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UNIT 1 - COMMUNICATION
SYLLABUS:
Basics of AM, FM and PM-block diagram, concepts of AM, FM modulator and AM, FM
demodulators – pulse modulation system- pulse amplitude modulation- sampling, quantization error.
NOTES:
Basics of AM
Theory of DSBFC:
In fig (1) observe that the carrier frequency remains same, but its amplitude varies according to
amplitude variations of the modulating signal.
Let us represent the modulating signal by em and it is given as,
em = Em sin 𝜔𝑚 t eq.1
and carrier signal can be represented by ec as,
ec = Ec sin 𝜔𝑐 t
Here
Using the above mathematical expressions for modulating and carrier signals. We can create a
new mathematical expression for the complete modulated wave. It is given as,
EAM = Ec + em
= Ec + Em sin 𝜔𝑚 t by putting em from eq.1
The instantaneous value of the amplitude modulated signal wave can be given as,
= EAM sin 𝜔𝑐 t
The ratio of maximum amplitude of modulating signal to maximum amplitude of carrier signal is called
modulation index. i.e.,
Em
Modulation index, m = eq. 4
Ec
Value of Em must be less than value of Ec to avoid any distortion in the modulated signal. Hence
maximum value of modulation index will be equal to 1 when Em = Ec. Minimum value will be zero.
When modulation index is expressed in percentage, it is also called percentage modulation.
The modulation carrier has new signals at different frequencies, called side frequencies or
sidebands. They occur above and below the carrier frequency.
i.e., fUSB = fc + fm
fLSB = fc - fm
Hence fc is carrier frequency and
fm is modulating signal frequency
fLSB is lower sideband frequency
and fuSB is upper sideband frequency
𝐸𝑚
We know that m = from eq. 4. Hence we have Em = mEc. Putting this value of Em in above eq
𝐸𝑐
we get,
eAM =( Ec + mEc sin 𝜔𝑚 𝑡) sin 𝜔𝑐 𝑡)
= Ec (1+ m sin 𝜔𝑚 𝑡) sin 𝜔𝑐 𝑡
= Ec sin 𝜔𝑐 𝑡+ mEc sin 𝜔𝑚 𝑡 sin 𝜔𝑐 𝑡 eq. 6
1 1
We know that sin(A)sin(B) = cos (A-B) - cos (A+B). Applying this result to last term
2 2
in above equation we get,
mEc mEc
eAM = Ec sin 𝜔𝑐 𝑡+ cos (𝜔𝑐 - 𝜔𝑚 )t - cos (𝜔𝑐 + 𝜔𝑚 )t eq. 7
2 2
In the above equation, the first term represents unmodulated carrier, the second term represents
lower sideband and last term represents upper sideband. Note that 𝜔𝑐 = 2𝜋𝑓𝑐 and 𝜔𝑚 = 2𝜋𝑓𝑚 . Hence
above equation can also be written as,
mEc mEc
eAM = Ec sin 2𝜋𝑓𝑐 𝑡+ cos 2𝜋 (𝑓𝑐 -𝑓𝑚 )t - cos2𝜋(𝑓𝑐 +𝑓𝑚 )t eq. 8
2 2
mEc mEc
= Ec sin 2𝜋𝑓𝑐 𝑡+ cos 2𝜋𝑓𝐿𝑆𝐵 t- cos 2𝜋𝑓𝑈𝑆𝐵 t eq. 9
2 2
From this equation we can prepare the frequency spectrum of AM wave as shown below in fig (2).
Ec
mEc mEc
2 2
fm fm
frequency
fc - fm fc fc + fm
fLSB fUSB
Fig (2) Frequency domain representation of AM wave
This contains full carrier and both the side bands, hence it is also called Double Sideband Full
Carrier (DSBFC) System. We know that bandwidth of the signal can be obtained by taking the
difference between highest and lowest frequencies. From above figure we can obtain bandwidth of AM
wave as,
BW = fUSB - fLSB
= 2fm eq. 10
Thus the bandwidth of AM signal is twice of the maximum frequency of modulating signal.
Fig (3) shows the AM waveform. This is also time domain representation of AM signal.
Fig (3)
It is clear from the above signal that the modulating signal rides upon the carrier signal. From
above figure we can write,
Frequency Modulation
Frequency modulation and phase modulation are also called as Angle Modulation. In both the
methods, amplitude of the carrier remains fixed and timing parameters such as frequency or phase is
modulated. Hence these techniques are also called Angle Modulation. In frequency modulation (FM),
the frequency of the carrier is varied according to amplitude variations in the modulating signal. But the
amplitude of the frequency modulated signal remains constant. The frequency modulated carrier by
sinusoidal modulation is shown in fig (4).
When the modulating signal has zero amplitude, then the carrier has frequency of 𝜔𝑐 or 𝑓𝑐 . As
the amplitude of the modulating signal increases, the frequency of the carrier increases. Similarly, as the
amplitude of the modulating signal is decreased, the frequency of carrier is also decreased.
Modulation index of FM is defined as the ratio of maximum frequency deviation (𝛿) to the
modulating frequency (fm) i.e.,
The maximum frequency deviation is the shift from center frequency fc when the amplitude of
modulating signal is maximum. The modulation index corresponding to maximum modulating
frequency is called Deviation ratio. i.e.,
Phase Modulation
Fig (5)
In this way, phase modulation produces frequency modulation. Since the amount of phase shift
is variable, the effect is as if the carrier frequency is varied. Thus phase modulation gives rise to
frequency modulation; therefore, many times phase modulation is often referred to as indirect method
of frequency modulation.
In the Fig. 5.32, the unmodulated carrier and the audio modulating voltage are shown separately.
If we combine them 90° out of phase, as shown in the Fig. 5.33 then a resultant vector is
obtained which is displaced from the original position of the unmodulated carrier by some angle.
As the audio signal goes through one complete cycle, its amplitude will vary and, with it, the angular
shift of the resultant carrier vector from the position it would occupy if no modulation voltage were
present. This is illustrated in the Fig (6). Note that the resultant vector [which is the phase modulated
wavcj first occupies a position to one side of the unmodulated carrier; then it moves to the other side. In
this way it fluctuates back and forth. From the shifting an equivalent FM arises.
Fig (6)
On the negative half of the modulating cycle, the resultant wave is shifted behind the position of
the unmodulated carrier. In effect this means a decrease in frequency. On the positive half, the resultant
is shifted to the opposite side, indicating that the phase of the carrier has advanced. This is equivalent to
an increase in frequency.
The above vector diagrams can be drawn only if the frequencies of the two vectors are nearly
equal. But this is not possible if audio signal is directly used, since carrier frequency is very much larger
than the audio frequency. Hence, to have frequencies near each other, first audio signal is made to
amplitude-modulate the carrier. From this AM., two sidebands are obtained which contain the original
intelligence of audio. Carrier is suppressed as it does not contain any information. Then two sidebands
are shifted in 900 and combined with carrier to obtain phase modulation. This is “Armstrong method”.
AM modulator
Fig (7)
AM Detector Circuits
The detector or demodulator obtains the original modulating signal from the IF signal.
Fig (8)
The most commonly used AM detector is simple diode detector as shown in fig (8). The AM
signal at fixed IF is applied to the transformer primary. The signal at secondary is half wave rectified by
diode D. This diode is the detector diode. The resistance R is load resistance to rectifier and C is the
filler capacitor. In the positive half cycle of AM signal, diode conducts and current flows through R,
whereas in negative half cycle, the diode is reverse biased and no current flows. Therefore only positive
half of the AM wave appears across resistance R as shown in fig (9b). The capacitor across R provides
low impedance at the carrier frequency and much higher impedance at the modulating frequency.
Therefore capacitor reconstructs the original modulating signal as shown in fig (9c) and high frequency
carrier is removed.
This is the distortion occurs in the output of diode detector because of unequal ac and dc load
impedance of the diode. The modulation index is defined as Em / Ec.
Fig (9)
Here 𝑍𝑚 is audio diode load impedance and Rc is the dc diode resistance. The audio load
resistance of the diode is smaller than the dc resistance. Hence the AF current Im is larger, in proportion
to dc current. This makes the modulation index in the demodulated wave relatively higher than that of
modulated wave applied at the detector input. This introduces the distortion due to over modulation in
the detected signal for modulation index near 100%. This is illustrated in fig (9). In the figure observe
that the negative peak of the detected signal takes place because of over modulation effect taking place
in detector.
Fig (10)
FM Modulators
We know that in FM, the frequency of the carrier is varied according to amplitude changes in
the modulating signal. The carrier frequency is generated by LC oscillators. The carrier frequency can be
changed by varying either the inductance or capacitance of the tank circuit. The devices like FET, BJT
and varactor diodes have the property that their reactance can be varied by varying the voltage across
them. Such devices can be used with LC tank circuits to vary the overall reactance. This reactance can be
Inductive or capacitive. The change in reactance changes the frequency of the oscillator.
Direct FM :
In this type of angle modulation, the frequency of the carrier is varied directly by the modulating
signal. This means, an instantaneous frequency deviation is directly proportional to amplitude of the
modulating signal.
Indirect FM:
In this type of angle modulation FM is obtained by phase modulation of the carrier.
Instantaneous phase of the carrier is directly proportional to the amplitude of the modulating signal.
Direct FM
Direct FM can be obtained by using FET and varactor diode.
Fig (11)
The capacitance of the varactor diode depends upon the fixed bias set by R1 and R2 and the AF
modulating signal. Either R1 or R2 is made variable so that the center carrier frequency can be adjusted
over a narrow range. The Radio Frequency Choke (RFC) has high reactance at the carrier frequency to
prevent the carrier signal from getting into the modulating signal circuits. At positive going modulating
signal adds to the reverse bias applied to the varactor diode D, which decreases its capacitance arid
increases the carrier frequency. A negative going modulating signal subtracts from the bias, increasing
the capacitance, which decreases the carrier frequency.
The frequency of the LC oscillator changes due to temperature effects. Hence crystals are used
In FM generators to provide frequency stability.
Indirect FM
Fig (12)
Fig (12) shows the circuit diagram of indirect FM generator. It consists of the varactor diode D1
in series with tuned L1 R1 network. The complete series and parallel network is in series with crystal
oscillator. The modulating signal is applied to varactor diode. The capacitance of varactor diode is
changed by modulating signal. This changes phase angle of the complete network. This creates phase
shift in the carrier signal from crystal oscillator. The instantaneous phase shift is directly proportional to
instantaneous amplitude of modulating signal.
Advantage:
The crystal oscillator is isolated from modulator. Hence frequency stability is more.
Disadvantages:
Capacitance versus voltage characteristic of varactor diode is nonlinear. This results in distortion in
the modulated waveform
Amplitude of modulating signal should be kept small to avoid distortion.
Armstrong method:
Basic Principle:
Narrowband FM (NBFM) signal is generated using phase modulation method. Then the NBFM
signal is converted to wideband FM (WBFM).
Block diagram: Fig (13) shows the block diagram of WBFM generation.
Fig (13)
Explanation:
When the signal is integrated and then phase modulated, the PM signal is actually FM signal.
This is because of integration of m(t). Similarly In above block diagram m(f) is integrated. Then
integrated signal is phase modulated. Hence the output of narrowband phase modulator is narrowband
FM.
The narrowband FM is then converted to wideband FM with the help of frequency multiplier.
The frequency multiplier increases the frequency deviation as well as carrier frequency. It is called
indirect method because FM is generated from PM.
Advantages:
Fig (14) shows the block diagram of Armstrong method which generates FM output based on
the indirect method just discussed.
Fig (14)
The crystal oscillator generates the carrier frequency f. This is highly stable frequency source.
The modulating signal is amplified and given to balanced modulator. The balance modulator generates
DSB AM signal at carrier frequency f. This DSB signal is phase shifted by 9(Y’ in the phase shifter. This
phase shifted AM signal is added (vector addition) with the carrier signal in the combining network. The
combining network produces the FM signal at its output.
Observe that AM signal is having frequency f with amplitude variations. Hence resultant vector
addition is phase modulated which is basically FM signal. The buffer isolates the crystal source from
combining network so that its stability is not disturbed.
FM Demodulators
The detection of FM is totally different compared to AM. The FM detector should be able to
produce the signal whose amplitude is proportional to the deviation in the frequency of FM signal. Thus
the job of FM detector is almost similar to frequency to voltage converter; here we will discuss these
types of FM detectors. Slope detectors, phase discriminator and ratio detector.
Fig (15) shows the circuit of balanced slope detector. It consists of two identical circuits
connected back to back. The FM signal is applied to the tuned LC circuit. Two tuned LC circuits are
connected in series. The inductance of this secondary tuned LC circuit is coupled with the inductance of
the primary (or input side) LC circuit Thus it forms a tuned transformer.
In Fig. 25.1, the upper tuned circuit is shown as T1 and lower tuned circuit is shown as T2. The
input side LC circuit is tuned to carrier frequency. T1 is tuned to f + 𝛿f, which represents highest
frequency. And lower LC circuit T2 is tuned to f - 𝛿f, which represents the minimum frequency of FM
signal. The input FM signal is coupled to T1 and T2 1800 out of phase. The secondary side tuned circuits
(T1 and T2) are connected to diodes D1 and D2 with RC loads. The total output Vout, is equal to
difference between V01 and V02, since they subtract (see Fig (15)). Fig (16) shows the characteristic of
the balanced slope detector. It shows V0, with respect to input frequency.
Fig (15)
For the other frequencies of input, the output (Vout) is produced according to the characteristic
Figexample
shown in Fig (16). For (16) if input frequency tries to increase above f then V01 will be greater than
V02 and net output Vout will be positive. it is desirable that the characteristics shown in Fig (16) should be
linear between f - 𝛿 f and f + 𝛿 f, then only proper detection will take place. The Linearity of the
characteristic depends upon alignment of tuning circuits and coupling characteristics of the tuned coils.
Fig (17)
PAM sampling occurs when the finite width pulses follow the modulating signal waveform. The
result is a frequency spectrum containing the modulating baseband along with a pair of sidebands
around each harmonic frequency of the pulse sampling Frequency.
Advantages of PAM
1. PAM waveform has pulses with varying amplitude and therefore power required to transmit them is
not constant.
2. This requires that the transmitter must be able to handle the power required to transmit pulse having
maximum amplitude.
Disadvantages of PAM
1. The bandwidth needed for transmission of PAM signal is very very large compared to its maximum
frequency content.
2. The amplitude of PAM pulses varies according to modulating signal. Thereafter interference of
noise is maximum for the PAM signal and this noise cannot be removed easily.
3. Since amplitude of PAM signal varies, this also varies the peak power required by the transmitter
with modulating signal.
Sampling Process
In a sampling process a continuous time signal is converted to an equivalent discrete tune signal.
The fig (18) shows how this conversion can be done. As shown in the Fig(18), switch position is
controlled by the sampling signal. The sampling signal is a periodic train of pulses of unit amplitude and
of period Ts. The time Ts is known as sampling time and during this time switch is closed so that
sampled signal is equal to the input signal. During remaining time switch is open and no input signal
appear at the output.
Fig (18)
Sampling Theorem
The sampling theorem states that, if a continuous signal f(t) has frequency/ frequencies in its
spectrum with fm is the highest frequency, then it is possible to convey all the information using
sampled signal with 2fm or more equally spaced samples per second.
Nyquist Criteria
Nyquist criteria deduces the minimum sampling rate. The nyquist rate is defined as the minimum
sampling rate required to represent complete information about continuous signal f(t) in its sampled
form, f..(t). Therefore, according to sampling theorem, the nyquist rate is
fs min = 2 fm
The maximum interval of sampling can be given as
1 1
Ts max = =
fs min 2fm
It is called Nyquist Interval or Nyquist Criteria. When Ts = 1/2fm, this amounts to 2fm
samples per second. This is called Nyquist rate of sampling and 1/Ts = fs = 2fm is called Nyquist
frequency. In simple words, it means that the signal must be sampled at least twice during each period of
cycle of its highest frequency component.
The minimum sampling frequency f equal to 2fm cannot be achieved in practice because of the
difficulty in realizing ideal filters. Practically we must use the sampling frequency which is more than
twice the maximum frequency in the baseband waveform. How much more is a matter that depends
upon the low-pass filter characteristic and how faithfully the baseband waveform must be reproduced.
Fig ()
The samples are taken at regular interval of time. Each sample is a pulse, whose amplitude is
determined by the amplitude of the variable at the instant of time at which the sample is taken. If
enough samples are taken, a reasonable approximation of the signal being sampled can be constructed at
the receiving end. This is known as “Pulse Amplitude Modulation” [PAM].