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Angle Modulation
θi (t + ∆t ) −θi (t )
= lim (4.2.3)
∆t →0 2π∆t
1 dθi (t )
=
2π dt
1
Thus according to equation (4.2.3), we may interpret the angle modulated signal s(t) as a
rotating phasor of length Ac and angle θi(t).The angular velocity of such a phasor is
dθi(t)/dt measured in radians per second, in accordance with equation (4.2.3).
In the simple case of an un-modulated carrier, the angle θi(t) is
θi (t ) = 2πf c t +ϕc
and the corresponding phasor rotates with a constant angular velocity equal to 2πfc. The
constant φc is the value of θi(t) at t = 0.
Now we can define the two types of angle modulation, namely phase modulation and
frequency modulation mathematically.
Phase Modulation (PM) is that form of angle modulation in which the angle θi(t) is
varied linearly with the message signal m(t),as shown by
θi (t ) = 2πf c t + k p m(t ) (4.2.4)
The term 2πfct represents the angle of the un-modulated carrier, and the constant kp
represents the phase sensitivity of the modulator, expressed in radians per volts on the
assumption that m(t) is a voltage waveform. We have assumed φc zero in the equation
(4.2.4).The phase-modulated signal s(t) is thus described in the time domain by
s (t ) = Ac Cos [ 2πf c t + k p m(t )] (4.2.5)
Frequency Modulation (FM) is that form of angle modulation in which the instantaneous
frequency fi (t) is varied linearly with the message signal m(t), as shown by
f i (t ) = f c + k f m(t ) (4.2.6)
The term fc represents the frequency of the un-modulated carrier, and the constant kf
represents the frequency sensitivity of the modulator, expressed in Hertz per volt on the
assumption that m(t) is a voltage waveform.
Integrating equation (4.2.6) with respect to time and multiplying the result by 2π,we get
t
θi (t ) = [2πf c t + 2πk f ∫ m(t ) dt ] (4.2.7)
0
The consequence of allowing the angle θi(t) to become dependent on the message signal
m(t) as in equation (4.2.4) or on its integral as in equation (4.2.7) is that the zero
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crossings of a PM or FM signal no longer have a perfect regularity in their spacing. This
is one important feature that distinguishes both PM and FM from an AM signal. Another
important difference is that the envelope of a PM or FM signal is constant (equal to the
carrier amplitude), whereas the envelope of an AM signal is dependent on the message
signal. Comparing equation (4.2.5) and (4.2.8), we can say that an FM signal can be
generated by first integrating m(t) and then using the result as input to phase modulator.
Conversely, a PM signal can be generated by first differentiating m(t) and then using the
result as input to a frequency modulator.
Ac cos( 2πf c t )
Ac cos( 2πf c t )
3
t
θi (t ) = 2π ∫ f i (t )dt
0
∆f
= 2πf c t + sin( 2πf m t ) (4.3.4)
fm
and
θi (t ) = 2πf c t + β sin( 2πf m t ) (4.3.6)
In physical sense, the parameter β represents the phase deviation of the FM signal, that is
the maximum departure of the angle θi(t) from the angle 2πfct of the unmodulated carrier;
hence β is represented in radians.
Using equations (4.2.1) and (4.3.6) the FM signal is given by
s (t ) = Ac Cos [2πf c t + βSin 2πf m t )] (4.3.7)
Depending on the value of the modulation index β, there are two cases of frequency
modulation:
• Narrowband FM, for which β is small compared to 1 rad. (usually β < 0.3 rad.)
• Wideband FM, for which β is large compared to one.
4
This equation (4.3.9) defines the approximate form of a narrowband FM signal produced
by a sinusoidal modulating signal Am cos(2πfmt).From this equation, we can construct the
following block diagram, which is narrowband FM modulator.
From equations (4.3.10) and (4.3.11), we can say that the essential bandwidth for AM
and narrowband FM is same, (that is 2 f m).
Ideally, an FM signal has a constant envelope and, for the case of sinusoidal signal of
frequency fm, the angle θi (t) is also sinusoidal with the same frequency. But the sinusoidal
signal produced by the narrowband modulator of figure 4.3 differs from this ideal
condition in two fundamental respects:
1. The envelope contains a residual amplitude modulation, and therefore varies with
time.
2. For a sinusoidal modulating wave, the angle θi (t) contains harmonic distortion
in the form of third- and higher-order harmonics of the modulation frequency fm.
However by restricting the value of β ≤ 0.3 radians, the effect of residual AM and
harmonic PM are limited to negligible levels.
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4.3.2 Wideband Frequency Modulation
Lets assume the carrier frequency fc is large enough (compared to the bandwidth of the
FM signal) to make the single-tone FM wave periodic. Now, we can rewrite the equation
(4.3.7) using the complex representation of band-pass signals as follows:
s(t) = Re[ Ac e j(2πfct + β sin(2πfmt )] (4.3.12)
Re [š (t) e j2πfct ] (4.3.13)
where š(t) is the complex envelope of the FM signal s(t), defined by
š (t) = Ac e jβ sin(2πfmt) (4.3.14)
š(t) is a periodic function of time with a fundamental frequency equal to the modulation
frequency fm. We may therefore expand š(t) in the form of a complex Fourier series as
follows:
∞
∑ c e j2πnfmt
n
š(t) = n = −∞ (4.3.15)
where the complex Fourier coefficient cn is given by
(1 / 2 ) fm
c n = fm
∫ −(1 / 2 ) fm
š(t) e –j2πnfmt dt (4.3.16)
(1 / 2 ) fm
= fm Ac
∫−(1 / 2 ) fm
e[jβ sin( 2πfm t ) – j2πnfm t] dt (4.3.17)
Let,
2πfm t = x
This gives
dt = dx/ 2πfm
as t → - 1/2fm, x → -π
as t → 1/2fm , x → π
6
Ac π
cn = 2π
∫π
−
e j(βsin x – nx) dx (4.3.18)
The integral on the right hand side of equation (4.3.18) is nth order Bessel function of the
first kind with argument β.This is denoted by Jn (β) and defined as
1 π
Jn (β) = 2π
∫π
−
e j(βsin x – nx) dx (4.3.19)
cn = Ac Jn (β) (4.3.20)
∑J n β
š(t) = Ac n = −∞ e j2πnfmt (4.3.21)
∑J n (β )
s(t) = Ac Re [ n = −∞ e [ j2π(fc + n fm ) t ] ] (4.3.22)
Interchanging the order of summation and taking the real part in the right hand side of
equation (4.3.22) results
∑J n ( β ) cos[2π ( f c + nf m )t ]
s(t) = Ac n = −∞ (4.3.23)
This is the required Fourier series representation of the single-tone FM signal s(t) for an
arbitrary value of β.
Taking Fourier transform on both sides of equation (4.3.23), we get
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∞
Ac
∑J n ( β )[δ ( f − f c − nf m ) + δ ( f + f c + nf m )]
S(f) = 2 n = −∞ (4.3.24)
Some important properties of Jn (β) are listed below. (For more details on Bessel
functions see page no. 218 : Advanced Engineering Mathematics – 8th edition, by Erwin
Kreyszic).
1. Jn (β) = (-1)n J-n (β), for all n,both positive and negative.
2. For small values of the modulation index β, we have
J0(β) ≈ 1
J1(β) ≈ β/2
Jn(β) ≈ 0, for n>2.
∑J
n = −∞
n
2
(β ) = 1
3.
1
1 +
BT = 2 ∆f + 2fm = 2 ∆f β (4.4.1)
We may also define the transmission bandwidth of FM in alternative form as follows:
The transmission bandwidth of an FM wave is the separation between the two
frequencies beyond which none of the side frequencies is greater than 1 percent of the
carrier amplitude obtained when the modulation is removed. That is, transmission
bandwidth, BT = 2 nmax fm , where fm is the modulation frequency and nmax is the largest
J ( β ) > 0.01
value of the integer n that satisfies the requirement n .
The following table shows the total number of significant side frequencies (including
both upper and lower sides) for different values of β, calculated on the basis of 1%
criterion described earlier.
8
Table 4.1: Number of significant side frequencies
of a wideband FM signal for varying modulation index.
9
Figure 4.4 : Simplified block diagram of Armstrong indirect FM wave generator. The
NBFM block here is given already in figure 4.3
The NBFM generated by Armstrong’s method has some distortion because of the
approximation of the approximations we made to derive the standard equation of FM.
The output of the Armstrong NBFM modulator, as a result, also has some amplitude
modulation. Amplitude limiting in the frequency multipliers removes most of the
distortion.
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1
ωo =
LC (4.5.1)
C = Co – k m(t) (4.5.2)
1
ωo =
km(t )
LC o 1 −
Co
(4.5.3)
1
1/ 2
km(t )
LC o 1 −
Co
= (4.5.4)
km(t )
Here, C o <<1,
So,
1 km(t )
ωo ≈ 1 +
LC o 2C o
(4.5.6)
1 kω c
ωc = kf =
LC o 2C o ,
If, and
ωo = ωc + kf m(t) (4.5.7)
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4.6 Demodulation of FM
ds (t ) d
= Ac cos 2πf c t + 2πk f ∫ m(t )dt
t
dt dt 0
= 0
(4.6.1)
This signal is both amplitude and frequency modulated, the envelope being
Ac [ 2πf c + 2πk f m(t )] [2πf c + 2πk f m(t )]
.As > 0 for all t, m(t) can be obtained by envelope
detection.
12
Several factors including noise cause Ac to vary. This variation in Ac should be removed
before applying the signal to FM receiver. This can be done with the help of bandpass
limiter.
13
Figure 4.6.4: Hard limiter input and corresponding output
From the above figures (4.6.2, 4.6.3, 4.6.4, 4.6.5), we know that the bandpass limiter
output to a sinusoid will be a square wave of unit amplitude regardless of the incoming
signal sinusoidal amplitude. Moreover, the zero crossings of the incoming sinusoid are
preserved in the output because when the input is zero, output is also zero. Thus an
frequency modulated sinusoidal input vi(t) = A(t) cosθ(t) results in a constant amplitude
angle modulated square wave vo(t) as shown in figure(4.6.4).
The output vo(t) of the hard limiter is +1 or -1, depending on whether vi (t ) = A(t ) cos θ (t )
is positive or negative. As, A(t)≥0, vo(t) can be expressed as a function of θ:
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vo(θ) = 1 for cosθ > 0
= -1 for cosθ < 0. (4.6.4)
Hence, vo as a function of θ is a periodic square wave function with period 2π, which can
be expressed in the form of Fourier series as follows:
4 1 1
v o (θ ) = cos θ − cos 3θ + cos 5θ + .........
π 3 5 (4.6.5)
[
v o [θ (t )] = vo 2πf c t + 2πk f ∫ m(t )dt ]
4
[ 1
] [ ]
cos 2πf c t + 2πk f ∫ m(t )dt − cos 3 2πf c t + 2πk f ∫ m(t )dt +
3
π 1
5
[ ]
cos 5 2πf c t + 2πk f ∫ m(t )dt + ..........
= (4.6.6)
The output therefore has the original FM signal plus a frequency-multiplied FM wave
with multiplication factors 3, 5, 7,…..The output of the hard limiter is passed through a
bandpass filter with center frequency fc , and bandwidth BFM. The filter output eo(t) is the
desired frequency-modulated carrier with constant amplitude,
eo (t ) =
4
π
[
cos 2πf c t + 2πk f ∫ m(t )dt ]
(4.6.7)
The bandpass filter not only maintains the constant amplitude of the frequency-
modulated carrier but also partially suppresses the channel noise.
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FM demodulation using PLL is the most widely used method because of their low cost
and best performance (when SNR is low).Consider the PLL shown in figure (4.6.6).
The output eo(t) of the loop filter H(s) acts as an input to VCO. The free running
frequency of VCO is set at carrier frequency fc Hz , that is ωc radians/sec. The
instantaneous frequency of the VCO is given by
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where c and B are constants of the PLL.
AB
AB sin(ω c t + θ i ) cos(ω c t + θ o ) = [ sin(θ i − θ o ) + sin( 2ω c t + θ i + θ o ]
2 (4.6.10)
The sum frequency term is suppressed by the loop filter. Hence, the effective input to the
loop filter is
1
AB sin[θ i (t ) − θ o (t )]
2 . (4.6.11)
If h(t) is the unit impulse response of the loop filter,
1
eo (t ) = h(t ) × AB sin[θ i (t ) − θ o (t )]
2
τ
1
AB ∫ h(t − x) sin[θ i ( x) − θ o ( x)]dx
2
= 0 (4.6.12)
• τ
θ (t ) = AK ∫ h(t − x) sin θ e ( x)dx
0 (4.6.13)
where K=(1/2)cB and θe(t) is the phase error, defined as
t
θ i (t ) = k f ∫ m(t )dt
0 (4.6.15)
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Hence,
t
θ o (t ) = k f ∫ m(t )dt − θ e
0 (4.6.16)
1 • kf
eo (t ) = θ o (t ) ≈ m(t )
c c (4.6.17)
In FM, the interference(the noise) increases linearly with frequency, and the noise power
in the receiver output is concentrated at higher frequencies.The PSD(power spectral
density) of an audio signal is concentrated at lower frequencies below 2.1 kHZ.Thus, the
noise PSD is concentrated at higher frequencies, where m(t) is weakest.In this very
situation, the effect of noise can greatly be minimized by a process shown in
figure(4.7.1).
At the transmitter, the weakest high frequency components(beyond 2.1 kHz) of the audio
signal m(t) are boosted before modulation by a preemphasis filter of transfer function
Hp(jω).At the receiver, the demodulator output is passed through a deemphasis filter of
transfer function Hd(jω) = 1/ Hp(jω).Thus the deemphasis filter undoes the preemphasis
by attenuating(deemphasizing) the higher frequency components beyond 2.1 kHz, and
thus restores the original signal m(t).The noise, however enters at the channel, and
therefore has not been preemphasized(boosted).But when it passes through the
deemphasis filter, the filter attenuates the higher frequency components of the noise,
where most of the noise power is concentrated.
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Figure 4.7.2: Pre-emphasis filter
Stereophonic FM Broadcasting:
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transmission, initially two baseband audio signals are generated form two microphones or
more than two microphones (grouped) situated some distance apart. These two audio
signals are designated as left channel signal (L) and right channel signal(R). These
signals are then frequency modulated and transmitted in such a fashion that the left
channel signal and the right channel signal are reproduced at the receiving side. The
advantage of such stereophonic transmission is that it yields at the receiver a more
“natural ” sound. The sound heard at the receiver is more nearly what the listener would
hear if he were located at the broadcasting studio itself, where his two ears would receive
somewhat different sound.
By the time commercial stereo broadcasting began to be contemplated, monophonic
broadcasting was well established, and there were already many millions of monophonic
receivers in use. Accordingly the FCC (Federal communication commission) ruled that
no proposed stereo scheme would be acceptable unless it were entirely compatible in the
sense that a standard FM receiver, without modification, would be able to receive a
monophonic version of a stereo transmission. Beside this the FCC required that the
bandwidth requirement for stereophonic transmission should not exceed the bandwidth
allocated to monophonic transmission. Now, let us analyze the working principle of the
commercially adopted stereo broadcasting FM system.
Transmitted Signal:
The figure shown below is the block diagram representation of a stereophonic FM
transmission system.
fsc=38KHz M(t)
fp=19KHz
At the transmitting side there are two microphones or two groups of microphones
generating two audio signals designated as L(t) and R(t) as shown in the figure above.
These signals are added and subtracted to generate the signal L(t)+R(t) and L(t)-R(t).
These sum and difference signals are each bandlimited to 15KHz by the filters that is not
shown explicitly in the figure. An oscillator provides a sinusoidal waveform, referred to
as a pilot carrier at a frequency fp=19KHz. The pilot carrier is then fed to the frequency
doubler to generate a sinusoidal subcarrier of frequency fsc=38KHz. The subcarrier and
the difference signal are applied to the input of the balanced modulator, and the output o f
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the balanced modulator is [L(t)-R(t)] Cos2πfsct. the output of the modulator, L(t)+R(t)
signal and the pilot carrier is added to generate the composite signal M(t), where,
M(t)=[ L(t)+R(t)]+ [L(t)-R(t)]Cos2πfsct + Kcos2πfpt.
In the above equation K is a constant that determines the level of the pilot carrier in
comparison with the other components of the composite signal. The power spectral
density of the signal M(t) is shown in the figure below.
At the receiver, the composite signal M(t) is recovered from the frequency modulated
carrier. The composite signal is then passed to different filters as shown in the figure
below.
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Figure: Block-diagram of a Stereo Receiver.
The signal [L(t)-R(t)] Cos2πfsct is recovered from the bandpass filter with lower cutoff
f1=23KHz and Upper cutoff f2= 53KHz. The pilot carrier is generated by passing the
composite signal through a narrow band bandpass filter with center frequency at 19KHz
and the transition bandwidth of 8KHz.. The L(t) +R(t) signal is recovered by passing the
composite signal through a lowpass(Baseband) filter with cut off at 15KHz. Now the
pilot carrier is applied to the frequency doubler to regenerate the sinusoidal subcarrier.
This subcarrier is used for synchronous demodulation of the DSBSC signal [L(t)-R(t)]
Cos2πfsct. The output of the synchronous demodulator is proportional to the difference
waveform [L(t)-R(t)] . From here we can see that the transmission of the pilot carrier
helped us in regeneration of the subcarrier. .
Now having available the sum signal [L(t)+R(t)]. And the difference signal [L(t)-R(t)],
then, by adding and subtracting the signals as shown in the figure above we can
regenerate the L(t) and R(t) signals respectively.
The above system is entirely compatible with the requirements of a monophonic receiver.
In such a receiver, the sum signal L(t)+R(t) passes through the Baseband filter while the
pilot carrier and the DSBSC signal do not. Hence these latter two signal contribute
nothing to the output of a monophonic receiver and neither do they interfere with the
receiver’s operation.
Interleaving:
A monophonic receiver makes use of only of the L+R signal in the stereo transmission. In
order that the monophonic receiver be as loud and disturbance free as possible, it is
necessary that the sum signal Vs=L+R, which modulates the FM carrier, be as large as
possible. If the only component of the modulation waveform were the sum signal VS, we
would be at a liberty to increase the peak excursion of VS to the point where the
corresponding peak instantaneous-frequency deviation of the carrier would
be±75KHZ(FCC recommendation). However, in order to accommodate to the
requirements of the stereo receiver, the composite modulating waveform must include as
well the DSBSC signal Vd=[L-R] Cos2πfsct. we now require the peak excursion of Vs+Vd
be no larger than the peak excursion previously allowed to Vs alone. These considerations
22
suggests that, when Vd is added to Vs, the level of Vs needs to be reduced and that as a
consequence, monophonic reception of a stereophonic transmission would be inferior to
monophonic reception of a monophonic transmission. But such is not the case. This
characteristic of the stereo system under consideration is known as INTERLEAVING.
Although the microphones L(t) and R(t) are physically apart, they will ordinarily not be
greatly different. After all , the microphones which generates the two signals are intended
to represent a person’s two ears. Hence, if naturalness in sound is to be achieved,
presumably the placement of the microphones at the studio will take this fact into
account. Thus we may expect that the levels of the signal output of the two microphones
will be comparable. The maximum excursion LM of L(t) and RM of R(t) will be about the
same. Therefore we set the maximum of the sum signal Vsm at:
Vsm=LM+RM≈2LM≈2RM
Turning now to the composite signal M(t), we note that Cos2πfsct rapidly oscillates
between +1 and –1, and ignoring temporarily the pilot carrier , we have that M(t)
oscillates rapidly between M(t)=2L(t)or 2R(t) thus the maximum excursion will be either
2LM or 2RM. Hence, in summary, we find that the addition of the difference signal Vd to
the sum signal Vs does not increase the peak signal excursion.
Now let us consider the effect of the pilot carrier. When pilot carrier is added, it increase
the peak excursion. Hence the addition of the pilot carrier calls for the reduction of the
sound signal modulation level. A low-level pilot carrier allows greater sound signal
modulation, while a high level pilot carrier eases the burden of extracting the pilot carrier
at the receiver. As an Engineering compromise, the FCC standards call for a pilot carrier
of such level that the peak sound modulation amplitude has to be reduced to about 90
percent of what would be allowed in the absence of a carrier.
******The End*****
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