Phase-locked loop (PLL) FM detector shown below

Phase-locked loop FM detector 77


From the previous block diagram,
x r (t ) = Ac cos(ω c t + φ (t ))
e0 (t ) = Av sin(ω c t + θ (t ))
Let the phase detector be described by

then,
e1 (t ) = Ac cos(ω c t + φ (t )) ⋅ Av sin(ω c t + θ (t )) ⋅ k d
k d Ac Av
= [ sin[ 2ω c t + φ (t ) + θ (t )] + sin[φ (t ) − θ (t )] ]
2

78
with the proper choice of lowpass filter, the output of the phase detector is
k d Ac Av
ed (t ) = sin[φ (t ) − θ (t )]
2
A VCO is an FM modulator with peak frequency deviation
dθ (t )
∆f = max
t dt
t
dθ (t )
where
dt
= k vco mˆ (t ) ∫
implies that θ (t ) = k vco m
0
ˆ (τ )dτ .

An equivalent nonlinear model is now shown

Nonlinear model of PLL FM demodulator 79


Let the PLL operate in the lock condition, i.e., θ (t ) ≅ φ (t ) , or that θ (t ) − φ (t )
is small. Then,
sin[φ (t ) − θ (t )] ≅ φ (t ) − θ (t )
and the linear approximation of the PLL is given by

Linear model of the phase-locked loop

Let the loop filter have the transfer function HLF(s) = 1, then
k d Ac Av µ
mˆ (t ) ≅ [φ (t ) − θ (t )]
2
= k t [φ (t ) − θ (t )]
80
Thus, the output of the vco is given by
t t
θ (t ) = k vco ∫ mˆ (τ )dτ = k t k vco ∫ [φ (τ ) − θ (τ )] dτ
0 0

Let k0 = ktkvco, then

dθ (t )
= k0 [ − θ (t ) + φ (t )]
dt
or
dθ (t )
+ k 0θ (t ) = k 0φ (t ),
dt

In the s-domain, assuming zero initial conditions,


( s + k 0 )Θ( s ) = k 0 Φ ( s ),
The closed-loop transfer function is therefore given by
Θ( s ) k0
H cl ( s ) = =
Φ ( s) s + k 0
81
The corresponding impulse response is
hcl (t ) = k 0 e − k0t u (t )
Let us now find out what happens when the loop gain k0 is increased, i.e.,
k0
lim H cl ( s ) = lim =1
k0 →∞ k0 →∞ s + k
0

Clearly, Θ( s ) → Φ( s )

or θ (t ) → φ (t ) , for large k0.

Example: Let the message be a step function, i.e., m(t) = Au(t), then
 t

x FM (t ) = Ac cos ω c t + kω ∫ Au (τ )dτ 

 0 
In this case, t
φ ( t ) = kω A∫ u( τ )dτ = kω A t
0

Ak ω Ak ω k 0
Θ( s ) = 2
In the s-domain, Φ ( s ) = s 2 and s (s + k 0 ) 82
The Laplace transform of m
ˆ (t ) is then given by
Akω k t  1 1 1  Akω  1 1 
M ( s ) = k t [ Φ( s ) − Θ( s )] =
ˆ = Akω k 0    − =  − .
s( s + k0 ) k
 0  s s + k 0 k vco  s s + k 0

Let k1 = kω/kvco, then in the time domain,


mˆ (t ) = Ak1u (t ) − Ak1e − k0t u (t ) = k1 m(t ) − Ak1e − k0t u (t )

ˆ (t ) → k1 m(t ).
Clearly, as t → ∞, the estimate m

Observation: This result is valid when the initial phase error is small.

Remark: A large loop gain k0 results in practical difficulties, hence, a different loop
filter has to be used.

Consider the loop filter described by


H LF ( s ) = ( s + a ) / s, a > 0

Then the output of the vco is given by


k 0 H LF ( s )
Θ( s ) = [ Φ ( s ) − Θ( s )]
s 83
Let φ (t ) − θ (t ) be small, then the closed-loop transfer function is
Θ( s ) k 0 H LF ( s ) k ( s + a)
H cl ( s ) = = = 2 0
Φ ( s ) s + k 0 H LF ( s ) s + k 0 s + k 0 a
Define Ψ ( s ) =ˆ Φ( s ) − Θ( s ), then

k 0 H LF ( s ) s2 s2
Ψ (s) = Φ( s) − Φ(s) = 2 Φ(s) = 2 Φ( s)
s + k 0 H LF ( s ) s + k0 s + k0 a s + 2ξω n s + ω n
2

where
ωn = k0 a

2ξω n = k 0
k0 k0 1 k0
ξ= = =
2ω n 2 k 0 a 2 a

Consider again the step function message m(t) = Au(t). Then


t t
φ (t ) = k ω ∫ Au (τ )dτ = Ak ω ∫ dτ = Ak ω t
0 0 84
In the complex frequency domain,
kω A
Φ( s) = 2
s

Let ∆ω ≡ kωA, then Φ(s) = ∆ω/s2


and ∆ω
Ψ ( s) =
s 2 + 2ξω n s + ω n2

If 0 < ξ < 1, then


ψ (t ) =
∆ω
ωn 1 − ξ 2
(
e −ξωnt sin ω n 1 − ξ 2 t )
and limψ (t ) = 0.
t →∞

Hence the steady-state phase error is zero.


A typical value of ξ is 0.707.
85
Superheterodyne Receiver
Definition: To heterodyne means to combine a radio frequency wave with a locally
generated wave of different frequency in order to produce a new frequency equal to
the sum or difference of the two.

Specifically, a superheterodyne receiver is one that performs the operations of


carrier frequency tuning of the desired signal, filtering it to separate it from
unwanted signals, in most instances, amplifying it to compensate for signal power
loss due to propagation medium.

Generic superheterodyne receiver


86
Example: Let m(t) modulate a sinusoidal carrier with frequency fc = 10 MHz. Let the
bandwidth of the modulated carrier be BT = 200 kHz and let fIF = 1 MHz. Let the
local oscillator local frequency be fLO = 11 MHz and let an interferer have its
spectrum located at 11.95 ≤ f ≤ 12.05 MHz. If no filter is used in the RF section,
then, at the output of the RF filter and of the mixer we have (for f ≥ 0 ).

Spectra when no RF filter is used


87