COURSE CODE : 5201

DIGITAL
COMMUNICATION

Sreenath Narayanan
Lecturer in ECE Dept.
SNPTC Kanhangad
 Module I
Pulse And Pulse Code Modulation
MODEL OF A DIGITAL COMMUNICATION
SYSTEM
Source
• The source can be an analog signal. Example: A Sound signal
Input Transducer
• This is a transducer which takes a physical input and converts it to an
electrical signal (Example: microphone). This block also consists of
an analog to digital converter where a digital signal is needed for
further processes.
• A digital signal is generally represented by a binary sequence.
Source Encoder
• The source encoder compresses the data into minimum number of
bits. This process helps in effective utilization of the bandwidth. It
removes the redundant bits unnecessary excess bits, i.e. zeroes.
Channel Encoder
• The channel encoder, does the coding for error correction. During the
transmission of the signal, due to the noise in the channel, the signal
may get altered and hence to avoid this, the channel encoder adds
some redundant bits to the transmitted data. These are the error
correcting bits.
Digital Modulator
• The signal to be transmitted is modulated here by a carrier. The
signal is also converted to analog from the digital sequence, in order
to make it travel through the channel or medium.
Channel
• The channel or a medium, allows the analog signal to transmit from
the transmitter end to the receiver end.
Digital Demodulator
• This is the first step at the receiver end. The received signal is
demodulated as well as converted again from analog to digital. The
signal gets reconstructed here.
Channel Decoder
• The channel decoder, after detecting the sequence, does some error
corrections. The distortions which might occur during the
transmission, are corrected by adding some redundant bits. This
addition of bits helps in the complete recovery of the original signal.
Source Decoder
• The resultant signal is once again digitized by sampling and
quantizing so that the pure digital output is obtained without the loss
of information. The source decoder recreates the source output.
Output Transducer
• This is the last block which converts the signal into the original
physical form, which was at the input of the transmitter. It converts
the electrical signal into physical output (Example: loud speaker).
Output Signal
• This is the output which is produced after the whole
process. Example − The sound signal received.
Advantages of Digital Communication

• The effect of distortion, noise, and interference is much less in

digital signals as they are less affected.
• Digital circuits are more reliable.
• Digital circuits are easy to design and cheaper than analog circuits.
• Signal processing functions such as encryption and compression are
employed in digital circuits to maintain the secrecy of the
information..
• The probability of error occurrence is reduced by employing error
detecting and error correcting codes.
• Combining digital signals using Time Division
Multiplexing TDM is easier than combining analog signals using
Frequency Division Multiplexing FDM.
• Digital signals can be saved and retrieved more conveniently than
analog signals.
• The capacity of the channel is effectively utilized by digital signals.
 Disadvantages of Digital Communication

• High transmission bandwidth is required for digital

communication.
• Digital communication needs synchronization
SAMPLING THEOREM
• During sampling process, a continuous-time signal is converted into
discrete-time signals by taking samples of continuous-time signal at
discrete time intervals.
• Statement 1: 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
– where 𝑓𝑠 is the sampling frequency (𝑓𝑠 =1/𝑇𝑠 , where 𝑇𝑠 is the sampling
period or sampling interval)
– 𝑓𝑚 is the modulating signal frequency

• Statement 2: The sampling theorem specifies the minimum-sampling rate

at which a continuous-time signal needs to be uniformly sampled so that
the original signal can be completely recovered or reconstructed by these
samples alone.
• The minimum sampling rate allowed by the sampling theorem (𝑓𝑠 ≥ 2𝑓𝑚 )
is called the Nyquistrate.
• If 𝑓𝑠 ≥ 2𝑓𝑚 the spectrum will not overlap each other.
• If 𝑓𝑠 = 2𝑓𝑚 the spectrum will not overlap but touches each other.
• If 𝑓𝑠 < 2𝑓𝑚 the spectrum will overlap each other.
TYPES OF MODULATION
PULSE MODULATION
• In Pulse modulation systems, some parameter of pulse train is varied in
accordance with the instantaneous value of the modulating signal.
• Various types of Pulse modulation techniques are
– Pulse Amplitude Modulation (PAM)
– Pulse Width Modulation (PWM)
– Pulse Position Modulation (PPM)
– Pulse Code Modulation (PCM)
PULSE AMPLITUDE MODULATION (PAM)
• In Pulse Amplitude Modulation (PAM) technique, the amplitude of the
pulse carrier varies, which is proportional to the instantaneous amplitude of
the message signal.
• The pulse amplitude modulated signal will follow the amplitude of the
original signal, as the signal traces out the path of the whole wave.
Pulse Width Modulation (PWM)
• In Pulse Width Modulation (PWM) or Pulse Duration Modulation (PDM)
or Pulse Time Modulation (PTM) technique, the width or the duration or
the time of the pulse carrier varies, which is 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.
Pulse Position Modulation (PPM)
• Pulse Position Modulation (PPM) is an analog modulation scheme in which,
the amplitude and the width of the pulses are kept constant, while the position
of each pulse varies according to the instantaneous value of the message
signal.
• Pulse position modulation is
done in accordance with the pulse
width modulated signal.
• Each trailing edge of the pulse
width modulated signal becomes the
starting point for pulses in PPM signal.
• Hence, the position of these pulses
is proportional to the width of the
PWM pulses.
Generation of Pulse Amplitude Modulation (PAM)
• A sample and hold circuit shown in fig. is used to produce flat top sampled
PAM .
• The sample and Hold 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 up to 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 .
 Demodulation of PAM Signals

Fig.1 : Block diagram of PAM demodulator

Fig.2 : A zero-order holding circuit Fig.3 : the output of a Low Pass filter (LPF)
• For pulse amplitude modulated (PAM) signals, the demodulation is
done using a Holding circuit. Fig.1 shows the block diagram of a
PAM demodulator.
• In this method, the received PAM signal is allowed to pass through a
Holding circuit and a low pass filter (LPF) as shown in fig.1.
• Fig.2 illustrates a very simple holding circuit.
• Here the switch ‘S’ is closed after the arrival of the pulse and it is
opened at the end of the pulse.
• In this way, the capacitor C is charged to the pulse amplitude value
and it holds this value during the interval between the two pulses.
• Hence, the sampled values are held as shown in fig.3.
• fter this the holding circuit output is smoothened in Low Pass filter
as shown in fig.3.

Drawbacks
• Very large bandwidth required for transmission.
• Interference of noise is maximum.
Generation of PWM Signal

Fig.1 : PWM Generator

Fig.2 : PWM Waveforms

• 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 .
• The leading edges of the PWM waveform coincide with the
falling edges of the ramp signal.
• The occurance of its trailing edges will be dependent on the
instantaneous amplitude of x(t).
Detection of PWM Signal

Fig.3 : PWM Detection Circuit

Fig.4 : Waveforms for PWM detection circuit
• 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.
• The regenerated pulses are applied to a reference pulse generator. It
produces a train of pulses with constant amplitude, constant width.
• 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 thereshold 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.
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.

Disadvantages of PWM
• Due to the variable pulse width, the pulses have variable power
contents.
• Bandwidth required for the PWM communication is large as
compared to PAM.
 Generation of PPM signal

Fig.1: Generation of PPM signal

• 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 generate from the PWM pulses as shown in
fig.2.
 Fig.2 : PPM pulses generated from PWM signal
• Here, it may be noted that with increase in the modulating voltage the
PPM pulses shift further with respect to reference.
• 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.
 Fig.3: Waveforms

• 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.
 Demodulation of PPM Signal

Fig.4: Detection of PPM signal

• 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.
PULSE-CODE MODULATION (PCM)
• Pulse Code modulation is the process by which an analog signal is
converted to digital form in order to be transmitted in digital means.
• i.e, In a PCM stream, the amplitude of the analog signal
is sampled regularly at uniform intervals, and each sample is quantized to
the nearest value within a range of digital steps.
• Pulse-code modulation (PCM) is a method used to digitally represent
sampled analog signals.
• PCM is in binary form, so there will be only two possible states high and
low(0 and 1).
• It is the standard form of digital audio in computers, compact discs, digital
telephony and other digital audio applications.
• The Pulse Code Modulation process is done in three steps
– Sampling
– Quantization
– Coding.
• Sampling process converts continuous amplitude signal into Discrete-time-
continuous signal. The Sampling process generates flat-top Pulse Amplitude
Modulated (PAM) signal. According to the Nyquist Theorem Sampling
frequency, Fs>=2*fmax
• Quantization approximates the analog sample values with the nearest quantization
values.
• The encoder encodes the quantized samples. Each quantized sample is encoded
into an 8-bit codeword
ADVANTAGES
• Uniform Transmission Quality
• Low Manufacturing Cost
• Good Performance Over Very poor
transmission paths
• Low noise

DISADVANTAGES
• Large Bandwidth required for
Transmission
• Rise in attenuation
• Quantizing and encoding circuits are
complex

APPLICATION
• In compact disk
• Digital telephony
• Digital audio applications
Basic Elements of PCM
• 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.
• Above fig. shows the block diagram of PCM which represents the basic elements of both
the transmitter and the receiver sections.
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
• This is the technique which helps to collect the sample data at instantaneous values of
message signal, so as to reconstruct the original signal.
• The sampling rate must be greater than twice the highest frequency component W of the
message signal, in accordance with the sampling theorem.
Quantizer
• Quantizing is a process of reducing the excessive bits and confining the data. The sampled
output when given to Quantizer, reduces the redundant bits and compresses the value.
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.
• These three sections LPF, Sampler, and Quantizer will act as an analog to digital
converter.
• Encoding minimizes the bandwidth used.
Regenerative Repeater
• This section increases the signal strength.
• The output of the channel also has one regenerative repeater circuit, to compensate the
signal loss and reconstruct the signal, and also to increase its strength.
Decoder
• The decoder circuit decodes the pulse coded waveform to reproduce the original signal.
• This circuit acts as the demodulator.
Reconstruction Filter
• After the digital-to-analog conversion is done by the regenerative circuit and the
decoder, a low-pass filter is employed, called as the reconstruction filter to get back the
original signal.
• Hence, the Pulse Code Modulator circuit digitizes the given analog signal, codes it and
samples it, and then transmits it in an analog form.
• This whole process is repeated in a reverse pattern to obtain the original signal.
 Quantization
• The digitization of analog signals involves the rounding off of the values
which are approximately equal to the analog values.
• The method of sampling chooses a few points on the analog signal and
then these points are joined to round off the value to a near stabilized
value. Such a process is called as Quantization.
Quantizing an Analog Signal

• 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-time signal.
• The figure above shows how an analog signal gets quantized.
• The blue line represents analog signal while the brown one represents the
quantized signal.
• Both sampling and quantization result in the loss of information.
• The quality of a Quantizer output depends upon the number of
quantization levels used. T
• 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.
• The following figure shows the resultant quantized signal which is the
digital form for the given analog signal.
• This is also called as Stair-case waveform, in accordance with its shape.
Types of Quantization
• There are two types of Quantization - Uniform Quantization and Non-
uniform Quantization.
• The type of quantization in which the quantization levels are uniformly
spaced is termed as a Uniform Quantization.
• The type of quantization in which the quantization levels are unequal and
mostly the relation between them is logarithmic, is termed as a Non-
uniform Quantization.
• There are two types of uniform quantization.
• They are Mid-Rise type and Mid-Tread type. The following figures
represent the two types of uniform quantization.
• Figure 1 shows the mid-rise type and figure 2 shows the mid-tread type of
uniform quantization.
• The Mid-Rise type is so called because the origin lies in the middle of a
raising part of the stair-case like graph. The quantization levels in this type
are even in number.
• The Mid-tread type is so called because the origin lies in the middle of a
tread of the stair-case like graph. The quantization levels in this type are
odd in number.
• Both the mid-rise and mid-tread type of uniform quantizers are symmetric
about the origin.
Nonuniform Quantization
• In nonuniform quantization, the step size is unequal. After the
quantization, the difference between an input value and its quantized
value is called the quantization error.
• In uniform quantization, the step size is equal. Therefore, some part of the
signal might not cover. This can increase quantization error.
• However, in case of nonuniform quantization, the step size changes so it
will have a minimum amount of error.
 Companding in PCM
• The word Companding is a combination of Compressing and Expanding, which
means that it does both.

• Nonuniform quantizers are difficult to make and expensive.

• An alternative is to first pass the speech signal through a nonlinearity before
quantizing with a uniform quantizer.
• The nonlinearity causes the signal amplitude to be Compressed.
• The input to the quantizer will have a more uniform distribution.
• At the receiver, the signal is Expanded by an inverse to the nonlinearity.
• The process of compressing and expanding is called Companding.
• This is a non-linear technique used in PCM which compresses the data at the
transmitter and expands the same data at the receiver.
• The effects of noise and crosstalk are reduced by using this technique.
• There are two types of Companding techniques. They are −
A-law Companding Technique
• Uniform quantization is achieved at A = 1, where the characteristic
curve is linear and no compression is done.
• A-law has mid-rise at the origin. Hence, it contains a non-zero value.
• The practically used value of µ is 87.56
• A-law companding is used for PCM telephone systems.

µ-law Companding Technique

• Uniform quantization is achieved at µ = 0, where the characteristic
curve is linear and no compression is done.
• µ-law has mid-tread at the origin. Hence, it contains a zero value.
• µ-law companding is used for speech and music signals.
• The practically used value of µ is 255
• µ-law is used in North America and Japan.
 Differential PCM

• For the samples that are highly correlated, when encoded by PCM
technique, leave redundant information behind.
• To process this redundant information and to have a better output, it is a
wise decision to take a predicted sampled value, assumed from its
previous output and summarize them with the quantized values.
• Such a process is called as Differential PCM DPCM technique.
• Redundant Information in PCM

• Samples taken at 4Ts , 5Ts and 6Ts are encoded to same value of (110) so it is
redundant.
• Difference between samples taken at 9Ts and 10Ts is only due to last bit and
first two bits are redundant.
• 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. below shows the transmitter of DPCM system.
• 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(nTs) 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., 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, substituing 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. below shows the block diagram of 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 signaling 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. below shows the analog signal x(t) and its staircase approximated
signal by the delta modulator.
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.(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. (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.(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) .

The delta modulation has two major drawbacks as under :

• Slope overload distortion
• Granular or idle noise
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 can not 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.
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 .
• In order to overcome the quantization errors due to slope
overload and granular noise, the step size (Δ) is made adaptive to variations
in the input signal x(t).
• Particularly in the steep segment of the signal x(t), the step size is increased.
And the step is decreased when the input is varying slowly.
• This method is known as Adaptive Delta Modulation (ADM).
• The adaptive delta modulators can take continuous changes in step size or
discrete changes in step size.
 Module II
Digital Carrier Modulation
 Baseband Transmission vs Passband Transmission
• In baseband pulse transmission, a data stream represented in the form of a
discrete pulse-amplitude modulated (PAM) signal is transmitted over a
lowpass channel.
• The baseband signals have an adequately large power at low frequencies. So
they can be transmitted over a pair of wires or coaxial cables.
• But it is not possible to transmit the baseband signals over radio links or
satellites because impracticably large antennas would be required to be used.
• Hence, the spectrum of the message signal has to be shifted to higher
frequencies.
• This is achieved by using the baseband digital signal to modulate a sinusoidal
carrier.
• This is called digital carrier modulation or digital passband communication.
• In digital passband transmission, the incoming data stream is modulated into
a carrier with fixed frequency and then transmitted over a band-pass channel.
• Passband digital transmission allows more efficient use of the allocated RF
bandwidth, and flexibility in accommodating different baseband signal
formats.
Elements of Band Pass (Pass band) Transmission System

• Pass-band modulation is the process by which an information signal is

converted to a sinusoidal waveform.
• Thus pass-band modulation can be defined as the process whereby the
amplitude, frequency and phase of an RF carrier, or a combination of
them, is varied in accordance with the information to be transmitted.
• The most common digital modulation formats: -
1- Amplitude shift keying (ASK).
2- Frequency shift keying (FSK).
3- Phase shift keying (PSK).
DIGITAL CARRIER MODULATION SCHEMES

• The techniques used to modulate digital information for transmission.

• In digital communications, the modulation process corresponds to
switching or keying the amplitude, frequency, or phase of a sinusoidal
carrier wave according to incoming digital data
• Three basic digital modulation techniques
– Amplitude-shift keying (ASK) or Binary amplitude shift keying
(BASK)
– Frequency-shift keying (FSK) or Binary frequency shift keying (BFSK)
– Phase-shift keying (PSK) or Binary phase shift keying (BPSK)
AMPLITUDE SHIFT-KEYING (ASK) OR BASK
• In this system, it shifts the amplitude of carrier signal according to the
binary symbols (1,0) and frequency & phase will be constant.
• In this system, generally binary symbols (1,0) modulate the amplitude of
the carrier. It is also known as on-off keying (ook)
FREQUENCY SHIFT-KEYING (FSK) OR BFSK
• In this system, it shifts the frequency of carrier signal according to the
binary symbols (1,0) and phase & amplitude will be un-effected.
• In this system, generally binary symbols (1,0) modulate the frequency of
carrier.
GENERATION OF BFSK SIGNAL
Here Ω represents the frequency shift
The above equations combinely may be written as
DETECTION OF BFSK
 SPECTRUM OF BFSK SIGNAL

BANDWIDTH OF BFSK SIGNAL

PHASE SHIFT-KEYING (PSK) OR BPSK
• In this system, it shifts the phase of carrier signal according to the binary
symbols (1,0) and frequency & amplitude will be un-effected.
• In this system, generally binary symbols (1,0) modulate the phase of
carrier.
• Here, the carrier undergoes two phase reversal such as 0° and 180°.
 BPSK modulation

• Here the system consists of NRZ encoder along with product modulator
and carrier generator.
• The binary message signal is fed to the bipolar NRZ level encoder that
converts the Binary data input into equivalent bipolar NRZ sequence m(t).
• This bipolar NRZ signal is fed to the balanced modulator along with the
carrier wave.
• Thus, the binary signal modulates the carrier wave that generates a phase
shifted modulated signal termed as BPSK signal.
• The above figure shows phase reversal when the bit sequence gets changed
either from 1 to 0 or from 0 to 1.
• When the bit sequence changes from 0 to 1 then we noticed a positive
phase change whereas, when the bit sequence changes from 1 to 0 then a
negative change of phase is noticed.
Expression for BPSK
• Let us consider the carrier wave is given as
s(t) = A cos (2πfct)
• The peak of the carrier wave is represented as A.
• The power dissipated is given as,
•

• A change in phase by 180° is noticed with the corresponding change in the

bit sequence.
• Assume the carrier for symbol 1 is given as

• Similarly, in the case of symbol 0, we have,

• π represents the phase shift of 180°
• As we know cos (ɸ + π) = – cos ɸ
• Thus, s2(t) can be written as,

• Hence, BPSK signal can be written as,

• b(t) = +1 in case of transmission of binary 1

• b(t) = -1 in case of transmission of binary 0.
 BPSK Demodulation

• Let us consider, the signal at the input of the receiver is

• The phase shift ɸ is based on the time delay in between transmitter and
receiver.
• The signal is then fed to a square law device that provides
cos2(2πfct + ɸ) as its output.
• Here, only the carrier of the signal is taken into consideration thus the
amplitude is neglected.
• As we know,

• Expanding the carrier as the above mathematical identity,

• Or we can write,

• Here dc level is showed by ½

• This signal is then fed to the BPF as we can see in the diagram above.
• This BPF has a centre frequency of 2fc, eliminates the dc level hence,
generates output as
cos 2 (2πfct + ɸ)
• This signal is then further fed to a frequency divider unit. As it is frequency
divider by 2 thus generates a carrier with frequency fc.
• i.e, cos 2 (2πfct + ɸ) /2 = cos (2πfct + ɸ)
• This carrier is then multiplied with the input signal,
s(t) × cos (2πfct + ɸ)

• This signal is then given to the integrator and bit synchronizer unit.
• The signal is integrated over the 1-bit period by the integrator by making
use of bit synchronizer.
• It manages the bit duration. After a completed bit duration, synchronizer
closes S2 and the output of the integrator acts as input to the decision
device.
Advantages of Phase shift keying
• It allows more efficient transmission of radio frequency signal.
• Better noise immunity is noticed in the case of BPSK technique.
• Less bandwidth is utilized by the BPSK signal in comparison to BFSK

Disadvantages of Phase shift keying

• Detection of a BPSK signal is quite complex.
• Phase discontinuity sometimes leads to variation in amplitude of the
signal.
Applications of Phase shift keying
• PSK modulation technique finds its applications in biometric operations,
Bluetooth connectivity, wireless local area networks and in telemetry
operations.
SPECTRUM OF BPSK SIGNAL

BANDWIDTH OF BPSK SIGNAL

Quadrature Phase Shift Keying (QPSK)
QPSK TRANSMITTER
• Figure shows the block diagram of QPSK transmitter.
• Here, the input binary sequence is first converted to a bipolar NRZ signal, b(t).
• The value of b(t)= +1v for logic 1 and b(t)=-1v for logic 0.
• The demultiplexer divides b(t) into two separate bit streams, named 𝑏𝑜 (t)
• and 𝑏𝑒 (t).
• Here 𝑏𝑜 (t) consists of only odd numbered bits and 𝑏𝑒 (t) consists of only even
numbered bits.
• The symbol duration of both of these odd and even numbered sequences is 2 𝑇𝑏
• The bit streams 𝑏𝑜 (t) , 𝑏𝑒 (t) are super imposed on carriers 𝑃𝑠 cos(2 𝜋𝑓𝑐 t) and
𝑃𝑠 sin(2 𝜋𝑓𝑐 t) respectively.
• The output of the multipliers are,
• 𝑆𝑒 t = 𝑏𝑒 (t) 𝑃𝑠 sin(2 𝜋𝑓𝑐 t)
• 𝑆𝑜 t = 𝑏𝑜 (t) 𝑃𝑠 cos(2 𝜋𝑓𝑐 t)
• These two signals are basically BPSK signals which are added together to form
QPSK signal, S t
• S t =𝑆𝑜 t + 𝑆𝑒 t
• S t =𝑏𝑜 (t) 𝑃𝑠 cos(2 𝜋𝑓𝑐 t) +𝑏𝑒 (t) 𝑃𝑠 sin(2 𝜋𝑓𝑐 t)
QPSK RECEIVER
• Figure shows the QPSK receiver.
• Here synchronous reception is used.
• Therefore it is necessary to generate local carriers cos(2 𝜋𝑓𝑐 t) and sin(2 𝜋𝑓𝑐 t).
• The received QPSK signal s(t) is raised to 4th power, i.e𝑠 4(t) .
• This signal is then filtered by using a BPF with center frequency 4𝑓𝑐 .
• The output of the bandpass filter is cos 4(2 𝜋𝑓𝑐 t)
• The frequency divider which divides the frequency by 4 and generates two
carrier signals cos(2 𝜋𝑓𝑐 t) and sin(2 𝜋𝑓𝑐 t).
• The incoming signal s(t) is applied to two demodulators consisting of a
multiplier and an integrator.
• Here, the integrator integrates the product signal over two bit interval (i.e.,
𝑇𝑠 =2𝑇𝑏 )
• At the end of this period, the output of the integrator is sampled .
• The output of the multiplier b(t) is the combination of odd and even sequences.
• The output of upper integrator is 𝑏𝑒 (t) 𝑃𝑠 𝑇𝑏
• The output of lower integrator is 𝑏𝑜 (t) 𝑃𝑠 𝑇𝑏
SPECTRUM AND BANDWIDTH OF QPSK
ADVANTAGES
• Bandwidth required by QPSK is reduced to half as compared to BPSK
• Information transmission rate of QPSK is higher
• Good noise immunity
• Low error probability

DISADVANTAGES
• More complex system
 Minimum Shift Keying (MSK)
Block Diagram of MSK Transmitter
Minimum shift keying (MSK) is a special type of continuous phase-frequency shift
keying (CPFSK) with modulation index h=0.5.
Waveform of MSK
Block Diagram of MSK Receiver
Advantages
• The MSK signal has continuous phase
• MSK waveform does not have amplitude variations.
• Interchannel interference is reduced.
• Main lobe of MSK is wider than that of qpsk.
Disadvantages
• Generation and detection are complex.
• The bandwidth requirement of MSK is 1.5 fb.
 Module III
Information Theory & Coding
INFORMATION CONTENT OF SYMBOL

• Let us consider a discrete memoryless source (DMS) denoted by X and

having alphabet {x1,x2,……….xm}. The information content of a symbol xi
denoted by I(xi) is defined by

• Where P(xi) is the probability of occurrence of symbol xi.

ENTROPY
• The Entropy H(X) is defined as the average information per source symbol or
Average information per individual message.
• For a given M messages x1, x2, x3,………..xm, the entropy is defined as

• Eg.
INFORMATION RATE

• If the time rate at which source X emits symbols, is r (symbol s), the
information rate R of the source is given by
• R = rH(X) b/s
• Here R is information rate.
• H(X) is Entropy or average information and r is rate at which symbols are
generated.
• Information rater is represented in average number of bits of information
per second.
CHANNEL CAPACITY

• The channel capacity C of the channel is defined as the maximum possible

rate of information that can be transmitted through the channel.
• For a given channel, I (X,Y) will be maximum for some of probabilities P(xi).
• This maximum value is the channel capacity.
• C = max {I(X,Y)} bits/ symbols
P(Xi)
• Thus C represent the maximum information that can be transmitted per
symbol over the channel.
 IMPORTANT TERMS USED IN ERROR CONTROL CODING
• Codeword: The encoded block of ‘n’ bits is called a codeword. It contains message
bitsb(k) and redundant check bits (q).

• Block length: The number of bits ‘n’ after coding is called the block length of the
code.

• Code rate: The code rate ‘r’ is defined as the ratio of message bits (k) and the
encoder output bits (n). Hence,

Code rate, r = k/n

• Hamming distance: The hamming distance (d) between the two code vectors is
equal to the number of elements in which they differ. Eg. Let X = 101 and Y = 110.

• Then hamming distance (d) between X and Y code vectors is 2.

• Minimum hamming distance: The smallest hamming distance between the valid
code vectors is termed as the minimum hamming distance (dmin)
Shannon’s-Hartley Theorem

• Shannon’s theorem gives the capacity of a system in the presence of noise. (The capacity
C (b/s) of the AWGN (Additive White Gaussian Noise) channel)
• C = 𝐵𝑙𝑜𝑔2(1 + SNR) b/s
Or
• C = 𝐵𝑙𝑜𝑔2(1 + S/N) b/s
• Where C is the channel capacity in bits per second
• B is the bandwidth of the channel in hertz
• S is the average received signal power over the bandwidth
• N is the average noise or interference power over the bandwidth
• S/N is the signal-to-noise ratio (SNR) or the carrier-to-noise ratio (CNR)

• The shannon’s capacity theorem states that a communication system will transmit
information with the small probability of errors when the information rate R is less than
or equal to channel capacity C. i.e, R ≤ C.
• A converse of these theorem states that it is not possible to transmit the message
without error if R > C
• Thus the capacity theorem predicts essentially error free transmission over a noisy
channel.
Need for Coding
Coding Requirement
 Shannon-Fano algorithm
• List the source symbols in order of decreasing probability
• Partition the set into two sets that are as close to equiprobables as
possible, and assign 0 to the upper set 1 to the lower set.
• Continue this process, ach time partitioning the sets with as nearly equal
probabilities
• as possible until further partitioning is not possible.
• Average codeword length L = 𝑚 𝑖=1 𝑃(𝑥𝑖 ) 𝑙𝑖
• Where 𝑙𝑖 is the code word length
• Eg. Codeword = 1001 then 𝑙𝑖 = 4
𝐻(𝑥)
• Code efficiency η =
𝐿
• An example of Shannon-Fano encoding is shown in Table given below.
• The table above shows the messages 𝑥1, 𝑥2, 𝑥3, 𝑥4 having probabilities
𝑃(𝑥1) = 0.5 , 𝑃(𝑥2) = 0.25 , 𝑃(𝑥3) = 0.125 , 𝑃(𝑥4) =0.125. Construct a
Shannon-Fano code for X, and calculate the efficiency of the code.

𝑥𝑖 𝑃(𝑥𝑖 ) Step 1 Step 2 Step 3 Code 𝑙𝑖

𝑥1 0.5 0 0 1
𝑥2 0.25 1 0 10 2
𝑥3 0.125 1 1 0 110 3
𝑥4 0.125 1 1 1 111 3

• H(X) = 𝑚 𝑖=1 𝑃(𝑥𝑖 )𝑙𝑜𝑔2 𝑃(𝑥𝑖 )

• 4𝑖=1 𝑃(𝑥𝑖 )𝑙𝑜𝑔2 𝑃(𝑥𝑖 )
• [ 𝑃(𝑥1 ) 𝑙𝑜𝑔2𝑃(𝑥1 ) + 𝑃(𝑥2 ) 𝑙𝑜𝑔2𝑃(𝑥2) + 𝑃(𝑥3) 𝑙𝑜𝑔2 𝑃(𝑥3 ) +
𝑃(𝑥4 ) 𝑙𝑜𝑔2𝑃(𝑥4)
• 0.5 𝑙𝑜𝑔2 0.5 + 0.25 𝑙𝑜𝑔2 0.25 + 0.125 𝑙𝑜𝑔2 0.125 + 0.125
𝑙𝑜𝑔2 0.125
• = 1.75
 𝑚
• L= 𝑖=1 𝑃(𝑥𝑖 ) 𝑙𝑖
• L = 4𝑖=1 𝑃(𝑥𝑖 ) 𝑙𝑖
• (0.5 × 1) + (0.25 × 2) + (0.125 × 3) + (0.125 × 3)
• 1.75
𝐻(𝑥)
• Code efficiency η =
𝐿
• 1.75 / 1.75 = 1
 ERROR DETECTION AND CORRECTION CODES
Types of Errors
• There may be three types of errors:
• Single bit error

• When only one bit in the data unit has changed.

• Multiple bits error

• when two or more nonconsecutive bits in the data unit have changed
• Burst error

• when two or more consecutive bits in the data unit have changed
ERROR DETECTION : PARITY BIT METHOD
• In this method one extra bit is sent along with the original bits to make number of
1s either even in case of even parity, or odd in case of odd parity.
• The sender while creating a frame counts the number of 1s in it.
• For example, if even parity is used and number of 1s is even then one bit with
value 0 is added. This way number of 1s remains even.
• If the number of 1s is odd, to make it even a bit with value 1 is added.

• The receiver simply counts the number of 1s in a frame. If the count of 1s is even
and even parity is used, the frame is considered to be not-corrupted and is
accepted. If the count of 1s is odd and odd parity is used, the frame is still not
corrupted.
• If a single bit flips in transit, the receiver can detect it by counting the number of
1s. But when more than one bits are error occurs, then it is very hard for the
receiver to detect the error.
ERROR DETECTION AND CORRECTION CODES :
CYCLIC REDUNDANCY CHECK (CRC)
• CRC is based on binary division. A sequence of redundant bits called CRC or
CRC remainder is appended at the end of a data unit such as byte.
• At the sender side, the data unit to be transmitted is divided by a predetermined
divisor (binary number) in order to obtain the remainder. This remainder is called
CRC.
• The CRC has one bit less than the divisor. It means that if CRC is of n bits,
divisor is of n+ 1 bit.
• The sender appends this CRC to the end of data unit such that the resulting data
unit becomes exactly divisible by predetermined divisor
• At the destination, the incoming data unit i.e. data + CRC is divided by the same
number (predetermined binary divisor).
• If the remainder after division is zero then there is no error in the data unit &
receiver accepts it.
• If remainder after division is not zero, it indicates that the data unit has been
damaged in transit and therefore it is rejected.
HAMMING CODE
Non-Systematic form of Hamming Code:
• In a non-systematic block code, it is not possible to identify message bits and
check bits. They are mixed in the block. The error detection and correction
capability of non-systematic hamming code can be explained below by an
example.
• Consider a data (message) block of 1 1 0 1. The hamming code adds three
parity bits to the message bits in such a way that both message bits (data bits)
and check bits (parity check bits) get mixed. The check bit locations are as
shown below.

• Here p1, p2 and p3represent the parity check bits to be added. D represents the
data (message) bits. Then we have

• The first parity bit, p1 provides even parity from a check of bit locations 3, 5
and 7. Here they are 1, 1 and 1 respectively. Hence p1 will therefore be 1 to
achieve even parity.
• The second parity bit, p2 checks locations 3, 6 and 7. Here they are 1, 0 and 1
respectively. Hence p2 will be 0 to achieve even parity.
• The third parity bit p3, checks locations 5, 6 and 7. Here they are 1, 0 and 1
respectively. Hence p3 will be 0 to achieve even parity.
• The resulting 7-bit code word generated is as below.

• Suppose that this code word is altered during transmission. Assume that location
5 changes from 1 to 0. Hence the received code word with error is given below.

• At the decoder, we have to evaluate the parity bits to determine where error
occurs. This is accomplished by assigning a 1 to any parity bit which is incorrect
and a 0 to the parity bit which is correct.
• We check p1 for locations 3, 5 and 7. Here they are 1, 0 and 1. For even
parity p1 should be 0, but we have received p1 as 1, which is incorrect. We
assign a 1 to p1.
• We check p2 for locations 3, 6 and 7. Here they are 1, 0 and 1 respectively.
For even parity p2 should be 0 and we have also received p2 as 0, which is
correct. We assign a 0 to p2.
• We check p3 for locations 5, 6 and 7. Here they are 0, 0 and 1 respectively.
For even parity p3 should be 1, but we have received p3 as 0, which is
incorrect. We assign a 1 to p3.
• The three assigned values result in the binary form of 1 0 1, which has a
decimal value of 5. This means that the bit location containing the error is
5. The decoder then change the 5th location bit from 0 to 1.
• • The hamming code is therefore capable of locating a single error. But it
fails if multiple errors occur in one data block.
CONVOLUTIONAL CODES

• Error-correcting codes (ECC) are a sequence of numbers generated by

specific algorithms for detecting and removing errors in data that has been
transmitted over noisy channels.
• ECCs can be broadly categorized into two types, block codes and
convolution codes.
• Convolution coding is a popular error-correcting coding method used to
improve the reliability (quality) of communication system.
• A message is convoluted, and then transmitted into a noisy channel.
• This convolution operation encodes some redundant information into the
transmitted signal, thereby improving the data capacity of the channel.
• Convolution codes are error detecting codes used to reliably transmit
digital data over unreliable communication channel system to channel
noise.
• It encodes the entire data stream, into a single codeword.
• It does not need to segment the data stream into blocks of fixed size
• It is a machine with memory
• For generating a convolutional code, the information is passed
sequentially through a linear finite-state shift register.
• Convolutional codes are generated using shift registers and Boolean
function generators (Adders).
• A convolutional code can be represented as (n, k, K) where
• k is the number of bits shifted into the encoder at one time.
Generally, k = 1.
• n is the number of encoder output bits corresponding to k information
bits.
• The code-rate, Rc = k/n .
• The encoder memory, a shift register of size K, is the constraint length.
• n is a function of the present input bits and the contents of K.
• Convolutional codes are typically decoded using the Viterbi algorithm
CONVOLUTIONAL CODES (ENCODER)
Let us consider a convolutional encoder with k = 1, n = 2 and K = 3.

Let us consider a convolutional encoder with k = 1, n = 2 and K = 3.

 LIMITATION OF FORWARD ERROR CORRECTION
(FEC) CODES
• Forward error correction (FEC) is an error correction technique to detect
and correct a limited number of errors in transmitted data without the
need for retransmission.
• In this method, the sender sends a redundant error-correcting code along
with the data frame. The receiver performs necessary checks based upon
the additional redundant bits. If it finds that the data is free from errors, it
executes error-correcting code that generates the actual frame. It then
removes the redundant bits before passing the message to the upper
layers.

• Its main limitation is that if there are too many errors, the frames need to
be retransmitted.
• Requires large bandwidth
• It does not work with analog communication
 INTERLEAVING
• It separates the codeword symbols in time.
• It transforms channel with memory to memoryless channel.
• It enables random error correcting codes in a burst noise channel.
• Interleaving is applied to coded message before transmission and
deinterleaving after reception.
• The burst of errors is spread out in time and are handled by decoder.

• Interleaver which shuffles code symbols over span of several block lengths is
known as block interleaver and one which shuffles over several constraint
lengths is known as convolutional interleaver.
• The span required in determined by burst duration. Figure depicts codewords
without and with interleaving.
• The 2 types of interleavers used in communications are Block, Convolutional
• Interleaving is frequently used in digital communication and storage
systems to improve the performance of forward error correcting codes.

BLOCK INTERLEAVER
• A block interleaver accepts coded symbols in blocks from encoder, shuffles
the symbols and then feeds the rearranged symbols to the data
modulator.
• The shuffling of block is accomplished by filling the columns of an M- row
by N - column ( M X N) array with encoded sequence.
• After the array is filled, these symbols are fed to the modulator one row at
a time and transmitted over the channel.
• At the receiver, the deinterleaver performs inverse operation, the symbols
are entered by rows and removed one column at a time.
• Example : As shown, 24 code symbols are place into the interleaver input.
The output sequence to the transmitter consists of code symbols removed
from array by rows.
• Block interleaver is further categorized into various subtypes such as
Matrix Interleaver, Helical Interleaver, Random Interleaver and Odd-Even
Interleaver.
CONVOLUTIONAL INTERLEAVER

• The code symbols are sequentially shifted into bank of "N" registers.
• Each successive register provides "J" symbols more storage than preceding
one.
• The zeroth register provides no storage.
• The new code symbol is shifted in while the oldest symbol is shifted out to
the modulator or transmitter.
 Module IV
Transmission Techniques And Error Control
 Data Transmission Methods
Simplex
• In simplex transmission mode, the communication between sender and
receiver occurs in only one direction (unidirectional).
• The sender can only send the data, and the receiver can only receive the
data. The receiver cannot reply to the sender.
• To take a keyboard / monitor relationship as an example, the keyboard can
only send the input to the monitor, and the monitor can only receive the input
and display it on the screen. The monitor cannot reply, or send any feedback,
to the keyboard.

Half Duplex
• The communication between sender and receiver occurs in both directions
(bi-directional) in half duplex transmission, but only one at a time.
• The sender and receiver can both send and receive the information, but only
one is allowed to send at any given time.
• For example, in walkie-talkies, the speakers at both ends can speak, but they
have to speak one by one. They cannot speak simultaneously.
Full Duplex
• In full duplex transmission mode, the communication between sender and
receiver can occur simultaneously.
• The sender and receiver can both transmit and receive at the same time.
• For example, in a telephone conversation, two people communicate, and
both are free to speak and listen at the same time.
 Synchronous and Asynchronous Data Transmission

• Data transmission is the process of sending data from the transmitter to

the receiver.
• There are two types of data transmission known as Parallel Transmission
and Serial Transmission.
• Serial transmission sends one bit at a time, sequentially over the
communication channel.
• In serial transmission, there is a single channel between the sender and
the receiver and the bits lined at the sending device for transmission go
one after the other sequentially.
• Parallel Transmission sends multiple bits over several parallel channels at
the same time.
• Synchronous and asynchronous transmission are two types of serial
transmission.
 Synchronous Transmission

• In this transmission, the transmitter clock and the receiver clock are in
synchronization, therefore, they run at the same rate.
• It transmits block by block or frame by frame at a single time within fixed time
intervals. Furthermore, it does not have overhead with extra header and footer
bits.
• In brief, synchronous transmission is fast, efficient, reliable and allows a large
amount of data transferring.
• Some examples of synchronous transmission are chat rooms, video
conferencing, telephone conversations, etc.
Asynchronous Transmission
• Asynchronous Transmission, also called as start/stop transmission, sends
data from sender to receiver using flow control method.
• It does not use a clock to synchronize data between the source and the
destination.
• This transmission sends one character or 8 bits at a time.
• Before transmission each character it sends the start bit.
• After sending the character it sends the stop bit.
• With the character bits and start and stop bits, the total number of bits in
10 bits.
• In brief, it is a simple and cost-effective transmission method.
• Some examples of asynchronous transmission are Emails, television, and
radios
MULTIPLEXING TECHNIQUES

• Multiplexing may be defined as a technique which allows many users to

share a common communication channel simultaneously.
• There are two major types of multiplexing techniques. They are
– Frequency Division Multiplexing (FDM)
– Time Division Multiplexing (TDM)
FREQUENCY DIVISION MULTIPLEXING (FDM)
• FDM is a analog multiplexing technique.
• The operation of frequency division multiplexing (FDM) is based on sharing
the available bandwidth of a communication channel among the signals to be
transmitted .
• This means that many signals are transmitted simultaneously with each signal
occupying a different frequency slot within a common bandwidth .
• Each signal to be transmitted modulates a different carrier . The modulation
can be AM,SSB, FM or PM .
• The modulated signals are then added together to form a composite signal
which is transmitted over a single channel .
FDM TRANSMITTER
• Each signal modulates a separate carrier .
• The modulator outputs are added together in a linear mixer or adder .
• 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 .
• The composite signal at the output of mixer is transmitted over the single
communication channel
FDM RECEIVER
• The composite signal is applied to a group of bandpass filters (BPF) .
• 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 .
ADVANTAGES
• Large no. of signals can be transmitted simultaneously.
• It does not need synchronization
• Demodulation of FDM is easy
DISADVANTAGES
• The channel must have a large bandwidth
• Intermodulation distortion takes place
• Large no. of modulators are needed
APPLICATION
• Telephone system
• TV Broadcasting
• AM and FM broadcasting
TIME DIVISION MULTIPLEXING (TDM)

• TDM is the digital multiplexing technique.

• In TDM, the channel/link is not divided on the basis of frequency but on the
basis of time.
• Total time available in the channel is divided between several users.
• Each user is allotted a particular a time interval called time slot or time slice
during which the data is transmitted by that user.
• Thus each sending device takes control of entire bandwidth of the channel
for fixed amount of time.
• In TDM all the signals to be transmitted are not transmitted simultaneously.
Instead, they are transmitted one-by-one.
• Thus each signal will be transmitted for a very short time. One cycle or
frame is said to be complete when all the signals are transmitted once on
the transmission channel.
ADVANTAGES
• Full available channel bandwidth can be utilized for each channel.
• Intermodulation distortion is absent.
• TDM circuitry is not very complex.
• The problem of crosstalk is not severe.
DISADVANTAGES
• Synchronization is essential for proper operation.
• Due to slow narrowband fading, all the TDM channels may get wiped out.
APPLICATIONS
• It used in ISDN (Integrated Services Digital Network) telephone lines.
• It is used in PSTN (public switched telephone network).
• TDM is used in Satelite Acess system
• Landline phone system
 SWITCHING TECHNIQUES
• In large networks there might be multiple paths linking sender and receiver.
Information may be switched as it travels through various communication
channels. The different types of switching techniques are
– Circuit Switching
– Packet Switching
– Message Switching
CIRCUIT SWITCHING
• Circuit switched network consists of a set of switches connected by physical links.
• In circuit switched network, two nodes communicate with each other over a
dedicated communication path.
• There is a need of pre-specified route from which data will travel and no other data
is permitted.
• Before starting communication, the nodes must make a reservation for the
resources to be used during the communication.
• In this type of switching, once a connection is established, a dedicated path exists
between both ends until the connection is terminated.
• The end systems, such as telephones or computers are directly connected to a
switch.
• When system A needs to communicate with system B, system A needs to request a
connection to system B that must be accepted by all switches as well as by B itself.
• This is called as setup phase in which a circuit is reserved on each link, and the
combination of circuits or channels defines a dedicated path.
• After the establishment of the dedicated circuit, the data transfer can take place.
• After all data has been transferred, the circuit is torn down.
ADVANTAGES
– The communication channel (once established) is dedicated.
DISADVANTAGES
– Possible long wait to establish a connection
– More expensive than any other switching techniques
– Inefficient use of the communication channel
PACKET SWITCHING
• Packet-switched describes the type of network in which relatively small units of
data called packets are routed through a network based on the destination address
contained within each packet.
• Breaking communication down into packets allows the same data path to be
shared among many users in the network.
• There are two methods of packet switching: Datagram and Virtual Circuit.
• In datagram network, each packet is routed independently through the network.
• Each packet carries a header that contains the full information about the
destination.
• When the switch receives the packet, the
destination address in the header of the packet
is examined; the routing table is consulted to
find the corresponding port through which the
packet should be forwarded.
• Virtual circuit packet switching establishes a fixed path between a source and
a destination to transfer the packets.
• It is also called as connection oriented network.
• Source and destination have to go through three phases in a virtual circuit
packet switching:
I. Setup phase
ii. Data transfer phase
iii. Connection release phase
• A logical connection is established when a sender sends a setup request to the
receiver and the receiver sends back an acknowledgement to the sender if the
receiver agree.
• All packets belonging to the same source
and destination travel the same path.
• The information is delivered to the
receiver in the same order as transmitted
by the sender.
ADVANTAGES
• Packet switching is cost effective
• Packet switching offers improved delay characteristics
• Many network users can share the same channel at the same time.
• Packet switching can maximize link efficiency.
DISADVANTAGES
• Protocols for packet switching are typically more complex.
• Implementation cost is high.
• If packet is lost, sender needs to retransmit the data.
MESSAGE SWITCHING
• In message switching, it is not necessary to establish a dedicated path
between transmitter and receiver.
• In this, each message is routed independently through the network.
• Each message carries a header that contains the full information about the
destination.
• Each intermediate device receives the whole message and buffers it until
there are resources available to transfer it to the next hop.
• If the next hop does not have enough resources to accommodate large size
message, the message is stored and switch waits.
• For this reason a message switching is sometimes called as Store and Forward
Switching.
• Message switching is very slow because of store-and-forward technique.
• Message switching is not recommended for real time applications like voice and
video.
ADVANTAGES
• Channel efficiency can be greater compared to circuit-switched systems, because
more devices are sharing the channel.
• Traffic congestion can be reduced, because messages may be temporarily stored
in route.
• Message priorities can be established due to store-and-forward technique.
• Message broadcasting can be achieved with the use of broadcast address
appended in the message
DISADVANTAGES
• Message switching is not compatible with interactive applications.
• Store-and-forward devices are expensive, because they must have large disks to
hold potentially long messages
ERROR CONTROL METHODS
• There are three types of techniques available to control the errors by Automatic
Repeat Requests (ARQ):

REQUIREMENTS FOR ERROR CONTROL MECHANISM

• POSITIVE ACK - When the receiver receives a correct frame, it should
acknowledge it.
• NEGATIVE ACK - When the receiver receives a damaged frame or a
duplicate frame, it sends a NACK back to the sender and the sender must
retransmit the correct frame.
• RETRANSMISSION - The sender maintains a clock and sets a timeout period.
If an acknowledgement of a data-frame previously transmitted does not arrive
before the timeout the sender retransmits the frame, thinking that the frame or
it’s acknowledgement is lost in transit.
 STOP-AND-WAIT ARQ
The following transition may occur in Stop-and-Wait ARQ
• The sender maintains a timeout counter.
• When a frame is sent, the sender starts the timeout
counter.
• If acknowledgement of frame comes in time, the sender
transmits the next frame in queue.
• If acknowledgement does not come in time, the sender
assumes that either the frame or its acknowledgement is
lost in transit (transmission path). Sender retransmits
the frame and starts the timeout counter.
• If a negative acknowledgement is received, the sender
retransmits the frame.
 GO-BACK-N ARQ
• Stop and wait ARQ mechanism does not utilize the
resources at their best. Till the acknowledgement is
received, the sender sits idle and does nothing. In Go-
Back-N ARQ method, both sender and receiver
maintain a window.
• The sending-window size enables the sender to send
multiple frames without receiving the
acknowledgement of the previous ones.
• The receiving-window enables the receiver to receive
multiple frames and acknowledge them.
• The receiver keeps track of incoming frame’s
sequence number.
• When the sender sends all the frames in window, it
checks up to what sequence number it has received
positive acknowledgement.
• If all frames are positively acknowledged, the sender
sends next set of frames.
• If sender finds that it has received NACK or has not
receive any ACK for a particular frame, it retransmits
all the frames after which it does not receive any
positive ACK.
 SELECTIVE REPEAT ARQ
• In Go-back-N ARQ, it is assumed that the
receiver does not have any buffer space for its
window size and has to process each frame as
it comes.
• This enforces the sender to retransmit all the
frames which are not acknowledged.
• In Selective-Repeat ARQ, the receiver while
keeping track of sequence numbers, buffers
the frames in memory and sends NACK for
only frame which is missing or damaged.
• The sender in this case, sends only packet for
which NACK is received.
 PUBLIC KEY CRYPTOGRAPHY

• A form of cryptography in which the key used to encrypt a message differs

from the key used to decrypt it.
• In public key cryptography, a user has a pair of cryptographic keys—a
public key and a private key.
• The private key is kept secret, while the public key may be widely
distributed.
• The two main branches of public key cryptography are:
1. Public key encryption
2. Digital signatures
 PUBLIC KEY ENCRYPTION

• A message encrypted with a recipient's public key cannot be decrypted by

anyone except the recipient possessing the corresponding private key.
• Actual algorithms - two linked keys:
• Step 1: The most common ones have the property that Alice and Bob each
own two keys, one for encryption and one for decryption
• Step 2: Alice publish a public key to send her a message. And has a private
key to decrypt it.
• Step 3: Now Bob send Alice a message using the public key and Alice
decrypt it using her private key.

Key Terms
Public and private keys
• This is a pair of keys that have been selected so that if one is used for
encryption, the other is used for decryption.
• The exact transformations performed by the algorithm depend on the
public or private key that is provided as input.
Plain text
• Information that can be directly read by humans or a machine.
• It is the readable message or data that is fed into the encryption algorithm
as input.
Cipher
• A cipher is a cryptographic algorithm

Cipher text
• Cipher text is encrypted text transformed from plaintext using an
encryption algorithm.
• Cipher text can't be read until it has been converted into plaintext
(decrypted) with a key.
• The decryption cipher is an algorithm that transforms the cipher text back
into plaintext.

Encryption
• The process of converting plaintext to ciphertext

Decryption
• The process of reverting ciphertext to plaintext
RSA (Rivest–Shamir–Adleman) Algorithm
 Digital Signature
• A digital signature is a mathematical technique used to validate the
authenticity and integrity of a message, software or digital document.

Model of Digital Signature

• Each person adopting this scheme has a public-private key pair.
• Generally, the key pairs used for encryption/decryption and signing/verifying
are different. The private key used for signing is referred to as the signature key
and the public key as the verification key.
• Signer feeds data to the hash function and generates hash of data.
• Hash value and signature key are then fed to the signature algorithm which
produces the digital signature on given hash. Signature is appended to the data
and then both are sent to the verifier.
• Verifier feeds the digital signature and the verification key into the verification
algorithm. The verification algorithm gives some value as output.
• Verifier also runs same hash function on received data to generate hash value.
• For verification, this hash value and output of verification algorithm are
compared. Based on the comparison result, verifier decides whether the digital
signature is valid.
• Since digital signature is created by ‘private’ key of signer and no one else can
have this key; the signer cannot reject signing the data as unauthorized in
future.
Sreenath Narayanan
Lecturer in ECE Dept.
SNPTC Kanhangad