Department of Electronics Engineering

Lab Manual
Of
Digital Communication Lab (ECC 308)
Academic Complex, 2nd Floor, Room No.215

INDIAN INSTITUTE OF TECHNOLOGY

(Indian School of Mines), DHANBAD
JHARKHAND-826004

1
 List of The Experiments
Page
Sl.No. Name of Experiments No.
To generate uniform and normal random variables, also find
1. 03-06
the mean and variance of the distribution.
To generate continuous sine waveform, square waveform, and
2. 07-10
saw-tooth waveform.
(a) Generation of Pseudo Noise (PN) random sequence using
3. hardware. 11-16
(b) Recovery of the clock from PN sequence using Hardware.
Study of PCM modulation, and demodulation using Hardware
4. 17-22
(Multisim).
To study Pulse Code Modulation and Demodulation using
5. 23-33
MATLAB Simulink
6. To study BPSK Modulation and plot BER using MATLAB 34-39

To study QPSK/4-QAM Modulation and plot BER using

7. 40-45
MATLAB
To study the Digital Signal transmission using Quadrature
8. 46-48
Amplitude Modulation (QAM) using MATLAB Tool
To generate the line codes for a 10-bit dataset in a MATLAB
9. 49-57
simulation environment.
To analyze the Delta Modulation and Demodulation using
10. 58-63
Multisim

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 EXPERIMENT NO. –1

AIM: -
a. To generate uniform random variables, also find mean and variance of the distribution.
b. To generate normal random variables, also find mean and variance of the distribution.

SOFTWARE USED: -
MATLAB R2016a

FUNCTIONS USED: -

 rand (1,n): - To generate a defined ‘n’ number of random variables having

uniform distribution in the rangeof [0,1].
 randn (1,n): - To generate a defined ‘n’ number of random variables
having normal distribution in therange (-∞, ∞).
 mean (_): - To find the mean of the generated data for the defined function.
 variance (_): - To find the variance of the generated data for the defined function.
 plot (a,b): - For a graphical representation of data with ‘a’ belonging to the x-axis
and ‘b’ belonging to the y-axis.
 subplot (l,m,n): - Used for compiling more than one plot where l,m,n
represents a total number of rows,columns, and figure position for the
divided grid.
 hist(a,b): - For bar chart representation of data with ‘a’ belonging to x-axis and ‘b’
belonging to y-axis.

THEORY: -

a. Uniform random number generation is being carried out with the help of “rand”
function.
b. The probabilitydensity function of variables should be uniform over the range of [0,1].

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 𝑛
∑𝑖=1 𝑎𝑖 𝑓𝑖
Mean is the average value of numbers generated, symbolized as ‘µ’ can be calculated from data as ∑𝑛 𝑓𝑖
and
𝑖=1

𝑎+𝑏
; where a,b are the distribution ranges for unifrom probability density function.
can be narrowed to
2

Theoretically mean of the random numbers should be 0.5, accuracy of which depends on the number of samples
generated.

Variance is an aspect which measures the deviation of random variables from its mean value. For a uniform
2
(𝑎−𝑏)
random function ranges from [a,b] variance can be calculated by . Theoretically variance of uniform
12

random numbers distributed in [0,1] as calculated from the given formula equals to 0.83̅ .

a) Generation of random numbers having normal(gaussian) probability density function is being carried out
with the help of “randn” function. The probability density function of variables ranges in (-∞, ∞) defined by
1 −(𝑥−𝜇)2
(𝑥) = 𝑒 2𝜎2 .
𝜎√2𝜋

Mean ‘µ’ for standard normal distribution function should be zero while the variance ‘σ2’ should be one.

4
MATLAB CODE: -

a) Code for generating uniform distribution random variables

clc
clear allclose
all

a = rand (1,5500);b =
mean (a);
c = var (a);
disp (b);
disp (c); figure (1);
subplot(211);plot
(a);
title ('Uniform Distribution of Random Variables');xlabel ('Sample Number')
ylabel ('Amplitude')
subplot(212); hist(a);
title ('Histogram of Uniform Distribution');xlabel ('Numerical
Value');
ylabel ('Total no. of Occurence');

b) Code for generating normal distribution random variables

clc
clear allclose
all

a = randn (1,5500);b = mean

(a);
c = var (a);
disp (b);
disp (c); figure (1);
subplot(211);plot
(a);
title ('Normal Distribution of Random Variables');xlabel ('Sample
Number')
ylabel ('Amplitude')
subplot(212); hist(a);
title ('Histogram of Normal Distribution');xlabel ('Numerical
Value');
ylabel ('Total no. of Occurence');

OUTPUT: -

a) For the generated random variables having uniform density function

Mean = 0.5037; Variance = 0.0825
b) For the generated random variables having normal density function
Mean = 0.0065; Variance = 0.9911

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 Figure 1(a)

Figure 1(b)

CONCLUSIONS: -

 With the generated number of random variables the histogram is close to

theoretical uniform and normaldistributions.
 For uniform random variables error in mean and variance is observed as 0.74% and
0.96% respectively.
 For normal random variables error in mean and variance is observed as 0.65% and
0.89% respectively.
 As the number of generated random variable increases the distribution
function, mean and varianceattains standard values.

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 EXPERIMENT 2.

AIM: -

a. To generate continuous sine waveform, square waveform and sawtooth waveform.

b. To generate discrete sine waveform, square waveform and sawtooth waveform.

SOFTWARE USED: -
Matlab R2016a

FUNCTIONS USED: -
 sin(2*pi*f*t): - It generates sine waveform with time period 1/f.
 square(t): - It generates square waveform with time period 2π for the elements of
time vector t.
 sawtooth(t): - It generates sawtooth waveform with time period 2π for the
elements of time vector t.
 stem(_,_): - It provides graphical representation of data in discrete manner.
 axis([a b c d]): - It sets the limit for current axes. The four element vector
a,b,c,d specifies minimum and maximum for x and y axis respectively.
 plot (a,b): - For graphical representation of data with ‘a’ belonging to x axis and
‘b’ belonging to y axis.
 subplot (l,m,n): - Used for compiling more than one plot where l,m,n
represents total number of rows, columns and figure position for the
divided grid.

THEORY:-
The sine function generates sine wave with a period of 1/f where ‘f’ is specified
frequency.
The sawtooth function generates a sawtooth wave with peaks at +/- ‘A’ and a period
of 2π . An optional width parameter specifies a fractional multiple of 2π at which the
signal's maximum occurs.

The square function generates a square wave with a period of 2π . An optional

parameter specifies duty cycle, thepercent of the period for which the signal is positive.

Waveforms generated can be amplified or attenuated by multiplying the function by

a scalar element. Timescaling is a technique that alters the time period of the signal
thus expanding or contracting the same.

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MATLAB CODE: -
a) To generate continuous waveforms
clc clear all
close all
t = 0:0.01:10
f = 3;
a = 3*sin(2*pi*f*t);
b = 5*square(3*t);
c = 4*sawtooth(3*t);
figure(1);
subplot(311);
plot(t,a);
title ('Sine waveform');
xlabel ('Time(continuous)');
ylabel ('Amplitude')
axis ([0 3 -6 6]);
subplot(312);
plot(t,b);
axis ([0 8 -7 7]);
title ('Square waveform');
xlabel ('Time(continuous)')
ylabel ('Amplitude')
subplot(313);
plot(t,c);
axis ([0 8 -5 5]);
title ('Sawtooth waveform');
xlabel ('Time(continuous)')
ylabel ('Amplitude')

b) To generate discrete waveforms

n = 0:0.05:10
f=2
a = 3*sin(2*pi*f*n)
b = 5*square(5*n);
c = 4*sawtooth(3*n);
figure(1);
subplot(311);
stem(n,a);
title ('Sine waveform');
xlabel ('Time(discrete)')
ylabel ('Amplitude')
axis ([0 3 -6 6]);
subplot(312);
stem(n,b);
axis ([0 8 -7 7]);
title ('Square waveform');
xlabel ('Time(discrete)')
ylabel ('Amplitude')
subplot(313);
stem(n,c);

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 axis ([0 8 -5 5]);
title ('Sawtooth waveform');
xlabel ('Time(discrete)')
ylabel ('Amplitude')

OUTPUT: - a) Generated continuous sine, square and sawtooth waveforms. Figure 2(a) b)
Generated discrete sine, square and sawtooth waveforms. Figure 2(b

OUTPUT: -

a) Generated continuous sine, square and sawtooth waveforms.

Figure 2(a)

b) Generated discrete sine, square and sawtooth waveforms

9
 Figure 2(b)

CONCLUSIONS: -
a. The process of time scaling and amplification is applied and observed
along with the frequency change in the generated signals.
b. The interval distribution for the continuous signal generation
should be low to obtain optimum waveforms.
c. The time spacing for discrete signal generation should be
observed for the functional values to be differentiable.

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 EXPERIMENT NO.-3

Aim: Generation of Pseudo Noise sequence and recovery of the clock.

Objective:
(i) Verification of maximal length sequence by changing the Tapping and SEED.
(ii) Verification of amplitude and frequency of recoverd clock with input clock.

Apparatus/ Software Required:

(i) MultisimSoftware
(ii) PC/ Laptop

Theory:
Pseudo-Noise (PN) sequences are commonly used to generate noise that is approximately "white". It has
applications in scrambling, cryptography, and spread-spectrum communications. It is also commonly referred to
as the Pseudo-Random Binary Sequence (PRBS). These are very widely used in communication standards these
days. The qualifier "pseudo" implies that the sequence is not truly random. Actually, it is periodic with a
(possibly large) period, and exhibits some characteristics of a random white sequence within that period.

Pseudo random noise generator built from Linear Feedback ShiftRegister (LFSR) with judicious selection of the
XOR taps feedback path. Pseudo random number generators generate a stream of numbers in a known pattern.
The pattern is typically very long and it is hard to recognize the sequence of numbers is ordered. LFSR is a linear
feedback shift register whose input bit is a linear function of previous function that contains the signal through
the register from one bit to the next most significant when it is clocked. Figure 1 shows the block diagram of
LFSR, a 4-bit LFSR generate 24-1 different non zero bit pattern by performing exclusive-OR gate on the outputs
of two or more of the flip-flops and feeding those outputs back in to the input of one of the flip-flops.

Clock recovery from the data stream is expedited by modifying the transmitted data. Wherever a serial
communication channel does not transmit the clock signal along with the data stream, the clock must be
regenerated at the receiver, using the timing information from the data stream. Clock recovery is a common
component of systems communicating over wires, optical fibers, or by radio.

Figure1: Block diagram of LFSR

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Pin Configuration of IC74164N:

Procedures:

1. Take the length of PN sequence (N).

2. Find no. of shift registers using2n-1=N, where n is the no. of shift registers.
3. Select IC 74164N as a shift register from the component library.
4. Make the connection according to the given circuit diagram.
5. Verify that PN sequence must repeat after every 2n-1 period.
6. Apply generated PN sequence to IC 74123 to get rising and falling edge pulses.
7. After xoring, pass this to BFP followed by a comparator to get original clock.

Figure 2: Circuit diagram to generate required PN sequence.

12
 Figure 3: Interfacing IC 74123 with pn sequence generator

Figure 4: BPF with cut-off frequency equal to the clock frequency

Figure 5: Comparator Circuit

13
Observation Table:

1. Take initial sequence as 0100, 0101 and 0110.

2. Find the output after every clock.
3. Make sure the output is repeated after 2n-1 clock.

Table I

clock QA QB QC QD Output

1. 0 1 0 0
2. ….. ….. ….. …..
3. ….. ….. ….. …..
4. ….. ….. ….. …..
5. ….. ….. ….. …..
… ….. ….. ….. …..
… ….. ….. ….. …..
… ….. ….. ….. …..
… ….. ….. ….. …..
… ….. ….. ….. …..
… ….. ….. ….. …..
… ….. ….. ….. …..
… ….. ….. ….. …..
… ….. ….. ….. …..

Table II

n Primitive Polynomial

1 1+x

2 1+x+x2

3 1+x+x3 , 1+x2+x3

4 1+x+x4, 1+x3+x4

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Results:

1. The above sequences are generated and verified.

2. If n registers are used then a PN sequence with 2n-1 bits are generated if it is a maximal length sequence.
3. Clock is recovered which has slight modification in amplitude and frequency because of poorly tuned
BPF.

Figure 6: Generated PN sequence in Multisim software.

Figure 7: Rising and falling edge pulses with respect to input clock.

15
Figure 8: Output of BPF and recovered clock.

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 EXPERIMENT 4

Aim: To study Pulse Code Modulation and Demodulation technique.

Apparatus and Component Required: IC 7474, IC 7486, IC 741, Resistors and capacitor, function
generator, and connecting wires

Theory:

The pulse code modulator technique samples the input signal x(t) at a sampling frequency. This
sampled variable amplitude pulse is then digitalized by the analog to digital converter. Figure (1) shows
the PCM generator.

Figure.(1): PCM modulator

In the PCM generator, the signal is first passed through sampler which is
sampled at a rate of (fs) where:
𝑓𝑠 ≥ 2𝑓𝑚

The output of the sampler x(nTs) which is discrete in time is fed to a q level quantizer. The quantizer
compares the input x(nTs) with it's fixed levels. It assigns any one of the digital level to x(nTs) that
results in minimum distortion or error. The error is called quantization error, thus the output of the
quantizer is a digital level called q(nTs). The quantized signal level q(nTs) is binary encode. The
encoder converts the input signal to v digits binary word.

Figure.(2) A sampled signal and the quantized levels

17
Figure.(3) shows the block diagram of the PCM receiver. The
receiver starts by reshaping the received pulses, removes the noise and
then converts the binary bits to analog. The received samples are then
filtered by a low pass filter; the cut off frequency is at fc.
fc= fm
where fm: is the highest frequency component in the original signal.

Figure. (3): PCM demodulator

It is impossible to reconstruct the original signal x(t) because of the

permanent quantization error introduced during quantization at the
transmitter. The quantization error can be reduced by the increasing
quantization levels. This corresponds to the increase of bits per
sample (more information). But increasing bits (v) increases the signaling rate and requires a large
transmission bandwidth. The choice of the
parameter for the number of quantization levels must be acceptable with
the quantization noise (quantization error). Figure.(4) shows the
reconstructed signal

18
Figure (4): The reconstructed signal

Procedure:

1. The given below circuit is used for 3 bit PCM.

2. In PCM we start with the sampling of input sinusoidal 1 kHz analog signal. In sampling we
multiply input signal with 10 kHz (two times greater than input signal frequency) square wave
signal. Sampling output is taken from common collector BJT.
3. Use sampling output as one input of comparator and the other input of comparator is different
potential of Vref. For 3 bit PCM 7 comparator is use. The output of comparator is in digital
form which store in D flip flop using 7474 IC.
4. The digital output is encoded using XOR gates. The composition of XOR gates is given in
encoder circuit. Here 3 outputs are because we encode each sampled value using 3 bit.
5. In demodulation we convert encoded digital data to analog data. The output of D/A converter
is not smooth. To make it smooth (or accurate) pass from low pass filter.

19
Circuit of Sampler

20
21
Result:

Precaution:

1. The connections should be made properly and tightly.

2. Check all the connections before switching ON the kit.

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 EXPERIMENT – 05

AIM: Study of PULSE CODE MODULATION (PCM) and its DEMODULATION Scheme through
SIMULINK Models on MATLAB.
COMPONENTS REQUIRED:

 Sinusoidal Wave Generator

 Pulse Generator
 Sample & Hold
 Uniform Encoder
 Integer To Bit Converter
 Bit To Integer Converter
 Uniform Decoder
 Analog Filter Design
 Scope(S)

Simulink Model of Pulse Code Modulation (PCM)

23
BLOCK(s) used in PCM scheme:

 SINOSOIDAL WAVE GENERATOR: The Sine Wave block generates a

multichannel real or complex sinusoidal signal, with independent amplitude,
frequency, and phase in each output channel. The block supports floating point
and signed fixed-point data types.

 PULSE GENERATOR: The Pulse Generator block generates square wave

pulses at regular intervals. The block waveform parameters, Amplitude, Pulse
Width, Period, and Phase delay, determine the shape of the output waveform.
The following diagram shows how each parameter affects the waveform. The
Pulse Generator block can emit scalar, vector, or matrix signals of any real data
type. To emit a scalar signal, use scalars to specify the waveform parameters.
To emit a vector or matrix signal, use vectors or matrices, respectively, to
specify the waveform parameters. Each element of the waveform parameters
affects the corresponding element of the output signal. For example, the first
element of a vector amplitude parameter determines the amplitude of the first
element of a vector output pulse. All the waveform parameters must have the
same dimensions after scalar expansion. The data type of the output is the same
as the data type of the Amplitude parameter.

24
 SAMPLE and HOLD: The Sample and Hold block acquires the input at the
signal port whenever it receives a trigger event at the trigger port (marked by
). The block then holds the output at the acquired input value until the next
triggering event occurs.

25
UNIFORM ENCODER: The Uniform Encoder block performs the following two
operations on each floating-point sample in the input vector or matrix:
o Quantizes the value using the same precision.
o Encodes the quantized floating-point value to an integer value.

26
 INTEGER TO BIT CONVERTER: The Integer to Bit Converter block maps
each integer (or fixed-point value) in the input vector to a group of bits in the
output vector. This block is single-rate and single-channel.

 BIT TO INTEGER CONVERTER: The Bit to Integer Converter block maps

groups of bits in the input vector to integers in the output vector. If M is the
Number of bits per integer parameter, then the block maps each group of M bits
to an Integer between 0 and 2M-1. As a result, the output vector length is 1/M
times the input vector length.

 UNIFORM DECODER: The Uniform Decoder block performs the inverse

27
 operation of the Uniform Encoder block, and reconstructs quantized floating-
point values from encoded integer input. Inputs can be real or complex Values
of the following six integer data types: unit 8, uint16, uint32, int8, int16.

 ANALOG FILTER DESIGN: The Analog Filter Design block designs and
implements a Butterworth, Chebyshev type I, Chebyshev type II, elliptic, or
bessel filter in a high pass, low pass, band pass, or band stop configuration. The
Analog Filter Design block uses a state-space filter representation, and applies
the filter using the State-Space (Simulink) block in the Simulink Continuous
library. All of the design methods use Signal Processing Toolbox functions to
design the filter.
 SCOPE: The Scope block displays its input with respect to simulation time. The
Scope block can have multiple axes (one per port); all axes have a common time
range with independent y-axes. The Scope allows you to adjust the amount of
time and the range of input values displayed.

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OBSERVATION:

 Output of Scope 2 & Scope 3

Fig.1 Sine wave & Pulse generator output

29
 Fig.2 Pulse generator & Uniform Encoder Output

 Output of Scope 4

Fig. 3 Uniform Encoder & Integer to Bit Converter Output

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 Output of Scope 1

Fig.4 Sine wave, Pulse generator, Sample & hold, Uniform

Encoder, Integer to Bit Converter output

 Final Output

Fig.5 Sine wave & Analog Filter Design output

31
 SINE WAVE & ANALOG FILTER DESIGN OUTPUT:

Fig.6 Demodulated Output Along With Input

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  RESULT AND DISCUSSION:
o From a setup for PCM block diagram done in MATLAB Simulink
Software, the first input, scope-2 is shown above. The result shows that
the input analog signal is a sine wave pattern and pulse generator.

o The result display in Output scope-4 is as shown above. This signal is Pulse
Amplitude Signal (PAM) which signals after the sampling and quantizing
processes. Next, this input will go through the process of sampling that
functionally converts the analog to digital signal by using the sample and
hold which its initial condition is set to 0. The result was also formed with
the quantizer at the end. The signal would go through a quantization process
that functionally measures the numerical values of the samples and gives
suitable scale.
o SCOPE-1 shows the Output 2 scope after multiplexing blocks. From the
observation, this is due to usage of multiplexer as a device that has multiple
inputs and shows in one output. The quantized PAM signal is converted to
a serial binary code before transmission.

o FINAL OUTPUT IN THE SCOPE is the output of PCM Demodulation

and Input Sine Wave. The Demodulated Output Is Obtained after passing
through the PCM modulated output to BIT TO INTEGER CONVERTER
& then through UNIFORM DECODER and finally passing it through the
ANALOG FILTER DESIGN BLOCK.

o Based on the theoretical concept of PCM, it shows that the whole result of
simulation MATLAB SIMULINK has achieved 100% accuracy.
 CONCLUSION:
The MATLAB SIMULINK of the PCM block diagram was studied and executed.
We have successfully done a performance analysis of Pulse Code Modulation and
the output has been depicted in the figures above. In this work, according to the
basics of the PCM system, every block is implemented sequentially in MATLAB
SIMULINK. Every function of PCM system is included in a single block of
MATLAB SIMULINK, which is very helpful for the students to understand the
whole PCM system.

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 EXPERIMENT NO: 6
Aim: To study BPSK Modulation and plot BER using MATLAB

Theory:
Digital modulation:
There are three basic types of modulation methods for the transmission of a digital signal. The
methods are based on three attributes of a sinusoidal signal, amplitude, frequency and phase. The
corresponding digital modulation methods are amplitude shift keying (ASK), frequency shift
keying (FSK) and phase shift keying (PSK).
Amplitude shift keying (ASK): Amplitude shift keying (ASK) is the simple form of digital
modulation. Digital input is unipolar NRZ signal. In ASK carrier amplitude is multiplied by high
amplitude for binary “1” or by low amplitude for a binary “0”. However, when the low amplitude
is 0 for binary “0” then the ASK is called On-Off keying or OOK which shown in Figure 1. In
OOK the amplitude modulated carrier signal can be written as
𝑣(𝑡) = 𝐴 𝑆𝑖𝑛(2𝜋𝑓𝑐 𝑡)………….(1)

Fig.1
Frequency shift keying (FSK): In frequency shift keying (FSK), the frequency of the carrier is
shifted between two discrete values, one representing binary “1” and representing binary “0” but
the carrier amplitude does not changes. The simple form of FSK is BFSK. The instantaneous vale
of the FSK signal is given by
𝑣(𝑡) = 𝐴 𝑆𝑖𝑛(2𝜋𝑓1 𝑡) + 𝐴𝑆𝑖𝑛(2𝜋𝑓2 𝑡)………(2)
Where, 𝑓1 and 𝑓2 are the frequencies corresponding to binary “1” and “0” respectively and𝑓1 >
𝑓2. From above equation, it is clear that the FSK signal can be considered to be comprising of
two ASK signal with carrier frequencies f1 and f2.
34
 Fig.2
Phase shift keying (PSK):
In Phase shift keying (PSK), the phase of the carrier is modulated to represent the binary values.
The carrier phase change between 0 and 𝜋 by the bipolar digital signal. Binary states “1” and “0”
are represented by the positive and negative polarity of the digital signal. The simplest form of
PSK is BPSK is shown in Figure 3. The instantaneous value of the digital signal can be written
as
𝑣(𝑡) = 𝐴𝑠𝑖𝑛(2𝜋𝑓𝑐 𝑡)
Where, 𝐴 = ±1; 𝐴 = 1 for binary state “1” and 𝐴 = −1 for binary state “0”.

Fig.3
With Binary Phase Shift Keying (BPSK), the binary digits 1 and 0 maybe represented by the
analog levels and respectively. The system model is as shown in the Figure below.

35
Fig.4: Simplified block diagram with BPSK transmitter-receiver

Channel Model:
The transmitted waveform gets corrupted by noise , typically referred to as Additive White
Gaussian Noise (AWGN).
Additive : As the noise gets ‘added’ (and not multiplied) to the received signal
White : The spectrum of the noise if flat for all frequencies.
Gaussian : The values of the noise follows the Gaussian probability distribution

function, with and .

Computing the probability of error

The received signal,

when bit 1 is transmitted and

when bit 0 is transmitted.

The conditional probability distribution function (PDF) of for the two cases are:

36
Figure: Conditional probability density function with BPSK modulation
Assuming that and are equally probable i.e. the threshold 0 forms
the optimal decision boundary.
 if the received signal is is greater than 0, then the receiver assumes was transmitted.
 if the received signal is is less than or equal to 0, then the receiver assumes was
transmitted.
i.e.

and

Probability of error given was transmitted

With this threshold, the probability of error given is transmitted is (the area in blue region):

where,

is the complementary error function.

Probability of error given was transmitted

Similarly the probability of error given is transmitted is (the area in green region):

37
Total probability of bit error

Given that we assumed that and are equally probable i.e. , the bit
error probability is,

Simulation model:

Matlab code for computing the bit error rate with BPSK modulation from theory and
simulation. The code performs the following:

(a) Generation of random BPSK modulated symbols +1′s and -1′s.

(b) Passing them through Additive White Gaussian Noise channel.

(c) Demodulation of the received symbol based on the location in the constellation.

(d) Counting the number of errors.

(e) Repeating the same for multiple Eb/No value.

38
Result:

Fig.5

39
EXPERIMENT 7

40
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45
 EXPERIMENT- 8
1. AIM : To study the Digital Signal transmission using Quadrature Amplitude Modulation (QAM)
using MATLB Tool

2. LEARNING OUTCOME: After the competing this experiment we can learn that

 How does change the Amplitude and phase of a higher frequency signal (carrier signal) with
16-symbols baseband binary polar pulse sequence (Digital Message signals) i.e. study of 16-
QAM.

 The effects of the channel Gaussian noise on the modulated signal i.e. study of the SNR of
QAM.

3. THEORY:
The Quadrature Amplitude Modulation (QAM) is mainly used to increase the transmission rate and
decrease he transmission bandwidth. This modulation scheme uses in the both analog and digital
version of the communication. Digital signal transmission using QAM has same concept of the
analog version of the QAM .But basic difference in the base band signal, the message signals are two
digital bit streams, (binary polar pulse sequence) instead of the two analog message signals in this
case. These two base band signals are modulated (m1 (t) and m2 (t)) by a carrier of the same frequency
(cos(wct)) but in quadrature phase(sin(wct)). A basic block diagram of QAM Modulator and
demodulator are shown in Fig.1.(a).

Fig.1. a) QAM Modulator and demodulator b) 16-QAM

46
 A Mathematic representation of the digital QAM is given by Eq.(1)
𝑝𝑖 (𝑡) = 𝑎𝑖 𝑝(𝑡)𝑐𝑜𝑠(𝑤𝑐 𝑡) + 𝑏𝑖 𝑝(𝑡)𝑠𝑜𝑠(𝑤𝑐 𝑡) (1)
= 𝑟𝑖 𝑝(𝑡)𝑐𝑜𝑠(𝑤𝑐 𝑡 − 𝜃𝑖 )) 𝑖 = 1,2 … … . 𝑀
𝑏
Where 𝑟𝑖 = √𝑎𝑖 2 + 𝑏𝑖 2 and 𝜃𝑐 = tan−1 𝑎𝑖 𝑚1 (𝑡) = 𝑎𝑖 𝑝(𝑡) 𝑎𝑛𝑑 𝑚2 (𝑡) = 𝑏𝑖 𝑝(𝑡)
𝑖
p(t) is properly shaped baseband pulse . The signal pi(t) can be generated using QAM . One possible
choice of ri and Θi for 16 pulses is shown in Fig.1.(b). The transmitting pulse pi(t can take on 16
distinct forms and is, therefore , a 16-ary pulse. Since M=16, each pulse can transmit the information
on 4 binary digits. This can be done as follows: there are 16 possible sequences of four binary digits
and there are 16 combinations (ai,bi) in Fig.1.(b). Thus, every possible 4-bit sequence is transmitted
by a particular (ai,bi) or (ri , Θi ). Therefore. One signal pulse 𝑟𝑖 𝑝(𝑡)𝑐𝑜𝑠(𝑤𝑐 𝑡 − 𝜃𝑖 )) transmits 4 bits.
The bit rate quadrated without increasing the bandwidth. The transmission rate can be increased
further by increasing the value of M.
After modulation, this modulated signal will be transmitted through a channel which having the
AWGN noise with energy to noise power spectrum density E/N =10 dB.
4. Apparatus and Component required:

 MATLAB TOOL

5. Experiment Step/Simulation Flow Chart:

 MATLAB FLOW CHART

1. Define the Parameters 3. Generate the binary data stream

 Size of the signal constellation  Matlab function: randi
 Number of bits per symbol
 Number of bit to process

2. Convert binary data to Integer-

Value signal and plot 4. Plot the binary data stream
 Matlab function: bi2de  Matlab function: xlabel; ylabel

6. Adding AWGN of Eb/N0 = 10 dB with

5. 16-QAM Modulation and plot Modulated signal, Constellation plotting
Constellation QAM  Matlab function: awgn , scatterplot
 Matlab function: qammod , scatterplot
47
6. Results and observation:

 Observe the generated binary data stream and compare this with its equivalent integers
signal.

 Observe Constellation plots of the modulated 16-QAM with and without AWGN channel
noise, and compare results of the both cases.

 Comments on the SNR of the modulated signal at case of the AWGN channel noise.

7. Questions:

 What are basic differences between the digital QAM and Analog QAM?
 Why it is called the Quadrature PSK (QPSK)?
 Write the some advantages of QAM over the other modulation schemes.
 How does it increase the transmission rate and increase the bandwidth efficiency with
increase of the bit in symbol of signal?

48
 EXPERIMENT 9

Aim: Generate the line codes for a 10-bit dataset in MATLAB simulation environment.
1. Learning Outcome: After the completing this experiment we becomes familiar to generate
line codes of a binary dataset in digital communication.
2. Theory: A computer network is used for communication of data from one station to another
station in the network. We have seen that analog or digital data traverses through a
communication media in the form of a signal from the source to the destination. The channel
bridging the transmitter and the receiver may be a guided transmission medium such as a wire
or a wave-guide or it can be an unguided atmospheric or space channel. But, irrespective of the
medium, the signal traversing the channel becomes attenuated and distorted with increasing
distance. Hence a process is adopted to match the properties of the transmitted signal to the
channel characteristics so as to efficiently communicate overthe transmission media. There are
two alternatives; the data can be either converted to digital or analog signal. Both the
approaches have pros and cons. What to be used depends on the situation and the available
bandwidth.
Now, either form of data can be encoded into either form of signal. For digital signaling,
the data source can be either analog or digital, which is encoded into digital signal, using
different encoding techniques. The basis of analog signaling is a constant frequency signal
known as a carrier signal, which is chosen to be compatible with the transmission media being
used, so that it can traverse a longdistance with minimum of attenuation and distortion. Data can
be transmitted using these carrier signals by a process called modulation, where one or more
fundamental parameters of the carrier wave, i.e. amplitude, frequency and phase are being
modulated by the source data. The resulting signal, called modulated signal traverses the media,
which is demodulated at the receiving end and the original signal is extracted. A line code is a
specific code (with precisely defined parameters) used for transmitting a digital signal over a
channel. Line coding is used in digital data transport –the pattern of voltage, current used to
represent digital data on a transmission link is called line encoding.

49
 3. Linear coding characteristics:

Fig.1 Line coding to convert digital data to digital

signal

Important parameters those characteristics line coding techniques are mentioned below.
No of signal levels: This refers to the number values allowed in a signal, known as signal levels, to
represent data. Figure 2 (a) shows two signal levels, whereas Fig. 2 (b) shows three signal levels to
represent binarydata.

Fig. 2 (a) Signal with two voltage levels, (b) Signal with three voltage levels
Bit rate versus Baud rate: The bit rate represents the number of bits sent per second, whereas the
baud rate defines the number of signal elements per second in the signal. Depending on the
encoding technique used, baud rate may be more than or less than the data rate.
DC components: After line coding, the signal may have zero frequency component in the spectrum
of the signal, which is known as the direct-current (DC) component. DC component in a signal is
not desirable because the DC component does not pass through some components of a
communication system such as a transformer. This leads to distortion of the signal and may create
error at the output. The DC component also results in unwanted energy loss on the line.
Signal Spectrum: Different encoding of data leads to different spectrum of the signal. It is
necessary to use suitable encoding technique to match with the medium so that the signal suffers
minimum attenuation and distortion as it is transmitted through a medium.
Synchronization: To interpret the received signal correctly, the bit interval of the receiver should be
50
exactlysame or within certain limit of that of the transmitter. Any mismatch between the two may
lead wrong interpretation of the received signal. Usually, clock is generated and synchronized from
the received signal with the help of a special hardware known as Phase Lock Loop (PLL). However,
this can be achieved if the received signal is self-synchronizing having frequent transitions
(preferably, a minimum of one transition per bit interval) in the signal.
Cost of Implementation: It is desirable to keep the encoding technique simple enough such that it
does not incur high cost of implementation.
4. Line Coding Techniques: Line coding techniques can be broadly divided into three broad
categories: Unipolar, Polar and Bipolar as shown in fig. 3.

Fig. 3 Three basic categories of line coding techniques

4.1 Unipolar: In unipolar encoding technique, only two voltage levels are used. It uses only one
polarity of voltage level as shown in Fig. 4. In this encoding approach, the bit rate same as data
rate. Unfortunately, DC component present in the encoded signal and there is loss of
synchronization for long sequences of 0’sand 1’s. It is simple but obsolete.

Fig. 4 Unipolar encoding with two voltage levels

There are two unipolar line coding schemes- i) Unipolar RZ, ii) Unipolar NRZ (ON-OFF Keying).
5.1. i) Unipolar RZ: In unipolar RZ, the waveform has zero value when ‘0’ is transmitted and

waveform has ‘A’ volts when ‘1’ is transmitted. In RZ form, the ‘A’ volts is presented for 𝑇𝑏,
2

period if symbol ‘1’ transmitted and remaining 𝑇𝑏, waveform returns to zero value, i.e., for unipolar
RZ form.
2

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 If symbol ‘1’ is transmitted, then we have
𝐴 𝑓𝑜𝑟 0 ≤ 𝑡 < 𝑇𝑏/2
𝑇
𝑥(𝑡) = { 0 𝑓𝑜𝑟 𝑏 ≤ 𝑡 < (1)
𝑇

2 𝑏
If symbol ‘0’ is transmitted, then
(𝑡) = 0 𝑓𝑜𝑟 0 ≤ 𝑡 < 𝑇𝑏 (2)
If the input bit pattern is 1101000011, then the unipolar RZ waveform will be

52
 Fig. 5 Unipolar RZ
ii) Unipolar NRZ (ON-OFF keying): In unipolar NRZ format, when symbol ‘1’ is to be
transmitted, the symbol has ‘A’ volts for full duration. When symbol ‘0’ is to be transmitted, the
signal has zero volts for complete symbol duration.
If symbol ‘1’ is transmitted
(𝑡) = 𝐴 𝑓𝑜𝑟 0 ≤ 𝑡 < 𝑇𝑏 (3)
If symbol ‘0’ is transmitted
(𝑡) = 0 𝑓𝑜𝑟 0 ≤ 𝑡 < 𝑇𝑏 (4)

Fig. 6 Unipolar NRZ

5.2. Polar: Polar encoding technique uses two voltage levels – one positive and the other one
negative. Four different encoding schemes shown in Fig. 2.4.6 under this category discussed
below:

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 Fig. 5 Encoding Schemes under polar category

5.2. i) Polar Return to Zero RZ: To ensure synchronization, there must be a signal transition in
each bit. Key characteristics of the RZ coding are: Three levels, Bit rate is double than that of data
rate, No dc component, Good synchronization, and Main limitation is the increase in bandwidth.
In polar RZ, symbol ‘1’ is represented by positive voltage polarity whereas symbol ‘0’ is represented
by negative voltage polarity. For RZ format pulse is transmitted only for half duration.
If ‘1’ is transmitted, then
+𝐴/2 𝑓𝑜𝑟 0 ≤ 𝑡 < 𝑇𝑏/2
𝑇
𝑥(𝑡) = { 0 𝑓𝑜𝑟 𝑏 ≤ 𝑡 < (5)
𝑇

2 𝑏
If ‘0’ is transmitted, then
−𝐴/2 𝑓𝑜𝑟 0 ≤ 𝑡 < 𝑇𝑏/2 0
𝑥(𝑡) = { 𝑇 (6)
𝑓𝑜𝑟 𝑏 ≤ 𝑡 < 𝑇

54
 2 𝑏

Fig. 7 Polar RZ
ii) Polar Non Return to zero (NRZ): The most common and easiest way to transmit digital signals
is to use two different voltage levels for the two binary digits. Usually a negative voltage is used

to represent one binary value and a positive voltage to represent the other. The data is encoded as
the presence or absence of a signal transition at the beginning of the bit time. As shown in the figure
below,in NRZ encoding, the signal level remains same throughout the bit-period. There are two
encoding schemes in NRZ: NRZ-L and NRZ-I.
ii) a) NRZ-L: For polar NRZ format, symbol ‘1’ is represented by negative polarity and symbol ‘0’
is represented by positive polarity.
If symbol ‘1’ is transmitted, then
(𝑡) = 𝐴/2 𝑓𝑜𝑟 0 ≤ 𝑡 < 𝑇𝑏 (7)
If symbol ‘0’ is transmitted, then
(𝑡) = −𝐴/2 𝑓𝑜𝑟 0 ≤ 𝑡 < 𝑇𝑏 (8)

Fig. 8 Polar NRZ-L

b) NRZ-I: For polar NRZ format, symbol ‘0’ is represented by no transition state and symbol‘1’
is represented by transition state.
If symbol ‘1’ is transmitted, then

55
(𝑡) = no transition 𝑓𝑜𝑟 0 ≤ 𝑡 < 𝑇𝑏 (7)
If symbol ‘0’ is transmitted, then
(𝑡) = transition 𝑓𝑜𝑟 0 ≤ 𝑡 < 𝑇𝑏 (8)

Fig. 9 Polar NRZ-L

The advantages of NRZ coding are:
Detecting a transition in presence of noise is more reliable than to compare a value to a
threshold. NRZ codes are easy to engineer and it makes efficient use of bandwidth.
Biphase: To overcome the limitations of NRZ encoding, biphase encoding techniques can be
adopted. Manchester and differential Manchester Coding are the two common Biphase techniques
in use, as shown in Fig. 8. In Manchester coding the mid-bit transition serves as a clocking
mechanism and also as data.
ii) Manchester Code: In the standard Manchester coding there is a transition at the middle of
each bit period. A binary ‘0’ corresponds to a low-to-high transition and a binary ‘1’ to a high-to-
low transition in the middle.

Fig. 10 Manchester Code

iii) Differential Manchester code: In Differential Manchester, inversion in the middle of each bit
is used for synchronization. The encoding of a ‘0’ is represented by the presence of a transition
both at the beginning and at the middle and ‘1’ is represented by a transition only in the middle of
the bit period.
56
 Fig. 11 Differential Manchester Code
Key characteristics are: Two levels, No DC component, Good synchronization, higher bandwidth
due to doubling of bit rate with respect to data rate. The bandwidth required for biphase techniques
are greater than that of NRZ techniques, but due to the predictable transition during each bit time,
the receiver can synchronize properly on that transition.
5.3 Bipolar NRZ: Bipolar encoding technique uses Alternate Mark Inversion Technique (AMI).
Unlike RZ 0-level is used to represent a ‘0’ and a binary 1’s are represented by alternating positive
and negative voltages, as shown in Fig.12.

Fig. 12 Bipolar (AMI) NRZ

5. Apparatus and Software: Computer with MATLAB software environment.
6. Results and observation: Generate waveforms for line codes.
Take amplitude A=1; Samples/bit (Tb) = 1000; bit width = clock period and

Input Bit Pattern: [1 1 0 1 0 0 0 0 1 1]

7. Questions:
I. Why do you need encoding of data before sending over a medium?
II. What are the four possible encoding techniques? Give examples.
III. Between RZ and NRZ encoding techniques, which requires higher bandwidth and why?
IV. How does Manchester encoding differ from differential Manchester encoding?
V. How Manchester encoding helps in achieving better synchronization?
VI. Distinguish between PAM and PCM signals?
VII. What is quantization error? How can it be reduced?

57
 EXPERIMENT 10

Aim:- To analyse the Delta modulation and Demodulation using Multisim

Theory

58
59
60
61
62
XFG1

XSC1 XSC2
U1 4 U4A
R2 R12
~1PR G
Ext T rig
10kΩ 100 Ω 2 1D 1Q 5 + T
_
A B C D
D1 R9 A B

COMPARATOR_VIRTUAL 5V V5 3 1CLK ~1Q 6 1kΩ + _ + _

20kHz ~1CLR
R11 5V 1
7474N

50kΩ V6
Key=A 100% 5V

C2
V2
0.033uF V4 5V
12 V
R7
R3 4
U3 3.3kΩ
100kΩ U2
R5 R6
2
6 10kΩ 100 Ω
D3
3 5V COMPARATOR_VIRTUAL
R8
5 1 7 741 R4 3.3kΩ
1kΩ
D2
V3 5V
12 V

R17

50kΩ R1
C3 100%
Key=A
10kΩ
1nF
C1
V10 V1
12 V 0.099uF V8 5V
4 U7 12 V R15
2
4
U5 3.3kΩ
U6
6
R19 R18 R13 R14
3 2
159.2kΩ 159.2kΩ 6 6.8kΩ 100 Ω
7 1 5 741 D5 R16
V9 3 5V COMPARATOR_VIRTUAL
R10
12 V 5 1 7 3.3kΩ
741
1kΩ
C4 D4
V7 5V
1nF 12 V

XSC3

Ext T rig
+
_
A B
+ _ + _

Fig. 1. Delta Modulation and Demodulation Circuit

63