Unit – I

Objectives:
➢ To introduce the concepts of modulation and AM modulation
technique

Syllabus:
Introduction, Need for modulation, Amplitude Modulation- Definition, Time domain
and frequency domain description, power relations in AM waves, Generation and
detection of AM Waves.

Outcomes:
Students will be able to
➢ Understand need of modulation

➢ analyze AM system
➢ Determine power for amplitude modulation scheme.
➢ Understand generation and detection of AM modulated waves
Introduction
Communication is a process of conveying message at a distance. If the distance
is involved is beyond the direct communication, the communication engineering
comes into the picture. The branch of engineering which deals with communication
systems is known as telecommunication engineering.
Telecommunication engineering is classified into two types based on
Transmission media. They are:
Line communication
Radio communication
In Line communication the media of transmission is a pair of conductors called
transmission line. In this technique signals are directly transmitted through the
transmission lines. The installation and maintenance of a transmission line is not
only costly and complex, but also overcrowds the open space.
In radio communication transmission media is open space or free space. In this
technique signals are transmitted by using antenna through the free space in the form
of EM waves.

Message Transmitter Channel Receiver Destination

` source

Fig.: Block diagram of Communication system

The communication system consists of three basic components.
• Transmitter
• Channel
• Receiver
Transmitter is the equipment which converts physical message, such as sound,
words, pictures etc., into corresponding electrical signal.
Receiver is equipment which converts electrical signal back to the physical
message.
Channel may be either transmission line or free space, which provides
transmission path between transmitter and receiver.
Modulation
Modulation is defined as the process by which some characteristics (i.e.
amplitude, frequency, and phase) of a carrier are varied in accordance with a
modulating wave.
Demodulation
Demodulation is the reverse process of modulation, which is used to get back
the original message signal. Modulation is performed at the transmitting end
whereas demodulation is performed at the receiving end.
In analog modulation sinusoidal signal is used as carrier where as in digital
modulation pulse train is used as carrier.
Need for modulation:
The reason why low frequency signals cannot be transmitted over long distances
through space is listed below:
1. Short Operating Range – When a wave has a large frequency, the
energy associated with it will also be large. Thus low frequency signals have
less power that does not enable them to travel over long distances.
2. Poor Radiation Efficiency – The radiation efficiency becomes very
poor for low frequency signals.
3. Mutual Interference – If all audio frequencies are send continuously
from different sources, they would all get mixed up and cause erroneous
interference air. If modulation is done, each signal will occupy different
frequency levels and can be transmitted simultaneously without any error.
4. Huge Antenna Requirement – For a effective signal transmission, the
sending and receiving antenna should be at least 1/4th of the wave length of
the signal. Thus, for small frequencies, the antenna will have kilometres of
length. But if the signal has the range of MegaHertz frequency, then the
antenna size would be less. The carrier wave cannot be used alone for
transmission purposes. Since its amplitude, frequency, and phase angle are
constant with respect to some preference.
TYPES OF MODULATION
Continuous wave modulation (CW): When the carrier wave is continuous in
nature the modulation process is known as continuous wave modulation. Pulse
modulation: When the carrier wave is a pulse in nature the modulation process is
known as continuous wave modulation Amplitude modulation (AM): A modulation
process in which the amplitude of the carrier is varied in accordance with the
instantaneous value of the modulating signal.

Amplitude Modulation- Time Domain and Frequency Representation

Amplitude modulation is defined as the process in which the amplitude of the
carrier signal is varied in accordance with the modulating signal or message signal.
Consider a sinusoidal carrier signal C (t) is defined as
C (t) = Ac Cos (2πfct +φ) t
Where Ac = Amplitude of the carrier signal
f c= frequency of the
carrier signal
φ= Phase angle.
For our convenience, assume the phase angle of the carrier signal is zero. An
amplitude-modulated (AM) wave S(t) can be described as function of time is given
by
S (t) = Ac [1+kam (t)] cos 2πfct

Where ka = Amplitude sensitivity of the modulator.

The amplitude modulated (AM) signal consists of both modulated carrier signal
and un modulated carrier signal.

There are two

requirements to maintain the envelope of AM signal is same as the shape of base
band signal.

➢ The amplitude of the kam(t) is always less than unity i.e., |kam(t)|<1
for all ‘t’
➢ The carrier signal frequency fc is far greater than the highest frequency
component W of the message signal m (t) i.e., fc>>W
Assume the message signal m (t) is band limited to the interval –W ≤ f ≤W M (f)

Fig. : Spectrum of message signal

The Fourier transform of AM signal S (t) is

S (f) = Ac/2 [δ(f-fc)+ δ(f+fc)]+kaAc/2[M(f-fc)+ M(f+fc)]

Fig : Spectrum of AM signal

The AM spectrum consists of two impulse functions which are located at f c and
-fc and weighted by Ac/2, two USBs, band of frequencies from fc to fc +W and band
of frequencies from -fc-W to –fc, and two LSBs, band of frequencies from fc-W to fc
and -fc to -fc+W.
 The difference between highest frequency component and lowest frequency
component is known as transmission bandwidth. i.e., BT = 2W
The envelope of AM signal is Ac [1+kam (t)].

SINGLE TONE MODULATION

In single-tone modulation modulating signal consists of only one frequency
component where as in multi-tone modulation modulating signal consists of more
than one frequency component.
S (t) = Ac [1+kam(t)] cos 2πfct ………..(i)
Let m (t) = Am cos 2πfmt
Substitute m (t) in equation (i)
S (t) = Ac [1+ka Am cos 2πfmt] cos 2πfct
Replace the term ka Am by m which is known as modulation index or modulation
factor.
Modulation index is defined as the ratio of amplitude of message signal to the
amplitude of carrier signal. i.e.,
µ= Am/Ac
S (t) = Ac cos (2πfct)+Acπ/2[cos2π(fc+fm)t]+ Acπ/2[cos2π(fc-fm)t]
Fourier transform of S (t) is
S(f) =Ac/2[π(f-fc) + π(f+fc)] +Acπ/4[π(f-fc-fm) + π(f+fc+fm)]
+ Acπ/4[π(f- fc+fm ) + π(f+fc-fm)]

Fig. :Spectrum of Single tone AM signal

POWER RELATIONS IN AM WAVES

 The standard time domain equation for single-tone AM signal is given by
S (t) = Accos (2πfct) +Acm/2[cos2 π(fc+fm) t] + Ac/2[cos2 π(fc-fm) t]
Power of any signal is equal to the mean square value of the signal
Carrier power Pc = Ac2/2
Upper Side Band power PUSB = Ac2m2/8
Lower Side Band power P LSB = Ac2m2/8
Total power PT = Pc + PLSB + PUSB
Total power PT = Ac2/2 + Ac2m2/8 + Ac2m2/8
PT = Pc [1+m2/2]

Transmission efficiency (η):-

Transmission efficiency is defined as the ratio of total side band power to
the total transmitted power.
i.e., PT =PSB/PT

Advantages of Amplitude modulation:-

• Generation and detection of AM signals are very easy
• It is very cheap to build, due to this reason it I most commonly used in
AM radio broad casting
Disadvantages of Amplitude of modulation:-
• Amplitude modulation is wasteful of power
• Amplitude modulation is wasteful of band width
Application of Amplitude modulation: -
AM Radio Broadcasting
Generation of AM waves: Square-Law Modulator and Switching
Modulator
There are two methods to generate AM waves
• Square-law modulator

Square-law modulator: -

.
 A Square-law modulator requires three features: a means of summing the carrier
and modulating waves, a nonlinear element, and a band pass filter for extracting the
desired modulation products. Semi-conductor diodes and transistors are the most
common nonlinear devices used for implementing square law modulators. The
filtering requirement is usually satisfied by using a single or double tuned filters.

When a nonlinear element such as a diode is suitably biased and operated in a

restricted portion of its characteristic curve, that is ,the signal applied to the diode is
relatively weak, we find that transfer characteristic of diode-load resistor
combination can be represented closely by a square law :
V0 (t) = a1Vi (t) + a2 Vi2(t) ……………….(i)
Where a1, a2 are constants
Now, the input voltage Vi (t) is the sum of both carrier and message signals i.e.,
Vi (t) =Accos 2πfct+m (t) ……………. (ii)
Substitute equation (ii) in equation (i)we get
V0 (t) =a1Ac [1+kam (t)] cos2πfct +a1m (t) +a2Ac2cos22πfct+a2m2 (t) ……….. (iii)
Where ka =2a2/a1
Now design the tuned filter /Band pass filter with center frequency fc and pass
band frequency width [Link] can remove the unwanted terms by passing this output
voltage V0(t) through the band pass filter and finally we will get required AM signal.
V0 (t) =a1Ac [1+2a2/a1 m (t)] cos2πfct
Assume the message signal m (t) is band limited to the interval –W ≤ f ≤ W

Fig .Spectrum of message signal

 The Fourier transform of output voltage VO (t) is given by
VO (f) = a1AC/2[δ(f-fc) + δ(f+fc)] +a2 AC [M (f-fc) + M (f+fc)]

Fig: Spectrum of AM signal

The AM spectrum consists of two impulse functions which are located at fc & -fc
and weighted by Aca1/2 & a2Ac/2, two USBs, band of frequencies from fc to fc +W
and band of frequencies from -fc-W to –fc, and two LSBs, band of frequencies from
fc-W to fc & -fc to -fc+W.

Demodulation of AM waves
There are two methods to demodulate AM signals. They are:
• Square-law detector
• Envelope detector
Square-law detector:-
A Square-law modulator requires nonlinear element and a low pass filter for
extracting the desired message signal. Semi-conductor diodes and transistors are the
most common nonlinear devices used for implementing square law modulators. The
filtering requirement is usually satisfied by using a single or double tuned filters.
When a nonlinear element such as a diode is suitably biased and operated in a
restricted portion of its characteristic curve, that is ,the signal applied to the diode is
relatively weak, we find that transfer characteristic of diode-load resistor
combination can be represented closely by a square law :
V0 (t) = a1Vi (t) + a2 Vi2 (t) ……………….(i)
Where a1, a2 are constants
 Now, the input voltage Vi (t) is the sum of both carrier and message signals
i.e.,Vi (t) = Ac [1+kam (t)] cos2πfct …………….(ii)
Substitute equation (ii) in equation (i) we get
V0 (t) = a1Ac [1+kam (t)] cos2πfct +1/2 a2Ac2 [1+2 kam (t) + ka2 m2 (t)]
[cos4πfct]………..(iii)
Now design the low pass filter with cutoff frequency f is equal to the required
message signal bandwidth. We can remove the unwanted terms by passing this
output voltage V0 (t) through the low pass filter and finally we will get required
message signal.
V0 (t) = Ac2 a2 m (t)
The Fourier transform of output voltage VO (t) is given by
VO (f) = Ac2 a2 M (f)

M (f)
Ac2 a2 M(0)

-W 0 W f

Fig: Spectrum of Output signal

Envelope Detector:
Rs

RL
AM signal C O/P

Fig.: Envelope detector

Envelope detector is used to detect high level modulated levels, whereas square-
law detector is used to detect low level modulated signals (i.e., below 1v). The
operation of the envelope detector is as follows. On a positive half cycle of the input
signal, the diode is forward biased and the capacitor C charges up rapidly to the peak
value of the input signal. When the input signal falls below this value, the diode
becomes reverse biased and the capacitor C discharges slowly through the load
resistor Rl . The discharging process continues until the next positive half cycle.
When the input signal becomes greater than the voltage across the capacitor, the
diode conducts again and the process is repeated.
The charging time constant RsC is very small when compared to the carrier
period 1/fc i.e.,
RsC << 1/fc
Where Rs = internal resistance of the voltage source.
C = capacitor
i.e., the capacitor C charges rapidly to the peak value of the signal.
The discharging time constant RlC is very large when compared to the charging
time constant i.e.,
1/fc << RlC << 1/W , Where Rl = load resistance value
W = message signal bandwidth i.e., the capacitor discharges slowly through the
load resistor.
Advantages
• It is very simple to design
• It is inexpensive
• Efficiency is very high when compared to Square Law detector
Disadvantage:
Due to large time constant, some distortion occurs which is known as diagonal
clipping i.e., selection of time constant is somewhat difficult
Application:
It is most commonly used in almost all commercial AM Radio receivers.
Assignment-Cum-Tutorial Questions

A. Questions testing the remembering / understanding level

I) Objective Questions:
1. The maximum power efficiency of an AM modulator is
(a) 25% (b) 50% (c) 33% (d) 100%
2. Square law modulator is
a. Is used for AM generation
b. Consists of four diodes connected in the form of ring
c. Is a product modulator
d. Is used for DSB SC ganeration

3. Frequency components of an AM wave (m = modulation index) are

[Link] frequency (ωc ) with amplitude A

b. Upper side band (ωc + ωm) having amplitude mA/2
c. Lower side band (ωc - ωm) having amplitude mA/2
d. All of the above
4. The requirements of carrier frequency are

a) precise frequency value b) low frequency drift

 c) both (a) and (b) d) neither (a) nor (b)

5. In an Amplitude Modulation
[Link] of the carrier varies constant b. Frequency of the carrier
remains
c. Phase of the carrier remains constant d. All of the above

6. If modulation index is greater than 1

[Link] baseband signal is not preserved in the envelope of the AM signal
b. The recovered signal is distorted
c. It is called over modulation
d. All of the above

7. Consider sinusoidal modulation in an AM systems. Assuming no over modulation

, the modulation index (μ) when the maximum and minimum values of the envelope,
respectively, are 3V and 1V is ____________.

8. The most suitable method for detecting a modulated signal (6+5coswmt) cos(wct)
is

A. Envelope detector B. Synchronous detector

C. Ratio detector D. Both A and B

9. AM demodulation techniques are

[Link] law demodulator b. Envelope detector
c. PLL detector d. Both a and b are correct

10. Suppose that the modulating signal is m(𝑡)=2cos(2𝜋𝑓𝑚𝑡) and the carrier signal is
c(𝑡)=𝐴𝐶cos(2𝜋𝑓𝑐𝑡).Which one of the following is a conventional AM signal
without over-modulation?
(a) (𝑡)=𝐴𝐶𝑚(𝑡)cos(2𝜋𝑓𝑐𝑡) b) (𝑡)=𝐴𝐶[1+𝑚(𝑡)]cos(2𝜋𝑓𝑐𝑡)
(c) (𝑡)=𝐴𝐶cos(2𝜋𝑓𝑐𝑡)+𝐴𝐶4𝑚(𝑡)cos(2𝜋𝑓𝑐𝑡)
(d) (𝑡)=𝐴𝐶cos(2𝜋𝑓𝑚𝑡)cos(2𝜋𝑓𝑐𝑡)+𝐴𝐶sin(2𝜋𝑓𝑚𝑡)sin(2𝜋𝑓𝑐𝑡)

II) Descriptive Questions

1. Defines modulation and explains need of modulation.
2. Represent single tone AM modulated wave in time and frequency domain.
3. Derive the total power in AM modulated signal.
4. Explain generation of square law AM modulated wave.
5. Explain the detection of the AM waves using envelope detector.
6. Explain generation of AM modulated wave using switching modulator.
B. Question testing the ability of students in applying the concepts.
I) Multiple choice Questions:
1. What is the carrier frequency in an AM wave when its highest frequency
component is 850Hz and the bandwidth of the signal is 50Hz?
a. 800 Hz b. 695 Hz c. 625 Hz d. 825 Hz

2. Calculate the power in one of the side band in Am modulation when the carrier
power is 124W and there is 80% modulation depth in the amplitude modulated
signal.

a. 89.33W b. 64.85W c. 79.36W d. 102 W

3. A 1 MHz sinusoidal carrier is amplitude modulated by a symmetrical square wave

of period 100 μsec. Which of the following frequencies will not be present in the
modulated signal?
(a) 990 kHz (b) 1010 kHz
(c) 1020 kHz (d) 1030 kHz

4. A message signal given by m(𝑡)=(1/2)cos𝜔1𝑡−(1/2)sin𝜔2𝑡 is amplitude

modulated with a carrier of frequency ωcto generate s(𝑡)=[1+𝑚(𝑡)]cos𝜔𝑐𝑡. What
is the power efficiency achieved by this modulation scheme?
(a) 8.33% (b) 11.11% (c) 20% (d) 25%

5 Which of the following demodulator (s) can be used for demodulating the signal
s(𝑡)=5(1+2cos200 𝜋𝑡)𝑐𝑜𝑠20000𝜋𝑡
(a) Envelope demodulator (b) Square-law demodulator
(c) Synchronous demodulator (d) None of the above

7. An AM signal is detected using an envelope detector. The carrier frequency and

modulation signal frequency are 1 MHz and 2 KHz respectively. An appropriate
value for the time constant of the envelope detector is
(a) 500 μsec (b) 20 μsec (c) 0.2 μsec (d) 1 μsec
8. A carrier is simultaneously modulated by two sine waves having modulation
indices of 0.4 and 0.3. The total modulation index will be

A. 0.1 B. 0.7 C. 0.5 D. 0.35

II) Problems:
1. A 2000 Hz audio signal having amplitude if 15V modulates a 100KHz
carrier which has a peak value of 25V when not modulated. Calculate
i) Modulation index
ii) Total power required for transmission
iii) What frequencies would show up in a spectrum analysis of modulated
wave.
2. The output signal from an AM modulator is given by
s (t) = 5 cos (1800πt) + 20 cos (2000πt) +5 cos (2200πt).
(i) Determine the message signal m (t) and the carrier c (t).
(ii) Determine the modulation index.
(iii) Determine the ratio of the power in the sidebands to the power in the carrier.

3. A carrier signal c (t) =5cos (2π106t) is modulated by message signal m (t) =cos
(4π103t) to generate an AM signal with μ=1. Calculate bandwidth and total
power.

4. An AM modulator is s (t) = 25(1+0.7 cos 5000πt – 0.3 cos 10000πt ) cos 5x106 πt.
(i) Determine the amplitudes and frequencies of the carrier and sidebands
(ii) Determine the effective modulation index.
(iii) Determine the bandwidth.

5. The AM signal S(t) = (1+µcos(2πfct) is to be detected by envelope detection.

What is the possible value of µ for which such detection possible? Sketch the
envelop detector output ofr µ=2.

6. The signal s(t)=[1+0.2 cos(2π(fm/3)t] cos(2πfct) is demodulated using a square

law demodulator having the characteristics So(t)=[s(t)+2]2. The output is then
filtered by an ideal LPF having cut off frequency at fm. determine the spectrum of
the demodulated signal and plot it in the frequency range -fm ≤ f≤ fm.

7. Using the message signal m(t)=1/(1+t2)

Determine and sketch the modulated waves for the following methods of
modulation:

a) Amplitude modulation with 50 percent modulation.

b) Double sideband- suppressed carrier modulation.

C) Previous GATE/IES Questions

1. A message signal m(t) = cos200pt + 4cos pt modulates the carrier c(t) = cos2πft
where f = 1 M Hz to produce an AM signal. For demodulating the generated AM
signal using an envelope detector, the time constant RC of the detector circuit
should satisfy. ( )
Gate 2011
(A) 0.5 ms < RC < 1ms (B) 1μs << RC < 0.5 ms
(C) RC << μs (D) RC >> 0.5 ms
2. Consider sinusoidal modulation in an AM system. Assuming no over modulation,
the modulation index (μ) when the maximum and minimum values of the
envelope, respectively, are 3 V and 1 V, is ________.
Gate 2014
3. Consider the signal s(t)= m(t)cos(wct)+ mˆ(t)cos(wct) where mˆ t denotes the
Hilbert
transform of m(t) and the bandwidth of m(t) is very small compared to wc. The
signal s(t) is a
(A) high-pass signal Gate 2015
(B) low-pass signal (C) band-pass signal
(D) double sideband suppressed carrier signal

4. Which of the following demodulator(s) can be used for demodulating the signal
S(𝑡) = 5(1 + 2cos200 𝜋𝑡)𝑐𝑜𝑠20000𝜋𝑡

(a) Envelope demodulator (b) Square-law demodulator

(c) Synchronous demodulator (d) None of the above

GATE 1993
5. The amplitude modulated wave form (𝑡) = [1 + 𝐾(𝑡)]cos𝜔𝐶𝑡 is fed to an ideal
envelope detector. The maximum magnitude of (𝑡) is greater than 1. Which of the
following could be the detector output ?

(a) m(𝑡) (b) 𝐴𝑐 2[1 + 𝐾(𝑡)]2

(c) |𝐴𝐶[1 + 𝐾𝑎𝑚(𝑡)]| (d) 𝐴𝐶|1 + 𝐾𝑎𝑚(𝑡)|2

GATE
2000

6. An AM signal is detected using an envelope detector. The carrier frequency and

modulation signal frequency are 1 MHz and 2 KHz respectively. An appropriate
value for the time constant of the envelope detector is

(a) 500 µsec (b) 20 µsec (c) 0.2 µsec (d) 1 µsec

GATE 2004
 Unit – II
CONTINUOUS WAVE MODULATION – II

Objectives:
➢ To introduce the concepts of DSBSC and SSBSC modulation
techniques and also to describe the effect of noise on analog
modulated signals

Syllabus:
AM DSBSC Modulation - Time domain and frequency domain description,
Generation of DSBSC Waves, Coherent detection of DSBSC Modulated waves,
Costas loop.
AM SSBSC Modulation- Frequency domain description, Time domain description,
Generation AM-SSB Modulated Waves-frequency discrimination, phase
discrimination, Demodulation of SSB Waves, Noise in AM Systems, Comparison of
AM techniques.

Outcomes:
Students will be able to
➢ Analyze AM DSB-SC systems
➢ Understand need of SSB modulation technique
➢ Determine power for various modulation schemes.
➢ Analyze the effect of noise in AM

DSBSC MODULATION
Double sideband-suppressed Carrier (DSB-SC) modulation, in which the
transmitted wave consists of only the upper and lower sidebands. Transmitted power
is saved through the suppression of the carrier wave, but the channel bandwidth
requirement is same as in AM (i.e. twice the bandwidth of the message signal).
Basically, double sideband-suppressed (DSB-SC) modulation consists of the
product of both the message signal m (t) and the carrier signal c(t),as follows:
S (t) =c (t) m (t)
S (t) = Ac cos(2πfct) m (t)
The modulated signal s (t) undergoes a phase reversal whenever the message
signal m (t) crosses zero. The envelope of a DSB-SC modulated signal is different
from the message signal.
The transmission bandwidth required by DSB-SC modulation is the same as
that for amplitude modulation which is twice the bandwidth of the message signal,
2W. Assume that the message signal is band-limited to the interval –W ≤f≤ W
 Fig: Spectrum of message signal

Fig: Spectrum of DSBSC wave

In single-tone modulation modulating signal consists of only one frequency

component where as in multi-tone modulation modulating signal consists of more
than one frequency components.
The standard time domain equation for the DSB-SC modulation is given by
S (t) = Ac cos(2πfct) m (t)………………… (1)
Assume m (t) = Am cos(2πfmt)……………….. (2)
Substitute equation (2) in equation (1) we will get
S (t) =Ac Am cos (2fct) cos (2fmt)
S (t) = Ac Am/2[cos 2π(fc-fm)t + cos 2π(fc+fm) t]…………… (3)

The Fourier transform of s (t) is

S (f) =Ac Am/4[δ (f-fc-fm) + δ (f+fc+fm)] + Ac Am/4[δ (f-fc+fm) + δ (f+fc+fm)]

Fig: Spectrum of Single tone DSBSC wave

Generation of DSB-SC waves- Balanced Modulator and Ring modulator

There are two methods to generate DSB-SC waves. They are
• Balanced modulator
• Ring modulator

Balanced Modulator:-

Fig: Balanced Modulator

One possible scheme for generating a DSBSC wave is to use two AM

modulators arranged in a balanced configuration so as to suppress the carrier wave,
as shown in above fig. Assume that two AM modulators are identical, except for the
sign reversal of the modulating signal applied to the input of one of the modulators.
Thus the outputs of the two AM modulators can be expressed as follows:
S1 (t) = Ac [1+kam (t)] cos 2fct and
S2 (t) = Ac [1- kam (t)] cos 2fct
Subtracting S2 (t) from S1 (t), we obtain
S (t) = S1 (t) – S2 (t)
S (t) = 2Ac kam (t) cos 2fct
Hence, except for the scaling factor 2ka the balanced modulator output is
equal to product of the modulating signal and the carrier signal
The Fourier transform of s (t) is
S (f) =kaAc [M (f-fc) + M (f+fc)]
Assume that the message signal is band-limited to the interval –W ≤f≤ W
 Fig: Spectrum of Baseband signal

Fig: Spectrum of DSBSC wave

Ring modulator:-

Fig: Ring modulator

One of the most useful product modulator, well suited for generating a DSB-
SC wave, is the ring modulator shown in above figure. The four diodes form ring in
which they all point in the same way-hence the name. The diodes are controlled by a
square-wave carrier c (t) of frequency fc, which applied longitudinally by means of
to center-tapped transformers. If the transformers are perfectly balanced and the
diodes are identical, there is no leakage of the modulation frequency into the
modulator output. On one half-cycle of the carrier, the outer diodes are switched to
their forward resistance rf and the inner diodes are switched to their backward
resistance rb .on other half-cycle of the carrier wave, the diodes operate in the
opposite condition.
 The square wave carrier c (t) can be represented by a Fourier series as follows:
∞
c (t)=4/π Σ (-1)n-1/(2n-1) cos [2πfct(2n-1)]
n=1
When the carrier supply is positive, the outer diodes are switched ON and the inner
diodes are switched OFF, so that the modulator multiplies the message signal by +1
When the carrier supply is positive, the outer diodes are switched ON and the inner
diodes are switched OFF, so that the modulator multiplies the message signal by +[Link] the
carrier supply is negative, the outer diodes are switched OFF and the inner diodes are switched
ON, so that the modulator multiplies the message signal by -1.
Now, the Ring modulator output is the product of both message signal m (t) and carrier
signal c (t).
S (t) =c (t) m (t)
∞
S (t) =4/π Σ (-1) n-1/ (2n-1) cos [2πfct (2n-1)] m (t)
n=1
For n=1
S (t) =4/π cos (2πfct) m (t)
There is no output from the modulator at the carrier frequency i.e the modulator output
consists of modulation products. The ring modulator is sometimes referred to as a double-
balanced modulator, because it is balanced with respect to both the message signal and the
square wave carrier signal.

The Fourier transform of s (t) is S (f) =2/π [M (f-fc) + M (f+fc)]

Coherent Detection of DSB-SC Waves:-

Fig: Coherent detection of DSBSC waves.

 The base band signal m (t) can be recovered from a DSB-SC wave s(t) by multiplying s(t)
with a locally generated sinusoidal signal and then low pass filtering the product. It is assumed
that local oscillator signal is coherent or synchronized, in both frequency and phase ,with the
carrier signal c(t) used in the product modulator to generate s(t).this method of demodulation is
known as coherent detection or synchronous demodulation.
The product modulator produces the product of both input signal and local oscillator and
the output of the product modulator v (t) is given by

v (t) =Á c cos (2πfct+Ø) s (t)

v t) =Ác cos (2πfct+Ø) Ac cos2πfct m (t)

v (t) =Ac Ác /2 cos(2πfct+Ø) m(t)+ Ac Ác/2 cosØ m(t)

The high frequency can be eliminated by passing this output voltage to the Low Pass Filter. Now
the Output Voltage at the Low pass Filter is given by

v0 (t) = Ac Ác /2 cosØ m (t)

The Fourier transform of vo (t) is

VO (f) = Ac Ác /2 cosØ M (f)

The demodulated signal is proportional to the message signal m (t) when the phase
error is constant. The Amplitude of this Demodulated signal is maximum when Ø=0, and
it is minimum (zero) when Ø=±π/2 the zero demodulated signal, which occurs for
Ø=±π/2 represents quadrature null effect of the coherent detector.
The demodulated signal is proportional to the message signal m (t) when the phase
error is constant. The Amplitude of this Demodulated signal is maximum when Ø=0, and
it is minimum (zero) when Ø=±π/2 the zero demodulated signal, which occurs for
Ø=±π/2 represents quadrature null effect of the coherent detector.

COSTA’S Loop
 Coherent detection of a DSB-SC modulated wave requires that the locally
generated carrier in the receiver be synchronous in both frequency and phase with the
oscillator responsible for generating the carrier in the transmitter. This is a rather
demanding requirement, all the more so since the carrier is suppressed from the
transmitted DSB-SC signal Ac m(t) cos (2πfct). One method of satisfying this
requirement is to use the Costas receiver shown in Fig. This receiver consists of two
coherent detectors supplied with the same input signal—namely, the incoming DSB-SC
wave but with two local oscillator signals that are in phase quadrature with respect to
each other.

Fig: COSTA’S Loop

The detector in the upper path is referred to as the in phase coherent detector or I-
channel, and the detector in the lower path is referred to as the quadrature-phase coherent
detector or Q-channel. To understand the operation of this receiver, suppose that the local
oscillator signal is of the same phase as the carrier wave used to generate the incoming
DSBSC wave. Under these conditions, we find that the I-channel output contains the
desired demodulated signal m(t).
From the discussion on coherent detection in Section, we know that the I-channel
output is proportional to and for small hence, the I-channel output remains essentially
unchanged so long as is small. But there will now be some signal, albeit small, appearing
at the Q-channel output, which is proportional to for small This Q-channel output will
have the same polarity as the I-channel output for one direction of local oscillator phase
drift and the opposite polarity for the opposite direction of Thus, by combining the I- and
Q-channel outputs in a phase discriminator (which consists of a multiplier followed by a
time-averaging unit), a dc control signal proportional to the phase drift is generated. With
negative feedback acting around the Costas receiver, the control signal tends to
automatically correct for the local phase error in the voltage-controlled oscillator.

SINGLE SIDE BAND (SSB) Modulation Wave:-

In suppressing the carrier, DSB-SC modulation takes care of a major limitation of

AM that pertains to the wastage of transmitted power. To take care of the other major
limitation of AM that pertains to channel bandwidth, we need to suppress one of the two
sidebands in the DSB-SC modulated wave. This modification of DSB-SC modulation is
precisely what is done in single sideband (SSB) modulation. In effect, SSB modulation
relies solely on the lower sideband or upper sideband to transmit the message signal
across a communication channel. Depending on which particular sideband is actually
transmitted, we speak of lower SSB or upper SSB modulation. Specifically, we start the
study of SSB modulation by first considering the simple case of a sinusoidal modulating
wave, and then we generalize the results to an arbitrary modulating signal in a step-by-
step manner.
the cosine term cos(2πfct) is replaced by the sine term sin(2πfct) and we may reformulate the
SSB modulated wave

We observe that the periodic signal m^(t)can be derived from the periodic modulating signal m(t)
simply by shifting the phase of each cosine term by -90o
 Fig: SSB Frequency Spectrum
SSB Generations:-

1. Frequency Discrimination Method

The frequency-domain description presented for SSB modulation leads us naturally to the
frequency discrimination method for generating as SSB modulated wave. Application of the
method, however, requires that the message signal satisfy two conditions.

1. The message signal m(t) has little or no low-frequency content; that is, the message
spectrum M(f) has “holes” at zero frequency. An important type of message signal with such a
property is an audio signal (speech or music). In telephone, for example, the useful Frequency
content of a speech signal is restricted to the band 0.3-3.4kHz, thereby creating an energy gap
from zero to 300Hz
2. The highest frequency component W of the message signal m(t) is much less that the
carrier frequency fc. Then, under these conditions, the desired sideband will appear in a non-
overlapping interval in the spectrum in such a way that it may be selected by an appropriate
filter.

Thus an SSB modulator based on frequency discrimination consists basically of product

modulator and a filter designed to pass the desired sideband of the DSBSC modulated wave at
the product modulator output and reject the other sideband. A block diagram of this modulator is
shown in Fig 1. The most severe requirement of this method of SSB generation usually arises
from the unwanted sideband, the nearest frequency component of which is separated from the
desired sideband by twice the lowest frequency component of the message signal. In designing
the band-pass filter in the SSB modulation scheme must satisfy two basic requirements:

1. The pass band of the filter occupies the same frequency range as the spectrum of the desired
SSB modulated wave.
2. The width of the guard band of the filter, separating the pass band from the stop band where
the unwanted sideband of the filter input lies, is twice the lowest frequency component of
the message signal.

We usually find that this kind of frequency discrimination can be satisfied only by using highly
selective filters, which can be realized using crystal resonators with a Q factor per resonator in
the range of 1000 to 2000.
 Fig: Block diagram of Frequency Discrimination Method

In the above figure, Frequency discrimination method of SSB modulated wave a) single
stage b)two stage When it is necessary to generate an SSB modulated wave occupying a
frequency band that is much higher than that of the message signal (e.g., translating a voice
signal to the high-frequency region of the radio spectrum), it becomes very difficult to design an
appropriate filter that will pass the desired sideband and reject the other using the simple
arrangement of Fig.2. In such a situation it is necessary to resort to a multiple modulation process
so as to ease the filtering requirement. This approach is illustrated in Fig 1b involving two stages
of modulation. The SSB modulated wave at the first filter output is used as the modulation wave
for the second product modulator, which produces a DSBSC modulated wave with a spectrum
that is symmetrically spaced about the second carrier frequency fc2. The frequency separation
between the sidebands of this DSBSC modulated wave is effectively twice the first carrier
frequency fc1, thereby permitting the second filter to remove the unwanted sideband.

2. Phase Discrimination Method

The second method for SSB generation, called the phase discrimination method, is
depicted in Fig. its implementation follows from the time-domain description of SSB waves
defined in Equations. This second SSB modulator consists of two parallel paths, one called the
in-phase path and the other called the quadrature path. Each path involves a product modulator.
The sinusoidal carrier waves applied to the two product modulators are in phase quadrature,
which is taken care of by simply using a -900 phase-shifter as shown in Fig.3 However, it
requires special attention is the wide-band phase-shifter, which is designed to produce the Hilbert
transform m^(t) in response to the incoming message signal m(t). The role of the quadrature path
embodying the wide-band phase shifter is merely to interfere with the in-phase path so as to
eliminate power in one of the two sidebands, depending on whether upper SSB or lower SSB is
the requirement. The two modulators are clearly quite different in their structures. In terms of
design challenge, the band-pass filters in the frequency discriminator of stands out as the
functional block that requires special attention. On the other hand, in the phase discriminator, it
is the wide-band phase shifter that requires special attention.

Fig: Phase Discrimination Method

Demodulation technique of SSB wave:-

Coherent Detection of SSB wave

The product modulator is a type of coherent SSB demodulator. To recover the
modulating signal from the SSB-SC signal, we require a phase coherent or synchronous
demodulator. The received SSB signal is first multiplied with a locally generated carrier signal.
The locally generated carrier should have exactly the same frequency as that of the suppressed
carrier. The product modulator multiplies the two signals at its input and the product signal is
passed through a low pass filter with a bandwidth equal to fm. At the output of the filter, we get
the modulating signal back.

Fig: Block Diagram of coherent SSB demodulator

 The combination of the remaining terms represents an SSB modulated wave with a
carrier frequency of 2fc; as such, it represents an unwanted component in the product modulator
output that is removed by low-pass filtering. The detection of SSB modulated waves, just
presented, assumes ideal conditions, namely, perfect synchronization between the local carrier
and that in the transmitter both in frequency and phase. The effect of a phase error in the local
generated carrier wave is to modify the detector output as follows.

Where the plus sign applies to an incoming SSB modulated wave containing only the
upper sideband and the minus sign applies to one containing only the lower sideband. Owing to
the phase error, the detector output Vo(t) contains not only the message signal m(t) but also its
Hilbert transform m(t). Consequently, the detector output suffers from phase distortion. This
phase distortion is usually not serious with voice communications because the human ear is
relatively in sensitive to phase distortion. The presence of phase distortion gives rise to what is
called the Donald Duck voice effect. In the transmission of music and video signals, on the other
hand, phase distortion in the form of a constant phase difference in all components can be
intolerable.
Noise:
Comparison of AM techniques:-

[Link]: Parameter DSB-FC (AM) DSB-SC SSB-SC

1 Power High Medium Less
2 Band Width 2fm 2fm fm
3 Carrier Suppression No Yes Yes
One Side band
4 Side band Transmission No No
Completely
5 Transmission Efficiency Minimum Moderate Maximum
6 Receiver Complexity Simple Complex Complex
7 Modulation Type Non-Linear Linear Linear
 Unit – III

Objectives:
➢ To introduce the concepts of AM Transmission and multiplexing

UNIT – III AM Transmitters

Classification of Transmitters, AM Transmitters: high level and low level AM transmitters,
Frequency division multiplexing and Time division multiplexing
Outcomes:
Students will be able to
➢ understand various blocks in AM transmitters

➢ Know the difference between the TDM and FDM

 AM TRANSMITTERS

CLASSIFICATIONOF TRANSMITTERS
Transmitters are mainly classified into two types
• Low level transmitters
• High level transmitters
AM TRANSMITTERS
Transmitters that transmit AM signals are known as AM transmitters. These
transmitters are used in medium wave (MW) and short wave (SW) frequency bands for AM
broadcast. The MW band has frequencies between 550 KHz and 1650 KHz, and the SW band
has frequencies ranging from 3 MHz to 30 MHz The two types of AM transmitters that are used
based on their transmitting powers are:
1. High Level
2. Low Level
HIGH LEVEL TRANSMITTER
High level transmitters use high level modulation, and low level transmitters use low
level modulation. The choice between the two modulation schemes depends on the transmitting
power of the AM transmitter. In broadcast transmitters, where the transmitting power may be of
the order of kilowatts, high level modulation is employed. In low power transmitters, where only
a few watts of transmitting power are required, low level modulation is used. High-Level and
Low-Level Transmitters Below figure show the block diagram of high-level and low-level
transmitters. The basic difference between the two transmitters is the power amplification of the
carrier and modulating signal.
 Fig. Block diagram of High level AM transmitter

In low-level modulation, the powers of the two input signals of the modulator stage are
not amplified. The required transmitting power is obtained from the last stage of the transmitter,
the class C power amplifier.
The various sections of the figure are:
1. Carrier oscillator.
2. Buffer amplifier.
3. Frequency multiplier.
4. Power amplifier
5. Audio chain.
6. Modulated class C power amplifier.
Carrier oscillator
The carrier oscillator generates the carrier signal, which lies in the RF range. The
frequency of the carrier is always very high. Because it is very difficult to generate high
frequencies with good Frequency stability, the carrier oscillator generates a sub multiple with the
required carrier frequency. This sub multiple frequency is multiplied by the frequency multiplier
stage to get the required carrier frequency. Further, a crystal oscillator can be used in this stage to
generate a low frequency carrier with the best frequency stability. The frequency multiplier stage
then increases the frequency of the carrier to its requirements.
Buffer Amplifier
The purpose of the buffer amplifier is twofold. It first matches the output impedance of
the carrier oscillator with the input impedance of the frequency multiplier, the next stage of the
carrier oscillator. It then isolates the carrier oscillator and frequency multiplier. This is required
so that the multiplier does not draw a large current from the carrier oscillator. If this occurs, the
frequency of the carrier oscillator will not remain stable.
Frequency Multiplier
The sub-multiple frequency of the carrier signal, generated by the carrier oscillator, is
now applied to the frequency multiplier through the buffer amplifier. This stage is also known as
harmonic generator. The frequency multiplier generates higher harmonics of carrier oscillator
frequency. The frequency multiplier is a tuned circuit that can be tuned to the requisite carrier
frequency that is to be transmitted.
Power Amplifier
The power of the carrier signal is then amplified in the power amplifier stage. This is the
basic requirement of a high-level transmitter. A class C power amplifier gives high power
current pulses of the carrier signal at its output.
Audio Chain
The audio signal to be transmitted is obtained from the microphone, as shown in figure.
The audio driver amplifier amplifies the voltage of this signal. This amplification is necessary to
drive the audio power amplifier. Next, a class A or a class B power amplifier amplifies the power
of the audio signal.
Modulated Class C Amplifier
This is the output stage of the transmitter. The modulating audio signal and the carrier
signal, after power amplification, are applied to this modulating stage. The modulation takes
place at this stage. The class C amplifier also amplifies the power of the AM signal to the
reacquired transmitting power. This signal is finally passed to the antenna. This radiates the
signal into space of transmission.
LOW LEVEL TRANSMITTER

Fig. Block diagram of low level AM transmitter

 The low-level AM transmitter shown in the above figure is similar to a high-level
transmitter, except that the powers of the carrier and audio signals are not amplified. These two
signals are directly applied to the modulated class C power amplifier. Modulation takes place at
the stage, and the power of the modulated signal is amplified to the required transmitting power
level. The transmitting antenna then transmits the signal.
Coupling of Output Stage and Antenna
The output stage of the modulated class C power amplifier feeds the signal to the
transmitting antenna. To transfer maximum power from the output stage to the antenna it is
necessary that the impedance of the two sections match. For this, a matching network is required.
The matching between the two should be perfect at all transmitting frequencies. As the matching
is required at different frequencies, inductors and capacitors offering different impedance at
different frequencies are used in the matching networks.
Multiplexing
Multiplexing is a method by which multiple analog or digital signals are combined into
one signal over a shared medium. There are three types of multiplexing techniques.
1. Time-division multiplexing
2. Frequency-division multiplexing
3. Wave length-division multiplexing
Frequency-division multiplexing (FDM)
FDM is an analog technique which is applied only when the bandwidth of the link is
greater than the combined bandwidth of the signals to be transmitted. Each sending device
generates signals which modulate at different carrier frequencies. To hold the modulated signal,
the carrier frequencies are separated by sufficient bandwidth.
The operation of frequency division multiplexing (FDM) is based on sharing the available
bandwidth of a communication channel among the signals to be transmitted.
FDM Transmitter
Each signal modulates a separate carrier. The modulator outputs will contain the sidebands of the
corresponding signals. The linear mixer is different from the normal mixers. Here the sum and
difference frequency components are not produced. But only the algebraic addition of the
modulated outputs will take place.

FDM Receiver
Each BPF has a center frequency corresponding to one of the carriers. The BPFs have an
adequate bandwidth to pass all the channel information without any distortion. Each filter will
pass only its channel and rejects all the other channels. The channel demodulator then removes
the carrier and recovers the original signal back.

Time division multiplexing

TDM is a communications process that transmits two or more streaming digital signals
over a common channel. In TDM, incoming signals are divided into equal fixed-length time
slots. After multiplexing, these signals are transmitted over a shared medium and reassembled
into their original format after de-multiplexing. Time slot selection is directly proportional to
overall system efficiency.

Since the synchronization of the signals is a must for a time division multiplexing system, a
synchronization pulse is used at the receiver side. It is used to decode the information from individual
channels in the proper way. The absence of synchronization will lead to severe interference among the
channels in almost all cases. The output is decoded and by using suitable filters, information from each
channel is separated.

Difference between TDM and FDM

[Link]: FDM TDM

1 The signals which are to be multiplexed The signal which is to be multiplexed,
are added in the time domain. But they they occupy the entire bandwidth in the
occupy different slots in the frequency time domain.
domain.
2 FDM is usually preferred for the analog TDM is preferred for the digital signal.
signals.
3 Synchronization is not required. Synchronization is required is required.
4 The FDM requires a complex circuitry at TDM circuitry is not very complex.
transmitter and receiver.
5 FDM suffers from the problem of In TDM the problem of crosstalk is not
 crosstalk due to imperfect Band Pass severe.
Filter
6 Due to bandwidth fading in the Due to fading only a few TDM
transmission medium, all the FDM channels will be affected.
channels are affected.

ASSIGNMENT CUM TUTORIAL QUESTIONS

A) Questions testing the understanding and remembering level of students.
I. Objective Questions
1. How many types of multiplexing techniques are
(a) One (b) Two (c) Three (d) Four
2. Time-Division multiplexing (TDM) is a digital technique of
(a) Encoding (b) Decoding (c) Multiplexing (d) De-multiplexing
3. Applications of Frequency-Division Multiplexing (FDM) are
(a) Television broadcasting (b) AM and FM radio stations
(c) Cellular Telephones (d) All of the above
4. FDM is an analog multiplexing technique that combines
(a) Digital signals (b) Analog signals (c) Both a and b (d) None

[Link] Questions

1. Explain the operation of high level AM transmitter

2. Define multiplexer
3. Explain the concept of Time Division Multiplexing.
4. Explain the concept of Frequency Division Multiplexing
5. Explain the operation of low level AM transmitter
6. Distinguish between the TDM & FDM multiplexing techniques?

Problems
1. Assume that a voice channel occupies a bandwidth of 4 kHz. We need to multiplex 10 voice
channels with gaurd bands of 500 Hz using FDM. Calculate the required bandwidth.
2. Twenty-four voice signals are to be multiplexed and transmitted over twisted pair. What is
the bandwidth required for FDM?
3. A signal m1(t) is band-limited to 3.6khz and 3 other signals m2(t), m3(t) and m4(t) are band-
limited to 1.2khz each. These signals are to be transmitted by means of TDM
(a) Set up a scheme for accomplishing this multiplexing requirement, with each signal
sampled at its Nyquist rate.
(b) What must be the speed of commutator (in samples per second)?
(c) Determine the minimum transmission bandwidth of the channel.
 Unit – IV

Objectives:
➢ To introduce the concepts of AM Receivers

UNIT - IV:AM Receivers

Receiver types – tuned radio frequency receiver, super heterodyne receiver,image frequency and
rejection ratio,RF section and receiver characteristics,AGC.
Outcomes:
Students will be able to
➢ Understand various blocks in AM receivers
➢ Analyze image frequency and rejection ratio
➢ Compare TRF receiver with super heterodyne receiver
 RECEIVERS
This process is used in the receivers to recover the original signal coming from the sender
end in modulating form. We can say that its function is opposite to that of modulation process.
As we have discussed earlier that signal transmission is done by superimposing the
signal on a carrier wave. This is done by using any one method from the number of available
modulation methods. Then after modulation these amplified signals are transmitted successfully
with the help of a transmitting antenna.
When the signals reach the destination i.e. at the receiver end, then the signal strength
will be very less. This weak signal is amplified with the help of other signals. After amplification
this signal is filtered from the other signals which were used earlier to modify it. When the signal
becomes ready for demodulation process, then the below steps are performed for demodulation.
These steps are basically the functions of the receiver:

1. Demodulating and amplifying the received signal.

2. Filtering of the original received signal from the non necessary signals.
3. Proper display of the received signal after the completion of demodulation process.

TYPES OF RECEIVERS
Receivers are basically available in two types:
1. Tuned Radio Frequency Receiver.
2. Super Heterodyne Receiver.
TUNED RADIO FREQUENCY RECEIVER
The RF signal (RF energy) is picked up by the antenna, as it is in all receivers, and goes
into the tuned circuit. The circuit diagrams above shows a parallel tuned circuit, which can be
tuned to the desired radio station (frequency) by the variable capacitor.
At the resonant frequency, the parallel tuned circuit has a high impedance (or
opposition to AC / RF currents), the effect of this is to generate a maximum AC / RF potential
difference across the tuned circuit.
The potential difference is rectified by the diode and DC is smoothed out by the
capacitor C. The end result is that the recovered audio is only the overall shape of the signal
which is applied to the ear piece / headphones which would have been the same shaped audio
signal used to modulate the transmitter hence you hear what was transmitted.

Fig: TRF Receiver Block Diagram

The Tuned Radio Frequency receiver is for AM signals and with the BFO ( beat
frequency oscillator switched in can also receive CW signals.
RF Amplifier
TRF has a RF Amplifier (also called a Radio Frequency Stage or (RF) Stage). This
stage amplifies the incoming signal from the antenna before it passes it to the detector or
demodulator stage.
The detector (also call the demodulator)
The detector recovers the AM signal and operates on the same principal as the crystal
set but has the advantage that the amplified signal is provided by the RF amplifier stage. Then
the recovered audio is passed to the Audio Amplifier stage.
Audio Amplifier
The Audio Amplifier stage simply makes the recovered audio signal louder so that it is
sufficient to hear in a loudspeaker.
BFO
The detector mixing with the BFO recovers the CW signal. As mentioned above the
detector recovers the AM signal and with the BFO switched in a CW signal can be resolved.
Then the recovered audio is passed to the Audio Amplifier stage.

SUPERHETERODYNE RADIO RECEIVER

 One of the most common forms of radio receiver is the superhet or superheterodyne
radio receiver. Virtually all broadcast radio receivers, as well as televisions, short wave receivers
and commercial radios use the superheterodyne principle as the basis of their operation. The
superheterodyne radio technique is used in most radios found around the home. Virtually all
transistor portable radios as well as television sets, hand portable radios and many others use the
superheterodyne principle.
Basic superheterodyne concepts
The superhet or superheterodyne radio operates by taking the signal on the incoming
frequency, mixing it with a variable frequency locally generated signal to convert it down to a
frequency where it can pass through a high performance fixed frequency filter before being
demodulated to extract the required modulation or signal. It is obviously necessary to look ay
this in more detail to understand the concept behind what goes on, but the main process in the
superheterodyne radio is that of mixing.

How the superheterodyne receiver works

In order to look at how a superhet or superheterodyne radio works, it is necessary to
follow the signal through it. In this way the processes it undergoes can be viewed more closely.
The signal that is picked up by the antenna passes into the receiver and enters a mixer. Another
locally generated signal, often called the local oscillator, is fed into the other port on the mixer
and the two signals are mixed. As a result new signal are generated at the sum and difference
frequencies.
The output from the mixer is passed into what is termed the intermediate frequency or
IF stages where the signal is amplified and filtered. Any of the converted signals that fall within
the pass band of the IF filter will be able to pass through the filter and they will also be amplified
by the amplifier stages. Any signals that fall outside the pass band of the filter will be rejected.
It is often helpful to look at a real example to illustrate how the process works. To see
how this operates in reality take the example of two signals, one at 1.0 MHz and another at
1.1MHz. If the IF filter is centred at 0.25 MHz, and the local oscillator is set to 0.75 MHz, then
the two signals generated by the mixer as a result of the 1.0 MHz signal fall at 0.25 MHz and
1.75MHz. Naturally the 1.75 MHz signal is rejected, but the one at 0.25 MHz passes through the
IF stages. The signal at 1.1 MHz produces a signal at 0.35 MHz and another at 1.85MHz. Both
of these fall outside bandwidth of the IF filter so the only signal to pass through the IF is that
from the signal on 1.0 MHz.

If the local oscillator frequency is moved up by 0.1 MHz to 0.85 MHz then the signal at
1.1 MHz will give rise to a signal at 0.25 MHz and another at 1.95 MHz. As a result the signal at
1.1 MHz giving rise to the 0.25 MHz signal after mixing will pass through the filter. The signal
at 1.0 MHz will give rise to a signal of 0.15 MHz at the IF and another at 1.85 MHz and both
will be rejected. In this way the receiver acts as a variable frequency filter, and tuning is
accomplished by varying the frequency of the local oscillator within the superhet or
superheterodyne receiver.
The advantage of the superheterodyne radio process is that very selective fixed
frequency filters can be used and these far out perform any variable frequency ones. They are
also normally at a lower frequency than the incoming signal and again this enables their
performance to be better and less costly.
Complete superheterodyne receiver
Having looked at the concepts behind the superheterodyne receiver it is helpful to look
at a block diagram of a basic superhet. Signals enter the front end circuitry from the antenna.
This contains the front end tuning for the superhet to remove the image signal and often includes
an RF amplifier to amplify the signals before they enter the mixer. Once the signals leave the
mixer they enter the IF stages. These stages contain most of the amplification in the receiver as
well as the filtering that enables signals on one frequency to be separated from those on the next.
Filters may consist simply of LC tuned transformers providing inter-stage coupling, or they may
be much higher performance ceramic or even crystal filters, dependent upon what is required.
 Once the signals have passed through the IF stages of the superheterodyne receiver, they
need to be demodulated. Different demodulators are required for different types of transmission,
and as a result some receivers may have a variety of demodulators that can be switched in to
accommodate the different types of transmission that are to be encountered. The output from the
demodulator is the recovered audio. This is passed into the audio stages where they are amplified
and presented to the headphones or loudspeaker.

Fig. Block diagram of a basic super heterodyne receiver

The diagram above shows a very basic version of the superhet or superheterodyne
receiver. Many sets these days are far more complicated.
RF SECTION AND RECEIVER CHARACTERISTICS
The RF stage provides
1. Increased sensitivity
2. Increased selectivity
3. Improved automatic volume control action.
4. Elimination of image frequency response.

Image Frequency and its rejection:

In a standard broadcast receiver the local oscillator frequency is made higher than the
incoming signal frequency for reasons that will become apparent. It is made equal at all times to
the signal frequency plus the intermediate frequency. Thus no fo = fsi +fi or fs = fo-fi matter what
the signal frequency may be. When fs and are fo mixed, the difference frequency , which is one
of the by-products, is equal to fi. As such , it is the only one passed and amplified by the IF stage.
 If a frequency fsi manages to reach the mixer, such that, fsi = fo + fi that is, fsi = fs +2fi
then this frequency will also produce, when mixed with. The relationship of these frequencies is
shown in above figure. This spurious intermediate-frequency signal will also be amplified by the
IF stage and will therefore provided interference. This has the effect of two stations being
received simultaneously and is naturally undesirable. The term fo is called the image frequency
and is defined as the signal frequency plus twice the intermediate frequency.
fsi = fs +2fi ----------(1)
The rejection of image frequency by a single-tuned circuit i.e, the ratio is the gain at the
signal frequency to the gain at the image frequency, is given by

--------------(2)
Where

-------------(3)

Automatic Gain Control:

The purpose of the Automatic Gain Control (AGC) / Automatic Voltage Control (AGC)
is to maintain a constant voltage level at the output even though the input voltage may vary. The
circuit can do this over a limited range eg. 30dB.

• Automatic gain control (AGC) is a mechanism wherein the overall gain of the radio
receiver is automatically varied according to the changing strength of the received signal.
This is done to maintain the output at a constant level.
• If the gain is not varied as per the input signal, consider a stronger input signal, then the
signal might probably be distorted with some of the amplifiers reaching saturation level.
• AGC is applied to the RF, IF and mixer stages, which also helps in improving the
dynamic range of the receiver antenna to 60-100 dB by adjusting the gain of the various
stages in the radio receiver.
 • The AGC derives dc bias voltage from the part of the detected signal to apply to the RF,
IF and mixer stages to control their gains. The transconductance and hence the gain of the
devices used in these stages of the receiver depends on the applied bias voltage or
current.
• When the overall signal level increases, the value of the applied AGC bias increase
leading to the decrease in the gain of the controlled stages.
• When there is no signal or signal with low value, there is minimum AGC bias which
results in amplifier generating maximum gain.
• AGC facilitates tuning to varying signal strength stations providing a constant output.

There are two types of AGC circuits:

1. Simple AGC 2. Delayed AGC

Simple AGC:

Fig: Simple AGC circuit

The above diagram shows the circuit diagram of linear diode detector with capacitor filter
and simpleAGC. In this circuit half wave rectifier voltage is developed across the load
instaneously positive bias voltage is applied . The capacitor ‘c’ is used to filter out RF
components due to which the output of capacitor ‘c’ only consists of Dc and modulating voltage
and obtain across the resistor R. The coupling capacitor Cc is used to require Dc components
and finally at the output of coupling capacitor only modulating voltage components are
available. To over come this frequency ccomponents we introduce an AGC filter .
The AGC filter mainly consists of two components
1.) Series resistance Ra
2.) Shunt capacitance Ca
The output of AGC only a positive DC voltage that is used in for AGC bias. If the time
constant RaCa is very high some lowest modulating frequency components from AGC bias. If
RaCa is very small the AGC bias follows the varying the series amplitude. Therefore, in
practical the AGC filter having time constant is in the range of 0.1-0.2 seconds.

Drawbacks of simple AGC circuit:

1. The simple AGC clock operates even for very weak signals. Due to that the gain
of receiver starts falling as soon as diode detector starts producing the output.
2. An ideal AGC signal must operates until the input carrier reaches to reasonable
pre-determined voltage subsequently the AGC must come into operation to
maintain the output level constant instead of variations of input voltage.
These two drawbacks are overcome by using delayed AGC

DELAYED AGC:
AGC bias is not applied to the amplifiers until signal strength crosses a predetermined
level, after which AGC bias is applied.
 Fig: Delayed AGC circuit

Operation:
When input carrier voltage is increasing the AGC bias produce due to rectification of
carrier voltage in diode detection D1 increases. When this rectifier bias magnitude exceeds the
magnitude of the positive cathode voltage of diode D2. The diode D2 stops conduction hence
the AGC works normally (works as simple AGC).
 Unit – V

Objectives:
➢ To introduce the concepts of FM modulation and demodulation technique

UNIT – V: Angle Modulation

Introduction to angle modulation, relation between frequency and phase modulations, Single
tone frequency modulation, narrow band FM, wide band FM, constant average power,
transmission bandwidth of FM wave, generation of FM waves: direct FM and Armstrong
method.

Outcomes:
Students will be able to
➢ Relate phase modulation and frequency modulation
➢ Explain single tone frequency modulation
➢ Identify the difference between NBFM & WBFM
➢ Classify methods of generation of FM.
 ANGLE MODULATION
Angle modulation: there are two types of Angle modulation techniques namely
• Phase modulation
• Frequency modulation
Phase modulation (PM) is that of angle modulation in which the angular argument θ (t) is
varied linearly with the message signal m(t), as shown by
θ (t) =2πfct+kpm(t)
where 2πfct represents the angle of the unmodulated carrier
kp represents the phase sensitivity of the modulator(radians/volt) The phase modulated wave
s(t)=Accos[2πfct+kpm(t)].

Frequency modulation (FM) is that of angle modulation in which the instantaneous frequency
fi(t) is varied linearly with the message signal m(t), as shown by
fi(t) =fc+kf m(t)
Where fc represents the frequency of the unmodulated carrier
kf represents the frequency sensitivity of the modulator(Hz/volt) .
The frequency modulated wave s (t)=Accos[2πfct+2πkf otm(t)dt]

FM wave can be generated by first integrating m(t) and then using the result as the input
to a phase modulator
PM wave can be generated by first differentiating m(t) and then using the result as the
input to a frequency modulator. Frequency modulation is a Non-linear modulation process.

This means that an FM signal can be generated by first integrating m(t) and the using
the result as the input to a phase modulator (see Fig 2.1).

Relation between Phase modulation and Frequency modulation:

Fig.1 : Illustrating the relationship between frequency modulation and phase modulation.

Single tone FM:

Consider m(t)=Amcos(2πfmt)
The instantaneous frequency of the resulting FM wave fi(t) =fc+kf Amcos(2πfmt)
= fc+f cos(2πfmt)
where f = kf Am is called as frequency deviation
θ (t) =2π  fi(t)dt =2πfct+f/fm sin(2πfmt) = 2πfct+β sin(2πfmt)
Where β= f/fm= modulation index of the FM wave

• When β<<1 radian then it is called as narrowband FM consisting essentially of a

carrier, an upper side-frequency component, and a lower side-frequency component.
• When β>>1 radian then it is called as wideband FM which contains a carrier and an
infinite number of side-frequency components located symmetrically around the
carrier.
Narrowband Frequency Modulation:
Consider an FM Signal resulting from the use of a sinusoidal modulating signal, expanding the
relation we get,
S(t)= Ac cos(2πfct) cos[βsin(2πfmt)]-Ac sin(2πfct) sin[βsin(2πfmt)]
Assuming that the modulation index β is small compared to 1 radian , we may use the following
approximation s
cos[βsin(2πfmt)] ≅1
sin[βsin(2πfmt)] ≅ βsin(2πfmt)

Hence, equation simplified to

S(t)= Ac cos(2πfct) -Ac sin(2πfct) [ βsin(2πfmt)]

Block diagram of a method for generating a Narrowband FM Signal :-

And the above equation is further simplied to
S(t)= Ac cos(2πfct) +0.5 βAc {cos(2π(fc+fm)t - cos(2π(fc-fm)t }
And the corresponding narrowband FM signal with phasor diagram is shown
below:

Fig.: Phasor diagram of Narrowband FM Signal

Frequency Spectrum Analysis of sinusoidal FM wave

The FM modulated signal in the time domain is given by

• From this equation it can be seen that the frequency spectrum of an FM waveform with a
sinusoidal modulating signal is a discrete frequency spectrum made up of components
spaced at frequencies of c± nm
• By analogy with AM modulation, these frequency components are called sidebands.
• We can see that the expression for s(t) is an infinite series. Therefore the frequency
spectrum of an FM signal has an infinite number of sidebands.
 • The amplitudes of the carrier and sidebands of an FM signal are given by the
corresponding Bessel functions, which are themselves functions of the modulation index

The following spectra show the effect of modulation index, , on the bandwidth of an FM signal,
and the relative amplitudes of the carrier and sidebands The envelope of an FM wave is constant,
so that the average power of such a wave dissipated in a 1-ohm resistor is also constant.

Fig. FM WAVE SIDE BANDS

WIDEBAND FM:
Parameters :
1. modulation index : Greater than 1
2. Maximum deviation: 75 kHz
3. Range of modulating frequency: 30 Hz to 15 kHz
4. Maximum modulation index: 5 to 2500
5. Bandwidth: large,about 15 times higher than BW of narrowband FM
6. Applications: entertainment broadcasting
7. Pre-emphasis and de-emphasis : is needed.
TRANSMISSION BANDWIDTH OF FM SIGNAL
In theory an FM signal contains an infinite number of side frequencies so that the band
width required to transmit such a signal is infinite in extent.

In practice, we find that FM signal is effectively limited to a finite number of significant

side frequencies.

In the case of an FM signal generated by a single tone modulating wave of frequency fm ,

The side frequencies that are separated from the carrier frequency fC by an amount
greater than the frequency deviation f decrease rapidly toward zero.

Specifically for large values of β , the bandwidth approaches, and is only slightly greater
 than the total frequency deviation

Carson’s Rule: Bandwidth is twice the sum of the maximum frequency deviation and
the modulating frequency.

BW=2(f+ fm)

The nominal BW  2f = 2 βfm.

Direct Method of FM Generation:-

In this approach a “Varactor diode” is used. A Varactor diode is a capacitor whose
capacitance changes with the applied voltage.

Therefore , if this capacitor is used in a tuned circuit of the oscillator and the message signal
is applied to it, the frequency of the tuned circuit ,and the oscillator will change in accordance
with the message signal(see diagram below).

Let the indicator in the tuned circuit be 0 L and the capacitance of the varactor diode is
given by C(t)=c0+k0m(t)
When m (t) =0, the frequency of the tuned circuit is given by:
In general if m(t) ≠ 0,

and using the approximations,

We obtain,

This is the relation for a frequency modulated signal.

Indirect method of generation of FM:

In indirect FM , the baseband signal is first integrated and then used to phase modulate a
crystal controlled oscillator. In order to minimize the distortion inherent in the phase
modulator, the maximum phase deviation or modulation index β is kept small. Thereby
resulting in a NBFM . this signal is next multiplied in frequency by means of a frequency
multiplier so as to produce the desired wideband FM Wave.

A frequency multiplier consists of a non-linear device followed by a Bandpass filter.

The instantaneous frequency in this method is fi(t)=fc+kfm(t)
For now it suffices to say that after bandpass filtering of the non-linear device’s output we
have a new FM Signal defined by

Whose instantaneous frequency is

fi(t)=nfc+nkfm(t).

We see that the non-linear processing circuit acts as a frequency multiplier. The frequency
multiplication ratio is determined by the highest power n in the input-output relation ,
characterizing the memoryless nonlinear device.

Block diagram of a frequency multiplier:

 Unit – VI

Objectives:
➢ To introduce the concepts of FM modulation and demodulation technique

UNIT – IV: Angle Modulation

Detection of FM waves: balanced frequency discriminator, FM transmitters, FM
receivers,noise in FM system, pre-emphasis and de-emphasis in FM. TDM.

Outcomes:
Students will be able to
➢ realize concept of frequency discrimination
➢ analyze FM modulation and demodulation schemes
➢ find the necessity of Pre-emphasis and De-emphasis.
➢ understand various blocks in FM transmitters and receivers

SLOPE DETECTOR AND BALANCED DETECTOR

The slope detector is the simplest type of FM detector. A schematic diagram of a slope
detector appears below:
 Fig. circuit diagram and waveforms

The operation of the slope detector is very simple. The output network of an amplifier
is tuned to a frequency that is slightly more than the carrier frequency + peak deviation. As
the input signal varies in frequency, the output signal across the LC network will vary in
amplitude because of the band pass properties of the tank circuit. The output of this amplifier
is AM, which can be detected using a diode detector.

The circuit shown in the diagram above looks very similar to the last IF amplifier and
detector of an AM receiver, and it is possible to receive NBFM on an AM receiver by
detuning the last IF transformer. If this transformer is tuned to a frequency of approximately
1 KHz above the IF frequency, the last IF amplifier will convert NBFM to AM.
In spite of its simplicity, the slope detector is rarely used because it has poor linearity.
To see why this is so, it is necessary to look at the expression for the voltage across the
primary of the tuned transformer in the sloped detector:

The voltage across the transformer's primary winding is related to the square of the
frequency. Since the frequency deviation of the FM signal is directly proportional to the
modulating signal's amplitude, the output of the slope detector will be distorted. If the
bandwidth of the FM signal is small, it is possible to approximate the response of the slope
detector by a linear function, and a slope detector could be used to demodulate an NBFM
signal
COMPARSION BETWEEN AM AND FM

In radio communication, the message signal wave (low frequency) is combined with
a carrier signal (high frequency). In this combination, one or more characteristics of the
carrier wave are varied with respect to message signal. This variation is termed as modulation
and it is needed so that message can be transmitted over long distances and no undesired
signal mixing takes place. Depending on several factors such as range, application and
budget, modulation can be casted into three types: Amplitude Modulation, Frequency
Modulation and Phase Modulation. Out of these three types, the former two are widely
known as they form a major commercially applicative part of radio communication. In this
article, we will discuss common difference between AM and FM which will enhance our
learning in terms of these two technologies.

1. Evolution: Formulated in the 1870s, AM is a relatively older modulation process

compared to FM which was found in the 1930s by Edwin Armstrong.
2. Technology: AM stands for amplitude modulation where the amplitude of the carrier is
modulated as per the message signal. The other aspects of the carrier wave such as frequency
phase etc. remain constant. On the other hand, FM means frequency modulation and in it
only frequency of the carrier wave changes while amplitude, phase etc. remain constant.
3. Frequency range of working: Amplitude modulation works between 540-1650 KHz
while FM works at 88-108MHz.
4. Power Consumption: FM based signal transmission consumes a higher amount of power
than an equivalent AM based signal transmission system.
5. AM vs FM: Signal Quality: Signal quality is a lot superior in FM than AM as
amplitude based signals are more susceptible to noise than those which use frequency.
Moreover, noise signals are difficult to filter out in AM reception whereas FM receivers
easily filter out noise using the capture effect and pre-emphasis, de-emphasis effects. In
capture effect, the receiver locks itself to catch stronger signal so that signals received are
more synced with that at the transmitting end.
In pre-emphasis, de-emphasis process, the signal is further amplified to a higher frequency at
sending end (pre-emphasis) and vice versa at receiver end (de-emphasis). These two
processes reduce down the chances of a signal to get mixed with other signals and make FM
more immune to noise than AM.
6. Fading: Fading refers to power variation during signal transmission. Due to fading, the
power with the signal received can vary significantly and reception wouldn’t be of a good
quality. Fading is more prominent in amplitude modulation as compared to frequency
modulation. That is why, AM radio channels often face the problem where sound intensity
varies while FM radio channels have constant good reception.
7. Wavelength Difference between AM and FM: AM waves work in the range of KHz
while in FM waves work in MHz range. As a result, AM waves have a higher wavelength
than the FM ones. A higher wavelength increases the range of AM signals as compared to
FM which have a limited area of coverage.
8. Bandwidth consumption: AM signals consume 30KHz of bandwidth for each while in
FM 80KHz is the bandwidth consumed by each signal. Hence, over a limited range of
bandwidth, more number of signals can be sent in AM than FM.
9. Circuit Complexity: Aforesaid, Amplitude Modulation is an older process and has a very
simple circuitry. On the other hand, frequency modulation requires a complicated circuitry
for transmission and reception of signal. The signals sent in FM are more modulated and
emphasized at the transmitter and they are thoroughly checked and corrected at the receiving
end. This is why circuitry for FM signals is very complicated.
10. Commercial Aspects: Setting up an AM based radio communication system is very
economic as there is no complicated circuitry and processes are easy to understand

FM TRANSMITTER

INDIRECT METHOD (PHASE SHIFT) OF MODULATION

The part of the Armstrong FM transmitter (Armstrong phase modulator) which is expressed
in dotted lines describes the principle of operation of an Armstrong phase modulator. It
should be noted, first that the output signal from the carrier oscillator is supplied to circuits
that perform the task of modulating the carrier signal. The oscillator does not change
frequency, as is the case of direct FM. These points out the major advantage of phase
modulation (PM), or indirect FM, over direct FM. That is the phase modulator is crystal
controlled for frequency.

Fig. Armstrong Modulator

The crystal-controlled carrier oscillator signal is directed to two circuits in parallel. This
signal (usually a sine wave) is established as the reference past carrier signal and is assigned a
value 0°.The balanced modulator is an amplitude modulator used to form an envelope of
double sidebands and to suppress the carrier signal (DSSC). This requires two input signals,
the carrier signal and the modulating message signal. The output of the modulator is
connected to the adder circuit; here the 90° phase-delayed carriers signal will be added back
to replace the suppressed carrier. The act of delaying the carrier phase by 90° does not change
the carrier frequency or its wave shape. This signal identified as the 90° carrier signal.
 Fig. Phasor diagram of Armstrong Modulator

The carrier frequency change at the adder output is a function of the output phase shift and is
found by. fc = Δθfs (in hertz) When θ is the phase change in radians and fs is the lowest
audio modulating frequency. In most FM radio bands, the lowest audio frequency is 50Hz.
Therefore, the carrier frequency change at the adder output is 0.6125 x 50Hz = ± 30Hz since
10% AM represents the upper limit of carrier voltage change, then ± 30Hz is the maximum
deviation from the modulator for PM.
The 90° phase shift network does not change the signal frequency because the
components and resulting phase change are constant with time. However, the phase of the
adder output voltage is in a continual state of change brought about by the cyclical variations
of the message signal, and during the time of a phase change, there will also be a frequency
change.

USING REACTANCE MODULATOR DIRECT METHOD

Fig. Reactance Modulator

The FM transmitter has three basic sections.

1. The exciter section contains the carrier oscillator, reactance modulator and the buffer
amplifier.
2. The frequency multiplier section, which features several frequency multipliers.
3. The power output section, which includes a low level power amplifier, the final power
amplifier, and the impedance matching network to properly load the power section with the
antenna impedance.
The essential function of each circuit in the FM transmitter may be described as follows.

The Exciter
1. The function of the carrier oscillator is to generate a stable sine wave signal at the rest
frequency, when no modulation is applied. It must be able to linearly change
frequency when fully modulated, with no measurable change in amplitude.
2. The buffer amplifier acts as a constant high-impedance load on the oscillator to help
stabilize the oscillator frequency. The buffer amplifier may have a small gain.
3. The modulator acts to change the carrier oscillator frequency by application of the
message signal. The positive peak of the message signal generally lowers the
oscillator's frequency to a point below the rest frequency, and the negative message
peak raises the oscillator frequency to a value above the rest frequency. The greater
the peak-to-peak message signal, the larger the oscillator deviation.
Frequency multiplier
Frequency multipliers are tuned-input, tuned-output RF amplifiers in which the output
resonant circuit is tuned to a multiple of the input frequency. Common frequency multipliers
are 2x, 3x and 4x multiplication. A 5x Frequency multiplier is sometimes seen, but its
extreme low efficiency forbids widespread usage. Note that multiplication is by whole
numbers only. There can not a 1.5x multiplier, for instance.
Power Output Section
The final power section develops the carrier power, to be transmitted and often has a low-
power amplifier driven the final power amplifier. The impedance matching network is the
same as for the AM transmitter and matches the antenna impedance to the correct load on the
final over amplifier.

FM RECEIVER
 Fig.: FM Receiver.

See filters, mixers, frequency changers, am modulation and amplifiers. The f.m. band covers
88-108 MHz. There are signals from many radio transmitters in this band inducing signal
voltages in the aerial. The rf amplifier selects and amplifies the desired station from the
many. It is adjustable so that the selection frequency can be altered. This is called TUNING.
In cheaper receivers the tuning is fixed and the tuning filter is wide enough to pass all signals
in the f.m. band. The selected frequency is applied to the mixer. The output of an oscillator is
also applied to the mixer. The mixer and oscillator form a FREQUENCY CHANGER circuit.
The output from the mixer is the intermediate frequency (i.f.) The [Link] a fixed frequency of
10.7 MHz. No matter what the frequency of the selected radio station is, the [Link] always
10.7 MHz. The [Link] is fed into the [Link]. The advantage of the I.F. amplifier is
that its frequency and bandwidth are fixed, no matter what the frequency of the incoming
signal is. This makes the design and operation of the amplifier much simpler. The amplified
[Link] is fed to the demodulator. This circuit recovers the audio signal and discards the r.f.
carrier. Some of the audio is fed back to the oscillator as an AUTOMATIC FREQUENCY
CONTROL voltage. This ensures that the oscillator frequency is stable in spite of
temperature changes. The audio signal voltage is increased in amplitude by a voltage
amplifier. The power level is increased sufficiently to drive the loudspeaker by the power
amplifier.

Pre-emphasis and De-emphasis

There is another way in which the SNR of an FM system may be increased . We saw in the
previous subsection that the PSD of the noise at the detector output has a square-law
dependence on frequency. On the other hand, the PSD of a typical message source is not
uniform, and typically rolls off at around 6 dB per decade. We note that at high frequencies
the relative message power is quite low, whereas the noise power is quite high (and is rapidly
increasing). It is possible that this situation could be improved by reducing the bandwidth of
the transmitted message (and the corresponding cutoff frequency of the baseband LPF in the
receiver), thus rejecting a large amount of the out-of-band noise. In practice, however, the
distortion introduced by low-pass filtering the message signal is unsatisfactory

Figure: Pre-emphasis and Deemphasis in FM System

Figure: PSD of: (a) noise at the output of FM receiver, (b) a typical message signal
A better solution is obtained by using the pre-emphasis and de-emphasis stages shown in Fig.
. The intention of this scheme is that (f) is used to artificially emphasize the high frequency
components of the message prior to modulation, and hence, before noise is introduced. This
serves to effectively equalize the low- and high-frequency portions of the message PSD such
that the message more fully utilizes the bandwidth available to it. At the receiver, (f) performs
the inverse operation by de-emphasizing the high frequency components, thereby restoring
the original PSD of the message signal.

Simple circuits that perform pre- and de-emphasis are shown in Fig. , along with their
respective frequency responses. Haykin shows that these circuits can improve the output SNR
by around 13 dB. In closing this section, we also note that Dolby noise reduction uses an
analogous pre-emphasis technique to reduce the effects of noise.
Noise in FM:-
The model of an FM receiver[1] is shown in Fig., where s(t) is the FM signal , and w(t) is
white Gaussian noise with power spectral density No/2. The bandpass filter is used to remove
any signals outside the bandwidth of fc+BT/2, and thus, the predetection noise at the receiver
is bandpass with a bandwidth of BT. Since an FM signal has a constant envelope, the limiter
is used to remove any amplitude variations. The discriminator is a device whose output is
proportional to the deviation in the instantaneous frequency (i.e., it recovers the message
signal), and the final baseband low-pass filter has a bandwidth of W and thus passes the
message signal and removes out-of-band noise.
The predetection signal is
x(t) = Acos[2πfcdt+2πkf ∫ m(T)dT]+nc(t)]cos(2πfct)-ns(t)sin(2πfct)
First, let us consider the signal power at the receiver output. When the predetection SNR is
high, it can be shown that the noise does not affect the power of the signal at the output.c
Thus,
ignoring the noise, the instantaneous frequency of the input signal is
fi = fc +kfm(t)
and the output of the discriminator (which is designed to simply return the deviation of the
instantaneous frequency away from the carrier frequency) is Kfm(t)
The output signal power is therefore
Ps = Kf2P
where P is the average power of the message signal.
Now, to calculate the noise power at the receiver output, it turns out that for high predetection
SNR the noise output is approximately independent of the message signal. In this case, we
only
have the carrier and noise signals present. Thus,

The phasor diagram of this is shown in Fig. From this diagram, we see that the instantaneous
phase is
Figure: Phasor diagram of the FM carrier and noise signals.
For large carrier power, then most of the time

On solving the noise equations finally, we have that at the output the SNR is

Since the transmitted power of an FM waveform is PT=A2/2

One should note that whereas equation suggests that output SNR for an FM system
can be increased arbitrarily by increasing β while keeping the signal power fixed, inspection
of equation shows this not to be strictly true. The reason is that if β increases too far, the
condition of equation that we are above threshold may no longer be true, meaning that
equation no longer provides an expression for the true SNR.

Time Division Multiplexing:-

TDM is widely used in digital communications, for example in the form of pulse code
modulation in digital telephony. In TDM, each message signal occupies the channel (e.g. a
transmission line) for a short period of time. The principle is illustrated below:
Switches SW1 and SW2 rotate in synchronism, and in effect sample each message input in a
sequence m1(t), m2(t), m3(t), m4(t), m5(t), m1(t), m2(t),…

The sampled value (usually in digital form) is transmitted and recovered at the ‘far end’ to
produce output m1(t)…m5(t). For ease of illustration consider such a system with 3 messages,
m1(t), m2(t) and m3(t), each a different DC level as shown below.

m1(t) V1

0 t

m2(t) V2

0 t
m3(t) V3

0 t

SW1
‘Sample’

t
Position 1 2 3 1 2 3
 V3
V2
V1
t
m1(t) m2(t) m3(t) m1(t) m2(t) m3(t) m1(t)

Channel
Time
Slots

1 2 3 1 2 3 1

Time slot

In this illustration the samples are shown as levels, i.e. V1, V2 or V3. Normally, these voltages
would be converted to a binary code before transmission as discussed below.

Note that the channel is divided into time slots and in this example, 3 messages are time-
division multiplexed on to the channel. The sampling process requires that the message
signals are a sampled at a rate fs  2B, where fs is the sample rate, samples per second, and B
is the maximum frequency in the message signal, m(t) (i.e. Sampling Theorem applies). This
sampling process effectively produces a pulse train, which requires a bandwidth much greater
than B.

Thus in TDM, the message signals occupy a wide bandwidth for short intervals of time. In
the illustration above, the signals are shown as PAM (Pulse Amplitude Modulation) signals.
In practice these are normally converted to digital signals before time division multiplexing.