DC Unit-1 Introduction

1.1 Block Diagram of Digital Communication System

The elements which form a digital communication system is represented by the
following block diagram for the ease of understanding.

Fig 1 Elements of Digital Communication System

Following are the sections of the 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.
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.

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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.

Output Signal
This is the output which is produced after the whole process.

Key takeaway
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.

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1.2 Advantages of Digital communication system over Analog communication
systems

PARAMETERS ANALOG COMMUNICATION DIGITAL COMMUNICATION

Definition Analog Communication is the Digital Communication is the

technology which uses Analog technology which uses digital
signal for the transmission of signal for the transmission of
information. information.

Noise and Get affected by Noise Immune from Noise and

Distortion Distortion

Error Probability Error Probability is high due to Error Probability is low

parallax.

Hardware Hardware is complicated and Hardware is flexible and less

less flexible than digital system. complicated than Analog
system.

Cost Low Cost High Cost

Bandwidth Low bandwidth requirement High bandwidth Requirement

Requirement

Power High power is required Low Power Requirement

Requirement

Portability Less Portable as the More portable due to

components are heavy compact equipments.

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PARAMETERS ANALOG COMMUNICATION DIGITAL COMMUNICATION

Modulation Used Amplitude and Angle Pulse coded Modulation or

Modulation PCM, DPCM etc.

Representation Analog signal can be Digital signal is represented by

of Signal represented by sine wave. square wave.

Signal Values Consists of continuous values Consists of discrete values

Example of Analog signal comprises of Digital signals are used in

Signal voice, sound etc. computers

1.3 Sampling theorem

Sampling is defined as, “The process of measuring the instantaneous values of

continuous-time signal in a discrete form.”
Sample is a piece of data taken from the whole data which is continuous in the time
domain.
When a source generates an analog signal and if that has to be digitized,
having 1s and 0s i.e., High or Low, the signal has to be discretized in time. This
discretization of analog signal is called as Sampling.

The following figure indicates a continuous-time signal x (t) and a sampled signal xs (t).
When x (t) is multiplied by a periodic impulse train, the sampled signal xs (t) is
obtained.

Fig 2 Sampled Signal

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Sampling Rate

To discretize the signals, the gap between the samples should be fixed. That gap can
be termed as a sampling period Ts.

Sampling Frequency=1/Ts=fs

Where,
 Ts is the sampling time
 fs is the sampling frequency or the sampling rate
Sampling frequency is the reciprocal of the sampling period. This sampling
frequency, can be simply called as Sampling rate. The sampling rate denotes the
number of samples taken per second, or for a finite set of values.

Nyquist Rate

Suppose that a signal is band-limited with no frequency components higher

than W Hertz. That means, W is the highest frequency. For such a signal, for effective
reproduction of the original signal, the sampling rate should be twice the highest
frequency.
Which means,

fS=2W

Where,
 fS is the sampling rate
 W is the highest frequency
This rate of sampling is called as Nyquist rate.

Sampling Theorem
The sampling theorem, which is also called as Nyquist theorem, delivers the theory of
sufficient sample rate in terms of bandwidth for the class of functions that are band
limited.
The sampling theorem states that, “a signal can be exactly reproduced if it is sampled
at the rate fs which is greater than twice the maximum frequency W.”
Let us consider a band-limited signal, i.e., a signal whose value is non-zero between
some –W and W Hertz.

Such a signal is represented as x(f)=0 for ∣f∣>W

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For the continuous-time signal x (t), the band-limited signal in frequency domain, can
be represented as shown in the following figure.

Fig 3 Band limited Signal

If the signal x(t) is sampled above the Nyquist rate, the original signal can be
recovered, and if it is sampled below the Nyquist rate, the signal cannot be recovered.
The following figure explains a signal, if sampled at a higher rate than 2w in the
frequency domain.

Fig 4 Fourier Transform of xs(t)

The above figure shows the Fourier transform of a signal xs (t).
If fs<2W
The resultant pattern will look like the following figure.

Fig 5 Output Waveform

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Here, the over-lapping of information is done, which leads to mixing up and loss of
information. This unwanted phenomenon of over-lapping is called as Aliasing.
Aliasing
Aliasing can be referred to as “the phenomenon of a high-frequency component in the
spectrum of a signal, taking on the identity of a low-frequency component in the
spectrum of its sampled version.”
Key Takeaways:
 If the signal x(t) is sampled above the Nyquist rate, the original signal can be
recovered, and if it is sampled below the Nyquist rate, the signal cannot be recovered.
 The over-lapping of information is done, which leads to mixing up and loss of
information. This unwanted phenomenon of over-lapping is called as Aliasing.
Examples:
The continuous-time signal x(t) = cos(200πt) is used as the input for a CD converter
with the sampling period 1/300 sec. Determine the resultant discrete-time signal
x[n].
Solution:
We know,
X[n] =x(nT)
= cos(200πnT)
= cos(2πn/3) , where n= -1,0,1,2……
The frequency in x(t) is 200π rad/s while that of x[n] is 2π/3.
Determine the Nyquist frequency and Nyquist rate for the continuous-time signal
x(t) which has the form of:
X(t) = 1+ sin(2000πt) + cos (4000πt)
Solution:
The frequencies are 0, 2000π and 4000π.
The Nyquist frequency is 4000π rad/s and the Nyquist rate is 8000π rad/s.
Consider an analog signal
Find the Nyquist rate?
Solution.
The frequency in the analog signal

The largest frequency is

The Nyquist rate is

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The analog signal
 What is the Nyquist rate for this signal?
 Using a sampling rate

. What is discrete time signal obtained after sampling?

 What is analog signal

we can reconstruct from the samples if we use ideal interpolation?

Solution.
 The frequency of the analog signal are

 For

For ,the folding frequency is

Hence is not effected by aliasing
Is changed by the aliasing effect

Is changed by the aliasing effect

So that normalizing frequencies are

The analog signal that we can recover is

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Which is different than the original signal
For
 Find the minimum sampling rate required to avoid aliasing.
 If

, what is the discrete time signal after sampling?

 If

, what is the discrete time signal after sampling?

 What is the frequency F of a sinusoidal that yields sampling identical to
obtained in part c?
Solution. a.
The minimum sampling rate is

And the discrete time signal is

b. if , the discrete time signal is

c. If Fs=75Hz , the discrete time signal is

d. For the sampling rate

in part in (c). Hence

So, the analog sinusoidal signal is

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Suppose a continuous-time signal x(t) = cos (Ø0t) is sampled at a sampling
frequency of 1000Hz to produce x[n]: x[n] = cos(πn/4)
Determine 2 possible positive values of Ø0, say, Ø1 and Ø2. Discuss if cos(Ø1t) or
cos(Ø2t) will be obtained when passing through the DC converter.
Solution:
Taking T= 1/1000s
cos(πn/4) =x[n] = x(nT) = cos (Ø0n/1000)

Ø1 is easily computed as
Ø1 = 250π
Ø2 can be obtained by noting the periodicity of a sinusoid:

Ø2n/1000)
Ø2 = 2250π
1.4 Signal reconstruction in time domain

Fig 6 (a) Sampler (b) Output

The sampled data signal is modified by the controller. The hold circuit than converts the
signal to analog form. The simplest hold circuit is ZOH (zero order hold) in which the
reconstructed signal acquires the same value as the last received sample for the entire
sampling period.
The basic sampler is shown in above figure (a) and output in figure (b). The high
frequency signal present in the reconstructed signal is filtered by the controller
elements which are the low pass in frequency behavior.

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The first or higher order holds have no advantage over the ZOH. In the first order hold
the last two signal samples are used to reconstruct the signal for the current sampling
period.
Data hold is a process of generating a continuous-time signal h(t) from a discrete time
signal x(kT). The signal x(kT) can be approximated by a polynomial as
h(kT+ )=an n+an-1 n-1+….+a
1 +a0 0≤ ≤ T
h(kT)=x(kT)
h(kT+ )=an n+an-1 n-1+….+a
1 +x(kT)
If the data hold circuit is an nth order polynomial it is called as nth order hold circuit.
Zero Order Hold (ZOH)
If n=0 in above equation the zero-order hold is obtained.
h(kT+ )= x(kT) 0≤ ≤ T k=0,1,2,3…….

Fig 7 Zero Order Hold

h(t)=x(0)[u(t)-u(t-T)]+x(t)[u(t-T)-u(t-2T)]+x(2T)[u(t-2T)-u(t-3T)]……..

=
Taking Laplace transform of above equation we get

L[u(t-kT)] =

L[h(t)]=L{ }

H(s)= =

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Gho(s)=

X*(s)=

The transfer function of ZOH is Gho(s)=

Key takeaway

The transfer function of ZOH is Gho(s)=

1.5 Practical and Flat Top Sampling

During transmission, noise is introduced at top of the transmission pulse which can be easily
removed if the pulse is in the form of flat top. Here, the top of the samples are flat i.e. they have
constant amplitude. Hence, it is called as flat top sampling or practical sampling. Flat top sampling
makes use of sample and hold circuit.

Fig 8 Flat top Sampled Signal

Fig 9 Flat top signal obtained after convolution

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 The sampled signal can be obtained by convolution of rectangular pulse p(t) with ideally
sampled signal say yδ(t) as shown in the diagram above

y(t) = p(t) x yδ(t)

Taking F.T of above signal we get

From convolution

Key takeaway

 This is indeed a strategy with the least noise disturbance to that of the measured signal.
 This is accomplished by making the sequencing mechanism or feature multiplies the input
message.
 This seems to be a convenient tool used to analyse signals.

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1.6 Sampling of Band-pass Signal

The bandpass signal x(t) which maximum bandwidth is 2W can be completely represented into
and recovered from its samples if it is sampled at the minimum rate of twice the bandwidth. The
spectrum is centred around frequency fc. The bandwidth is 2W. Thus, the frequencies in the
bandpass signals are from fc-W to fc+W. The bandpass signal is represented in its in-phase and
quadrature components.

Fig 10 Spectrum of Bandpass Signal

x1(t) = in-phase component

xQ(t) = Quadrature component
Then signal x(t) can be written as
x(t) = x1(t)cos(2πfct)-xQ(t) sin (2 πfct)
The spectrum of the above signal is in between -W to +W. This is shown in below figure

Fig 11 Spectrum of in-phase and quadrature components of Bandpass signal x(t)

The reconstructed signal is then given by

X(t)= πfc(t- )]
From above equation we can conclude that

= x(nTs)

Ts =
Key takeaway
For band pass signal with bandwidth 2W the minimum sampling rate is twice the bandwidth.

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1.7 Aliasing Problem
 When fs>2 fm due to aliasing there is loss of information as seen from the above figure. The
signal components do not possess fm. Some components are outside the bandwidth.
 In order to avoid aliasing the anti-aliasing filters are used. The cut off frequency fs/2 of the filter
needs to be higher than the bandwidth of the system.
 There is dead time in the system due to conversion of signals. Hence, the sampling interval is
selected in such a manner that the stability limit of the closed loop control system as the sampling
interval is increased.
 As the signal is converted to digital form errors occur. As the sampling interval increases the
error also increase.

1.8 Uniform and Non-uniform quantization

Quantization noise is a model of quantization error introduced by quantization in the analog-to-

digital conversion (ADC). It is a rounding error between the analog input voltage to the ADC and
the output digitized value. The noise is non-linear and signal-dependent.

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.

Fig 12 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.

Fig 13 Non-Uniform Quantization

Key takeaway

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.

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1.9 Signal to Quantization ratio of Quantized Signal.

The signal-to-noise ratio is

SNR = average signal power /average noise power

For uniform quantization noise

average signal power ≈ am2p (a ≈ 1 2 )

quantization error ≈ 1/3 (mp/L)2

SNR ≈ cL2 = c22m where m is the number of bits in the PCM sample, so L = 2m. c is a constant

SNR grows exponentially with the number of bits.

If we measure SNR in dB,

SNRdB = 10 log10(c22m) = 10 log10(c) + 2m log102 = (α + 6m)dB

where α = 10 log10 c.

Key takeaway

Increasing n by one bit improves SNR by 6 dB! One bit quadruples SNR.

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