SAMPLING THEOREM

Statement: A continuous time signal can be represented in its samples and can be
recovered back when sampling frequency fs is greater than or equal to the twice
the highest frequency component of message signal. i. e.

fs≥2fm

Proof: Consider a continuous time signal x(t). The spectrum of x(t) is a band
limited to fm Hz i.e. the spectrum of x(t) is zero for |ω|>ωm.

Sampling of input signal x(t) can be obtained by multiplying x(t) with an impulse
train δ(t) of period Ts. The output of multiplier is a discrete signal called sampled
signal which is represented with y(t) in the following diagrams:

Here, you can observe that the sampled signal takes the period of impulse. The
process of sampling can be explained by the following mathematical expression:

Sampled signal
y(t)=x(t).δ(t)......(1)
The trigonometric Fourier series representation of δ(t) is given by

Where
Substitute above values in equation 2.

Substitute δ(t) in equation 1.

Take Fourier transform on both sides.

To reconstruct x(t), you must recover input signal spectrum X(ω) from sampled
signal spectrum Y(ω), which is possible when there is no overlapping between the
cycles of Y(ω).

Possibility of sampled frequency spectrum with different conditions is given by

the following diagrams:
Aliasing Effect:

The overlapped region in case of under sampling represents aliasing effect, which
can be removed by

 considering fs >2fm

 By using anti-aliasing filters.

Sampling Techniques:

Basically, there are three types of sampling techniques such as:

(i) Instantaneous sampling

(ii) Natural sampling

(iii) Flat top sampling

Out of these three, instantaneous sampling is called ideal sampling whereas

natural sampling and flat-top sampling are called practical sampling methods.

Now, let us discuss three different types of sampling techniques in detail.

(i) Instantaneous sampling:

In this type of sampling, the sampling function is a train of impulses. Fig.1(b)

shows this sampling function.

Fig.1 : (a) Baseband signal, (b) impulse train, (c) functional diagram of a switching
sampler, (d) sampled signal

x(t) is the input signal (i.e., signal to be sampled) as shown in fig.1(a).

Fig.1(c) shows a circuit to produce instantaneous or ideal sampling. This circuit is

known as the switching sampler.

The working principle of this circuit is quite easy. The circuit simply consists of a
switch. Now if we assume that the closing time ‘t’ of the switch approaches zero,
then the output g(t) of this circuit will contain only instantaneous value of the
input signal x(t).
Since the width of the pulse approaches zero, the instantaneous sampling gives a
train of impulses of height equal to the instantaneous value of the input signal x(t)
at the sampling instant.

We know that the train of impulses may be represented as,

This is known as sampling function and its waveform is shown in fig.1(b).

The sampled signal g(t) is expressed as the multiplication of x(t) ans δTs(t). Thus,

Or

The Fourier transform of the ideally sampled signal given by above equation may
be expressed as,

Natural Sampling:

As we have already discussed, the instantaneous sampling results in the samples

whose width τ approaches zero. Due to this, the power content in the
instantaneously sampled pulse is negligible. Thus, this method is not suitable for
transmission purpose.

Natural sampling is a practical method. In this type of sampling, the pulse has a
finite width equal to τ.

Let us consider an analog continuous-time signal x(t) to be sampled at the rate of

fs Hertz.
Here it is assumed that fs is higher than Nyquist rate such that sampling theorem is
satisfied.
Again, let us consider a sampling function c(t) which is a train of periodic pulses of
width τ and frequency equal to fs Hz.
Fig.2 shows a functional diagram of a natural sampler.

Fig.2 : A functional diagram of a natural sampler

With the help of this natural sampler, a sampled signal g(t) is obtained by
multiplication of sampling function c(t) and input signal x(t).

Now, according to fig.2, we have when c(t) goes high, the switch ‘S’ is closed.
Therefore,

g(t) = x(t) when c(t) = A

g(t) = 0 when c(t) = 0

where A is the amplitude of c(t).

The waveforms of signals x(t), c(t) and g(t) have been illustrated in fig.3(a), (b) and
(c) respectively.
Fig.3 : (a) Continuous time signal x(t), (b) Sampling function waveform i.e.,
periodic pulse train, (c) Naturally sampled signal waveform g(t)

Now, the sampled signal g(t) may also be described mathematically as

g(t) = c(t) . x(t)

Here, c(t) is the periodic train of pulse of width t and frequency fs.
We know that the exponential Fourier series for any periodic waveform is
expressed as,

Also, for the periodic pulse train of c(t), we have,

So, we have

Now, it may be noted that since c(t) is a rectangular pulse train, therefore Cn for
this waveform will be expressed as,

here T = pulse width = τ

and fn = harmonic frequency

But here, fn = nfs

or,

Hence,

Therefore, the Fourier series representation for c(t) will be given as,
Now, substituting the value of c(t) in the equation of g(t), we get,

This is required time-domain representation for naturally sampled signal g(t).

Now, to get the frequency-domain representation of the naturally sampled signal

g(t), let us take its Fourier transform as,

Recall the frequency-shifting property of Fourier transform which states that

Therefore,

Now, since fn = nfs = harmonic frequency

Therefore,

Hence, we write Spectrum of naturally sampled signal:

This equation shows that the spectra of x(t) i.e., X(f) are periodic in f s and are
weighed by the sinc function.
Fig.4 illustrates some arbitrary spectra for x(t) and corresponding spectrum G(f).
Fig.4 :(a) Spectrum of continuous-time signal x(t), (b) Spectrum of naturally
sampled signal

Flat Top Sampling or Rectangular Pulse Sampling

Flat top sampling like natural sampling is also a practically possible sampling
method. But natural sampling is little complex whereas flat top sampling is quite
easy.

In flat-top sampling or rectangular pulse sampling, the top of the samples remains
constant and is equal to the instantaneous value of the baseband signal x(t) at the
start of sampling.

The duration or width of each sample is τ and sampling rate is equal to fs = 1 / Ts.
Fig.5(a) shows the functional diagram of a sample and hold circuit which is used to
generate the flat top samples.

Fig.5 :(a) A sample and hold circuit to generate flat top samples (b) A general
waveform of flat top sampling
Fig. 5(b) shows the general waveform of flat top samples. From fig.5(b), it may be
noted that only starting edge of the pulse represents instantaneous value of the
baseband signal x(t).

Also the flat top pulse of g(t) is mathematically equivalent to the convolution of
instantaneous sample and a pulse h(t) as depicted in fig.6.

Fig.6 : Convolution of any function with delta function is equal to that function

This means that the width of the pulse in g(t) is determined by the width of h(t)
and the sampling instant is determined by delta function.

In fig. 5(b), the starting edge of the pulse represents the point where baseband
signal is sampled and width is determined by function h(t). Therefore, g(t) will be
expressed as,

This equation has been explained in fig.6.

Now, from the property of delta function, we know that for any function f(t)

This property is used to obtain flat top samples. It may be noted that to obtain flat
top sampling, we are not applying the above equation directly here i.e., we are
applying a modified form of the above equation.

Thus, in this modified equation, we are taking s(t) in place of delta function δ(t).

Observe that δ(t) is a constant amplitude delta function whereas s(t) is a varying
amplitude train of impulses. This means that we are taking s(t) which is an
instantaneously sampled signal and this is convolved with function h(t).

Therefore, on convolution of s(t) and h(t), we get a pulse whose duration is equal
to h(t) only but amplitude is defined by s(t).

Now, we know that the train of impulses may be represented mathematically as,
The signal s(t) is obtained by multiplication of baseband signal x(t) and δTs(t).
Thus,

Now, sampled signal g(t) is given as

According to shifting property of delta function, we know that

Hence,

This equation represents value of g(t) in terms of sampled value x(nT s) and
function h(t – nTs) for flat top sampled signal.
Now, we have

Taking Fourier transform of both sides of above equation, we get

We know that S(f) is given as

Therefore,
Thus, spectrum of flat top sampled signal:

Fig.8 : (a) Baseband signal x(t), (b) Instantaneously sample signal s(t), (c) Constant
pulse width function h(t), (d) Flat top sampled signal g(t) obtained through
convolution of h(t) and s(t)
 PULSE AMPLITUDE MODULATION (PAM)

Pulse amplitude modulation is a type of modulation in which the amplitudes of

regularly spaced rectangular pulses vary according to instantaneous value of the
modulating or message signal.

In fact, the pulses in a PAM signal may be of flat top type or natural type or ideal
type.

Out of all the three pulse amplitude modulation methods, the flat top PAM is most
popular and is widely used.

The reason for using flat top PAM is that during the transmission, the noise
interferes with the top of the transmitted pulses and this noise can be easily
removed if the PAM pulse has flat top.

However, in case of natural samples PAM signal, the pulse has varying top in
accordance with the signal variation.

Now, when such type of pulse is received at the receiver, it is always contaminated
by noise.

Then it becomes quite difficult to determine the shape of the top of the pulse and
thus amplitude detection of the pulse is not exact.Due to this, errors are introduced
in the received signal.

Therefore, flat top sampled PAM is widely used.

Fig.1 shows the sample and hold circuit to produce flat top sampled PAM and the
waveform for flat top sampled PAM.
Fig.1 : (a) Sample and hold circuit generating flat top sampled PAM, (b) Waveforms
of flat top sampled PAM

Principle of Working:

A sample and hold circuit shown in fig.1 is used to produce Flat top sampled PAM.
The working principle of this circuit is quite easy.

The sample and Hold (S/H) circuit consists of two field effect transistors (FET)
switches and a capacitor.

The sampling switch is closed for a short duration by a short pulse applied to the
gate G1 of the transistor.

During this period, the capacitor ‘C’ is quickly charged upto a voltage equal to the
instantaneous sample value of the incoming signal x(t).

Now, the sampling switch is opened, and the capacitor ‘C’ holds the charge.

The discharge switch is then closed by a pulse applied to gate G2 of the other
transistor.

Due to this, the capacitor ‘C’ is discharged to zero volts. The discharges switch is
then opened and thus capacitor has no voltage.

Hence, the output of the sample and hold circuit consists of a sequence of flat top
samples as shown in fig.1(b).
Mathematical Analysis:

In a flat top PAM, the top of the samples remains constant and is equal to the
instantaneous value of the baseband signal x(t) at the start of sampling.

The duration or width of each sample is τ and sampling rate is equal to,

From fig.1 (b), it may be noted that only starting edge of the pulse represents
instantaneous value of the baseband signal x(t).

Also, the flat top pulse of g(t) is mathematically equivalent to the convolution of
instantaneous sample and a pulse h(t) as depicted in fig.2.

Fig.2 : Convolution of any function with delta function is equal to that function

This means that the width of the pulse in g(t) is determined by the width of h(t)
and the sampling instant is determined by the delta function.

In fig.1 (b), the starting edge of the pulse represents the point where baseband
signal is sampled, and width is determined by function h(t).

Therefore, g(t) will be expressed as,

………… (1)

This equation has been explained in fig.3 below.

Fig.3 : (a) Baseband signal x(t), (b) Instantaneously sample signal s(t), (c) Constant
pulse width function h(t), (d) Flat top sampled PAM signal g(t) obtained through
convolution of h(t) and s(t).

Now, from the property of delta function, we know that for any function f(t),

…………… (2)
This property is used to obtain flat top samples. It may be noted that to obtain flat
top sampling, we are not applying the equation (2) directly here i.e., we are
applying a modified form of equation (2). This modified equation is equation (1).

Thus, in this modified equation, we are taking s(t) in place of delta functions δ(t).
Observe that δ(t) is a constant amplitude delta function whereas s(t) is a varying
amplitude train of impulses. This means that we are taking s(t) which is an
instantaneously sampled signal and this is convolved with function h(t) as in
equation (1). Therefore, on convolution of s(t) and h(t), we get a pulse whose
duration is equal to h(t) only but amplitude is defined by s(t).

Now, we know that the train of impulses may be represented mathematically as,
The signal s(t) is obtained by multiplication of baseband signal x(t) and δTs (t).
Thus,

Or,

Now, sampled signal g(t) is given as equation (1)

According to shifting property of delta function, we know that,

Hence,

This equation represents value of g(t) in terms of sampled value x(nTs) and
function h (t – nTs) for flat top sampled signal. Now, again from equation (1), we
have

Taking Fourier transform of both sides of above equation, we get

We know that S(f) is given as

Therefore,
Thus, spectrum of flat top PAM signal:

Here, H(f) is the Fourier transform of the rectangular pulse. The spectrum of this
rectangular pulse is shown in fig.4(b).

Let the spectrum of s(t) be the rectangular pulse train as shown in fig.4(a) and the
spectrum of h(t) i.e., H(f) is shown in fig.4(b).

Fig.4: (a) Spectrum of some arbitrary signal. The signal is sampled at fs and
maximum frequency in the signal is fm, (b) Spectrum of flat top signal. The dotted
curve is H(f) = τ sin c(fτ)

we know that,

Thus, according to above equation, we can plot the spectrum G(f) as shown in
fig.4(b).

It may be observed in fig.4(b) that higher frequencies in S(f) are attenuated due to
roll-off characteristics of the ‘sinc’ pulse. This effect is popularly known as aperture
effect. An equalizer is needed to overcome this effect.
 PULSE WIDTH MODULATION (PWM)

In PWM, the width of the modulated pulses varies in proportion with the
amplitude of modulating signal. The waveforms of PWM is shown in fig.1 below.

Fig.1 : PWM signal

As we can observe, the amplitude and the frequency of the PWM wave remain
constant. Only the width changes.

That is why the information is contained in the width variation. This is similar to
FM.

As the noise is normally additive noise, it changes the amplitude of the PWM signal.

At the receiver, it is possible to remove these unwanted amplitude variations very

easily by means of a limiter circuit.

As the information is contained in the width variation, it is unaffected by the

amplitude variations introduced by the noise. Thus, the PWM system is more
immune to noise than the PAM signal.

Generation of PWM Signal:

The block diagram of a PWM signal generator is shown in fig.2 below. This circuit
can also be used for the generation of PPM signal.
Fig.1 : PWM and PPM Generator

 A sawtooth generator generates a sawtooth signal of frequency fs, and this

sawtooth signal in this case is used as a sampling signal.
 It is applied to the inverting terminal of a comparator.
 The modulating signal x (t) is applied to the non-inverting terminal of the
same comparator.
 The comparator output will remain high as long as the instantaneous
amplitude of x (t) is higher than that of the ramp signal.
 This gives rise to a PWM signal at the comparator output as shown in fig.2 .

Fig.2 : Waveforms

Here, it may be noted that the leading edges of the PWM waveform coincide with
the falling edges of the ramp signal. Thus, the leading edges of PWM signal are
always generated at fixed time instants.

However, the occurrence of its trailing edges will be dependent on the

instantaneous amplitude of x(t). Therefore, this PWM signal is said to be trail edge
modulated PWM.

Detection of PWM Signal:

The circuit for the detection of PWM signal is shown in fig.3 below.
Fig.3: PWM Detection Circuit

The working operation of the circuit may be explained as under:

 The PWM signal received at the input of the detection circuit is

contaminated with noise. This signal is applied to pulse generator circuit
which regenerates the PWM signal.
 Thus, some of the noise is removed and the pulses are squared up.
 The regenerated pulses are applied to a reference pulse generator. It
produces a train of constant amplitude, constant width pulses.
 These pulses are synchronized to the leading edges of the regenerated PWM
pulses but delayed by a fixed interval.
 The regenerated PWM pulses are also applied to a ramp generator. At the
output of it, we get a constant slope ramp for the duration of the pulse. The
height of the ramp is thus proportional to the width of the PWM pulses.
 At the end of the pulse, a sample and hold amplifier retains the final ramp
voltage until it is reset at the end of the pulse.
 The constant amplitude pulses at the output of reference pulse generator
are then added to the ramp signal.
 The output of the adder is then clipped off at a threshold level to generate a
PAM signal at the output of the clipper.
 A low pass filter is used to recover the original modulating signal back from
the PAM signal. The waveforms for this circuit have been shown in fig.4.
Fig.4: Waveforms for PWM detection circuit

Advantages of PWM

 Less effect of noise i.e., very good noise immunity.

 Synchronization between the transmitter and receiver is not essential
(Which is essential in PPM).
 It is possible to reconstruct the PWM signal from a noise, contaminated
PWM, as discussed in the detection circuit. Thus, it is possible to separate
out signal from noise (which is not possible in PAM).

Disadvantages of PWM

 Due to the variable pulse width, the pulses have variable power contents.
Hence, the transmission must be powerful enough to handle the maximum
width, pulse, though the average power transmitted can be as low as 50% of
this maximum power.
 In order to avoid any waveform distortion, the bandwidth required for the
PWM communication is large as compared to bandwidth of PAM.
 PULSE POSITION MODULATION (PPM)

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 derived from the PWM pulses as shown in fig.1.

Here, it may be noted that with increase in the modulating voltage the PPM pulses
shift further with respect to reference.

Fig.1 : PPM pulses generated from PWM signal

The vertical dotted lines drawn in fig.1 are treated as 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.

Generation of PPM Signal

The PPM signal can be generated from PWM signal as shown in fig.2 (a).
Fig.2 : Generation of PPM signal

The PWM pulses obtained at the comparator output are applied to a monostable
multivibrator. The monostable is negative edge triggered.

Hence, corresponding to each trailing edge of PWM signal, the monostable output
goes high.

It remains high for a fixed time decided by its own RC components.

Thus, as the trailing edges of the PWM signal keep shifting in proportion with the
modulating signal x(t), the PPM pulses also keep shifting, as shown in fig.3.

Fig.3Demodulation of PPM Signal

The PPM demodulator block diagram has been shown in fig.4 .

Fig.4 The operation of the demodulator circuit may be explained as under:

  The noise corrupted PPM waveform is received by the PPM demodulator
circuit.
 The pulse generator develops a pulsed waveform at its output of fixed
duration and applies these pulses to the reset pin (R) of a SR flip-flop.
 A fixed period reference pulse is generated from the incoming PPM
waveform and the SR flip-flop is set by the reference pulses.
 Due to the set and reset signals applied to the flip-flop, we get a PWM signal
at its output.
 The PWM signal can be demodulated using the PWM demodulator.

Key Differences Between PAM, PWM and PPM

1. A factor that generates a key difference between PAM, PWM and PPM,
is that in PAM amplitude of the pulse shows proportionality with the
amplitude of modulating signal.

In PWM width of the pulses shows proportionality with the amplitude

of the message signal. Whereas in PPM the position of the pulses is
proportional to the amplitude of analog modulating signal.

2. PAM technique shows low immunity towards the noise. As against

PWM and PPM has low noise interference factor because their noise
immunity is high.

3. In PAM and PWM techniques transmitter and receiver, synchronization

is not required. But PPM technique needs synchronization between
transmitter and receiver section.

4. The SNR is low in case of pulse amplitude modulated signals while it is

moderate in case of pulse width modulated signals and highest in case
of pulse position modulated signals.

5. The transmission power in case of PAM and PWM is variable due to

variation in amplitude and width respectively. However, it is constant
in case of PPM because both amplitude and width are constant in case
of PPM.

6. In PAM, transmission channel bandwidth relies on the pulse width.

But in case of PWM and PPM, rise time of the pulse is a factor on which
the transmission channel bandwidth depends.
7. As the position of the pulses is changed in PPM, therefor it
requires synchronization pulses. While both PAM and PWM do not
require synchronization pulses.

8. All PAM, PWM and PPM show similarity in implementation with AM,
FM and PM respectively.

9. The bandwidth requirement is low in case of PAM but is

comparatively high in case of PWM and PPM.
 TIME DIVISION MULTIPLEXING (TDM)

Definition: A multiplexing technique by which multiple data signals can be

transmitted over a common communication channel in different time slots is
known as Time Division Multiplexing (TDM).

It allows the division of the overall time domain into various fixed length time
slots. A single frame is said to be transmitted when it’s all signal components gets
transmitted over the channel.

Theory of TDM:

As we know, multiplexing allows the transmission of several signals over a

common channel. However, one may need to differentiate between the various
signal for proper data transmission. So, in time division multiplexing, the
complete signal gets transmitted by occupying different time slots.

The name itself is indicating here that basically time division is performed in order
to multiplex multiple data signals.

Let us have a look at the figure below in order to have a better understanding of
the TDM process.

As we can see that source A, B and C wants to transmit data through a common
medium. Thus, the signal from the 3 sources, is divided into multiple frames each
having their fixed time slot. Here, 3 units from each source are taken into
consideration, that jointly form the actual signal.

A frame is transmitted at a time that is composed of one unit of each source. As

these units are entirely different from each other thus the chances of unnecessary
signal mixing can be eliminated.
When a frame gets transmitted over the particular time slot, the next frame uses
the same channel to get transmitted and the process is further repeated until the
completion of the transmission.

Here, we have taken the example of 3 different sources, but one can perform
multiplexing of n source signals. It is noteworthy here that units of a single source
must be equivalent to the total number of source signals to be transmitted.

Both analog and digital signals can be multiplexed using time division
multiplexing, but its processing technique allows the multiplexing of digital signals
conveniently rather than analog one.

TDM system:

The figure below shows the block diagram of a TDM system employing both
transmitter and receiver section.

The technique efficiently utilizes the complete channel for data transmission hence
sometimes known as PAM/TDM. This is so because a TDM system uses a pulse
amplitude modulation. In this modulation technique, each pulse holds some short
time duration allowing maximal channel usage.

Here at the beginning, the system consists of multiple LPF depending on the
number of data inputs. These low pass filters are basically anti-aliasing filters that
eliminate the aliasing of the data input signal.

The output of the LPF is then fed to the commutator. As per the rotation of the
commutator the samples of the data inputs are collected by it. Here, fs is the rate of
rotation of the commutator, thus denotes the sampling frequency of the system.
Suppose we have n data inputs, then one after the other, according to the rotation,
these data inputs after getting multiplexed transmitted over the common channel.

Now, at the receiver end, a de-commutator is placed that is synchronized with the
commutator at the transmitting end. This de-commutator separates the time
division multiplexed signal at the receiving end.

The commutator and de-commutator must have same rotational speed so as to

have accurate demultiplexing of the signal at the receiving end. According to the
rotation performed by the de-commutator, the samples are collected by the LPF
and the original data input is recovered at the receiver.

The way a TDM works:

Let fm be the maximum signal frequency and fs is the sampling frequency then

Thus, the time duration in between successive sample is given as,

Rewriting in terms of fm

Now, as we have considered that there are N input channels, then one sample is
collected from each of the N samples.

Hence, each interval will provide us with N samples and the spacing between the
two is given as

We know pulse frequency is basically the number of pulses per second and is given
by
For a TDM signal pulse per second is the signalling rate denoted as ‘r’.

Thus,

Implementation of TDM

The technique of time division multiplexing can be implemented in basically two

ways:

1. Synchronous TDM: In this technique, the time slots are assigned at the
beginning, irrespective of the idea about the presence of data at the source. This
leads to the wastage of the channel capacity. As in the absence of any data unit, that
particular time slot gets entirely wasted.

As it leads to exhaustion of the time slot in the absence of any data unit thus, the
technique is considered to be efficient.

2. Asynchronous TDM: It is also termed as statistical or intelligent

TDM technique as it eliminates the drawback of wastage of time slot present in
synchronous TDM.
Here, a particular frame is transmitted by the transmitting end only when it gets
completely filled by the data units. It exhibits higher efficiency than that of
synchronous TDM technique as it requires smaller transmission time and ensures
better bandwidth utilization.

Advantages of TDM

 Simple circuit design.

 It uses entire channel bandwidth for the transmission of the signal.

 The problem of Intermodulation distortion is not present in TDM.

 Pulse overlapping can sometimes cause crosstalk, but it can be reduced by

utilizing guard time. Thus, is not much serious.

Disadvantages of TDM

 The transmitting and receiving section must be properly synchronized in

order to have proper signal transmission and reception.

 Slow narrowband fading can wipe out all the TDM channels.

Applications of Time division multiplexing

TDM finds its application mainly in a digital communication system, in cellular

radio and in satellite communication system.
 PULSE CODE MODULATION

Pulse-code modulation or PCM is known as a digital pulse modulation technique.

In fact, the pulse-code modulation is quite complex as compared to the Analog
pulse modulation techniques i.e. PAM, PWM and PPM, in the sense that the
message signal is subjected to a great number of operations.

In PCM an Analog signal or information is converted into a binary sequence,

i.e.,’ 1’s and ‘0’s. The output of a PCM resembles a binary sequence.
The following figure shows an example of PCM output with respect to
instantaneous values of a given sine wave.

PCM produces a series of numbers or digits instead of a pulse train,. Each one of
these digits, in binary code, represent the approximate amplitude of the signal
sample at that instant.

In Pulse Code Modulation, the message signal is represented by a sequence of

coded pulses. This message signal is achieved by representing the signal in discrete
form in both time and amplitude.

Elements of a PCM System:

Fig.1 shows the basic elements of a PCM system .

(a): PCM Transmitter

(b) : Transmitter Path

(c) : Receiver

Fig.1 : The basic elements of a PCM System

It consists of three main parts i.e. ,

1. Transmitter
2. Transmission path
3. Receiver
The essential operation in the transmitter of a PCM system are :

1. Sampling
2. Quantizing
3. Encoding
As discussed earlier, sampling is the operation in which an analog (continuous-
time) signal is sampled according to the sampling theorem resulting in a discrete-
time signal .

The quantizing and encoding operations are usually performed in the same circuit
which is known as an analog-to-digital converter (ADC) .

The essential operations in the receiver of a PCM system are :

1. Regeneration of impaired signals

2. Decoding and demodulation of the train of quantized samples
These operations are usually performed in the same circuit which is known as a
digital-to-analog converter (DAC) .
Further, at intermediate points, along the transmission route from the transmitter
to the receiver, regenerative repeaters are used to reconstruct (i.e. regenerate) the
transmitted sequence of coded pulses in order to combat the accumulated effects
of signal distortion and noise.

As discussed before, the quantization refers to the use of a finite set of amplitude
levels and the selection of a level nearest to a particular sample value of the
message signal as the representation for it. In fact, this operation combined with
sampling, permits the use of coded pulses for representing the message signal.
Thus, it is the combined use of quantizing and coding that distinguishes pulse code
modulation from Analog modulation techniques.

Few Important Points:

Now, let us summarize PCM in the form of few points as under:

1. PCM is a type of pulse modulation like PAM, PWM or PPM but there is
an important difference between them i.e., PAM, PWM or PPM are
Analog pulse modulation systems whereas PCM is a digital pulse
modulation system.
2. This means that the PCM output is in the coded digital form. It is in the
form of digital pulses of constant amplitude, width, and position.
3. The information is transmitted in the form of code words. A PCM
system consists of a PCM encoder (transmitter) and a PCM decoder
(receiver) .
4. The essential operations in the PCM transmitter are sampling,
quantizing, and encoding.
5. All the operations are usually performed in the same circuit called as
Analog-to digital converter.
6. The PCM is not modulation in the conventional sense. Because in
modulation, one of the characteristics of the carrier is varied in
proportion with the amplitude of the modulating signal. Nothing of that
sort happen in PCM.

1. PCM Transmitter:

Fig.2 shows a practical block diagram of a PCM generator or transmitter.

Fig.2: PCM Transmitter

In PCM transmitter, the signal x(t) is first passed through the low-pass filter of cut-
off frequency fm Hz. This low-pass filter blocks all the frequency components above
fm Hz. This means that now the signal x(t) is bandlimited to fm Hz.
The sample and hold circuit then sample this signal at the rate of fs.
Sampling frequency fs is selected sufficiently above Nyquist rate to avoid aliasing
i.e.,
fs ≥ 2fm
In fig.2, the output of sample and hold circuit is denoted by x(nTs). This
signal x(nTs) is discrete in time and continuous in amplitude.
A q-level quantizer compares input x(nTs) with its fixed digital levels.
It then assigns any one of the digital levels to x(nTs) which results in minimum
distortion or error. This error is called quantization error.
Thus, output of quantizer is a digital level called xq(nTs).
Now the quantized signal level xq(nTs) is given to binary encoder. This encoder
converts input signal to ‘v’ digits binary word. This encoder is also known as
digitizer.
In addition to these, there is an oscillator which generates the clocks for sample
and hold circuit and parallel to serial converter.

In PCM, sample and hold, quantizer and encoder combinely form an Analog to
digital converter (ADC).

2. PCM Transmission Path:

The path between the PCM transmitter and receiver over which the PCM signal
travel, is known as PCM transmission path and it is shown in Fig.3.
Fig.3 : PCM Transmission path

The most important feature of PCM system lies in its ability to control the effects of
distortion and noise when the PCM wave travels on the channel.

This is accomplished by means of using a chain of regenerative repeaters as shown

in fig.3.

Such repeaters are spaced close enough to each other on the transmission path.

The repeaters perform three basic operations such as : quantization, timing and
decision making.

Hence, each repeater reproduces the clean and noise free PCM signal.

This improves the performance of PCM in presence of noise.

Repeater:

Fig.4 shows the block diagram of a repeater.

Fig.4: block diagram of regenerative repeater

The amplitude quantizer shapes the distorted PCM wave to compensate for the
effects of amplitude and phase distortions.

The timing circuit produces a periodic pulse train which is derived from the input
PCM pulses.

This pulse train is then applied to the decision-making device.

The decision-making device uses this pulse train for sampling the equalized PCM
pulses.

The sampling is carried out at the instants where the signal to noise ratio is
maximum.

The decision device decides about whether the equalized PCM wave at its input has
0 value or 1 value at the instant of sampling.

Such a decision is made by comparing equalized PCM with a reference level called
decision threshold.

At the output of the decision device, we get a clean PCM signal without any noise.

PCM Receiver

Fig.5 : PCM Receiver

The regenerator at the start of PCM receiver reshapes the pulse and removes the
noise.

This signal is then converted to parallel digital words for each sample.

Now, the digital word is converted to its Analog value denoted as xq(t) with the
help of a sample and hold circuit.
This signal, at the output of sample and hold circuit is allowed to pass through a
low-pass reconstruction filter to get the original message signal x(t) .
 DIFFERENTIAL PULSE CODE MODULATION

It may be observed that the samples of a signal are highly correlated with each
other. This is since any signal does not change fast. Which means, its value from
present sample to next sample does not vary by a large amount.

The adjacent samples of the signal carry the same information with a little
difference.

When these samples are encoded by a standard PCM system, the resulting encoded
signal contains some redundant information.
Redundant Information in PCM

Fig.1 shows a continuous time signal x(t) by dotted line. This signal is sampled by
flat top sampling at intervals Ts , 2Ts , 3Ts ….. nTs .

Fig.1 : Illustration of redundant information in PCM

The sampling frequency is selected to be higher than Nyquist rate.

The samples are encoded by using 3-bit (7 levels) PCM.

The sample is quantized to the nearest digital level as shown by small circles in
fig.1.

The encoded binary value of each sample is written on the top of the samples.

We can observe from fig.1 that the samples taken at 4Ts, 5Ts and 6Ts are encoded to
same value of (110).
This information can be carried only by one sample.

But three samples are carrying the same information means that it is redundant.
We consider another example of samples taken at 9Ts and 10Ts. The difference
between these samples only due to last bit and first two bits are redundant, as they
do not change.
If this redundancy is reduced, then overall bit rate will decrease, and number of
bits required to transmit one sample will also be reduced.

This type of digital pulse modulation technique is called as Differential Code

Modulation (DPCM).

Working Principle:

The differential pulse code modulation works on the principle of prediction. The
value of the present sample is predicted from the past samples.

The prediction may not be exact but it is very close to the actual sample value.

Fig.2 shows the transmitter of DPCM system.

Fig.2 : A Differential pulse code modulation

The sampled signal is denoted by x(nTs) and predicted signal is denoted by xˆ(nTs).
The comparator finds out the difference between the actual sample value x(nT s)
and predicted sample value xˆ(nTs).
This is known as prediction error and it is denoted by e(nTs).
It can be defined as ,

e(nTs) = x(nTs) – xˆ(nTs)……………………….(1)

The predicted value is produced by using a prediction filter.

The quantizer output signal gap eq(nTs) and previous prediction is added and given
as input to the prediction filter.This signal is called xq(nTs).
This makes the prediction more and more close to the actual sampled signal.
We can observe that the quantized error signal eq(nTs) is very small and can be
encoded by using small number of bits.
Thus number of bits per sample are reduced in DPCM.

The quantizer output can be written as ,

eq(nTs) = e(nTs) + q(nTs)………………………..(2)

Here, q(nTs) is the quantization error.
As shown in fig.2, the prediction filter input xq(nTs) is obtained by sum xˆ(nTs) and
quantizer output. i.e.,
xq(nTs) = xˆ(nTs) + eq(nTs)……………………..(3)
Substituting the value of eq(nTs) from eq.(2) in the above eq. (3) , we get,
xq(nTs) = xˆ(nTs) + e(nTs) + q(nTs) ………………….(4)
eq.(1) is written as,

e(nTs) = x(nTs) – xˆ(nTs)

∴ e(nTs) + xˆ(nTs) = x(nTs)
Therefore, substituting the value of e(nTs) + xˆ(nTs) from the above equation into
eq. (4),

we get,
xq(nTs) = x(nTs) + q(nTs) …………………..(5)

Reception of DPCM Signal

Fig.3 shows the block diagram of DPCM receiver.

Fig.3 : DPCM Receiver

The decoder first reconstructs the quantized error signal from incoming binary
signal.
The prediction filter output and quantized error signals are summed up to give the
quantized version of the original signal.

Thus the signal at the receiver differs from actual signal by quantization
error q(nTs), which is introduced permanently in the reconstructed signal.
Advantages of DPCM

 As the difference between x(nTs) and xˆ(nTs) is being encoded and

transmitted by the DPCM technique, a small difference voltage is to be
quantized and encoded.
 This will require less number of quantization levels and hence less number
of bits to represent them.
 Thus signaling rate and bandwidth of a DPCM system will be less than that
of PCM.
 DELTA MODULATION

In PCM the signalling rate and transmission channel bandwidth are quite large
since it transmits all the bits which are used to code a sample. To overcome this
problem, Delta modulation is used.

Working Principle:

Delta modulation transmits only one bit per sample. Here, the present sample
value is compared with the previous sample value and this result whether the
amplitude is increased or decreased is transmitted.

Input signal x(t) is approximated to step signal by the delta modulator.This step
size is kept fixed.

The difference between the input signal x(t) and staircase approximated signal is
confined to two levels, i.e., +Δ and -Δ.

Now, if the difference is positive, then approximated signal is increased by one

step, i.e., ‘Δ’. If the difference is negative, then approximated signal is reduced by ‘Δ’
.

When the step is reduced, ‘0’ is transmitted and if the step is increased, ‘1’ is
transmitted.

Hence, for each sample, only one binary bit is transmitted.

Fig.1 shows the analog signal x(t) and its staircase approximated signal by the
delta modulator.

Fig.1. Delta Modulation Waveform

Mathematical Expressions:

The error between the sampled value of x(t) and last approximated sample is given
as:

Where e(nTs) = error at present sample

x(nTs) = sampled signal of x(t)

If we assume u(nTs) as the present sample approximation of staircase output, then

Let us define a quantity b( nTs) in such a way that,

This means that depending on the sign of error e( nTs) , the sign of step size Δ is
decided. In other words we can write

Also if b (nTs) =+Δ then a binary ‘1’ is transmitted

and if b (nTs) =-Δ then a binary ‘0’ is transmitted
Here Ts = sampling interval.
Transmitter

Fig. 2 (a) shows the transmitter. It is also known as Delta modulator.

Fig.2 (a) Delta Modulation Transmitter

It consists of a 1-bit quantizer and a delay circuit along with two summer circuits.

The summer in the accumulator adds quantizer output (±Δ) with the previous
sample approximation. This gives present sample approximation. i.e.,

The previous sample approximation u[(n-1)Ts ] is restored by delaying one sample

period Ts .
The samples input signal x(nTs ) and staircase approximated signal xˆ(nTs ) are
subtracted to get error signal e(nTs ).
Thus, depending on the sign of e(nTs ), one bit quantizer generates an output of +Δ
or -Δ .
If the step size is +Δ, then binary ‘1’ is transmitted and if it is -Δ, then binary ‘0’ is
transmitted.

Receiver:

At the receiver end also known as delta demodulator, as shown in fig.2 (b) ,
it comprises of a low pass filter(LPF), a summer, and a delay circuit. The predictor
circuit is eliminated here and hence no assumed input is given to the demodulator.

Fig.2 (b) Delta Modulation Receiver

The accumulator generates the staircase approximated signal output and is

delayed by one sampling period Ts.
It is then added to the input signal.

If the input is binary ‘1’ then it adds +Δ step to the previous output (which is
delayed).

If the input is binary ‘0’ then one step ‘Δ’ is subtracted from the delayed signal.
Also, the low pass filter smoothens the staircase signal to reconstruct the original
message signal x(t).

Advantages of Delta Modulation

The delta modulation has certain advantages over PCM as under :

1. Since, the delta modulation transmits only one bit for one sample,
therefore the signalling rate and transmission channel bandwidth is
quite small for delta modulation compared to PCM .
2. The transmitter and receiver implementation is very much simple for
delta modulation. There is no analog to digital converter required in
delta modulation.

Disadvantages of Delta Modulation

The delta modulation has two major drawbacks as under :

1. Slope overload distortion

2. Granular or idle noise
Now, we will discuss these two drawbacks in detail.

1. Slope Overload Distortion

This distortion arises because of large dynamic range of the input signal.

Fig.1 : Quantization errors in Delta modulation

We can observe from fig.1 , the rate of rise of input signal x(t) is so high that the
staircase signal cannot approximate it, the step size ‘Δ’ becomes too small for
staircase signal u(t) to follow the step segment of x(t).
Hence, there is a large error between the staircase approximated signal and the
original input signal x(t).

This error or noise is known as slope overload distortion.

To reduce this error, the step size must be increased when slope of signal x(t) is
high.

Since, the step size of delta modulator remain fixed, its maximum or minimum
slopes occur along straight lines. Therefore, this modulator is known as Linear
Delta Modulator (LDM).

2. Granular or Idle Noise

Granular or Idle noise occurs when the step size is too large compared to small
variation in the input signal.

This means that for very small variations in the input signal, the staircase signal is
changed by large amount (Δ) because of large step size.

Fig.1 shows that when the input signal is almost flat , the staircase signal u(t) keeps
on oscillating by ±Δ around the signal.

The error between the input and approximated signal is called granular noise.
The solution to this problem is to make the step size small.

Difference between delta modulation (DM) and pulse code modulation

(PCM):

 PCM stands for pulse code modulations while in DM is stands for delta
modulations.

 In PCM feedback does not exist in transmitter or receiver while in delta

modulation feedback existed in the transmitter.

 PCM requires the highest transmitter bandwidth while DM requires the

lowest transmitter bandwidth.

 PCM is complex in terms of complexity of implementation, whereas DM is

simple in terms of complexity of implementation.

 In PCM per sample 4,8, or 16 bits are used while in DM only one's bit is used
per sample.
 PCM may be a technique wont to digitally represent sampled analog signals
while in DM convert digital to Analog and Analog to digital converter.

 PCM is costly, DM is cheap.

 PCM has a good signal to noise ratio while in DM has a poor signal to noise
ratio.

 PCM is mostly used in video telephony and audio telephony, DM is mostly

used in speeches as well as images.

 PCM signals are required encoder and decoder both sides while in DM
signals can modulate and demodulates.

 In PCM Quantization error depends on the number of levels, while in DM

slope overload distortion is present.