Pre-Emphasis, De-Emphasis,

Pulse Modulation
COMMUNICATION SYSTEM (TC-306), SPRING, 2025
Pre-Emphasis
Pre-Emphasis
De-Emphasis
Phase Locked Loop
Phase Locked Loop
Phase Locked Loop
Phase Locked Loop
Phase Locked Loop
Phase Locked Loop

Figure 5-5
Phase Locked Loop
Pulse Modulation
 In pulse modulation, a series of on-off pulses serve as the carrier wave that is subsequently modulated.
 The characteristics of pulse i.e, pulse amplitude, pulse width and pulse position are changed in
accordance to random message signal.
Pulse Amplitude Modulation (PAM)
Pulse amplitude modulation is the basic form of pulse modulation. In this modulation, the signal is
sampled at regular intervals and each sample is made proportional to the amplitude of the modulating
signal. The pulse sampling time and pulse on-time remains constant.
Pulse Amplitude Modulation (PAM)
There are three types of sampling techniques:
1. Impulse sampling.
2. Natural sampling.
3. Flat Top sampling.
1. Impulse/Ideal Sampling: Impulse sampling can be performed by multiplying input signal x(t) with
impulse train of period 'T'. Here, the amplitude of impulse changes with respect to amplitude
of input signal x(t). The output of sampler is given by
Pulse Amplitude Modulation (PAM)
Limitations of Impulse Sampling:
1. It is ideal sampling or impulse sampling so cannot be used practically because pulse width is zero and
the generation of impulse train is not possible practically.
2. Noise interference is high.
3. Less power is used to transmit such sampled signal so it cannot be transmitted over long distance.
Natural Sampling:
Natural sampling is similar to impulse sampling, except the impulse train is replaced by pulse train of
period T.
Pulse Amplitude Modulation (PAM)
Natural Sampling:
Pulse Amplitude Modulation (PAM)
Flat-Top Sampling:
Pulse Amplitude Modulation (PAM)
Flat-Top Sampling: Its demerit is Aperture effect. To eliminate aperture effect, pulse width is minimized
and equalizer is used.
Pulse Width Modulation
Pulse Width Modulation (PWM) or Pulse Duration Modulation (PDM) or Pulse Time Modulation (PTM) is
an analog modulating scheme in which the duration or width or time of the pulse carrier varies proportional to the
instantaneous amplitude of the message signal.
The width of the pulse varies in this method, but the amplitude of the signal remains constant. Amplitude limiters
are used to make the amplitude of the signal constant.
 Pulse Width Modulation
There are three variations of PWM as shown in figure. They are:
•Trailing Edge PWM: The leading edge of the pulse being constant, the trailing edge varies according to the message
signal.
•Leading Edge PWM: The trailing edge of the pulse being constant, the leading edge varies according to the message
signal.
•Leading/Trailing Edge PWM: The center of the pulse being constant, the leading edge and the trailing edge varies
according to the message signal.

Generation of PWM:
Pulse Width Modulation
•A triangular function generator generates a triangle signal of frequency fs, and this signal in this case is used as a
sampling signal which is applied to the non-inverting terminal of a comparator.
•The modulating signal x (t) is applied to the non-inverting terminal of the same comparator and added with the
triangular signal shown in the figure.
• The reference voltage is applied to the inverting terminal of a comparator.
•The comparator output will remain high as long as the adder output is higher than that of the reference voltage
signal otherwise the comparator output will be low.
•This gives rise to a PWM signal at the comparator output as shown in figure .
Pulse Position Modulation
 In PPM the amplitude and width of the pulses is kept constant
but the position of each pulse is varied in accordance with
the amplitudes of the sampled values of the modulating signal.
 The position of the pulses is changed with respect to the
Position of reference pulses
 The PPM pulses can be divided from the PWM pulses.
With increase in the modulating voltage the PPM pulses shift
further with respect to reference.
 The vertical treated dotted lines are reference lines to
measure the shift in position of PPM pulses.
 The PPM pulses marked 1, 2 and 3 in fig.1 go away from their
respective reference lines. This is corresponding to increase in
the modulating signal amplitude.
 Then, as the modulating voltage decreases, the PPM pulses 4, 5, 6, 7 come progressively closer to their
respective reference lines.
Pulse Position Modulation
 The PPM signal can be generated from PWM signal
as shown in fig.2.
 P.W.M is the Pulse width modulation that generates
pulses of varying widths.
 On the contrary, PPM is known as the pulse position modulation
which generates a pulse of fixed amplitude with respect
to specific time intervals.
 The PPM Signal can be generated from PWM signal.
The PWM pules obtained at the comparator output
are applied to a mono stable multivibrator.
 Hence corresponding to each trailing edge of PWM signal,
the mono stable output goes high. It remains high for a
fixed time decided by its own RC comparator.
 Thus as the trailing edges of the PWM signal keep shifting in
proportion with the modulating signal x(t), the PWM pulses also keep shifting as shown in the figure 3.
 All the PPM pulses have the same with and amplitude. The information is conveyed via changing the portion of
pulses.
Pulse Position Modulation
Pulse Code Modulation
The block diagram of PCM is given below which represents the basic elements of both the transmitter and the
receiver sections.
Pulse Code Modulation
The transmitter section of a Pulse Code Modulator circuit consists of Sampling, Quantizing and Encoding, which
are performed in the analog-to-digital converter section. The low pass filter prior to sampling prevents aliasing of the
message signal.
The basic operations in the receiver section are regeneration of impaired signals, decoding, and reconstruction of
the quantized pulse train.
Low Pass Filter
This filter eliminates the high frequency components present in the input analog signal which is greater than the
highest frequency of the message signal, to avoid aliasing of the message signal.
Sampler:
All real world signals are analog in nature. Analog signal is the continuously varying with respect to amplitude as well
as time. On the other hand, computers understand digital language so these analog signals must be converted into
digital signals so that computer can process the signals. Sampling is the technique which converts analog signal into
discrete signal.
 Pulse Code Modulation
Sampler: The distance between the two samples in discrete signal
is fixed, and the time variation is fixed so the amplitude is taken at
the fixed interval of time. Time axis is discretized but the amplitude
is continuous.
To reconstruct the signal, it is important to take samples as much
as possible or to take the time interval between the samples less.
Sample: It is the estimated value of signal at any instant of time.
In order to do sampling, the analog signal x(t) is multiplied with
Impulse train and the sampled or discrete signal x’(t) is obtained.
Pulse Code Modulation
Sampling Rate
To discretize the signals, the gap between the samples should be fixed. That gap can be termed as
a sampling period Ts.
Where Ts is the sampling time and
Fs is the sampling rate of sampling frequency.
Sampling frequency is the reciprocal of the sampling period. This sampling frequency, can be simply called
as Sampling rate. The sampling rate denotes the number of samples taken per second, or for a finite set of
values.
For an analog signal to be reconstructed from the digitized signal, the sampling rate should be highly
considered. The rate of sampling should be such that the data in the message signal should neither be lost
nor it should get over-lapped.
Pulse Code Modulation
 Pulse Code Modulation
For a bandlimited signal in time domain, the energy will be limited in the frequency domain.

Sampling Theorem

The sampling theorem, which is also called as Nyquist theorem, delivers the theory of sufficient sample rate in terms of bandwidth for the class of
functions that are bandlimited.

The sampling theorem states that, “a signal can be exactly reproduced if it is sampled at the rate fs which is greater than twice the maximum
frequency W.”

To understand this sampling theorem, let us consider a band-limited signal, i.e., a signal whose value is non-zero between some –W and W Hertz.

For the continuous-time signal, x(t) the band-limited signal in frequency domain, can be represented as shown in the following figure.
 Pulse Code Modulation
 For case 1, the signal can be easily reconstructed using both the ideal as well as practical low pass filter. It is known as sampling theorem when fs>2fm.

 For case 2, the signal can be reconstructed using ideal low pass filter which does not exist practically. Such as case is known as Nyquist rate when fs=2fm
 Pulse Code Modulation

We can observe from the above pattern that the over-lapping of information is done, which leads to mixing up and loss
of information. This unwanted phenomenon of over-lapping is called as Aliasing.
Aliasing
Aliasing can be referred to as “the phenomenon of a high-frequency component in the spectrum of a signal, taking on
the identity of a low-frequency component in the spectrum of its sampled version.”
The corrective measures taken to reduce the effect of Aliasing are −
•In the transmitter section of PCM, a low pass anti-aliasing filter is employed, before the sampler, to eliminate the
high frequency components, which are unwanted.
•The signal which is sampled after filtering, is sampled at a rate slightly higher than the Nyquist rate.
This choice of having the sampling rate higher than Nyquist rate, also helps in the easier design of the reconstruction
filter at the receiver.
Pulse Code Modulation
Quantizer: The quantizing of an analog signal is done by discretizing the signal with a number of
quantization levels. Quantization is representing the sampled values of the amplitude by a finite set of
levels, which means converting a continuous-amplitude sample into a discrete-amplitude signal.

Where x[KTs] is the sampled input and xq [KTs] is the quantized output. Quantizer assigns n different
levels to the amplitude of sampled output by assigning a nearest level. The number of levels are
dependent upon the number of bits the encoder can encode.
If encoder encodes 2 bits, then the 4 levels will be assigned to the amplitude of the sampled output.
Further, if the encode encodes 4 bits then the 16 levels will be assigned to the amplitude of the sampled
output. Hence, the more the no: of levels, the quantization error will be less and the signal will be
represented with more precision.
 Pulse Code Modulation

Quantization error: For any system, during its functioning, there is always a difference in the values of its input and
output. The processing of the system results in an error, which is the difference of those values.
The difference between an input value and its quantized value is called a Quantization Error.
Pulse Code Modulation
Quantization: Both sampling and quantization result in the loss of information. The quality of a Quantizer output depends upon the
number of quantization levels used. The discrete amplitudes of the quantized output are called as representation
levels or reconstruction levels. The spacing between the two adjacent representation levels is called a quantum or step-size.
As shown in the figure, if 2 bit encoder is considered, then 4 levels are
taken i.e., 10,20,50 and 70. The step size is found as 20 so 20, 40, 60 and
80 are taken as the step size between the levels. It is shown that when
the original signal has the amplitude value of 15 which is less than the
step size 20 so the assigned quantized level is 10, when the original
signal has the value of 24.5 which is greater than the step size 20 so it
has assigned the quantized level of 30, when the original signal has the
value of 42.5 which is greater than the step size 40 so it has assigned the
quantized level of 50, when the original signal has the value of 55.5
which is less than the step size 60 so it has assigned the quantized level
of 50, when the original signal has the value of 62.5, which is greater
than the step size of 60 so it has assigned the quantized level of 70, when
the original signal has the value of 69.9, which is greater than the step
size of 60 so it has assigned the quantized level of 70.
 Pulse Code Modulation
The table below shows the original signal values, quantized levels, quantization error and bit which encoder has used to represent
each quantized level.
Pulse Code Modulation
Encoder
The digitization of analog signal is done by the encoder. It designates each quantized level by a binary code. The sampling done here is the
sample-and-hold process. When the more number of quantization levels are introduced to reduce the quantization error then the burden on
encoder will occur because more number of bits will be required to encode a given signal. Hence, the transmission frequency of encoder will be
increased so it is necessary to keep the quantization levels minimum or less than the threshold levels. The output of the encoder is in terms of
binary bits so the this digital data is converted into electrical signal using line coding techniques so that the signal can pass through the
communication channel. The encoder also introduces the parity bits into the signal so that error detection and error correction can be done at the
receiver.
Regenerative Repeaters:
Now the signal is transmitted into the channel where it gets added with the noise and amplitude and phase errors will be
introduced in the signal. That signal is then fed into the series of regenerative repeaters that increases the signal strength to
compensate the signal loss and reconstruct the signal. The regenerative repeaters contain equalizer which removes amplitude
variations and phase delays. The second block in regenerative repeaters is the regenerator which is connected with the clock.
The clock is synchronized with that of the transmitter to generate the positive and zero voltage levels.
Receiver:
The reverse operation of transmitter is applied at the receiver side to get back the original signal. The signal from the channel is
fed into the regenerative circuit to amplify the strength of signal. Then the signal is fed into the decoder where the parity bits
are removed to detect and correct the errors. The decoder will convert the signal into the amplitude levels. The signal is then
fed into the reconstruction filter which is a low pass filter to reverse the operation of sampler and the original signal is received
at the destination.