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Manual

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EC8512

AIM: -

To generate amplitude modulated wave and demodulate the AM wave using Envelope

Detector

APPARATUS REQUIRED: -

1. Transistor BC107 1

2. Diode IN4007 1

3. Function Generator 0-2MHz 2

4. DSO 100MHz 1

5. Resistor 50kΩ,15kΩ,10kΩ, Each 1

4.7k Ω, 1k Ω Each 1

6. Capacitor 470pF,100nF, Each 1

0.1mF

7. Probe 2

8. Bread Board 1

9. Regulated Power Supply 0-30v 1

THEORY:

Amplitude Modulation is defined as a process in which the amplitude of the carrier

wave c(t) is varied linearly with the instantaneous amplitude of the message signal m(t).

The demodulation circuit is used to recover the message signal from the incoming AM

wave at the receiver. An envelope detector is a simple and yet highly effective device that is well

suited for the demodulation of AM wave, for which the percentage modulation is less than

100%.Ideally, an envelope detector produces an output signal that follows the envelop of the input

signal wave form exactly; hence, the name.

PROCEDURE: -

2. A modulating signal is given as input to the circuit.

3. Now increase the amplitude of the modulating signal to the required level.

4. The amplitude and the time duration of the modulating signal are observed using CRO.

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Message Signal

Carrier Signal

Demodulated wave

Tabular Column:

1 Message signal

2 Carrier signal

3 AM wave Vmax Vmin

4 Demodulated wave

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5. Finally the amplitude modulated output is observed from the output of amplitude

modulator stage and the amplitude and time duration of the AM wave are noted down.

6. Calculate the modulation index by using the formula and verify them. The final

demodulated signal is viewed using an CRO at the output. Also the amplitude and time

duration of the demodulated wave are noted down.

Calculation:

Modulation Index:

The modulation index m is calculated that indicates by how much the modulated variable varies

around its unmodulated level. It relates to the variations in the amplitude of the carrier signal.

RESULT: -

Thus the amplitude modulated wave was generated and the modulation index is

verified. Also, the modulated wave is demodulated by passing through the envelope detector

circuit.

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Fig.2.1Circuit Diagram

Fig.2.1Pin Diagram

EC8512

AIM: -

To construct a frequency modulation circuit using a sinusoidal input waveform and to

measure the Modulation Index.

APPARATUS REQUIRED: -

1. NE 566 1

2. Function Generator 0-2MHz 1

3. DSO 100MHz 1

4. Resistor 2.5kΩ, 5.6kΩ, Each 1

39kΩ

5. Capacitor 0.01μF 2

6. Probe 1

7. Bread Board 1

8. Regulated Power Supply + or - 12v 1

THEORY:

Frequency modulation (FM) is a method of impressing data onto an alternating-current

(AC) wave by varying the instantaneous frequency of the wave.

The NE/SE566 Function Generator is a general purpose voltage-controlled oscillator designed for

highly linear frequency modulation. The circuit provides simultaneous square wave and triangle

wave outputs at frequencies up to 1MHz. A typical connection diagram is shown in Figure 2.1 The

control terminal (Pin 5) must be biased externally with a voltage (Vc) in the range

Where VCC is the total supply voltage. In Figure 2, the control voltage is set by the voltage divider

formed with R2 and R3. The modulating signal is then AC coupled with the capacitor C2. The

modulating signal can be direct coupled as well, if the appropriate DC bias voltage is applied to

the control terminal. The frequency is given approximately by

and R1 should be in the range 2k< R1<20k. A small capacitor (typically 0.001uF) should be

connected between Pins 5 and 6 to eliminate possible oscillation in the control current source. The

value of C1 is 1nF.

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Tabular Column:-

1 Message signal

2 Carrier signal

3 FM signal

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PROCEDURE: -

2. The free running frequency fo is measured at pin 3.

3. The modulating input is given to pin 5 through a coupling capacitor and the

corresponding changes in digital storage oscilloscope is noted.

4. For various amplitudes and the modulating signal, corresponding DSO readings were

noted.

5. Frequency deviation and modulation index were calculated.

6. Frequency modulated output was drawn on a graph sheet.

Calculation:

The Frequency modulation experiment demonstrates some of the principles of VCO operation

using the NE566 integrated circuit by implementing a Frequency Modulation Circuit.

The modulation index h is calculated that indicates by how much the modulated variable

varies around its unmodulated level. It relates to the variations in the frequency of the carrier signal.

∆𝒇

𝜷=

𝒇𝒎

Bandwidth:

𝑩𝑾 = 𝟐 ∗ (∆𝒇 + 𝒇𝒎 )

RESULT: -

Thus the FM modulation wave were generated and the obtained

Modulation index :

Bandwidth :

COMMUNICATION SYSTEMS LAB MANUAL Page 8

EC8512

Fig.3.1Circuit Diagram

EC8512

AIM: -

To study the process of time division multiplexing and to perform multiplex of two

set of signals.

APPARATUS REQUIRED: -

1. CL 100 NPN 1

2. CK 100 PNP 1

3. Function Generator 0-2MHz 2

4. DSO 100MHz 1

5. Resistor 33k Ω 1

6. Probe 2

7. Bread Board 1

8. Regulated Power Supply 0-30v 1

THEORY:

The Sampling Theorem provides the basis for transmitting the information contained in a

band limited message signal m (t) as a sequence of samples of m(t) taken uniformly at a rate that

is usually slighter higher than the nyquist rate. An important feature of the sampling process is a

conservation of time. That is, the transmission the message samples engages the communication

channel s for only a fraction of the sampling interval on a periodic basis, and in this way some of

the time interval between adjacent samples is cleared for use by other independent message sources

on a time shared basis. We there by obtain a time division multiplexing (TDM) system, which

enables the joint utilization of a common communication channel by a plurality of independent

message sources without mutual interference among them.

The TDM system is highly sensitive to dispersion in the common channel, that is, to variations

of amplitude with frequency or lack of proportionality of phase with frequency. Accordingly,

accurate equalization of both magnitude and phase response of a channel is necessary to ensure a

satisfactory operation of the system. Unlike FDM, TDM is immune to nonlinearities in the

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Tabular Column:-

1. Message signal 1

2. Message signal 2

3. Clock Signal

channel as a source of cross talk. The reason for this is, the different message signals are not

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simultaneously applied to the channel. The primary advantage of TDM is that several channels of

information can be transmitted simultaneously over a single cable.

PROCEDURE: -

1. Connect the circuit as per the diagram

2. Give two different amplitudes inputs from function generated to the emitters of both the

transistors

3. Connect the common clock signal to the back of both the transistors and observed in the

CRO, we can see the wave form.

4. The output is taken from the collector of both the transistors and the waveform is plotted

in graph.

5. Their positions and identification can be highlighted by reducing the other signal

amplitudes to zero and then gradually increasing them to observe them occupying their

positions.

RESULT: -

Thus, the two different signals are interleaved in their respective time slots without

overlapping each other using a single channel.

EX. NO. 4 SIGNAL SAMPLING AND RECONSTRUCTION

EC8512

Fig.4.1Circuit Diagram

RECONSTRUCTION FILTER

EC8512

AIM: -

To analyze an implementation of a sample and hold (S/H) circuit..

APPARATUS REQUIRED: -

1. IC 741 1

2. Function Generator 0-2MHz 3

3. DSO 100MHz 1

4. Resistor 33k Ω 1

5. Probe 2

6. Bread Board 1

7. Regulated Power Supply 0-30v 1

THEORY:

The analog signal can be converted to a discrete time signal by a process called sampling.

The sampling theorem for a band limited signal of finite energy can be stated as,

‘’A band limited signal of finite energy, which has no frequency component higher than W

Hz is completely described by specifying the values of the signal at instants of time separated by

1/2W seconds.’’

It can be recovered from the knowledge of samples taken at the rate of 2W per second.

In the Circuit, the switching rate is controlled by a clock signal whose frequency should satisfy the

Nyquist sampling criterion. The switch is a FET whose gate is controlled by the clock pulse.

Buffers are placed at the input and output to isolate the circuit.

When the switch S is closed, the capacitor C is charged to the value of the input voltage,

the sample stage. Afterwards the switch is opened and the capacitor retains its charge, the hold

stage. Signal is reconstructed after passing the sampled signal into the low pass filter.

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Tabular Column:-

1. Message signal

2. Clock Signal

3. Sampled

4. Reconstructed Signal

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PROCEDURE: -

2. Give the continuous time message signal from the functional generator.

3. Connect the clock signal to the Gate of FET and observe the sampled signal in the CRO.

Analyze by varying the clock signal frequency.

4. The sampled waveform can be reconstructed by passing it through an low pass filter.

RESULT: -

Thus the continuous-time signals are sampled and then signals are reconstructed from the samples

at the receiver side.

EC8512

Formula:

Symbol error rate = No. of symbols in error after detection / No. of symbols transmitted

EC8512

AIM:

To study the SNR performance of Pulse Code Modulation (PCM)

APPARATUS REQUIRED: -

1. Desktop Computer 1

2. Matlab software 1

THEORY

Pulse code modulation (PCM) is a digital scheme for transmitting analog data. The

signals in PCM are binary; that is, there are only two possible states, represented by logic 1

(high) and logic0 (low). Using PCM, it is possible to digitize all forms of analog data, including

full-motion video, voices, music, telemetry, and virtual reality. To obtain PCM from an analog

waveform at the source of a communications circuit, the analog signal amplitude is sampled at

regular time intervals. The sampling rate, or number of samples per second, is several times

the maximum frequency of the analog waveform in cycles per second or hertz. The

instantaneous amplitude of the analog signal at each sampling is rounded off to the nearest of

several specific, predetermined levels. This process is called quantization. The number at each

level can be represented by three, four, five, or six binary digits (bits) respectively. The output

of a pulse code modulator is thus a series of binary numbers, each represented by some power

of 2bits.

At the destination of the communications circuit, a pulse code demodulator converts the

binary numbers back into pulses having the same quantum levels as those in the modulator.

These pulses are further processed to restore the original analog waveform.

PROGRAM:

EC8512

Am=2;

fm=1000; %Input signal frequency

fs=OSR*2*fm; %Sampling frequency

x=Am*cos(2*pi*fm*n/fs); %generation of the sinusoidal signal

Mlen=length(MM);

for ii=1:Mlen

M=MM(ii)

nbit(ii)=ceil(log(M)/log(2)); % number of bits used for quantisation

qv=[-Am:delta:Am]; % quantisation values

y=round((x+Am)/delta)*delta-Am; %quantised value

t=n/fs;

figure;

plot(t,x,'r-'); %ploting analog signal

title('Input and Sampled signals'); xlabel('time in seconds'); ylabel('amplitude');

axis([t(1), t(end), -1.1*Am, 1.1*Am])

hold on;

stairs(t,y,'b-') % ploting staircase signal- sampled signal

legend('Original Signal ','PCM wave')

qe=x-y;

snr(ii)=10*log10((x*x')/(qe*qe'));

end

Nbits=0:log10(max(MM))/log10(2)

snr_ideal=6*Nbits+1.72

figure;

plot(nbit, snr,'b',Nbits, snr_ideal,'r');

legend('Measured','Theoritical ')

title('SNR Vs No. of bits used for quantisation');

xlabel('No. of bits'); ylabel('SNR in dB');

EC8512

ALGORITHM:

1. Generate a sinusoidal wave with higher sampling rate and estimate the signal power

2. Identify the quantization values and their boundaries of quantization levels with

respect to number of levels

3. Using above, quantise the samples of the sinusoidal wave

4. Find the error between quantised signal and the original signal

5. Find error power and calculate the SNR for the respective number of output levels

6. Change the number of levels and repeat the steps 2-5.

7. Estimate theoretical SNR

8. Plot SNR vs number of bits and compare the results

Observation:

No. of bits:

SNR:

RESULT:

Thus SNR performance of Pulse Code Modulation (PCM) was studied using Matlab

software.

EX.NO.6 DELTA MODULATION AND DEMODULATION

EC8512

EC8512

AIM:

To design and simulate the delta modulation and demodulation process using Matlab

software.

APPARATUS REQUIRED: -

1. Desktop Computer 1

2. Matlab software 1

THEORY

A 1-bit DPCM coder is known as a delta modulator (DM). In other words, DM codes the

differences in the signal amplitude instead of the signal amplitude itself. Yet another name for DM

is pulse width modulation. A delta-modulation encoder is shown in Figure 1; it is known as a single

integration modulator.

The input signal is compared to the integrated output pulses and the delta (difference)

signal is applied to the quantizer. The quantizer generates a positive pulse when the difference

signal is negative, and a negative pulse when the difference signal is positive. This difference

signal moves the integrator step by step closer to the present value input, tracking the derivative

of the input signal.

A delta modulation decoder has to integrate the modulated signal and low pass filter the

output of the integrator as shown in figure 6.1

PROGRAM:

EC8512

% Message Signal

a=2;

t=0.2*pi/50:0.1:2*pi;

x=a*sin(t);

L=length(x);

subplot(4,1,1); plot(x,'r'); axis([0 L -3 3]);

grid on;

legend('Message signal');

title('Message Signal');

% Delta Modulation

delta=0.2;

xn=0;

for i=1:L

if x(i)>xn(i)

d(i)=1;

xn(i+1)=xn(i)+delta;

else

d(i)=0;

xn(i+1)=xn(i)-delta;

end

end

subplot(4,1,2); plot(x,'r'); axis([0 L -3 3]);

hold on; stairs(xn);

grid on;

legend('Message signal','Stair case approximated signal');

% Delta DeModulation

yn=0;

for i=1:length(d)

if d(i)==0

yn(i+1)=yn(i)-delta;

else

yn(i+1)=yn(i)+delta;

end

end

subplot(4,1,3); stairs(d,'black'); axis([0 L -3 3]); grid on;

legend('Delta Modulation');

title('Delta Modulation');

subplot(4,1,4); plot(yn,'b'); axis([0 L -3 3]); grid on;

legend('Demodulation Signal');

title('Delta Demodulation');

ALGORITHM:

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RESULT:

Thus, the Delta modulation and demodulation process was simulated and executed.

EC8512

Program:

sample=50;

binary=randint(1,binarysize,2);

%% Message

output=[];

for i=1:1:binarysize

if binary(i) == 1

output=[output ones(1,sample)];

else

output=[output zeros(1,sample)];

end

end

subplot(3,3,1)

plot(output);

axis([0 binarysize*sample -2 2]);

grid on;

xlabel(' samples');

ylabel('amplitude');

title(' Message');

%% NRZ Unipolar

output=[];

for i=1:1:binarysize

if binary(i) == 1

output=[output ones(1,sample)];

else

output=[output zeros(1,sample)];

end

end

subplot(3,3,2)

plot(output);

axis([0 binarysize*sample -2 2]);

grid on;

xlabel(' samples');

ylabel('amplitude');

title(' NRZ Unipolar');

%% NRZ polar

output=[];

for i=1:1:binarysize

if binary(i) == 1

output=[output ones(1,sample)];

else

output=[output -1*ones(1,sample)];

end

end

EC8512

AIM:

To write a MATLAB program to study the different types of line encoding schemes.

APPARATUS REQUIRED: -

1. Desktop Computer 1

2. Matlab software 1

THEORY:

UNIPOLAR:

In unipolar format (also known as on-off signaling), symbol 1 is represented by a transmitting

pulse, whereas symbol 0 is represented by switching off the pulse. When the pulse occupies the

full duration of a symbol, the unipolar format is said to be of the nonreturn-to-zero (NRZ) type.

When it occupies a fraction (usually one-half) of the symbol duration, it is said to be return-to-zero

(RZ) type.

POLAR:

In polar format, a positive pulse is transmitted for symbol 1, and a negative pulse for symbol 0.

Unlike unipolar waveform, a polar waveform has no DC component, provided that the 0s and 1s

in the input data occur in equal proportion.

BIPOLAR:

In bipolar format (also known as pseudoternary signaling), positive and negative pulses are used

alternately for the transmission of 1s, and no pulses for the transmission of 0s.

MANCHESTER:

In Manchester format (also known as bi phase baseband signaling), symbol 1 is represented by

transmitting a positive pulse for one-half of the symbol duration, followed by a negative pulse

subplot(3,3,3)

EC8512

plot(output);

axis([0 binarysize*sample -2 2]);

grid on;

xlabel(' samples');

ylabel('amplitude');

title(' NRZ polar');

%% NRZ Bipolar

output=[];

temp=1;

for i=1:1:binarysize

if binary(i) == 1

output=[output temp*ones(1,sample)];

temp=-1*temp;

else

output=[output zeros(1,sample)];

end

end

subplot(3,3,4)

plot(output);

axis([0 binarysize*sample -2 2]);

grid on;

xlabel(' samples');

ylabel('amplitude');

title(' NRZ Bipolar');

%% RZ Unipolar

output=[];

for i=1:1:binarysize

if binary(i) == 1

output=[output ones(1,sample/2) zeros(1,sample/2)];

else

output=[output zeros(1,sample)];

end

end

subplot(3,3,5)

plot(output);

axis([0 binarysize*sample -2 2]);

grid on;

xlabel(' samples');

ylabel('amplitude');

title(' RZ Unipolar');

%% RZ polar

output=[];

for i=1:1:binarysize

if binary(i) == 1

output=[output ones(1,sample/2) zeros(1,sample/2)];

else

output=[output -1*ones(1,sample/2) zeros(1,sample/2)];

end

end

subplot(3,3,6)

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for the remaining half of the symbol duration; for symbol 0, these two pulses are transmitted in

reverse order. It has no DC component.

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plot(output);

axis([0 binarysize*sample -2 2]);

grid on;

xlabel(' samples');

ylabel('amplitude');

title(' RZ polar');

%% RZ Bipolar

output=[];

temp=1;

for i=1:1:binarysize

if binary(i) == 1

output=[output temp*ones(1,sample/2) zeros(1,sample/2)];

temp=-1*temp;

else

output=[output zeros(1,sample)];

end

end

subplot(3,3,7)

plot(output);

axis([0 binarysize*sample -2 2]);

grid on;

xlabel(' samples');

ylabel('amplitude');

title(' RZ Bipolar');

%% Manchester

output=[];

for i=1:1:binarysize

if binary(i) == 1

output=[output ones(1,sample/2) -1*ones(1,sample/2)];

else

output=[output -1*ones(1,sample/2) ones(1,sample/2)];

end

end

subplot(3,3,8)

plot(output);

axis([0 binarysize*sample -2 2]);

grid on;

xlabel(' samples');

ylabel('amplitude');

title(' Manchester');

EC8512

RESULT

Thus, the MATLAB program for different types of line encoding process is written and

executed.

EC8512

EC8512

AIM:

To design the digital modulations techniques and simulate it using simulink.

APPARATUS REQUIRED: -

1. Desktop Computer 1

2. Matlab software 1

THEORY

BPSK is a method for modulating a binary signal onto a complex waveform by shifting the

phase of the complex signal. In digital baseband BPSK, the symbols 0 and 1 are modulated to the

complex numbers exp(jt) and -exp(jt), respectively, where t is a fixed angle.

is modulated onto a sinusoidal carrier whose frequency shifts between different values.

Differential Phase Shift Keying or DPSK modulation signal records changes in a binary

stream. Changes in bits will result in a π phase shift. Phase-shift keyed signals cannot be detected

incoherently as they use coherent detection. A coherent detector has two inputs for which one is a

reference signal and the other is the modulated signal to be demodulated. However in this case a

partially coherent detection is used here where a one-bit delay is use.

EC8512

EC8512

Formula:

Generation: 𝑠(𝑡) = 𝑠𝑖1 𝜑1 (𝑡)

1, 𝑏𝑖 = 1

𝑠𝑖1 = {

−1, 𝑏𝑖 = 0

2𝐸

𝜑1 (𝑡), = √ 𝑇 𝑏 cos(2𝜋𝑓𝑐 𝑡) , 0 ≤ 𝑡 < 𝑇𝑏 ,

𝑏

Detection:

𝑇

𝑥𝑖 = ∫0 𝑏 𝑠(𝑡) 𝜑1 (𝑡)𝑑𝑡,

1, 𝑥 ≥ 0

𝑏̂ = { 𝑖

0, 𝑥𝑖 < 0

Generation:

1. Generate binary random number over a desired length (minimum of 10);

2. Select bit rate and generate polar NRZ wave for random bit sequence

a. Generate a impulse train with period equal to bit period

b. Replace the impulse polarity in accordance with si1.

c. Generate a pulse with a length equal to bit period

d. Convolve the pulse with the impulse train that will yield NRZ wave

3. Select bit energy and generate the carrier signal 𝜑1 (𝑡).

4. Multiply the polar NRZ wave with the carrier signal to obtain the BPSK signal si(t)

Detection

5. Multiply the si(t) with the local copy of 𝜑1 (𝑡) and integrate over one bit period.

6. Sample the sum at end each bit period and obtain xi1.

7. Detect the transmitted bits as “1” if xi is greater than zero and as “0” ” if xi is lesser than zero.

EC8512

EC8512

Formula:

Generation: 𝑠(𝑡) = 𝑠𝑖1 𝜑1 (𝑡) + 𝑠𝑖2 𝜑2 (𝑡)

1

𝑠𝑖1 [ ], 𝑏𝑖 = 1

[𝑠 ] = { 0

𝑖2 0

[ ], 𝑏𝑖 = 0

1

2𝐸

𝜑1 (𝑡), = √ 𝑇 𝑏 cos(2𝜋𝑓𝑐1 𝑡) , 0 ≤ 𝑡 < 𝑇𝑏 ,

𝑏

2𝐸

𝜑2 (𝑡), = √ 𝑇 𝑏 cos(2𝜋𝑓𝑐2 𝑡) , 0 ≤ 𝑡 < 𝑇𝑏 ,

𝑏

Detection:

𝑇

𝑥𝑖1 = ∫0 𝑏 𝑠(𝑡) 𝜑1 (𝑡)𝑑𝑡,

𝑇

𝑥𝑖2 = ∫0 𝑏 𝑠(𝑡) 𝜑2 (𝑡)𝑑𝑡,

𝑙 = 𝑥𝑖1 − 𝑥𝑖2

1, 𝑙 ≥ 0

𝑏̂ = {

0, 𝑙 < 0

Generation:

1. Generate binary random number over a desired length (minimum of 10);

2. Select bit rate and generate unipolar NRZ wave for random bit sequence

a. Generate a impulse train with period equal to bit period

b. Replace the impulse polarity in accordance with bi.

c. Generate a pulse with a length equal to bit period

d. Convolve the pulse with the impulse train that will yield unipolar NRZ wave

3. Select bit energy and generate the carrier signal 𝜑1 (𝑡).

4. Generate the inverted output for the unipolar NRZ wave by using NOT.

5. Multiply the unipolar NRZ wave with the carrier signal 𝜑1 (𝑡) and multiply the inverted

unipolar NRZ wave with the carrier signal 𝜑2 (𝑡). Add the two streams to obtain the BfSK

signal si(t)

Detection

6. Multiply the si(t) with the local copy of 𝜑1 (𝑡) and integrate over one bit period.

7. Sample the sum at end each bit period and obtain xi1.

8. Again multiply the si(t) with the local copy of 𝜑2 (𝑡) and integrate over one bit period.

9. Sample the sum at end each bit period and obtain xi2.

10. Subtract xi2 from xi1 to get l.

11. Detect the transmitted bits as “1” if l is greater than zero and as “0” ” if l is lesser than zero.

RESULT

Thus, the different digital modulation techniques- BFSK,BPSK and DPSK was designed

and simulated.

EC8512

Formula:

Bit error rate = No. of bits in error after detection / No. of bits transmitted

Program:

n=7; k=4;

blk_nos=20000;

Nbits=blk_nos*k; % Number of bits in simulation

snrdB=[0:10];

P= [ 1 1 0 ; % Parity Matrix

0 1 1 ;

1 1 1 ;

1 0 1];

In= eye(4); % Identity Matrix

Ink=eye(3);

G=[P,In]; % Generator Matrix

H=[Ink, P']; % Parity Check Matrix

errpat=[ 0 0 0 0 0 0 0 ;

1 0 0 0 0 0 0 ;

0 1 0 0 0 0 0 ;

0 0 1 0 0 0 0 ;

0 0 0 1 0 0 0 ;

0 0 0 0 1 0 0 ;

0 0 0 0 0 1 0 ;

0 0 0 0 0 0 1 ];

syndrom=errpat*H';

m=reshape(bs, k,[] )'; % Blocking of bits into 4 bits in each block

ct=reshape(x', 1,[]); % Transmitted bit sequence

ctx=ct;

ctx(ct==0)=-1;

EC8512

AIM:

To study the BER performance of Linear Block Code using matlab software

APPARATUS REQUIRED: -

1. Desktop Computer 1

2. Matlab software 1

Theory:

The input to the encoder is binary information sequence at a rate R bits/sec. There are

mainly two types of channel encoding techniques namely Block coding and Convolutional coding.

In block coding, a block of k information bits is encoded into a block of n bits known as codeword

(n>k). So for k bits there could be total 2k possible code words. The code rate defined as the ratio

Rc= k/n is a measure of amount of redundancy introduced by block coding.

In convolution coding each k bit information symbol to be encoded and transformed into

n bit called as codeword such that n>k and transformation is a function of the last L information

symbols where L is the constraint length of the code. The codeword can be generated using finite

state shift register approach. Thus code rate Rc would be same as that of block codes. Hence a

good code is the one that ensure a certain error correcting capability at minimum Rc or maximum

output encoder rate R/Rc.

EC8512

for ii=1:length(snrdB)

snr(ii)=10^(snrdB(ii)/10)

r=awgn(ctx, snr(ii), 'measured');

cr=r;

cr(r>=0)=1;

cr(r<=0)=0;

y=reshape(cr', n, [])';

s=mod( y*H', 2);

e0=[];

for jj=1:blk_nos

for kk=1:2^(n-k)

if(syndrom(kk,:)==s(jj,:))

ro_match=kk;

break;

end

end

e0=[e0; errpat(ro_match,:)];

end

xcat=mod( y+e0, 2);

mcat=xcat(:, n-k+1:end);

bcat=reshape(mcat', 1,[]);

BER_wo_EC(ii)=length(find(ct~=cr))/length(ct)

BER_with_EC(ii)=length(find(x~=xcat))/length(ct)

end

semilogy(snr, BER_with_EC, 'r*-', snr, BER_wo_EC, 'b-+')

legend('BER_with_EC', 'BER_wo_EC,');

xlabel('SNR (dB)'); ylabel('BER'); title('BER Performance of Linear Block

Code');

EC8512

Algorithm:

2. Convert them into block of 7 bits to form it as message block

3. Generate generator matrix

4. Multiply the message block with generator matrix and get codeword blocks

5. Convert back codeword blocks into bit stream.

6. Generate noise to yield the required SNR with respect to the energy in the codeword bit

stream.

7. Decode the codeword without error detection and correction. Compare the decoded bits

with transmitted and calculate BER without error control

8. Convert the codeword bit stream into blocks again. Apply parity check matrix to calculate

syndrome

9. Identify the respective error pattern and add with codeword to generate error corrected

codeword

10. Separate message words and get the output message bit stream. Compare this input

message bit stream and calculate BER with error control

11. Repeat the steps 6-10 for different SNR and plot BER Vs SNR curve with and without

error control coding

Observation:

Number of blocks on simulation :

Number of bits used:

Message word length =

Codeword length =

Generator Matrix=

SNR:

BER (w/o EC):

BER (with EC):

RESULT

Thus, the BER performance of Linear Block Code using matlab software was performed

and coding gain obtained for 10−4 BER is

EC8512

EC8512

AIM:

To design and simulate the operation of equivalent base-band binary phase shift keying

(BPSK) direct sequence spread spectrum (DS/SS) system.

APPARATUS REQUIRED: -

1. Desktop Computer 1

2. Matlab software 1

THEORY

A system may be required to provide a form of secure communication in a hostile

environment such that the transmitted signal is not easily detected or recognized by unwanted

listeners. This requirement is fulfilled by a signaling technique called “Spread-Spectrum

modulation”.

1. Spread Spectrum is a means of transmission in which the data of interest occupies a

bandwidth in excess of the minimum bandwidth necessary to send the data.

2. The spectrum spreading is accomplished before transmission through the use of a code that

is independent of the data sequence. The same code is used in the receiver to de-spread the

received signal so that the original data may be recovered.

In a direct sequence spread spectrum technique, two stages of modulation are used. First, the

incoming data sequence is used to modulate a wideband code. This code transforms the

narrowband data sequence into a noise-like wideband signal. The resulting wideband signal

undergoes a second modulation using a phase shift keying technique.

EC8512

EC8512

RESULT

EC8512

Thus the equivalent base-band binary phase shift keying (BPSK) direct sequence spread

spectrum (DS/SS) system is designed and simulate.

EC8512

AIM:

To design and simulate the operation of equivalent base-band binary phase shift keying

(BPSK) direct sequence spread spectrum (DS/SS) system.

APPARATUS REQUIRED: -

1. Desktop Computer 1

2. Matlab software 1

THEORY

communication channel. Ideally, if the channel is ideal (without and channel distortion and

additive noise), we can demodulate the signal perfectly at the output without causing any error.

However, in practice, all the channels are non-ideal and noisy in nature. So, to recover the original

signal after demodulation, an equalization filter that minimizes the error between original

transmitted signal and demodulated is used.

Several algorithms like Least Mean Square (LMS), Recursive Least Mean Square (RLMS),

Normalized Least Mean Square (NLMS) etc., has been proposed to perform this operation of

equalization.

EC8512

PROGRAM

LMS ALGORITHM

clc;

clear all;

close all;

d=[ 0 1 0 1 1 ];

t=linspace(0,10,1000);

n=length(t);

b=2*d-1; % Convert unipolar to bipolar

Nsb=n/length(d)% Number of samples per bit

bb=repmat(b',1,Nsb)% replicate each bit Nsb times

bw=bb'; % Transpose the rows and columns

bw=bw(:)' ; % Data sequence samples

x=bw+randn[‘,length(d)*0.01];%adding noise

w=zeros(1,n)

mu=0.2;

for i=1:n

e(i)=bw(i)-w(i)*x(i);

w(i+1)=w(i)+(mu*e(i)*x(i));

end

for i=1:n

y(i)=w(i).*bw(i);

end

subplot(2,2,1),plot(bw);

ylabel('original signal');

subplot(2,2,2),plot(x);

ylabel('signal added with noise');

subplot(2,2,3),plot(e);

ylabel('error');

subplot(2,2,4),plot(y);

ylabel('adaptive equalizer output');

EC8512

RESULT

EC8512

Thus the distortion introduced by the channel on the transmitted signal on the received

samples are mitigated using LMS algorithm.

Rough Work

EC8512

Rough Work

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