Lecture Notes On

Communication Engineering

Topics Covered:

1. Amplitude Modulation Techniques

2. Angle Modulation
Fourier Transform

The Fourier transform is the extension of the Fourier series to the general class of signals (periodic and
nonperiodic). Here, as in Fourier series, the signals are expressed in terms of complex exponentials of
various frequencies, but these frequencies are not discrete. Hence, in this case, the signal has a
continuous spectrum as opposed to a discrete spectrum. Fourier Transform of a signal x(t) can be
expressed as:

∫ x(t) e
− j 2π ft
F [x(t)] = X(f) = dt
−∞

x (t) ⇔ X(f) represents a Fourier Transform pair

The time-domain signal x(t) can be obtained from its frequency domain signal X(f) by Fourier
inverse defined as:

∞
x(t) = F−1 [ X (f)] = ∫ X (f)e
j 2π ft
df
−∞

When frequency is defined in terms of angular frequency ω ,then Fourier transform relation
can be expressed as:

∫ x(t) e
− jωt
F [x(t)] = X(ω ) = dt
−∞

and

∞
1
x(t) = F−1 [ X (ω )] = ∫ X (ω )e
jωt
dω
2π −∞

Properties of Fourier Transform

Let there be signals x(t) and y(t) ,with their Fourier transform pairs:

x(t) ⇔ X(f)
y(t) ⇔ Y(f) then,

1. Linearity Property
ax (t) + by(t) ⇔ aX(f) + bY(f) , where a and b are the constants

2. Duality Property
X (t) ⇔ x(− f )or
X(t) ⇔ 2π X(−ω)

3. Time Shift Property

x(t− t0 ) ⇔ e− j 2π ft0 X (f)
 4. Time Scaling Property
1 ⎛f ⎞
x(at) ⇔ X⎜ ⎟
a ⎝a⎠
5. Convolution Property: If convolution operation between two signals is defined as:
∞
x(t) ⊗ y(t) = ∫ x (τ )x ( t − τ ) dτ , then
−∞

x (t) ⊗ y(t) ⇔ X(f) Y(f)

6. Modulation Property
e j 2π f0t x(t) ⇔ X(f − f0 )
7. Parseval’s Property
∞ ∞

∫ x(t) y∗ (t) dt = ∫ X (f) Y (f)df

∗

−∞ −∞

8. Autocorrelation Property: If the time autocorrelation of signal x(t) is expressed as:

∞
Rx (τ ) = ∫ x(t) x∗ (t − τ )dt
−∞
,then

2
Rx (τ ) ⇔ X (f)
9. Differentiation Property:
d
x (t) ⇔ j 2π fX (f)
dt

Response of a linear system

The reason what makes Trigonometric Fourier Series expansion so important is the unique
characteristic of the sinusoidal waveform that such a signal always represent a particular frequency.
When any linear system is excited by a sinusoidal signal, the response also is a sinusoidal signal of
same frequency. In other words, a sinusoidal waveform preserves its wave-shape throughout a linear
system. Hence the response-excitation relationship for a linear system can be characterised by, how the
response amplitude is related to the excitation amplitude (amplitude ratio) and how the response phase
is related to the excitation phase (phase difference) for a particular frequency. Let the input to a linear
system be :

vi ( t , ωn ) = Vn e jωnt

Then the filter output is related to this input by the Transfer Function (characteristic of the Linear
Filter): H (ω n ) = H (ω n ) e
− jθ (ω n )
, such that the filter output is given as

vo ( t , ω n ) = Vn H ( ω n ) e ( n
j ω t − jθ (ω n ) )
Normalised Power

While discussing communication systems, rather than the absolute power we are interested in another
quantity called Normalised Mean Power. It is an average power normalised across a 1 ohm resistor,
averaged over a single time-period for a periodic signal. In general irrespective of the fact, whether it is
a periodic or non-periodic signal, average normalised power of a signal v(t) is expressed as :

T
2
1
P = lim ∫ v ( t )dt
2
T →∞ T −T
2

Energy of signal

For a continuous-time signal, the energy of the signal is expressed as:

∞
E= ∫ x (t)dt
2

−∞

A signal is called an Energy Signal if

0<E<∞
P=0

A signal is called Power Signal if

0< P<∞
E =∞

Normalised Power of a Fourier Expansion

If a periodic signal can be expressed as a Fourier Series expansion as:

v ( t ) = C0 + C1 cos ( 2π f0t ) + C2 cos ( 4π f0t ) + ...

Then, its normalised average power is given by :

T
2
1
v2 ( t )dt
T →∞ T ∫
P = lim
−T
2

Integral of the cross-product terms become zero, since the integral of a product of orthogonal signals
over period is zero. Hence the power expression becomes:

C12 C22
P = C0 + 2
+ + ...
2 2
By generalisation, normalised average power expression for entire Fourier Series becomes:
 Cn2 ∞
P = C0 + ∑2
+ ...
n=1 2

In terms of trigonometric Fourier coefficients An‘s, Bn‘s, the power expression can be written as:

∞ ∞
P = A02 + ∑ An2 + ∑ Bn2
n=1 n=1

In terms of complex exponential Fourier series coefficients Vn’s, the power expressions becomes:

∞
P = ∑ VnVn*
n=−∞

Energy Spectral Density(ESD)

It can be proved that energy E of a signal x(t) is given by :

∞ ∞ 2

E= ∫
−∞
x 2 (t)dt = ∫
−∞
X (f) df → Parseval’s Theorem for energy signals

∫
2
So, E = ψ (f)df , where ψ (f) = X (f) → Energy Spectral Density
−∞

The above expression says that ψ (f) integrated over all of the frequencies, gives the total energy of the
signal. Hence Energy Spectral Density (ESD) quantifies the energy contribution from every frequency
component in the signal, and is a function of frequency.

Power Spectral Density(PSD)

It can be proved that the average normalised power P of a signal x(t),such that xτ (t) is a truncated and
⎧ −τ τ⎫
⎪ x (t); <t< ⎪
periodically repeated version of x(t) such that xτ (t) = ⎨ 2 2⎬ is given by :
⎪⎩ 0; elsewhere ⎭⎪

τ τ 2
2 2
1 1
P = lim ∫τ x ( t )dt = τlim τ ∫τ X τ ( t ) dt → Parseval’s Theorem for power signals
2
τ →∞ τ −
→∞
−
2 2

∞ 2
Xτ (f)
So, P = ∫ S(f)df , where S (f) =τ lim
−∞
JJJG ∞
τ
→ Power Spectral Density

The above expression says that S(f)integrated over all of the frequencies, gives the total normalised
power of the signal. Hence Power Spectral Density (PSD) quantifies the power contribution from every
frequency component in the signal, and is a function of frequency.
Expansion in Orthogonal Functions

Let there be a set of functions g 1 (x), g 2 (x), g 3 (x), ..., g n (x) , defined over the interval x1 < x < x 2 and
such that any two functions of the set have a special relation:
x2

∫ g (x) g
x1
i j (x) dx = 0 .

The set of functions showing the above property are said to be an orthogonal set of functions in the
interval x1 < x < x 2 . We can then write a function f (x) in the same interval x1 < x < x 2 , as a linear
sum of such g n (x) ’s as:

f (x) = C 1 g 1 (x) + C 2 g 2 (x) + C 3 g 3 (x) + ... + C n g n (x) , where Cn’s are the numerical coefficients

The numerical value of any coefficient Cn can be found out as:

x2

∫ f (x) g (x)dx
x1
n

Cn = x2

∫g
2
n (x) dx
x1

x2
In a special case when the functions g n (x) in the set are chosen such that
∫g (x) dx =1, then such a
2
n
x1

set is called as a set of orthonormal functions, that is the functions are orthogonal to each other and each
one is a normalised function too.
 Amplitude Modulation Systems

In communication systems, we often need to design and analyse systems in which many independent
message can be transmitted simultaneously through the same physical transmission channel. It is
possible with a technique called frequency division multiplexing, in which each message is translated in
frequency to occupy a different range of spectrum. This involves an auxiliary signal called carrier
which determines the amount of frequency translation. It requires modulation, in which either the
amplitude, frequency or phase of the carrier signal is varied as according to the instantaneous value of
the message signal. The resulting signal then is called a modulated signal. When the amplitude of the
carrier is changed as according to the instantaneous value of the message/baseband signal, it results in
Amplitude Modulation. The systems implementing such modulation are called as Amplitude modulation
systems.

Frequency Translation

Frequency translation involves translating the signal from one region in frequency to another region. A
signal band-limited in frequency lying in the frequencies from f1 to f2, after frequency translation can be
translated to a new range of frequencies from f1’ to f2’. The information in the original message signal at
baseband frequencies can be recovered back even from the frequency-translated signal. The
advantagesof frequency translation are as follows:

1. Frequency Multiplexing: In a case when there are more than one sources which produce band-
limited signals that lie in the same frequency band. Such signals if transmitted as such
simultaneously through a transmission channel, they will interfere with each other and cannot
be recovered back at the intended receiver. But if each signal is translated in frequency such
that they encompass different ranges of frequencies, not interfering with other signal spectrums,
then each signal can be separated back at the receiver with the use of proper filters. The output
of filters then can be suitably processed to get back the original message signal.
2. Practicability of antenna: In a wireless medium, antennas are used to radiate and to receive the
signals. The antenna operates effectively, only when the dimension of the antenna is of the
order of magnitude of the wavelength of the signal concerned. At baseband low frequencies,
wavelength is large and so is the dimension of antenna required is impracticable. By frequency
translation, the signal can be shifted in frequency to higher range of frequencies. Hence the
corresponding wavelength is small to the extent that the dimension of antenna required is quite
small and practical.
3. Narrow banding: For a band-limited signal, an antenna dimension suitable for use at one end of
the frequency range may fall too short or too large for use at another end of the frequency
range. This happens when the ratio of the highest to lowest frequency contained in the signal is
large (wideband signal). This ratio can be reduced to close around one by translating the signal
to a higher frequency range, the resulting signal being called as a narrow-banded signal.
Narrowband signal works effectively well with the same antenna dimension for both the higher
end frequency as well as lower end frequency of the band-limited signal.
4. Common Processing: In order to process different signals occupying different spectral ranges
but similar in general character, it may always be necessary to adjust the frequency range of
operation of the apparatus. But this may be avoided, by keeping the frequency range of
operation of the apparatus constant, and instead every time the signal of interest beingtranslated
down to the operating frequency range of the apparatus.
Amplitud
de Modulatioon Types:

1. Double-sidebaand with carrier (DSB+C)

D
2. D
Double-sideba and suppressed carrier (DSB-SC)
3. Single-sidebannd suppressedd carrier (SSB
B-SC)
4. V
Vestigial sidebband (VSB)

Double-siideband with
h carrier (DS
SB+C)

Let there be a sinusoiidal carrier signal

s t , of frequen
c (t) = A C o s(2 π f c t) ncy fc . Withh the conceptt of
amplitudee modulation,, the instantaaneous amplittude of the carrier signal will be moduulated (changged)
proportionnally accordinng to the insttantaneous am
mplitude of th
he baseband or modulatinng signal x(t).. So
the expresssion for the Amplitude
A M
Modulated (AMM) wave becoomes:

s(t) = [ A + x(tt)] Cos(2π fc t) = E(t)Coss(2π fc t)

E (t) = A + x(tt)

The time varying ampplitude E(t) ofo the AM w wave is calledd as the envvelope of the AM wave. The
envelope of the AM waave has the saame shape as the message signal or baseeband signal.

Figuree 3 Amplitude modulation

m tim
me-domain plot

Modulatioon Index (ma):

) It is definedd as the meassure of extent of amplitudee variation about unmodulaated
maximumm carrier ampplitude. It is i also called as depth of o modulatioon, degree off modulationn or
modulatioon factor.
 x(t) max
ma =
A
On the basis of modulation index, AM signal can be from any of these cases:

I. m a > 1 : Here the maximum amplitude of baseband signal exceeds maximum carrier

amplitude, x(t) max > A. In this case, the baseband signal is not preserved in the AM envelope,
hence baseband signal recovered from the envelope will be distorted.
II. m a ≤ 1 : Here the maximum amplitude of baseband signal is less than carrier amplitude

x(t) max ≤ A . The baseband signal is preserved in the AM envelope.

Spectrum of Double-sideband with carrier (DSB+C)

Let x(t) be a bandlimited baseband signal with maximum frequency content fm. Let this signal
modulate a carrier c (t) = A C os(2 π f c t) .Then the expression for AM wave in time-domain is given by:

s (t) = [ A + x (t) ] Cos (2π f c t)

= ACos(2π f c t) + x(t) Cos(2π f c t)

Taking the Fourier transform of the two terms in the above expression will give us the spectrum of the
DSB+C AM signal.

1
ACos(2π f c t) ⇔ [δ (f + fc ) + δ (f − fc )]
2
1
x(t) Cos(2π f c t) ⇔ [ X (f + fc ) + X(f − fc )]
2

So, first transform pair points out two impulses at f = ± f c , showing the presence of carrier signal in
the modulated waveform. Along with that, the second transform pair shows that the AM signal
spectrum contains the spectrum of original baseband signal shifted in frequency in both negative and
positive direction by amount f c . The portion of AM spectrum lying from f c to f c + f m in positive
frequency and from − f c to − f c − f m in negative frequency represent the Upper Sideband(USB). The
portion of AM spectrum lying from f c − f m to f c in positive frequency and from − f c + f m to − f c in
negative frequency represent the Lower Sideband(LSB). Total AM signal spectrum spans a frequency
from f c − f m to f c + f m , hence has a bandwidth of 2 f m .

Power Content in AM Wave

By the general expression of AM wave:

s(t) = ACos(2π fct) + x(t)Cos(2π fct)

Hence, total average normalised power of an AM wave comprises of the carrier power corresponding to
first term and sideband power corresponding to second term of the above expression.
 Ptotal = Pcarrier + Psideband
T /2
1 A2
T −T∫/2
Pcarrier = lim A 2
Cos 2
(2π f c t)dt =
T →∞ 2
T /2
1 1
Psideband = lim
T →∞
∫
T −T /2
x 2 (t)Cos 2 (2π f c t)dt = x 2 (t)
2

In the case of single-tone modulating signal where x(t) = Vm Cos(2π fmt) :

A2
Pcarrier =
2
V 2
Psideband = m
4
A 2 Vm 2
Ptotal = Pcarrier + Psideband = +
2 4
⎡ ma ⎤2
⇒ Ptotal = Pcarrier ⎢1 + ⎥
⎣ 2 ⎦

Where, ma is the modulation index given as m a = V m .

A

Net Modulation Index for Multi-tone Modulation: If modulating signal is a multitone signal
expressed in the form:

x(t) = V1 Cos(2π f1t) + V2 Cos(2π f2t) + V3 Cos(2π f3t) +... + Vn Cos(2π fnt)

⎡ 2 2 2 2
⎤
Then, Ptotal = Pcarrier ⎢1 + m1 + m 2 + m3 ... m n ⎥
⎣ 2 2 2 2 ⎦

Where m1 = V1 , m 2 = V 2 , m 3 = V 2 ,..., m n = V n
A A A A

Generation of DSB+C AM by Square Law Modulation

Square law diode modulation makes use of non-linear current-voltage characteristics of diode.
This method is suited for low voltage levels as the current-voltage characteristic of diode is highly non-
linear in the low voltage region. So the diode is biased to operate in this non-linear region for this
application. A DC battery Vc is connected across the diode to get such a operating point on the
characteristic. When the carrier and modulating signal are applied at the input of diode, different
frequency terms appear at the output of the diode. These when applied across a tuned circuit tuned to
carrier frequency and a narrow bandwidth just to allow the two pass-bands, the output has the carrier
and the sidebands only which is essentially the DSB+C AM signal.
 Figurre 4 Current-vooltage characterristic of diode

F
Figure 5 Squaree Law Diode Modulator
M

T non-linearr current voltaage relationshhip can be wriitten in generral as:

The

i = av + bv 2

n this applicattion v = c (t) + x(t)

In

So
Cos(2π f c t) + x(t)] + b[AC
i = a[AC Cos(2π f c t) + x(t)]2
⇒ i = a ACos(2
A A2 Cos 2 (2π f c t) + b x 2 (t) + 2bA x(t) Cos(2π f c t)
π f c t)) + a x(t) + bA
bbA2 A2
bA
⇒ i = a ACos(2
A π f c t)
t + a x(t) + Cos(2π (2
( f c ) t) + os(2π f c t)
+ b x 2 (t) + 2bA x(t) Co
2 2

Out of the above frequuency terms, only the boxxed terms haave the frequeencies in the passband off the
tuned circcuit, and hencce will be at thhe output of thhe tuned circu
uit. There is ccarrier frequeency term andd the
sideband term
t which comprise essenntially a DSB B+C AM signal.
Demodulation of DSB+C by Square Law Detector

It can be used to detect modulated signals of small magnitude, so that the operating point may be
chosen in the non-linear portion of the V-I characteristic of diode. A DC supply voltage is used to get a
fixed operating point in the non-linear region of diode characteristics. The output diode current is hence

Figure 6 Square Law Detector

given by the non-linear expression:

i = avFM (t) + bv2FM (t)

Where vFM (t) =[A+ x(t)]Cos(2π fct)

This current will have terms at baseband frequencies as well as spectral components at higher
frequencies. The low pass filter comprised of the RC circuit is designed to have cut-off frequency as the
highest modulating frequency of the band limited baseband signal. It will allow only the baseband
frequency range, so the output of the filter will be the demodulated baseband signal.

Linear Diode Detector or Envelope Detector

This is essentially just a half-wave rectifier which charges a capacitor to a voltage to the peak voltage of
the incoming AM waveform. When the input wave's amplitude increases, the capacitor voltage is
increased via the rectifying diode quickly, due a very small RC time-constant (negligible R) of the
charging path. When the input's amplitude falls, the capacitor voltage is reduced by being discharged by
a ‘bleed’ resistor R which causes a considerable RC time constant in the discharge path making
discharge process a slower one as compared to charging. The voltage across C does not fall appreciably
during the small period of negative half-cycle, and by the time next positive half cycle appears. This
cycle again charges the capacitor C to peak value of carrier voltage and thus this process repeats on.
Hence the output voltage across capacitor C is a spiky envelope of the AM wave, which is same as the
amplitude variation of the modulating signal.
 Figure 7 E
Envelope Detector

Double Sideband Sup

ppressed Carrrier(DSB-SC
C)

If the carrrier is supprressed and only

o the sideebands are trransmitted, thhis will be a way to savving
transmitteer power. Thhis will not affect the innformation content
c of thhe AM signaal as the carrrier
componennt of AM sign nal do not caarry any inforrmation aboutt the basebannd signal variaation. A DSB
B+C
AM signaal is given by::

AC π fct) + x(t)Cos(2π fct)

sDSB+C (t) = ACos(2
So, the exxpression for DSB-SC
D wheere the carrierr has been supppressed can bbe given as:

sDSB−SC (t) = x(t)Cos(2

x( π fct)
t
Thereforee, a DSB-SC signal is obtaained by simmply multiplyiing modulatinng signal x(t)) with the carrrier
signal. Thhis is accompllished by a prroduct modu
ulator or mixeer.

Figure 8 P
Product Modulaator

Difference from the thee DSB+C beiing only the aabsence of carrrier componeent, and sincee DSBSC has
still both the
t sidebandss, spectral spaan of this DSB
BSC wave is still f c − f m tto f c + f m , hence
h has a
bandwidthh of 2 f m .

Generatio
on of DSB-SC
C Signal
 A circuit which can prroduce an outtput which is tthe product of
o two signals input to it is called a prodduct
modulatorr. Such an output when thee inputs are thhe modulatingg signals and the carrier sig
gnal is a DSBBSC
signal. On
ne such produuct modulatorr is a balancedd modulator.

Balanced modulator:

v1 = Coos(2π fc t) + x(t)
x
v2 = Coos(2π fc t) − x(t)
x

For diode D1,the nonlinnear v-i relationship becom

mes:

i1 = av1 + bv12 = a[Coos(2π fct) + x(t)]

x + b[Coss(2π fct) + x(t
(t)]2

Similarly,, for diode D2,

i2 = av2 + bv22 = a[Coos(2π fct) − x(t)]

x + b[Coss(2π fct) − x((t)]2

vi = v3 − v4 = (i1 − i2 ) R
Now, ((substituting for
f i1and i2)
⇒ vi = 2 R[ax(tt) + 2 bx(t) Coos(2π fc t)]

This voltaage is input to the bandpaass filter centrre frequency fc and bandw width 2fm. Hennce it allows the
componennt correspond ding to the seccond term of tthe vi, which is our desiredd DSB-SC siggnal.

Demodullation of DSB
BSC signal

Synchronous Detectionn: DSB-SC signal is generated at the transmitter

t byy frequency up-translating
u the
baseband spectrum by the carrier frrequency fc . Hence
H the oriiginal basebannd signal can be recoveredd by
frequencyy down-transllating the reeceived moduulated signal by the sam me amount. Recovery
R cann be
achieved by
b multiplyin
ng the receiveed signal by syynchronous carrier
c signal aand then low--pass filteringg.
 Figure 9 Synchronous Detection of DSBSC

Let the received DSB-SC signal is :

r(t) = x(t)Cos(2π fct)

So after carrier multiplication, the resulting signal:

e (t) = x(t) Cos(2π f c t).Cos(2π f c t)

⇒ e (t) = x(t) Cos 2 (2π f c t)
1
⇒ e (t) = x(t) [1 + Cos(2π (2 f c ) t) ]
2
1 1
⇒ e (t) = x(t) + x(t) Cos(2π (2 f c ) t)
2 2

The low-pass filter having cut-off frequency fm will only allow the baseband term 1 x(t) , which is in the
2
pass-band of the filter and is the demodulated signal.

Single Sideband Suppressed Carrier (SSB-SC) Modulation

The lower and upper sidebands are uniquely related to each other by virtue of their symmetry about
carrier frequency. If an amplitude and phase spectrum of either of the sidebands is known, the other
sideband can be obtained from it. This means as far as the transmission of information is concerned,
only one sideband is necessary. So bandwidth can be saved if only one of the sidebands is transmitted,
and such a AM signal even without the carrier is called as Single Sideband Suppressed Carrier signal. It
takes half as much bandwidth as DSB-SC or DSB+C modulation scheme.

For the case of single-tone baseband signal, the DSB-SC signal will have two sidebands :

The lower side-band: Cos(2π(fc −fm)t) = Cos(2π fmt)Cos(2π fct) + Sin(2π fmt)Sin(2π fct)

And the upper side-band: Cos(2π(fc +fm)t) = Cos(2π fmt)Cos(2π fct) − Sin(2π fmt)Sin(2π fct)
If any one of these sidebands is transmitted, it will be a SSB-SC waveform:

s(t)SSB = Cos(2π fmt)Cos(2π fct) ± Sin(2π fmt)Sin(2π fct)

Where (+) sign represents for the lower sideband, and (-) sign stands for the upper sideband. The
modulating signal here is x(t) = Cos(2π fmt) , so let xh (t) = Sin(2π fmt) be the Hilbert Transform
⎛ π⎞
of x (t) . The Hilbert Transform is obtained by simply giving ⎜ − ⎟ to a signal. So the expression
⎝ 2⎠
for SSB-SC signal can be written as:

s(t)SSB = x(t)Cos(2π fct) ± xh (t)Sin(2π fct)

⎛ π⎞
Where xh (t) is a signal obtained by shifting the phase of every component present in x (t) by ⎜ −
⎟.
⎝ 2⎠
Generation of SSB-SC signal

Frequency Discrimination Method:

 
Figure 10 Frequency Discrimination Method of SSB‐SC Generation 

The filter method of SSB generation produces double sideband suppressed carrier signals (using a
balanced modulator), one of which is then filtered to leave USB or LSB. It uses two filters that have
different passband centre frequencies for USB and LSB respectively. The resultant SSB signal is then
mixed (heterodyned) to shift its frequency higher.

Limitations:

I. This method can be used with practical filters only if the baseband signal is restricted at its
lower edge due to which the upper and lower sidebands do not overlap with each other. Hence
it is used for speech signal communication where lowest spectral component is 70 Hz and it
may be taken as 300 Hz without affecting the intelligibility of the speech signal.
II. The design of band-pass filter becomes quite difficult if the carrier frequency is quite higher
than the bandwidth of the baseband signal.

Phase-Shift Method:
 Figure 11 Phase shift method of SSB--SC generation
n

The phhase shifting method

m of SSSB generationn uses a phasee shift techniqque that causees one of the side
s
bands tot be cancelleed out. It uses two balanceed modulatorrs instead of oone. The balaanced modulaators
effectivvely eliminatte the carrierr. The carrier oscillator is
i applied dirrectly to the upper balannced
modulaator along wiith the audio modulating ssignal. Then both b the carrier and modu ulating signal are
shiftedd in phase byy 90o and appplied to the second,
s loweer, balanced modulator.
m Thhe two balannced
modulaator output are
a then added together allgebraically. The T phase shhifting action causes one side s
band to o be cancelled
d out when thhe two balanced modulatorr outputs are ccombined.

Demoodulation off SSB-SC Siggnals:

The baseband or modulating

m signal
s x(t) can be recovvered from thhe SSB-SC signal by ussing
synchrronous detectiion techniquee. With the heelp of synchroonous detectioon method, th he spectrum of an
SSB-SC signal centtered about , is rretranslated to
o the basedannd spectrum which
w is centeered
about . The process
p of synnchronous detection involv ves multiplicaation of the received
r SSB-SC
signal with a locally
y generated caarrier.

Incom
ming SSB-SC
C Multiplie
er x(
Low (t)
w Pass 
Filter (LPF) 

The ou
utput of the multiplier
m will be

or

or

or

or
When ed(t)is passed through a low-pass filter, the terms centre at ±ωc are filtered out and the output
of detector is only the baseband part i.e. 1 x(t) .
2

Vestigial Sideband Modulation(VSB)

SSB modulation is suited for transmission of voice signals due to the energy gap that exists in the
frequency range from zero to few hundred hertz. But when signals like video signals which contain
significant frequency components even at very low frequencies, the USB and LSB tend to meet at
the carrier frequency. In such a case one of the sidebands is very difficult to be isolated with the help
of practical filters. This problem is overcome by the Vestigial Sideband Modulation. In this
modulation technique along with one of the sidebands, a gradual cut of the other sideband is also
allowed which comes due to the use of practical filter. This cut of the other sideband is called as the
vestige. Bandwidth of VSB signal is given by :

BW = ( fc + fv ) −( fc − fm) = fm + fv

Where fm →bandwidth of bandlimited message signal

fv →width of the vestige in frequency

 Angle Modulation

Angle modulation may be defined as the process in which the total phase angle of a carrier wave is
varied in accordance with the instantaneous value of the modulating or message signal, while amplitude
of the carrier remain unchanged. Let the carrier signal be expressed as:

c(t) = ACos(2π f c t + θ )

Where φ = 2π f c t + θ → Total phase angle

θ → phase offset

f c → carrier frequency

So in-order to modulate the total phase angle according to the baseband signal, it can be done by either
changing the instantaneous carrier frequency according to the modulating signal- the case of Frequency
Modulation, or by changing the instantaneous phase offset angle according to the modulating signal- the
case of Phase Modulation. An angle-modulated signal in general can be written as

u (t ) = ACos (φ (t ))

where, φ (t) is the total phase of the signal, and its instantaneous frequency f i (t) is given by

1 d
fi (t ) = φ (t )
2π dt

Since u(t) is a band-pass signal, it can be represented as

u ( t ) = ACos ( 2π f ct + θ ( t ) )

and, therefore instantaneous frequency fi becomes :

1 d
fi (t ) = fc + θ (t )
2π dt

For angle modulation, total phase angle can modulated either by making the instantaneous frequency or
the phase offset to vary linearly with the modulating signal.

Let m(t) be the message signal, then in a Phase Modulation system we implement to have

θ ( t ) = θ +k p m ( t ) and with constant fc, we get (t) linearly varying with message signal.

and in an Frequency Modulation system letting phase offset θ be a constant, we implement to have

f i ( t ) = f c + k f m ( t ) , to get (t) linearly varying with message signal

where kp and kfare phase and frequency sensitivity constants.

The maximum phase deviation in a PM system is given by:

Δθ max = k p m ( t ) max

And the maximum frequency deviation in FM is given by:

Δf max = k f m ( t ) max
Δωmax = 2π k f m ( t ) max

Single Tone Frequency Modulation

(
The general expression for FM signal is s ( t ) = ACos ωct + k f m(t) dt
∫ )
So for the single tone case, wheremessage signal is m ( t ) = VCos (ωm t )

⎛ k fV ⎞
Then s (t) = ACos ⎜ ωc t + Sin(ωmt ) ⎟
⎝ ωm ⎠

⇒ s (t) = ACos (ωc t + m f Sin(ωm t ) )

kfV Δω
Here m f = = → Modulation Index
ωm ωm

Types of Frequency Modulation

High frequency deviation =>High Bandwidth=> High modulation index=>Wideband FM

Small frequency deviation =>Small Bandwidth=> Small modulation index=>Narrowband FM

Carson’s Rule

It provides a rule of thumb to calculate the bandwidth of a single-tone FM signal.

Bandwidth = 2 ( Δf + f m ) = 2 (1 + m f ) f m

If baseband signal is any arbitrary signal having large number of frequency components, this rule can be
modified by replacing m f by deviation ratio D.
Then the bandwidth of FM signal is given as: Bandwidth = 2 (1 + D ) f max

Spectrum of a Single-tone Narrowband FM signal

A single-tone FM modulated signal is mathematically given as:

s (t) = ACos (ωc t + m f Sin(ωmt ) )

⇒ s(t) = ACos (ωc t ) Cos(m f Sin(ωm t )) − A Sin(ωc t )Sin( m f Sin(ωm t ))

Since for narrowband FM modulation index mf<<1, sowe approximate as:

Cos( m f Sin(ωmt )) ≈ 1 and Sin( m f Sin(ωmt )) ≈ m f Sin(ωmt )

s (t) = ACos (ωc t ) − Am f Sin(ωc t ) Sin(ωmt )

And the expression s(t) becomes: Am f
⇒ s(t) = ACos(ωc t ) + {Cos(ωc + ωm ) t − Cos(ωc − ωm ) t}
2

The above equation represents the NBFM signal. This representation is similar to an AM
signal, except that the lower sideband frequency has a negative sign.

Spectrum of a Single-tone Wideband FM signal

A single-tone FM modulated signal is mathematically given as:

s (t) = ACos (ωc t + m f Sin(ωmt ) )

⇒ s(t) = ACos (ωc t ) Cos(m f Sin(ωm t )) − A Sin(ωc t )Sin( m f Sin(ωm t ))

The FM signal can be expressed in the complex envelope form as:

jωc t + jm f Sin (ωm t )

s (t) = Re ⎡⎣ Ae ⎤
⎦
jm f Sin (ωm t )
⇒ s(t) = Re ⎡⎣ Ae * e jωct ⎤⎦
 * e jωct ⎤⎦
⇒ s(t) = Re ⎡⎣s(t)

jm f Sin (ωm t ) 1
 = Ae
Where s(t) , which is a periodic function of period .
fm

The Fouries series expansion of this periodic function can be written as:
 ∞
 =
s(t) ∑Ce
n =−∞
n
j 2π nf m t

Where Cn spectral coefficients are given by

1
2 fm

Cn = f m ∫1
s (t) e− j 2π nfmt dt
−
2 fm
1
2 fm
⎡e jm f Sin (ωmt ) − j 2π nfmt ⎤ dt
⇒ Cn = Af m ∫1
⎣ ⎦
−
2 fm

Substituting x = 2π f mt , the above equation becomes,

π
A
∫π ⎡⎣e ⎤ dx
jm f Sin (x) − jnx
Cn = ⎦
2π −

As the above expression is in the form of nth order Bessels function of first kind :

π
1
∫π ⎡⎣e ⎤ dx ,
jm f Sin (x) − jnx
J n (m f ) = ⎦
2π −

therefore we can write Cn = AJ n (m f )

∞
 =
So, s(t) ∑ AJ
n =−∞
n (m f )e j 2π nf mt

Hence the FM signal in complex envelope form can be written as:

⎡ ∞ j 2π nf t +ω t ⎤
s (t) = A * Re ⎢ ∑ J n (m f )e ( m c ) ⎥
⎣ n =−∞ ⎦

⎡ ∞ ⎤
s (t) = A * ⎢ ∑ J n (m f ) Cos(2π nf mt + ωc t ) ⎥
⎣ n =−∞ ⎦

This is the Fourier series representation of Wideband Single-tone FM signal. Its Fourier Transform can
be written as:

⎡ ∞ ⎤
S(f) = A * ⎢ ∑ J n (m f ) {δ (f + f c + nf m ) + δ (f − f c − nf m )}⎥
⎣ n =−∞ ⎦

The spectrum of Wideband Single-tone FM signal indicates the following:

1. WBFM has infinite number of sidebands at frequencies (f c ± nf m ) .

 2. Sppectral amplitude values depends
d upon J n (m f ) .
3. The o significant sidebands ddepends uponn the modulaation index m f .For m f <<
T number of <1,
J 0 (m f ) and J1 (m f ) are only
o significaant, whereas for m f >> 1 , many signiificant sidebaands
exxists.

Methods of Generatin
ng FM wave

Direct FM
M: In this method
m the caarrier frequenncy is directlly varied inacccordance wiith the incom
ming
message signal
s to prod
duce a frequenncy modulated signal.

Indirect FM: This metthod was firstt proposed byy Armstrong. In thismethodd, the modulaating wave is first
f
used to produce
p a naarrow-band FMwave,
F annd frequency multiplicatioon is next used
u to increease
thefrequen
ncy deviationn to the desireed level.

Direct Method or Parrameter Variiation Methood

In this meethod, the baseband or mo odulating signnal directly modulates

m thee carrier. Thee carrier signaal is
generatedd with the helpp of an oscilllator circuit. TThis oscillatoor circuit usess a parallel tuuned L-C circcuit.
Thus the frequency
f of oscillation off the carrier geeneration is governed
g by thhe expression n:

1
ωc =
LC

The carrieer frequency isi made to vaary in accordaance with the baseband or modulating signal
s by makking
either L or
o C depend uponu to the baaseband signaal. Such an oscillator whose frequency isi controlled byb a
modulatinng signal volttage is calledd as Voltage C Controlled Oscillator. Thee frequency of
o VCO is varried
accordingg to the modullating signal simply by puutting shunt vooltage variablle capacitor (vvaractor/variccap)
with its tuuned circuit. The
T varactor diode is a sem miconductor diode whose junction capaacitance channges
with dc bias voltage. TheT capacitorr C is made m much smaller than the varaactor diode capacitance Cd so
that the RF
R voltage frrom oscillatorr across the ddiode is smalll as compareed to reversee bias dc volttage
across thee varactor diode.

Figure 12 Varaactor diode method of FM gene
eration(Direct M
Method)
 k 1
= k ( vD ) 2
−
Cd =
vD
vD = Vo + x ( t )
1
ωi =
Lo ( Co + Cd )
1
⇒ ωi =
⎛ − ⎞
1
Lo ⎜ Co + kvD 2 ⎟
⎝ ⎠

Drawbacks of direct method of FM generation:

1. Generation of carrier signal is directly affected by the modulating signal by directly controlling
the tank circuit and thus a stable oscillator circuit cannot be used. So a high order stability in
carrier frequency cannot be achieved.
2. The non-linearity of the varactor diode produces a frequency variation due to harmonics of the
modulating signal and therefore the FM signal is distorted.

Indirect method or Armstrong method of FM generation

A very high frequency stability can be achieved since in this case the crystal oscillator may be used as a
carrier frequency generator. In this method, first of all a narrowband FMis generated and then frequency
multiplication is used to cause required increased frequency deviation.The narrow band FM
wave is then passed through a frequency multiplier to obtain the wide band FM wave. Frequency
multiplication scales up the carrier frequency as well as the frequency deviation. The crystal controlled
oscillator provides good frequency stability. But this scheme does not provide both the desired
frequency deviation and carrier frequency at the same time. This problem can be solved by using
multiple stages of frequency multipliers and a mixer stages.

Figure 13 Narrow Band FM Generation 
FM Demodulators

In order to be able to demodulate FM, a receiver must produce a signal whose amplitude varies as
according to the frequency variations of the incoming signals and it should be insensitive to any
amplitude variations in FM signal. Insensitivity to amplitude variations is achieved by having a high
gain IF amplifier. Here the signals are amplified to such a degree that the amplifier runs into limiting. In
this way any amplitude variations are removed. Generally a FM demodulator is composed of two parts:
Discriminator and Envelope Detector.Discriminator is a frequency selective network which converts
the frequency variations in an input signal in to proportional amplitude variations. Hence when it is
input with an FM signal, it can produce an amplitude modulated signal. But it does not generally alter
the frequency variations which were there in the input signal. So the output of a discriminator is a both
frequency and amplitude modulated signal. This signal can be fed to the Envelope Detectorpart of FM
demodulator to get back the baseband signal

Figure 14  Slope detector 

Figure 15 Frequency response of slope detector 

Slope detector: A very simplest form of FM demodulation is known as slope detection or

demodulation. It consists of a tuned circuit that is tuned to a frequency slightly offset from the carrier of
 the signal.As the frequency of the signals varies up and down in frequency according to its modulation,
so the signal moves up and down the slope of the tuned circuit. This causes the amplitude of the signal
to vary in line with the frequency variations. In fact, at this point the signal has both frequency and
amplitude variations.It can be seen from the diagram that changes in the slope of the filter, reflect into
the linearity of the demodulation process.The linearity is very dependent not only on the filter slope as it
falls away, but also the tuning of the receiver - it is necessary to tune the receiver frequency to a
point where the filter characteristic is relatively linear. The final stage in the process is to demodulate
the amplitude modulation and this can be achieved using a simple diode circuit. One of the most
obvious disadvantages of this simple approach is the fact that both amplitude and frequency variations
in the incoming signal appear at the output. However, the amplitude variations can be removed by
placing a limiter before the detector. The input signal is a frequency modulated signal. It is applied to
the tuned transformer (T1, C1, C2 combination) which is offset from the centre carrier frequency. This
converts the incoming signal from just FM to one that has amplitude modulation superimposed upon the
signal. This amplitude signal is applied to a simple diode detector circuit, D1. Here the diode provides
the rectification, while C3 removes any unwanted high frequency components, and R1 provides a load.

PLL FM demodulator / detector:When used as an FM demodulator, the basic phase locked loop can
be used without any changes. With no modulation applied and the carrier in the centre position of the
pass-band the voltage on the tune line to the VCO is set to the mid position. However, if the carrier
deviates in frequency, the loop will try to keep the loop in lock. For this to happen the VCO frequency
must follow the incoming signal, and in turn for this to occur the tune line voltage must vary.
Monitoring the tune line shows that the variations in voltage correspond to the modulation applied to
the signal. By amplifying the variations in voltage on the tune line it is possible to generate the
demodulated signal.The PLL FM demodulator is one of the more widely used forms of FM
demodulator or detector these days. Its suitability for being combined into an integrated circuit, and the
small number of external components makes PLL FM demodulation ICs an ideal candidate for many
circuits these days.

Figure 16 PLL FM Demodulator 

Superheterodyne principle 

In early days TRF receivers were used to detect the baseband signal from modulated RF signal.
The performance of such receiver varies as the incoming RF frequency varies. This is because
it uses single conversion technique. Later double conversion technique (frequency of incoming
RF signal changes two times) is used by some receiver as shown in figure 3.8. These are
known as superheterodyne receiver. The main idea behind the design of such receiver is that:
whatever may be the frequency of the incoming RF signal, the output after first conversion
always produces a fixed frequency known as intermediate frequency. Due to this the
performance of receiver remains same for all type of incoming RF signal.
Pulse modulaton
Digital Transmission is the transmittal of digital signals between two or more points in a
communications system. The signals can be binary or any other form of discrete-level digital pulses.
Digital pulses can not be propagated through a wireless transmission system such as earth’s
atmosphere or free space.
Alex H. Reeves developed the first digital transmission system in 1937 at the Paris
Laboratories of AT & T for the purpose of carrying digitally encoded analog signals, such as the
human voice, over metallic wire cables between telephone offices.

Advantages & disadvantages of Digital Transmission

Advantages
--Noise immunity
--Multiplexing(Time domain)
--Regeneration
--Simple to evaluate and measure
Disadvantages
--Requires more bandwidth
--Additional encoding (A/D) and decoding (D/A) circuitry

Pulse Modulation
-- Pulse modulation consists essentially of sampling analog information signals and then
converting those samples into discrete pulses and transporting the pulses from a source to a destination
over a physical transmission medium.
--The four predominant methods of pulse modulation:
1) pulse width modulation (PWM)
2) pulse position modulation (PPM)
3) pulse amplitude modulation (PAM)
4) pulse code modulation (PCM).
Pulse Width Modulation
--PWM is sometimes called pulse duration modulation (PDM) or pulse length modulation (PLM), as
the width (active portion of the duty cycle) of a constant amplitude pulse is varied proportional to the
amplitude of the analog signal at the time the signal is sampled.
--The maximum analog signal amplitude produces the widest pulse, and the minimum analog signal
amplitude produces the narrowest pulse. Note, however, that all pulses have the same amplitude.
Pulse Position Modulation
--With PPM, the position of a constant-width pulse within a prescribed time slot is varied according to
the amplitude of the sample of the analog signal.
--The higher the amplitude of the sample, the farther to the right the pulse is positioned within the
prescribed time slot. The highest amplitude sample produces a pulse to the far right, and the lowest
amplitude sample produces a pulse to the far left.
Pulse Amplitude Modulation
--With PAM, the amplitude of a constant width, constant-position pulse is varied according to the
amplitude of the sample of the analog signal.
--The amplitude of a pulse coincides with the amplitude of the analog signal.
--PAM waveforms resemble the original analog signal more than the waveforms for PWM or PPM.
Pulse Code Modulation
--With PCM, the analog signal is sampled and then converted to a serial n-bit binary code for
transmission.
--Each code has the same number of bits and requires the same length of time for transmission
Pulse Modulation
--PAM is used as an intermediate form of modulation with PSK, QAM, and PCM, although it is
seldom used by itself.
--PWM and PPM are used in special-purpose communications systems mainly for the military but are
seldom used for commercial digital transmission systems.
--PCM is by far the most prevalent form of pulse modulation and will be discussed in more detail.
Pulse Code Modulation
--PCM is the preferred method of communications within the public switched telephone network
because with PCM it is easy to combine digitized voice and digital data into a single, high-speed
digital signal and propagate it over either metallic or optical fiber cables.
--With PCM, the pulses are of fixed length and fixed amplitude.
--PCM is a binary system where a pulse or lack of a pulse within a prescribed time slot represents
either a logic 1 or a logic 0 condition.
--PWM, PPM, and PAM are digital but seldom binary, as a pulse does not represent a single binary
digit (bit).

PCM system Block Diagram

--The band pass filter limits the frequency of the analog input signal to the standard voice-band
frequency range of 300 Hz to 3000 Hz.
--The sample- and- hold circuit periodically samples the analog input signal and converts those
samples to a multilevel PAM signal.
--The analog-to-digital converter (ADC) converts the PAM samples to parallel PCM codes, which are
converted to serial binary data in the parallel-to-serial converter and then outputted onto the
transmission linear serial digital pulses.
--The transmission line repeaters are placed at prescribed distances to regenerate the digital pulses.
--In the receiver, the serial-to-parallel converter converts serial pulses received from the transmission
line to parallel PCM codes.
--The digital-to-analog converter (DAC) converts the parallel PCM codes to multilevel PAM signals.
--The hold circuit is basically a low pass filter that converts the PAM signals back to its original
analog form.

The block diagram of a single-channel, simplex (one-way only) PCM system.

PCM Sampling:
--The function of a sampling circuit in a PCM transmitter is to periodically sample the continually
changing analog input voltage and convert those samples to a series of constant- amplitude pulses that
can more easily be converted to binary PCM code.
--A sample-and-hold circuit is a nonlinear device (mixer) with two inputs: the sampling pulse and the
analog input signal.
--For the ADC to accurately convert a voltage to a binary code, the voltage must be relatively constant
so that the ADC can complete the conversion before the voltage level changes. If not, the ADC would
be continually attempting to follow the changes and may never stabilize on any PCM code.
--Essentially, there are two basic techniques used to perform the sampling function
1) natural sampling
2) flat-top sampling

--Natural sampling is when tops of the sample pulses retain their natural shape during the sample
interval, making it difficult for an ADC to convert the sample to a PCM code.
--The most common method used for sampling voice signals in PCM systems is flat- top sampling,
which is accomplished in a sample-and-hold circuit.
-- The purpose of a sample-and-hold circuit is to periodically sample the continually changing analog
input voltage and convert those samples to a series of constant-amplitude PAM voltage levels.
Sampling Rate
--The Nyquist sampling theorem establishes the minimum Nyquist sampling rate (fs) that can be used
for a given PCM system.
--For a sample to be reproduced accurately in a PCM receiver, each cycle of the analog input signal
(fa) must be sampled at least twice.
--Consequently, the minimum sampling rate is equal to twice the highest audio input frequency.
--Mathematically, the minimum Nyquist sampling rate is:
fs ≥ 2fa
--If fs is less than two times fa an impairment called alias or foldover distortion occurs.

Quantization and the Folded Binary Code:

Quantization
--Quantization is the process of converting an infinite number of possibilities to a finite number of
conditions.
--Analog signals contain an infinite number of amplitude possibilities.
--Converting an analog signal to a PCM code with a limited number of combinations requires
quantization.
Folded Binary Code
--With quantization, the total voltage range is subdivided into a smaller number of sub-ranges.
--The PCM code shown in Table 10-2 is a three-bit sign- magnitude code with eight possible
combinations (four positive and four negative).
--The leftmost bit is the sign bit (1 = + and 0 = -), and the two rightmost bits represent magnitude.
-- This type of code is called a folded binary code because the codes on the bottom half of the table
are a mirror image of the codes on the top half, except for the sign bit.
Quantization
--With a folded binary code, each voltage level has one code assigned to it except zero volts, which
has two codes, 100 (+0) and 000 (-0).
--The magnitude difference between adjacent steps is called the quantization interval or quantum.
--For the code shown in Table 10-2, the quantization interval is 1 V.
--If the magnitude of the sample exceeds the highest quantization interval, overload distortion (also
called peak limiting) occurs.
--Assigning PCM codes to absolute magnitudes is called quantizing.
--The magnitude of a quantum is also called the resolution.
--The resolution is equal to the voltage of the minimum step size, which is equal to the voltage of the
least significant bit (Vlsb) of the PCM code.
--The smaller the magnitude of a quantum, the better (smaller) the resolution and the more accurately
the quantized signal will resemble the original analog sample.

--For a sample, the voltage at t3 is approximately +2.6 V. The folded PCM code is
sample voltage = 2.6 = 2.6
resolution 1
--There is no PCM code for +2.6; therefore, the magnitude of the sample is rounded off to the nearest
valid code, which is 111, or +3 V.
--The rounding-off process results in a quantization error of 0.4 V.
--The likelihood of a sample voltage being equal to one of the eight quantization levels is remote.
--Therefore, as shown in the figure, each sample voltage is rounded off (quantized) to the closest
available level and then converted to its corresponding PCM code.
--The rounded off error is called the called the quantization error (Qe).
--To determine the PCM code for a particular sample voltage, simply divide the voltage by the
resolution, convert the quotient to an n-bit binary code, and then add the sign bit.

Linear input-versus-output transfer curve

resolution
Qe =
2

Linear transfer function

Quantization Quantization error (Qe)

1) For the PCM coding scheme shown in Figure 10-8, determine the quantized voltage, quantization
error (Qe) and PCM code for the analog sample voltage of + 1.07 V.
A) To determine the quantized level, simply divide the sample voltage by resolution and then round
the answer off to the nearest quantization level:

+1.07V = 1.07 = 1
1V
The quantization error is the difference between the original sample voltage and the quantized level, or
Qe = 1.07 -1 = 0.07
From Table 10-2, the PCM code for + 1 is 101.

Dynamic Range (DR): It determines the number of PCM bits transmitted per sample.
-- Dynamic range is the ratio of the largest possible magnitude to the smallest possible magnitude
(other than zero) that can be decoded by the digital-to-analog converter in the receiver. Mathematically,

=
DR
Vmax
=
Vmax
= 2n − 1 = (
DR( dB ) 20 log 2n − 1 )
Vmin resolution
 =20 log Vmax
Vmin
where DR = dynamic range (unitless)
Vmin = the quantum value
Vmax = the maximum voltage magnitude of the DACs
n = number of bits in a PCM code (excluding the sign bit)
For n > 4
DR = 2n − 1 ≈ 2n
(
DR( dB ) ≈ 20 log 2=
n
)
− 1 20n log 2 ≈ 6n

Signal-to-Quantization Noise Efficiency

--For linear codes, the magnitude change between any two successive codes is the same.
--Also, the magnitude of their quantization error is also same.
The maximum quantization noise is half the resolution. Therefore, the worst possible signal voltage-
to-quantization noise voltage ratio (SQR) occurs when the input signal is at its minimum amplitude
(101 or 001). Mathematically, the worst-case voltage SQR is
resolution
SQR = resolution = Vlsb =2 Qe =
Qe V lsb /2 2

For input signal minimum amplitude

SQR = minimum voltage / quantization noise

Vmin resolution
SQR(= =
min )
= 2
Qe Qe
For input signal maximum amplitude
SQR = maximum voltage / quantization noise

Vmax
SQR( max ) =
Qe
SQR is not constant where R = resistance (ohms)
SQR (dB) = 10 log v2 /R v = rms signal voltage (volts)
(q 2 /12)/R q = quantization interval (volts)
v2 /R = average signal power (watts)
(q 2 /12)/R = average quantization noise power (watts)
Linear vs. Nonlinear PCM codes

Linear Nonlinear
Companding
--Companding is the process of compressing and then expanding
--High amplitude analog signals are compressed prior to txn. and then expanded in the receiver
--Higher amplitude analog signals are compressed and Dynamic range is improved
--Early PCM systems used analog companding, where as modern systems use digital companding.

Basic companding process

Analog companding

PCM system with analog companding

--In the transmitter, the dynamic range of the analog signal is compressed, and then converted o a
linear PCM code.
--In the receiver, the PCM code is converted to a PAM signal, filtered, and then expanded back to its
original dynamic range.

-- There are two methods of analog companding currently being used that closely approximate a
logarithmic function and are often called log-PCM codes.
The two methods are 1) µ-law and
2) A-law

µ-law companding
 Vmax ln 1 + µ in 
V
 Vmax 
Vout =
ln (1 + µ )

where Vmax = maximum uncompressed analog input amplitude(volts)

Vin = amplitude of the input signal at a particular instant of time (volts)
µ = parameter used tio define the amount of compression (unitless)
Vout = compressed output amplitude (volts)

A-law companding
--A-law is superior to µ-law in terms of small-signal quality
--The compression characteristic is given by
 A| x | 1 where y = Vout
 1 + log A , 0 ≤| x |≤ A x = Vin / Vmax
y =
1 + log( A | x |) 1
 , ≤| x |≤ 1
 1 + log A A
Digital Companding: Block diagram refer in text book.
--With digital companding, the analog signal is first sampled and converted to a linear PCM code, and
then the linear code is digitally compressed.
-- In the receiver, the compressed PCM code is expanded and then decoded back to analog.
-- The most recent digitally compressed PCM systems use a 12- bit linear PCM code and an 8-bit
compressed PCM code.

Digital compression error

--To calculate the percentage error introduced by digital compression
%error=12-bit encoded voltage - 12-bit decoded voltage X 100
12-bit decoded voltage
PCM Line speed
--Line speed is the data rate at which serial PCM bits are clocked out of the PCM encoder onto the
transmission line. Mathematicaly,
Line speed= samples X bits
second sample
Delta Modulation
--Delta modulation uses a single-bit PCM code to achieve digital transmission of analog signals.
--With conventional PCM, each code is a binary representation of both the sign and the magnitude of a
particular sample. Therefore, multiple-bit codes are required to represent the many values that the
sample can be.
--With delta modulation, rather than transmit a coded representation of the sample, only a single bit is
transmitted, which simply indicates whether that sample is larger or smaller than the previous sample.
--The algorithm for a delta modulation system is quite simple.
--If the current sample is smaller than the previous sample, a logic 0 is transmitted.
--If the current sample is larger than the previous sample, a logic 1 is transmitted.

Differential DM
--With Differential Pulse Code Modulation (DPCM), the difference in the amplitude of two successive
samples is transmitted rather than the actual sample. Because the range of sample differences is
typically less than the range of individual samples, fewer bits are required for DPCM than
conventional PCM.
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Analog Transmission
of Digital Data:
ASK, FSK, PSK, QAM
Required reading:
Garcia 3.7

CSE 3213, Fall 2010

Instructor: N. Vlajic
 2
Why Do We Need Digital-to-Analog Conversion?!

1) The medium/channel is band pass, and/or

2) Multiple users need to share the medium.
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Modulation of Digital Data

Modulation – process of converting digital data or a low-pass

analog to band-pass (higher-frequency) analog signal

Digital-to-analog modulation.

Analog-to-analog modulation.

Carrier Signal – aka carrier freq. or modulated signal - high freq.

signal that acts as a basis for the information signal
• information signal is called modulating signal

bandpass channel

freq
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Modulation of Digital Data (cont.)

Digital-to-Analog – process of changing one of the characteristic

Modulation of an analog signal (typically a sinewave) based
on the information in a digital signal
• sinewave is defined by 3 characteristics (amplitude,
frequency, and phase) ⇒ digital data (binary 0 & 1)
can be represented by varying any of the three
• application: transmission of digital data over
telephone wire (modem)

Types of Digital-to-Analog
Modulation
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Modulation of Digital Data: ASK
ASK – strength of carrier signal is varied to represent binary 1 or 0
• both frequency & phase remain constant while amplitude changes
• commonly, one of the amplitudes is zero

⎧A 0cos(2πfc t), binary 0 ⎧0, binary 0

s(t) = ⎨ = ⎨
⎩A 1 cos(2πfc t), binary 1 ⎩Acos(2πfc t), binary 1

+A
Is this picture,
from the textbook,
entirely correct?! -A

• demodulation: only the presence or absence of a sinusoid in a

given time interval needs to be determined
• advantage: simplicity
• disadvantage: ASK is very susceptible to noise interference –
noise usually (only) affects the amplitude, therefore ASK is the
modulation technique most affected by noise
• application: ASK is used to transmit digital data over optical fiber
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Modulation of Digital Data: ASK (cont.)

Example [ ASK ]

vd(t)

vc(t)

vASK(t)

vd(f) vc(f) f

How does the frequency spectrum of vASK(t) look like!?

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Modulation of Digital Data: ASK (cont.)
ASK-Modulated Signal: Frequency Spectrum
1
cosA ⋅ cosB = (cos(A - B) + cos(A + B))
2

Carrier signal: v c (t) = cos(2πfc t) = cos(ωc t) , where 2πfc=ωc

⎡1 2 2 2 ⎤
Digital signal: v d (t) = A ⋅ ⎢ + cosω0 t − cos3ω0 t + cos5ω0 t − ...⎥
(unipolar!!!) ⎣2 π 3π 5π ⎦

Modulated signal: v ASK (t) = v c (t) ⋅ v d (t) =

⎡1 2 2 2 ⎤
= cosωc t ⋅ ⎢ + cosω0 t − cos3ω0 t + cos5ω0 t − ...⎥ =
⎣2 π 3π 5π ⎦
1 2 2
= cosωc t + cosωc t ⋅ cosω0 t - cosωc t ⋅ cos3ω0 t + ... =
2 π 3π
1 1
= cosωc t + [cos(ωc − ω0 )t + cos(ωc + ω0 )t ] -
2 π
1
− [cos(ωc − 3ω0 )t + cos(ωc + 3ω0 )t ] + ...
3π

ωd_max ωc ω ωc-ωd_max ωc ωc+ωd_max ω

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Modulation of Digital Data: FSK

FSK – frequency of carrier signal is varied to represent binary 1 or 0

• peak amplitude & phase remain constant during each bit interval

⎧Acos(2 πf1t), binary 0

s(t) = ⎨
⎩Acos(2 πf2 t), binary 1

f1<f2
+A

-A

• demodulation: demodulator must be able to determine which of

two possible frequencies is present at a given time
• advantage: FSK is less susceptible to errors than ASK – receiver
looks for specific frequency changes over a number
of intervals, so voltage (noise) spikes can be ignored
• disadvantage: FSK spectrum is 2 x ASK spectrum
• application: over voice lines, in high-freq. radio transmission, etc.
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Modulation of Digital Data: FSK (cont.)

Example [ FSK ]

vd(t)

vc1(t)

vc2(t)

vFSK(t)

ωd_max ω1 ω2 ω ω1-ωd_maxω1 ω2 ω +ω
2 d_max
ω
 10
Modulation of Digital Data: FSK (cont.)
FSK-Modulated Signal: Frequency Spectrum
Digital signal: v d (t) - modulated with ω1 , and
v d ' (t) = 1- v d (t) - modulated with ω2

Modulated signal: v FSK (t) = cosω1t ⋅ v d (t) + cosω 2 t ⋅ (1 − v d (t)) =

⎡1 2 2 2 ⎤
= cosω1t ⋅ ⎢ + cosω0 t − cos3ω0 t + cos5ω0 t − ...⎥ +
⎣2 π 3π 5π ⎦
⎡1 2 2 2 ⎤
+ cosω2 t ⋅ ⎢ − cosω0 t + cos3ω0 t − cos5ω0 t − ...⎥ =
⎣2 π 3π 5π ⎦
= ...
1 1
= cosω1t + [cos(ω1 − ω0 )t + cos(ω1 + ω0 )t ] -
2 π
1
− [cos(ω1 − 3ω0 )t + cos(ω1 + 3ω0 )t ] + ... +
3π
1 1
cosω 2 t − [cos(ω2 − ω0 )t + cos(ω2 + ω0 )t ] -
2 π
1
+ [cos(ω2 − 3ω0 )t + cos(ω2 + 3ω0 )t ] + ... +
3π
 11
Modulation of Digital Data: PSK
PSK – phase of carrier signal is varied to represent binary 1 or 0
• peak amplitude & freq. remain constant during each bit interval
• example: binary 1 = 0º phase, binary 0 = 180º (πrad) phase
⇒ PSK is equivalent to multiplying carrier signal by +1 when the
information is 1, and by -1 when the information is 0

2-PSK, or ⎧Acos(2 πfc t), binary 1

s(t) = ⎨ +A
Binary PSK, ⎩Acos(2 πfc t + π ), binary 0
since only 2
different phases
⎧Acos(2 πfc t), binary 1 -A
are used. s(t) = ⎨
⎩- Acos(2 πfc t), binary 0

• demodulation: demodulator must determine the phase of received

sinusoid with respect to some reference phase
• advantage: ƒ PSK is less susceptible to errors than ASK, while it
requires/occupies the same bandwidth as ASK
ƒ more efficient use of bandwidth (higher data-rate)
are possible, compared to FSK !!!
• disadvantage: more complex signal detection / recovery process,
than in ASK and FSK
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Modulation of Digital Data: PSK (cont.)

Example [ PSK ]

vd(t)

vc(t)

vPSK(t)

ωd_max ωc ω ωc-ωd_max ωc ωc+ωd_max ω

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Modulation of Digital Data: PSK (cont.)

PSK Detection – multiply the received / modulated signal

1
± Acos(2 πfc t) by 2*cos(2πfct)
cos 2 A = (1+ cos 2A )
2 • resulting signal
2Acos 2 (2πfc t) = A[1+ cos(4 πfc t)] , binary 1

- 2Acos 2 (2πfc t) = − A[1+ cos(4 πfc t)] , binary 0

• by removing the oscillatory part with a

low-pass filter, the original baseband signal
(i.e. the original binary sequence) can be
easily determined
 14
Modulation of Digital Data: PSK (cont.)
Information 1 0 1 1 0 1

+A
Baseband
Signal
0 T 2T 3T 4T 5T 6T
-A
sender
+A
Modulated
Signal T 2T 4T 5T
0 3T 6T
x(t)
-A

A cos(2πft) -A cos(2πft)

A {1 + cos(4πft)} -A {1 + cos(4πft)} Signal shifted above / below zero level.

After multiplication
+A
at receiver
0 T 2T 3T 4T 5T 6T
receiver x(t) cos(2πfct)
-A

+A
Baseband
signal discernable
after smoothing 0 T 2T 3T 4T 5T 6T
-A
 15
Modulation of Digital Data: PSK (cont.)
Facts from Modulation Theory

If Wc/2 Data rate = 2*Wc/2 = B [bps]

Baseband signal x(t) with
bandwidth Wc/2
f
then B
Wc
Modulated signal
x(t)cos(2πfct) has
bandwidth Wc Hz f
fc-B fc fc+B

• If bandpass channel has bandwidth Wc [Hz],

ƒ then baseband channel has Wc/2 [Hz] available, so
ƒ modulation system supports 2*(Wc/2) = Wc [pulses/second]

ƒ recall Nyqyist Law: baseband transmission system of bandwidth Wc [Hz]

can theoretically support 2 Wc pulses/sec

How can we recover the factor 2 in supported data-rate !?

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Modulation of Digital Data: PSK (cont.)
QPSK = 4-PSK – PSK that uses phase shifts of 90º=π/2 rad ⇒ 4
different signals generated, each representing 2 bits
⎧Acos(2πfc t), binary 00
⎪
⎪Acos(2πfc t + π ), binary 01
⎪ 2
s(t) = ⎨
⎪Acos(2πfc t + π ), binary 10
⎪ 3π
⎪Acos(2πfc t + ), binary 11
⎩ 2

• advantage: higher data rate than in PSK (2 bits per bit

interval), while bandwidth occupancy remains the same
• 4-PSK can easily be extended to 8-PSK, i.e. n-PSK
• however, higher rate PSK schemes are limited by the
ability of equipment to distinguish small differences in
phase
 17
Modulation of Digital Data: QAM
Quadrature – uses “two-dimensional” signalling
Amplitude • original information stream is split into two sequences that
Modulation consist of odd and even symbols, e.g. Bk and Ak
(QAM) 1 0 1 1 0 1 …
1 -1 1 1 -1 1 …
B1 A1 B2 A2 B3 A3 …
• Ak sequence (in-phase comp.) is modulated by cos(2 πfc t)
Bk sequence (quadrature-phase comp.) is modulated by
sin(2πfc t)
• composite signal A cos(2 πf t) + B sin(2 πf t) is sent
k c k c
through the channel

Ak x Yi(t) = Ak cos(2πfct)

cos(2πfct) + Y(t) = Ak cos(2πfct) + Bk sin(2πfct)

Transmitted
Bk x Yq(t) = Bk sin(2πfct) Signal

sin(2πfct)

• advantage: data rate = 2 bits per bit-interval!

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Modulation of Digital Data: QAM (cont.)

Example [ QAM ]

vd(t)
Bk

sin(ωct)

Ak

cos(ωct)
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Modulation of Digital Data: QAM
QAM Demodulation • by multiplying Y(t) by 2 ⋅ cos(2 πfc t) and then low-
pass filtering the resultant signal, sequence Ak is
obtained
• by multiplying Y(t) by 2 ⋅ sin(2 πfc t) and then low-
pass filtering the resultant signal, sequence Bk is
obtained

Lowpass
A k cos(2 πfc t) + Bk sin(2 πfc t) = Y(t) x filter Ak
(smoother)
2cos(2πfct)
2Akcos2(2πfct)+2Bk cos(2πfct)sin(2πfct)
= Ak {1 + cos(4πfct)}+Bk {0 + sin(4πfct)}
smoothed to zero
Lowpass
x filter Bk
1 (smoother)
cos 2 (A) = (1+ cos(2A))
2 2sin(2πfct)
1 2Bk sin2(2πfct)+2Ak cos(2πfct)sin(2πfct)
sin2 (A) = (1− cos(2A)) = Bk {1 - cos(4πfct)}+Ak {0 + sin(4πfct)}
2
sin(2A) = 2sin(A)cos(A) smoothed to zero
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Signal Constellation
Constellation Diagram – used to represents possible symbols that may
be selected by a given modulation scheme as
points in 2-D plane
• X-axis is related to in-phase carrier: cos(ωct)
ƒ the projection of the point on the X-axis defines
the peak amplitude of the in-phase component
• Y-axis is related to quadrature carrier: sin(ωct)
ƒ the projection of the point on the Y-axis defines
the peak amplitude of the quadrature component
• the length of line that connects the point to
the origin is the peak amplitude of the signal
element (combination of X & Y components)
• the angle the line makes with the X-axis is the
phase of the signal element
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Modulation of Digital Data: QAM
QAM cont. – QAM can also be seen as a combination of ASK & PSK
( ) cos(2πf t + tan
1
2 2 2 Bk
Y(t) = A k cos(2πfc t) + Bk sin(2πfc t) = A k + Bk c
-1
)
Ak

Bk
(-A,A) (A, A)

4-level QAM Ak

(-A,-A) (A,-A)
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Modulation of Digital Data: QAM
16-level QAM – the number of bits transmitted per T [sec] interval
can be further increased by increasing the number
of levels used
• in case of 16-level QAM, Ak and Bk individually can
assume 4 different levels: -1, -1/3, 1/3, 1
• data rate: 4 bits/pulse ⇒ 4W bits/second

( ) cos(2πf t + tan
1
2 2 2 Bk
Y(t) = A k cos(2πfc t) + Bk sin(2πfc t) = A k + Bk c
-1
)
Ak
Bk

Ak and Bk individually
can take on 4 different values;
Ak the resultant signal can take
on (only) 3 different values!!!

In QAM various combinations of amplitude and phase are employed

to achieve higher digital data rates.

Amplitude changes are susceptible to noise ⇒ the number of phase shifts used
by a QAM system is always greater than the number of amplitude shifts.