FATIMA MICHEAL COLLEGE OF ENGINEERING AND

TECHNOLOGY
DEPARTMENT OF ELECTRONICS AND COMMUNICATION
ENGINEERING

LABORATORY MANUAL R-2021

Semester/Branch : V semester ECE

Academic Year : 2022-2023 (EVEN SEM)

Lab code & title : EC3461-COMMUNICATION SYSTEMS

LABORATORY

Lab incharge : Mrs. P. Jebane (AP/ECE)

EC3461 COMMUNICATION SYSTEMS LABORATORY L T P C 0 0 3 1.5

LIST OF EXPERIMENTS:
1. AM Modulator and Demodulator
2. FM Modulator and Demodulator
3. Pre Emphasis and De-Emphasis
4. Signal sampling & TDM
5. Pulse Code Modulation and Demodulation
6. Pulse Amplitude Modulation and Demodulation
7. Pulse Position Modulation and Demodulation
8. Digital Modulation – ASK, PSK, FSK
9. Delta Modulation and Demodulation
10. Simulation of ASK, FSK, and BPSK generation and detection schemes
11. Simulation of DPSK, QPSK and QAM generation and detection schemes
12. Simulation of Linear Block and Cyclic error control coding schemes

TOTAL: 45 PERIODS
S.No Date Experiment Mark Staff’s
signature
1. AM Modulator and Demodulator

2. FM Modulator and Demodulator

3. Pre Emphasis and De-Emphasis

4. A.Signal sampling

B.TDM

5. Pulse Code Modulation and Demodulation

6. Pulse Amplitude Modulation and

Demodulation
7. Pulse Position Modulation and
Demodulation
8. Digital Modulation – ASK

B.PSK

C.FSK

9. Delta Modulation and Demodulation

10. Simulation of ASK, FSK, and BPSK

generation schemes

11. Simulation of DPSK, QPSK and QAM

generation schemes
12. Simulation of Linear Block and Cyclic error
control coding schemes
 1. AMPLITUDE MODULATION & DEMODULATION

Aim:

1. To study the operation of Amplitude Modulation by observing the AM wave and

to measure its modulation index for various modulating voltages.
2. To perform Demodulation of AM wave.

Apparatus:

Sl. No Name of the component Specifications or Quantity

range

1 Amplitude Modulation and ------- 1

Demodulation trainer kit

2 CRO 20MHz 1

3 Function generator 1MHz 1

4 Connecting wires ---- As per required

number

Theory:
Amplitude Modulation:
Modulation is defined as the process by which some characteristics of a carrier signal is
varied in accordance with a modulating signal. The base band signal is referred to as the
modulating signal and the output of the modulation process is called as the modulated signal.

Amplitude modulation is defined as the process in which the amplitude of the carrier
wave is varied with the instantaneous values (voltages) of the message signal. This technique is
also known as DSBFC that is Double Side Band with Full Carrier scheme. the general equation
of Amplitude Modulated signal is given by
𝑠𝐴𝑀 (𝑡) = 𝐴𝑐 (1 + 𝑘𝑎 𝑚(𝑡)) cos 2𝜋𝑓𝑐 𝑡.
If the type of Modulation is single-tone modulation then message signal is replaced by single
tone 𝑚(𝑡) = 𝐴𝑚 cos 2𝜋𝑓𝑚 𝑡.AM signal observed practically is Single-tone AM signal.

Demodulation:
The process of detection provides a means of recovering the modulating Signal from
modulating signal. Demodulation is the reverse process of modulation. The detector circuit is
employed to separate the carrier wave and eliminate the side bands. Since the envelope of an AM
wave has the same shape as the message, independent of the carrier frequency and phase,
demodulation can be accomplished by extracting envelope. An increased time constant RC
results in a marginal output follows the modulation envelope. A further increase in time constant
the discharge curve become horizontal if the rate of modulation envelope during negative half
cycle of the modulation voltage is faster than the rate of voltage RC combination ,the output fails
to follow the modulation resulting distorted output is called as diagonal clipping : this will occur
even high modulation index. The depth of modulation at the detector output greater than unity
and circuit impedance is less than circuit load (RL > Zm) results in clipping of negative peaks of
modulating signal. It is called “negative clipping
The envelope of the modulating wave has the same shape as the base band message
provided the following two requirements are satisfied
1. The carrier frequency fc must be much greater than the highest frequency components
fm of the message signal m (t) i.e. fc >> fm.
2. The modulation index must be less than unity. If the modulation index is greater than
unity, the carrier wave becomes over modulated.
Applications of Amplitude Modulation:
Amplitude modulation is used in a variety of applications. Even though it is not as widely used
as it was in previous years in its basic format it can nevertheless still be found.

 Broadcast transmissions: AM is still widely used for broadcasting on the long,

medium and short wave bands. It is simple to demodulate and this means that radio
receivers capable of demodulating amplitude modulation are cheap and simple to
manufacture. Nevertheless many people are moving to high quality forms of transmission
 like frequency modulation, FM or digital transmissions.

 Air band radio: VHF transmissions for many airborne applications still use AM. . It is
used for ground to air radio communications as well as two way radio links for ground
staff as well.

 Single sideband: Amplitude modulation in the form of single sideband is still used for
HF radio links. Using a lower bandwidth and providing more effective use of the
transmitted power this form of modulation is still used for many point to point HF links.

 Quadrature amplitude modulation: AM is widely used for the transmission of data in

everything from short range wireless links such as Wi-Fi to cellular telecommunications
and much more. Effectively it is formed by having two carriers 90° out of phase.
Circuit diagram:
Model graph:
Experimental procedure:

Modulation:

1. Generate an AF signal (message signal) of 1 KHz frequency and 2V peak-peak amplitude

using function generator.
2. Switch on the trainer. Observe the output of the Carrier generator and measure its
amplitude and frequency.
3. Now connect AF signal to AF I/P and carrier signal to Carrier I/P of the modulator as
shown in the circuit diagram.
4. Observe the Modulator O/P signal at AM O/P by making necessary changes in AF I /P.
5. Vary the amplitude of message signal (by keeping frequency of message signal as
constant) and observe the effect on AM signal by calculating Modulation Index (μ).
6. Vary the frequency of message signal (by keeping amplitude of message signal as
constant) and observe the effect on AM signal by calculating Modulation Index (μ).

Demodulation:

1. Connect the Output of Amplitude Modulator to the input of demodulator circuit.

2. Now observe the output of demodulator and measure its frequency.
3. Check the phase difference between message and demodulated signals.

Tabular form: Modulation

Carrier voltage Vc= constant

Sl. No Message amplitude 𝑽𝒎𝒂𝒙 − 𝑽𝒎𝒊𝒏

Vmax Vmin 𝝁= 𝑿 𝟏𝟎𝟎%
/voltage Vm 𝑽𝒎𝒂𝒙 + 𝑽𝒎𝒊𝒏
Demodulation:

Sl. No Message frequency Demodulated signal

fm frequency

Result &Discussions: Thus the Amplitude modulation and demodulation is performed

practically.

1. As the amplitude of message signal increases, the Modulation index increases and vice
versa.
2. When message signal and carrier signal are in-phase it represents Vmax.
3. When message signal and carrier signal are out of phase it represents Vmin.
4. The phase difference between Message signal and demodulated signal are not same.
 2. FM MODULATION & DEMODULATION

Aim:

1. To study the operation of Frequency Modulation by observing the FM wave.

2. To measure its modulation index for various modulating voltages and frequencies.

Apparatus:

Sl. No Name of the component Specifications or Quantity

range

1 frequency Modulation and ------- 1

Demodulation trainer kit

2 CRO 20MHz 1

3 Function generator 1MHz 1

5 Connecting wires and Probes ---- As per required

number

Theory:
Angle modulation is a technique in which the angle of the carrier is varied with
instantaneous values of message signal. Angle Modulation has been divided into two types.
i. Phase Modulation. ii. Frequency Modulation.
Frequency modulation is defined as the process in which is the frequency of the carrier wave is
 varied with the instantaneous values (voltages) of the message signal.
𝑡
FM signal is given by 𝑆𝐹𝑀(𝑡) = 𝐴𝑐 cos(2𝜋 𝑓𝑐 𝑡 + 2𝜋𝑘𝑓 ∫0 𝑚(𝑟) 𝑑𝑟)
FM has been divided into two types Wideband FM and Narrow Band FM. FM range is 88Hz to
108MHz.FM has both advantages and Disadvantages
Advantages:
1. FM is more immune to Noise compared to AM. Hence there is a significant amount of
increase in Signal-to-Noise Ratio at the output.
2. It operates at Very High Frequency.
3. Amplitude of the Frequency Modulated signal is almost unaffected.
4. Frequency allocation allows for a Guard-Band which reduces adjacent Channel
Interference.

Applications of FM:
1. We commonly see AM and FM in radio broadcasting. FM transmission provides a
superior sound quality than that of AM, but with a reduced coverage. It is because of the
high Bandwidth it offers. And so most radio music stations use FM to provide good
quality sound.
2. AM is used for video signals in TV‟s, ranges from (535K Hz- 1750KHz).
FM is used for sound signals in TV‟s, ranges from (88MHz-108MHz).
Model graph:

Experimental procedure:
Modulation:

1. Switch on the trainer kit.

2. Connect CRO to the FM output and observe carrier frequency (and amplitude) of the
wave at that point without any AF input.
3. Connect around (3 kHz to 10 kHz) with 2.4 volts sine wave (AF signal) to the AF input
of the frequency Modulator.
4. Now observe the FM output on the first channel of CRO and adjust the amplitude of AF
signal to get clear FM wave form.
5. Vary the modulating amplitude and frequency of the message signal and observe the
effect on FM output.
6. Calculate modulation index in the first case by changing frequency of message signal (by
keeping amplitude as constant).
7. Calculate modulation index in the second case by changing amplitude of message
signal(by keeping frequency as constant)

Demodulation:

1. Connect the FM output to the FM input of Demodulator circuit .Observe the

demodulated wave (second channel of CRO).
2. By vary the potentiometer provided in the demodulator section and see the output.
Tabular form: Modulation
Message frequency fm= constant

Sl. No Message Fmax Fmin 𝑭𝒎𝒂𝒙 − 𝑭𝒎𝒊𝒏 ∆𝒇

∆𝒇 = 𝖰=
amplitude 𝟐 𝒇𝒎

/voltage Vm

Demoulation:

Sl.No Message frequency Demodulated frequency

Result & Discussions:

Thus the Frequency modulation and demodulation are performed practically.

1. Modulation index β decreases with respect to frequency of message signal when the
amplitude of message signal is kept constant.

2. Modulation index β increases with respect to amplitude of message signal when the
frequency of message signal is kept constant.
 3. PRE-EMPHASIS & DE-EMPHASIS

Aim:

To observe the effect of Pre-emphasis and De-emphasis on a given signal.

Apparatus:

Sl. No Name of the component Specifications or range Quantity

1 Resistors 7.5KΩ,6.8KΩ 1 each

2 Capacitors 10nF,0.1µF 1each

3 Function generator 1MHz 1

4 CRO 20MHz 1

5 Connecting wires ---- As per required

number

Theory:
The noise has greater effect on high frequencies than on the lower ones. Thus, if the higher frequencies
were artificially boosted at the transmitter and correspondingly cut at the receiver, an improvement in
noise immunity could be expected, thereby increasing the SNR ratio. This boosting of the higher
modulating frequencies at the transmitter is known as pre-emphasis and the compensation at the receiver
is called as de-emphasis.
Circuit diagram:
Model graph:
Experimental procedure:

1. Connect the circuit diagram as per pre-emphasis circuit shown in figure.

2. Apply a sinusoidal signal of 20mV as input signal to pre-emphasis circuit.
3. Then by increasing the input signal frequency from 500Hz to 20 kHz. Observe the output
voltage Vo and calculate gain in dB as 20 log 𝑉𝑜 .
𝑉𝑖

4. Similarly apply a sinusoidal signal of 5V, 100Hz to de-emphasis circuit vary the input
signal frequency from 100Hz to 20 KHz and calculate gain in dB as 20 log 𝑉𝑜/𝑉𝑖 by observing output
voltage Vo.
5. Plot pre-emphasis and de-emphasis curves.

Tabular form:
Pre-emphasis: Vi = 20mV.

Sl. No Frequency of input Output voltage Gain in dB

signal (Hz) Vo (volts) Av = 𝟐𝟎 𝐥𝐨𝐠 𝑽𝒐
𝑽𝒊

De-emphasis: Vi = 5V.

Sl. No Frequency of input Output voltage Gain in Db

signal (Hz) Vo (volts) Av = 𝟐𝟎 𝐥𝐨𝐠 𝑽𝒐
𝑽𝒊
Result & Discussions:
The effect of Pre-emphasis and De-emphasis on a given input signal is observed.
 4. A. SAMPLING AND RECONSTRUCTION

Aim:

To study the signal sampling and its reconstruction.

Apparatus:

Sl.No Name of the component Specifications or Quantity

range

1 Sampling theorem verification trainer - 1

kit-DCLT001

2 CRO 20MHz 1

3 Connecting wires and patch chords -- Based on requirement

Theory:
An analog Source of information produces an output that can have any one of a
continuum of possible value at any time. The sound pressure from an Orchestra playing music is
an example for analog source. There no of analog sources a signal generator is another analog
source. An analog signal is an electrical waveform that can have any one of continuous
amplitudes at any time. Voltage and current are examples of CT signals.
A digital source can be defined as the one which generates digital signals most sources
are analog in nature and by using some mechanism an analog source can be converted into a
digital source. For example, temperature is an analog quantity, but when combined with a
thermostat with output values of on or off, the combination may be considered as a digital
source.
A digital signal may be defined as an electrical waveform having one of a finite set of
possible amplitudes at any time. i.e, a binary signal is a digital signal. A communication system
is required to transport an information bearing signal from a source to destination through a
communication channel. Basically, a communication system may be analog or digital type. In an
analog communication system, the information-bearing signal is continuously varying in both
amplitude and time, and it is used directly to modify some characteristic of a sinusoidal carrier
wave, such as amplitude, phase or frequency. In Digital Communication system, on the other
hand the m(t) is processed so that it can be represented by a sequence of discrete messages.
Need for Digital Communications: The growth of Digital Communications is largely due to the
following reasons:
1. Digital communications provide improved reliability.
2. The availability of wide band channels provided by geo-stationary satellites, Optical fibers and
Co-axial cables.
3. The ever increasing availability of integrated Solid-state technology, which has made it
possible to increase system complexity by orders of magnitude in a cost effective manner.
As we observed the advantages of digital communications, there is every possible need for
converting the analog signal to digital form for compatibility. Three basic operations are
combined to convert an analog signal to a digital signal by
Sampling: In the sampling process only sample value of the analog signal at uniformly spaced
discrete instant of time are extracted and retained. i.e a continuous time signal is converted into a
discrete signal.
Quantizing: In this the nearest level in a finite set of discrete levels approximates each sample
value.
Encoding: In encoding, the selected level is represented by a code word that consists of a
prescribed number of code elements.
Sampling theorem: The analog signal can be converted to a discrete time signal by a process
called sampling. The Sampling theorem for a band limited (W Hz) signal of finite energy can be
stated as follows that
 „‟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 knowledge of samples taken at the rate of 2W
samples per second.
Experimental Procedure:

1. Observe the 2 KHz continuous signal on CRO by connecting any channel of CRO to 2
KHz input on the trainer kit.
2. Connect the 2 KHz 5V p-p signal generated onboard to the ANALOG INPUT, by means
of the patch-cords provided.
3. Change FR SEL switch to observe INTERNAL SAMPLING FREQUENCY at TP26 as
the switch position changes the corresponding LED at the output of Binary counter
glows (for 32 KHz, 16 KHz, 8 KHz, 4 KHz or 2 KHz). Choose 16 KHz signal as the
sampling signal with sampling frequency fs = 16 KHz.
4. Connect the Sampling frequency 16 KHz signal in the INTERNAL mode, by means of
the shorting pin provided.
5. By means of DIP switch setting, as indicated in the Duty Cycle Table vary the duty cycle
of the sampling frequency signal from 10% to 90% in the discrete steps of 10% each.
6. Observe the effect of duty cycle on INTERNAL SAMPLING FREQUENCY in each
case, the corresponding model graphs are given in Graph3.1 and Graph 3.2.
7. Keeps the position of DIP switch setting for 50% Duty Cycle for the INTERNAL
SAMPLING FREQUENCY.
8. Now observe the Sampled signal at S4 (TP32) for 30% duty cycle with fs = 16 KHz and
draw the corresponding sampled signal (count the number of samples with respect to 2
KHz).
SAMPLE AND HOLD OUTPUT:
9. Observe the Sample and Hold amplifier output at TP34 and draw the corresponding
signal.
RECONSTRUCTION:
10. Connect sampled output at TP32 to INPUT (S8, S9 and S10) of SECOND, FOURTH
AND SIXTH ORDER low pass filter to reconstruct original signal. Draw the
reconstructed signal for FOURTH order low pass filter at TP38.

11. Connect sample and hold output at TP34 to INPUT (S8, S9 and S10) of SECOND,
FOURTH AND SIXTH ORDER low pass filter to reconstruct original signal. Draw the
reconstructed signal for FOURTH order low pass filter at TP38.
Model graph:
Tabulation:
Amplitude Time Frequency
Input
Sampling output
Reconstruction
output

Result & discussions:

1. Sampling of a signal and its reconstruction is performed practically.

2. The effect of duty cycle of the sampling frequency on the sampled signal is observed.

3. The effect of duty cycle on Sample and hold output signal is observed.
 4. b . TIME DIVISION MULTIPLEXING

Aim:

To study the Time division multiplexing and De multiplexing using Pulse

AmplitudeModulation and De-modulation and to reconstruct the signals using filters.

Apparatus:

Sl.No Name of the component Specifications or Quantity

range

1 DCLT002-PAM-TDM trainer kit ---- 1

2 CRO (0-30)MHz 1

3 Connecting wires and probes patch ---- Required number

chords

Theory:
The TDM is used for transmitting several analog message signals over a
communication channel by dividing the time frame into slots, one slot for each
message signal. The four input signals, all band limited by the input filters are
sequentially sampled, the output of which is a PAM waveform containing samples of
the input signals periodically interlaced in time. The samples from adjacent input
message channels are separated by Ts/M, where M is the number of input channels. A
set of M pulses consisting of one sample from each of the input M-input channels is
called a frame. At the receiver the samples from individual channels are separated by
carefully synchronizing and are critical part in TDM. The samples from each channel
are filtered to reproduce the original message signal. There are two levels of
synchronization. Frame synchronization is necessary to establish when each group of
samples begins and word synchronization is necessary to properly separate the
samples within each frame. Besides the
space diversity & frequency diversity there is another method of sending multiple analog signalson a
channel using TIME DIVISION MULTIPLEXING & DEMULTIPLEXING Technique.

Experimental procedure:
 Multiplexing:

1. Observe four input signals at pins S1, S2, S3 and S4 and the Sync level S0 and clock
signals for CH0, CH1, CH2 and CH3 on the trainer kit.
2. Connect either any two channel inputs of 250 Hz, 500 Hz, 1 KHz, 2 KHz or all inputs to
the input of transmitter at CH0, CH1, CH2 and CH3 respectively.
3. Observe multiplexed data at TDX (Transmitter Data) of PAM-TDM transmitter. Thus
the signal is a multiplexed version of two signals or four signals.

De-Multiplexing:

1. Connect TXCLOCK (Transmitter clock) to RXCLOCK (Receiver clock).

2. Connect TXCH0 (Transmitter sync) to RXCH0 (Receiver sync).
3. Connect the TXD (Transmitter data) to RXD (Receiver data).
4. Observe multiplexed data at TDX for four input signals, Transmitter clock at TXCLOCK
and transmitter sync at TXSYNC.
5. Observe the de multiplexed signals at the receiver across the output of fourth order low
pass filter at CH0, CH1, CH2 and CH3.

Tabulation:
Amplitude Time Frequency
Input
Modulation output
Reconstruction
output
Results & Discussions: Thus the Time Division Multiplexing and de multiplexing of analogsignals
using PAM-TDM trainer kit is performed and the waveforms are observed.
 5.PCM MODULATION & DEMODULATION

Aim:

1. To study the Generation of the Pulse Code Modulation Waveform.

2. To perform Demodulation of PCM wave.

Apparatus:

Sl. No Name of the component Specifications or Quantity

range

1 PCM trainer kit ------- 1

2 CRO 20MHz 1

3 Digital Multi Meter ---- 1

4 Connecting wires, Probes and Patch ---- As per required

chords. number

Theory:

Pulse Code Modulation:

Pulse Code Modulation is a Digital modulation which different from continuous
modulation schemes like Amplitude Modulation and Frequency Modulation. PCM is a technique
used to convert analog signals into binary form. In the absence of noise and distortion it is
possible to completely recover a continuous analog modulated signal at the receiving end. But in
real time they suffer from transmission distortion and noise to an appreciable extent. In the PCM
system, groups of pulses or codes are transmitted which represent binary numbers corresponding
to modulating signal voltage levels. Recovery of the transmitter information does not depend on
the height, width, or energy content of the individual pulses, but only on their presence or
absence. Since it is relatively easy to recover pulses under these conditions, even in the presence
of large amounts of noise and distortion, PCM systems tend to be very immune to interference
and noise. Regeneration of the pulse enroute is also relatively easy, resulting in a system that
produces excellent for long- distance communication.

PCM Encoding:
The encoding process generates a binary code number corresponding to Modulating signal
voltage level to be transmitted for each sampling interval. Any one of the codes like binary,
ASCII etc, may be used as long as it provides a sufficient number of different symbols to
represent all of the levels to be transmitted. Ordinary binary number will contain a train of „1‟
and „0‟ pulses with a total of log 2n pulses in each number. (N is no of levels in the full range).
This system is very economical to realize, because it corresponds exactly to the process of
analog-to-digital (A/D) conversion.
Quantization:
The first step is in the PCM system is to quantize the modulating signal. The modulating signal
can assume an infinite number of different levels between the two limit values which define the
range of the signal in PCM. A code number is transmitted for each level sampled in the
modulating signal. If the exact number corresponding to the exact voltage were to be transmitted
for every sample, an infinitely large number of different code symbols would be needed.
Quantization has the effect of reducing this infinite number of levels to a relatively small number
which can be coded without difficulty.
In the quantization process, the total range of the modulating signal is divided up into a number
of small sub ranges. The number will depend on the nature of the modulating signal and will
form as few as 8 to as many as 128 levels. A number that is an integer power of two is generally
chosen because of the ease of generating binary codes. A new signal is generated by producing,
for each sample, a voltage level corresponding to the midpoint level of the sub range in which
the sample falls. Thus if a range of 0 to 5V were divided into 128, 5/128v sub ranges, and the
signal sampled when it was 3V, the Quantizer would put a voltage of 2.96 V and hold that level
until the next sampling time. The result is a stepped waveform which follows the counter of the
original modulating signal with each step synchronized to the sampling period.

Block diagram Description:

The block diagram of PCM modulation and Demodulation is shown ion front panel given
in the circuit diagram.
Modulating signal:
A variable sinusoidal frequency generator is provided with frequency from 0 Hz to 500
Hz. To see the actual bits that are transmitted through the communication channel, a variable DC
(-5V to +5V) source is provided on the panel.

PCM Encoding:
The modulating signal is applied to the input of Analog-to-Digital Converter which
performs the two functions of Quantization and Encoding, Producing a 8-bit binary coded
number. The signal is to be transmitted i.e, modulating signal is sampled at regular intervals. If
the maximum amplitude +5V is represented by 8-bits the 1LSB corresponds to Vx1/128 =
5/128=39mV and MSB represents the sign .So the values of the sampled signal at the output of
Analog-to-Digital converter are 00000000, 00111111, 01111111, 00000000, 10111111,
11111111, 10111111, 00000000.
To transmit all the bits in one channel, actually it is often sent as binary number back to
front by parallel to serial converter. i.e, 00000001, 11111100, 11111110 to make demodulating
easier. A parallel to serial converter transmits the code bits in serial fashion.
PCM Decoding:
At the receiver end the received data will be in serial form. The serial data is converted
back to parallel form by serial to parallel converter and passes the bits to a Digital- to- Analog
converter for decoding which has in-built sample and hold amplifier which maintains the pulse
level for the duration of the sampling period, recreating the pulse level for the duration of the
sampling period, recreating the staircase waveform which is approximation of modulating signal.
A low pass filter may be used to reduce the quantization noise and to yield the original
modulating signal.
Experimental procedure:

1. Connect the AC Adapter to the mains and the other side to the Experimental Trainer.
Power on the trainer kit.
2. Measure the Sampling Clock Frequency.
3. For visual convenience a DC Variable voltage is provided as Modulating signal. i.e.
Variable DC Output is connected to the analog input. The LEDs of ADC glow according
to the coded values of the strength of the DC signal.
4. LED „ON‟ represents „1‟ & „OFF‟ represents‟0‟.
5. As the DC voltage varies, the corresponding output data varies from 00000000 to
11111111. DMM is used to measure the DC voltage provided at the input. The readings
are tabulated as given in the tabular column.
6. Now by applying Variable frequency DC output at analog input of the modulator the
corresponding PCM modulated and demodulated signals can be observed and plotted.

Tabular form:

Sl. No Digital Output

DC Variable input voltage MSB bits LSB bits

(mV)

1 5 0000 0000
Result &Discussions: Thus the Pulse code modulation and demodulation is performed
practically.
 6.PULSE AMPLITUDE MODULATION

Aim:

To study the Pulse Amplitude Modulation and Demodulation using FT1503 Trainer.

Apparatus:

Sl.No Name of the component Specifications or Quantity

range

1 PAM Modulation and Demodulation - 1

experimental kit.( Built in DC Power
supply +/- 5V /350 mA, ICs provided
on the board TL084.LM565, 74LS163
each 1no, Provided with 2mm
Sockets.)

2 CRO 20MHz 1

3 Connecting wires, Probes and patch -- Based on requirement

chords

Theory:

Pulse Modulation may be used to transmit analog information, such as continuous speech
or data. It is a system in which continuous waveforms are sampled at regular
intervals.Information regarding the signal is transmitted only at the sampling times, together with
any synchronizing pulse that may be required. At the receiving end, the original waveforms may
be reconstructed from the information regarding the samples, if these are taken frequently
enough. Despite the fact that information about the signal is not supplied continuously, As in
Amplitude and Frequency modulation , the resulting receiver output can have regenerate the
analog information signal.

Pulse Modulation may be subdivided broadly into two categories, Analog and Digital. In
the former, the indication of sample Amplitude may be continuously variable, while in the later a
code which indicates the sample amplitude to the nearest predetermined level is sent. Pulse
Amplitude modulation is a form of analog communication which is discussed in the following
section.

Pulse Amplitude Modulation

In this we have a train of fixed width of pulses. The amplitude of each pulse is made
proportional to the amplitude of the modulating signal at that instant. In the PAM generation
circuit, Synchronous clock is applied to the base of the transistor. Modulating signal is applied to
the (unipolar positive) is given to the collector of the transistor. The output of the transistor
(Collector current) varies in accordance with the amplitude of the modulating signal voltage
resulting in modulated output.

Pulse Amplitude Demodulation

The Demodulation of Pulse Amplitude Modulation is quite a simple process. PAM signal
is fed to a Low Pass Filter, from which the Demodulating signal emerges, whose amplitude at
any time is proportional to the PAM at that time. This signal is given to an inverting amplifier to
amplify its level. The demodulated output is almost equal amplitude with the modulating signal
but is in phase shifted due to the modulation, demodulation process.

1 KHz Signal source (AF)

The circuit shown in the following figure uses two op-amps, one acting as non-inverting
integrator and the other one as inverting integrator. The two op-amps are connected in cascade to
form a feedback loop .the circuit oscillates with sinusoidal output. The sinusoidal oscillation
frequency is f = 1/2π RC. In practice the resistor R1 is made slightly larger than the other resistors
to ensure a sufficient positive feedback for oscillations. The two zener diodes Vz, used to bound
the output of the inverting integrator, so as to stabilize the amplitude of Oscillations.

Synchronous Clock generator

We are generating synchronous clock by using PLL technique .The NE 565 IC is a phase
locked loop, which is widely used in application such as frequency multiplication and synthesis
etc. This PLL device comprises of 4 basic elements phase comparator, low pass filter, error
correction amplifier and VCO. The VCO is a free running multi vibrator whose center frequency
is determined by an external timing capacitor and external resistor. It‟s center frequency can also
be shifted to either side by application of an input voltage to the appropriate terminal of the IC
.the frequency deviation is directly proportional to the input voltage and hence it is called a
“Voltage Controlled Oscillator”. The VCO output is presented to a phase detector where its
phase is compared with that of the input signal .The detector produces a DC output whose
magnitude is directly proportional to the phase difference. The output of VCO is divided
digitally by a number of times the multiplication is designed. Here the BC107 transistor acts as
an interface to drive the logic circuit. The sub divided frequency is given to phase comparator
which is a synchronous output.

Procedure:

1. Connect the AC adapter to the mains and the other side to the Experimental Trainer.
2. Observe the modulating signal generated by the 1 KHz Signal Source (AF) and note
down the peak to peak amplitude and Time Period.
3. Observe the Carrier signal generated by 8 KHz Synchronous Clock Generator and
measure the amplitude and frequency.
4. Apply the modulating signal generator output and Synchronous clock generator output to
the PAM modulator
5. The testing procedure is given by the following figure

6. By varying the amplitude of the modulating signal, depth of modulation varies.

7. During demodulation, connect PAM output to the input of the PAM demodulator and
observe the output of PAM demodulator.
8. Following figure shows the testing procedure.
Circuit Diagram:
Model graphs:
Tabulation:
Amplitude Time Frequency
Input
Sampling output
Reconstruction
output

Result & Discussions:

Pulse Amplitude Modulation and Demodulation is performed using FT1503 trainer. And the
modulated and demodulated waveforms were observed.
 7. PULSE POSITIONMODULATION - DEMODULATION

Aim:

1. To study the generation of Pulse Position Modulated and Demodulated signals using
PPM trainer kit DCT3206.

Apparatus:

Sl.No Name of the component Specifications or Quantity

range

1 PPM trainer kit DCT3206 ---- 1

2 CRO (0-30)MHz 1

3 Connecting wires and probes patch ---- Required number

chords

Theory:
Pulse modulation is used to transmit analog information. In this system continuous wave
forms are sampled at regular intervals. Information regarding the signal is transmitted only at the
sampling times together with synchronizing signals. At the receiving end, the Original signal
may be reconstructed from the information regarding the samples. Pulse Modulation may be
subdivided into two types, Analog and Digital. In analog the indication of sample amplitude is
the nearest variable. In Digital the information is a code.

The pulse position Modulation is one of the methods of the Pulse Time Modulation. PPM
is generated by changing the position of a fixed time slot. The amplitude and width of the pulses
is kept constant, while the position of each pulse, in relation to the position of the recurrent
reference pulse is valid by each instances sampled value of the modulating wave. Pulse position
Modulation into the category of Analog communication.PPM has the advantage of requiring
constant transmitter power output, but the disadvantage of depending on transmitter receiver
synchronization. PPM may be obtained very simply from PWM. However, in PWM the locations
of the leading edges are fixed, whereas those of the trailing edges are not. Their position depends
on pulse width, which is determined by signal amplitude at that instant. Thus, it may be said that
the trailing edges of PWM pulses are, in fact , position modulated. This has positive going pulses
corresponding to the trailing edge of an un modulated pulse is counted as zero displacement
other trailing edges will arrive earlier or later. They will therefore have a time displacement other
than zero. This time displacement is proportional to the instantaneous value of the signal voltage.
The differentiated pulses corresponding to the leading edges are removed with a diode clipper or
rectifier, and the remaining pulses, is position-modulated.
Circuit Diagram: Description

Modulating signal generator:

A 4.096 MHz clock is used to derive the modulating signal, which is generated by an
oscillator circuit comprising a 4.096 MHz crystal and three 74HC04 (U9) inverter gates. This
4.096 MHz clock is then divided down in frequency by a factor of 4096,by Binary counter
74HC4040(U4) , and 2 KHz square wave on pin no.15. The frequency is selectable by means of
SW1. This goes to input of fourth order low pass filter U 3(TL072) is used to produce sine wave
from the square wave. The amplitude of this sine wave can be varied.
Modulation:
The circuit uses 555IC (U1) a mono stable multi vibrator to perform the PPM. The
message signal is given to pin no.5 & at pin no.2 the pulse carrier is of 32KHz frequency is
connected internally & PWM, Differentiated PWM outputs are available at pins TP4,TP5.This
differentiated output is fed to the 555 IC(U2) in mono stable mode pin no.2. The PPM output is
available at pin no 6.
Model graphs:
Circuit Diagram:
Experimental procedure:

Procedure for PPM: modulation

1. Observe the signal generated by the Modulating signal generator at pin TP1 by
connecting any channel of the CRO by keeping frequency in 1 KHz position and
amplitude pot in max position.
2. Observe the pulse carrier signal at pin no2 TP3 of the 555 IC (U 1) measure its amplitude
and time period.
3. Now interconnect TP1 of modulating signal generator with TP2 of 555IC (U1) using
connecting wire.
4. Switch on the power supply, observe the PPM output at TP6 in CH1 of CRO with respect
to modulating signal in CH2 of CRO. Plot the PPM output wave carefully
5. By varying the amplitude and frequency of sine wave by varying amplitude pot and
frequency selection switch to 2 KHz and observe PPM output.
Demodulation:
1. Connect PPM output generated in step no 9. As input to the Low Pass Filter in the
Demodulation circuit at pin no TP7.

2. Switch on the power supply and observe the demodulated output at TP8 in CH1 of
CRO with respect to original signal at pin TP2 of 555IC(U1) in CH2 of CRO. Thus
the recovered signal is true replica of modulating signal in terms of frequency.
3. As the amplitude of LPF output is less, connect this output to an A.C amplifier and
observe the demodulated wave at pinTP10 by varying gain of the amplifier. This is
amplified Demodulated output.
4. Repeat the same procedure for 2 KHz modulating signal.
Tabulation:
Amplitude Time Frequency
Input
Sampling output
Reconstruction
output

Result & Discussions: The generation of PPM modulation-Demodulation is

studied using DCT3206 trainer kit.
 8.A. ASK GENERATION AND DETECTION

Aim:
To generate ASK Modulated wave and To Demodulate the ASK signal.

Apparatus Required:

Sl. No Name of the component Specifications or Quantity

range

1 ASK trainer kit ------- 1

2 CRO 20MHz 1

3 Connecting wires, Probes and Patch ---- As per required

chords. number

Theory:
Amplitude Shift Keying:
When it is required to transmit digital signals of the sinusoidal carrier is varied in
accordance with the incoming digital data since the digital data is in discrete steps, the
modulation of band pass sinusoidal carrier is also done in discrete steps. Therefore, this type of
modulation is called switching or signaling. If the amplitude of the carrier is switched depending
on the incoming digital signal then it is called amplitude shift keying (ASK). This is similar to
analog amplitude modulation.Amplitude shift keying (ASK) or ON-OFF keying is the simplest
digital modulation technique. In this method, there is only one unit energy carrier and sit is
switched on (or) off depending upon the input binary sequence. The ASK waveform can
represented as S(t) = 2Ps cos (2fot), (to transmit „1‟). To transmit symbol „0‟, the signal s(t)
= 0 where Ps is power dissipated and fo is carrier frequency. The Original Signal is first
transmitted and quantized as with PCM. If the sample currently being coded is above the
previous sample, then a binary bit is set to logic „1‟ .If the sample is lower than the previous
sample then the bit is set low.
Circuit Diagram:
Model graphs:
Experimental procedure:
Modulation:
1. Switch on the power supply.
2. The carrier frequency (sinusoid) is selected at carrier generation and is given to carrier
input TPL1.
3. The data clock duty cycle is adjusted by the potentiometer P1 and is given to the
modulation input at TP5. The data clock at TP6 is observed and connected to TP5.
4. By applying carrier input and digital system stream input to the double balanced
modulation the output ASK waveform is observed.
5. The ASK output can be adjusted by the gain adjustment potentiometer TP5.

Demodulation:
1. The ASK input is given to the input of rectifier.
2. This rectified signal is passed through low pass filter to remove carrier wave.
3. This out coming waveform is given to the data squaring circuit which sets up a threshold.
If the input to this circuit is greater than threshold it is set as +5V otherwise 0V.
4. The demodulated output at TP1 8 is observed.

Result &Discussions: Thus the ASK modulation and demodulation is performed practically
and the waveforms are plotted.
 8.B FSK GENERATION AND DETECTION
Aim:
1. To generate FSK modulated wave.
2. To generate Demodulated FSK signal.
3. To generate the NRZ-L signal.
Apparatus:

Sl. No Name of the component Specifications or Quantity

range

1 FSK trainer kit ------- 1

2 CRO 20MHz 1

3 Connecting wires, Probes and Patch ---- As per required

chords. number

Theory:
Frequency Shift Keying:
When a digital signal is to be transmitted over a long distance, it needs continuous wave
modulation.

A carrier of frequency „fo‟ is used for modulation. Then the digital signal modulates some
parameter like frequency, phase or amplitude of the carrier. The carrier „fo‟ has some deviation
in frequency. The deviation is called bandwidth of the channel. Thus the channel has to transmit
some range of frequency. Hence such a type of transmission is called band pass transmission
and the communication channel is called band pass channel.

When it is required to transmit digital signals on band pass channel the amplitude, frequency (or)
phase of the sinusoidal carrier is varied in accordance with the incoming digital data. Since the
digital data is in discrete steps, the modulation of band pass sinusoidal carrier is also done in
discrete steps. Hence this type of modulation is called switching or signaling. If the frequency
of the sinusoidal carrier is switched on depending on the incoming digital signal, then it is called
frequency shift keying (FSK). This is similar to analog frequency modulation.

In FSK the frequency of carrier is shifted according to binary symbol. That is there are 2
different frequency symbols according to binary symbols. Let there be a frequency shift by ,
then we can write the following equations.

If b (t) = 1, sH (t) = √2Ps Cos (2fo + ) t

If b(t) = 0, sL (t) = 2Ps Cos (2fo + ) t

Circuit Diagram:
Model graphs:
Experimental procedure:
Modulation:
1. The power supply is switched ON and data selection switch is set to the desired code.
2. The switch is set (DATA ON-OFF) ON position. The 8 bit word pattern is observed on
TP12.
3. Observe the data clock at TP1 and also observe NRZ (l) at TP2, RZ at TP3, and NRZ (M)
at TP4.
4. The patch cords are connected. The corresponding output (when data is logic „1‟
frequency is high) at TP8.
5. Repeat the steps for other inputs and observe the corresponding FSK outputs.
6. The data selections are changed and repeat the steps 3 to 6 & observe corresponding FSK
outputs.

Demodulation:

1. The patch chords are connected. The incoming FSK input is observed.
2. The output of square wave converter is available at TO10. The serial output data is
available at TP11.
3. Repeat steps 1, 2 for other serial data inputs and outputs and the corresponding serial data
is observed. The outputs are replica of the original inputs.

Result &Discussions: Thus the FSK modulation and demodulation is performed practically
and the waveforms are plotted.
 8.C. PSK GENERATION AND DETECTION
Aim:
To study the operation of phase shift keying modulation and demodulation
Apparatus:

Sl. No Name of the component Specifications or Quantity

range

1 PSK trainer kit-3210 ------- 1

2 CRO 20MHz 1

3 Connecting wires, Probes and Patch ---- As per required

chords. number
Theory:
Phase Shift Keying:
To transmit the Digital data from one place to another, we have to choose the
Transmission medium. The simplest possible method to connect the Transmitter to the Receiver
with a piece of wire. This works satisfactorily for short distances in some cases. But for long
distance communication with the aircraft, ship, vehicle this is not feasible. Here we have to opt
for the Radio Transmission.
It is not possible to send the digital data directly over the antenna because the antennae
of practiced size works on very high frequencies, much higher than our data transmission rate.
To be able to transmit the data over antennae, we have to modulate the signal i.e, phase,
frequency or amplitude etc is varied in accordance with the digital data. At receiver we separate
the signal from digital information by the process of demodulation. After this process we are left
with high frequency signal (carrier signal) which we discard & the digital information, which we
utilize. Modulation also allows different data streams to be transmitted over the same channel
(Transmission medium). This process is called as „Multiplexing‟ & result in a considerable
saving in Bandwidth no of channels to be used, Also it increases the channel efficiency. As
variation of particular parameter variation of the carrier wave give rise to various modulation
techniques. Some of the basic modulation techniques are ASK, FSK, PSK, DPSK &QPSK.
PSK is a form of angle modulated, constant amplitude digital modulation. Digital
communication become important with the expansion of the use of computers and data
processing and have continued to develop into a major industry providing the interconnection of
computer peripherals and transmission of data between distant sites.
PSK is a relatively new system in which the carrier may have phase which is fixed for
one level and is shifted by 1800 for another level. PSK is similar to PSK in many aspects, as in
PSK, frequency of carrier is shifted according to the modulating square wave.
Circuit Description:
In this trainer IC XR 2206 is a basic wave form generator which generates Sine, Square,
and Triangle Waveforms. The sine wave generated by this XR 2206 IC is used as carrier signal
to the system. This square wave is used as a clock input to a decade counter (IC 7490) which
generates the modulating data outputs. The digital signal is applied to the modulation input for
PSK generation is bipolar i.e., have equal positive and negative voltage levels. When the
modulating input is negative the output of the modulator is a sine wave in phase with the carrier
input. Whereas for the positive voltage levels, the modulator is a sine wave which is shifted out
of phase by 180o from the carrier input compared to the differential data stream. This happens
because the carrier input is now multiplied by the negative constant level. +Thus the output
changes in phase when a change in polarity of the modulating signal results. The functional
blocks of the PSK Modulator & Demodulator.
Modulation:
IC CD 4051 is an Analog multiplexer to which carrier is applied with and without 180 o
phase shift to the two multiplex inputs of the IC. Modulating data input is applied to its control
input. Depending up on the level of the controlling signal, carrier signal applied with or without
phase shift is steered to the output. The 180o phase shift to the carrier signal is carried by an
operational amplifier using 741 IC.
Demodulation:
During the demodulation the PSK signal is converted into a +5 volts Square wave signal
using a transistor and is applied to one input of an EX-OR gate. To the second input of the gate
carrier signal is applied after conversion into a +5 volts signal. So the EX-OR gate output is
equivalent to the modulating data signal.

Circuit Diagram:
Model graphs:
Experimental procedure:
Modulation:
1. The trainer is switched ON. The carrier signal is observed at TP1.
2. Data outputs (D1, D2, D3, and D4) are observed.
3. Carrier output TP1 is connected to carrier input of PSK modulator at
TP2 using patchchord.
4. Connect data input „D1‟ to input of PSK modulator at TP3. The PSK
output waveform isobserved on CRO on channel – 1 & corresponding
data input on channel – 2.
5. These steps are repeated for D2, D3 & D4 and the corresponding PSK output are
observed.

Demodulation:

1. Connect the PSK modulation output TP6 to the PSK input of Demodulator at TP4.
2. Connect carrier output TP1 to the carrier input of PSK modulator at TP5.
3. Observe the PSK modulated output at TP7 on CRO at Channel-1 and
corresponding dataoutput on channel-2.
4. The demodulated output is true replica of data outputs D2, D3, and D4.

Result &Discussions: Thus the PSK modulation and demodulation is

performed practicallyand the waveforms are plotted.
 9.DELTA / ADAPTIVE DELTA MODULATION AND DEMODULATION
AIM
To study the delta / Adaptive delta modulation and demodulation

EQUIPMENTS REQUIRED
a. DM / ADM kit
b. Two Channel 20MHz Oscilloscope
c. Patch Chords, Oscilloscope probe

THEORY
DELTA MODULATOR
The modulator comprises of comparator, quantizer and Integrator. The input base band
sinusoidal signal and its quantized approximated signals (feedback signal from integrator) are
applied to comparator. A comparator as its name suggests simply makes a comparison between
inputs. The comparator gives a TTL signal is then latched into a D-flip-flop which is clocked by
selected clock rates. The binary data stream from the flip flop is transmitted to receiver and is fed
to the integrator. The integrator output is then connected to the negative terminal of the voltage
comparator.
DELTA DEMODULATOR
The demodulator comprises of simple, integrator and low pass filter. The receive delta modulator
signal is applied to integrator; its output tries to follow the analog signal. The integrator output
contains sharp edges which are smoothened out by the 4th order low pass filter.

PROCEDURE
1. Connection are given as per the given block diagram.
2. To give a modulating input and sampled input (square wave form) to the input block.
3. To verify the output using CRO.
4. The output as given to the input of de modulated block and taken the output reading
5. Plot the graph

Tabulation:
Amplitude Time Frequency
Input
Transmitter output
Receiver output
RESULT
Thus the study of delta / Adaptive delta modulation and demodulation verified successfully.
 10. SIMULATION OF ASK, FSK, AND BPSK GENERATION SCHEMES

AIM
To study the FSK, PSK, BPSK Modulation using MATLAB code & observe the output
waveform.

APPARATUS REQUIRED:
Matlab software
PC

THEORY
FSK is one method used to overcome the bandwidth limitation of the telephone system so that
digital data can be sent over the phone lines. The basic idea of FSK is to represent 1s and 0s by
two different frequencies within the telephone bandwidth. The standard frequencies for a full
duplex 300 baud FSK Modulator & Demodulator in the originate modes are 1070 Hz for a 0
(called a space) and 1270 Hz for a 1 (called a mark). In the answer mode, 2025 Hz is a 0 and
2225 Hz is a 1. The relationship of these FSK frequencies and the telephone bandwidth is
illustrated in figure 1. Signals in both the originate and answer bands can exist at the same time
on the phone line and they do not interfere with each other because of the frequency separation.

WORKING OF FSK
In FSK, the carrier frequency is shifted in steps or levels corresponding to the levels of the digital
modulating signal. In the case of a binary signal, two carrier frequencies are used, one
corresponding to binary '0' (i.e space) and the other to a binary 1 (i.e mark). An example of a
digital data.

%Modulation ASK
g = [1 0 1 0 1];
f = 2;
t=0:2*pi/99:2*pi;
cp=[];sp=[];
mod=[];mod1=[];bit=[];
for n=1:length(g);
if g(n)==0;
die=zeros(1,100);
se=zeros(1,100);
else g(n)==1;
die=2*ones(1,100);
se=ones(1,100);
end
c=sin(f*t);
cp=[cp die];
mod=[mod c];
bit=[bit se];
end
ask=cp.*mod;
subplot(2,1,1);plot(bit,'LineWidth',1.5);grid on;
title('Binary Signal');
axis([0 100*length(g) -2.5 2.5]);
subplot(2,1,2);plot(ask,'LineWidth',1.5);grid on;
title('ASK modulation');
axis([0 100*length(g) -2.5 2.5]);

%>>>>>>>>> MATLAB code for binary FSK modulation and de-modulation >>>>>>>%

clc;
clear all;
close all;

x=[ 1 0 0 1 1 0 1]; % Binary Information

bp=.000001; % bit period
disp(' Binary information at Trans mitter :');
disp(x);

%XX representation of transmitting binary information as digital signal XXX

bit=[];
for n=1:1:length(x)
if x(n)==1;
se=ones(1,100);
else x(n)==0;
se=zeros(1,100);
end
bit=[bit se];

end
t1=bp/100:bp/100:100*length(x)*(bp/100);
subplot(3,1,1);
plot(t1,bit,'lineWidth',2.5);grid on;
axis([ 0 bp*length(x) -.5 1.5]);
ylabel('amplitude(volt)');
xlabel(' time(sec)');
title('transmitting information as digital signal');

%XXXXXXXXXXXXXXXXXXXXXXX Binary-FSK modulation XXXXXXXXXXXXXXXXXXXXXXXXXXX%

A=5; % Amplitude of carrier signal
br=1/bp; % bit rate
f1=br*8; % carrier frequency for information as 1
f2=br*2; % carrier frequency for information as 0
t2=bp/99:bp/99:bp;
ss=length(t2);
m=[];
for (i=1:1:length(x))
if (x(i)==1)
y=A*cos(2*pi*f1*t2);
else
y=A*cos(2*pi*f2*t2);
end
m=[m y];
end
t3=bp/99:bp/99:bp*length(x);
subplot(3,1,2);
plot(t3,m);
xlabel('time(sec)');
ylabel('amplitude(volt)');
title('waveform for binary FSK modulation coresponding binary information');

%XXXXXXXXXXXXXXXXXXXX Binary FSK demodulation XXXXXXXXXXXXXXXXXXXXXXXXXXXXX

mn=[];
for n=ss:ss:length(m)
t=bp/99:bp/99:bp;
y1=cos(2*pi*f1*t); % carrier siignal for information 1
y2=cos(2*pi*f2*t); % carrier siignal for information 0
mm=y1.*m((n-(ss-1)):n);
mmm=y2.*m((n-(ss-1)):n);
t4=bp/99:bp/99:bp;
z1=trapz(t4,mm) % intregation
z2=trapz(t4,mmm) % intregation
zz1=round(2*z1/bp)
zz2= round(2*z2/bp)
if(zz1>A/2) % logic lavel= (0+A)/2 or (A+0)/2 or 2.5 ( in this case)
a=1;
else(zz2>A/2)
a=0;
end
mn=[mn a];
end
disp(' Binary information at Reciver :');
disp(mn);

%XXXXX Representation of binary information as digital signal which achived

%after demodulation XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
bit=[];
for n=1:length(mn);
if mn(n)==1;
se=ones(1,100);
else mn(n)==0;
se=zeros(1,100);
end
bit=[bit se];

end
t4=bp/100:bp/100:100*length(mn)*(bp/100);
subplot(3,1,3)
plot(t4,bit,'LineWidth',2.5);grid on;
axis([ 0 bp*length(mn) -.5 1.5]);
ylabel('amplitude(volt)');
xlabel(' time(sec)');
title('recived information as digital signal after binary FSK demodulation');

%>>>>>>>>> MATLAB code for binary PSK modulation and de-modulation >>>>>>>%

clc;
clear all;
close all;

x=[ 1 0 0 1 1 0 1]; % Binary Information

bp=.000001; % bit period
disp(' Binary information at Trans mitter :');
disp(x);

%XX representation of transmitting binary information as digital signal XXX

bit=[];
for n=1:1:length(x)
if x(n)==1;
se=ones(1,100);
else x(n)==0;
se=zeros(1,100);
end
bit=[bit se];

end
t1=bp/100:bp/100:100*length(x)*(bp/100);
subplot(3,1,1);
plot(t1,bit,'lineWidth',2.5);grid on;
axis([ 0 bp*length(x) -.5 1.5]);
ylabel('amplitude(volt)');
xlabel(' time(sec)');
title('transmitting information as digital signal');

%XXXXXXXXXXXXXXXXXXXXXXX Binary-PSK modulation XXXXXXXXXXXXXXXXXXXXXXXXXXX%

A=5; % Amplitude of carrier signal
br=1/bp; % bit rate
f=br*2; % carrier frequency
t2=bp/99:bp/99:bp;
ss=length(t2);
m=[];
for (i=1:1:length(x))
if (x(i)==1)
y=A*cos(2*pi*f*t2);
else
y=A*cos(2*pi*f*t2+pi); %A*cos(2*pi*f*t+pi) means -A*cos(2*pi*f*t)
end
m=[m y];
end
t3=bp/99:bp/99:bp*length(x);
subplot(3,1,2);
plot(t3,m);
xlabel('time(sec)');
ylabel('amplitude(volt)');
title('waveform for binary PSK modulation coresponding binary information');

%XXXXXXXXXXXXXXXXXXXX Binary PSK demodulation XXXXXXXXXXXXXXXXXXXXXXXXXXXXX

mn=[];
for n=ss:ss:length(m)
t=bp/99:bp/99:bp;
y=cos(2*pi*f*t); % carrier siignal
mm=y.*m((n-(ss-1)):n);
t4=bp/99:bp/99:bp;
z=trapz(t4,mm) % intregation
zz=round((2*z/bp))
if(zz>0) % logic level = (A+A)/2=0
%becouse A*cos(2*pi*f*t+pi) means -A*cos(2*pi*f*t)
a=1;
else
a=0;
end
mn=[mn a];
end
disp(' Binary information at Reciver :');
disp(mn);

%XXXXX Representation of binary information as digital signal which achived

%after PSK demodulation XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
bit=[];
for n=1:length(mn);
if mn(n)==1;
se=ones(1,100);
else mn(n)==0;
se=zeros(1,100);
end
bit=[bit se];
end
t4=bp/100:bp/100:100*length(mn)*(bp/100);
subplot(3,1,3)
plot(t4,bit,'LineWidth',2.5);grid on;
axis([ 0 bp*length(mn) -.5 1.5]);
ylabel('amplitude(volt)');
xlabel(' time(sec)');
title('received information as digital signal after binary PSK
demodulation');

%>>>>>>>>>>>>>>>>>>>>>>>>>> end of program >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>%

%>>>>>>>>>>>>>>>>>>>>>>>>>> end of program >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>%

Result:
Thus the experiment was verified successfully.
 11.SIMULATION OF DPSK, QPSK, QAM GENERATION SCHEMES
AIM:
To generate a DPSK,QPSK,QAM modulated waveform and to simulate using MATLAB

THEORY
QPSK MODULATOR
Quaternary phase shift keying (QPSK), or quadrature PSK as it is sometimes called, is
another form of angle-modulated, constant-amplitude digital modulation. QPSK is an M-ary
encoding technique where M = 4 (hence, the name “quaternary,” meaning “4" ). With QPSK
four
output phases are possible for a single carrier frequency. Because there are four different output
phases, there must be four different input conditions. Because the digital input to a QPSK
modulator is a binary (base 2) signal, to produce four different input conditions, it takes more
than
a single input bit. With two bits, there are four possible conditions: 00, 01, 10 and
11. Therefore, with QPSK, the binary input data are combined into groups of two bits called
dibits. Each dibit code generates one of the four possible output phases. Therefore, for each
two - bit dibit clocked into the modulator, a single output change occurs. Therefore, the rate of
change at the output (baud rate ) is onehalf of the input bit rate.
A block diagram of QPSK modulator is shown in above Figure. Two bits (a dibit) are
clocked into the bit splitter. After both bits have been serially inputted, they are simultaneously
parallel outputted. One bit is directed to the I channel and the other to the Q channel. The 1- bit
modulates a carrier that is in phase with the reference oscillator (hence, the name “I” for “in
phase” channel), and the Q bit modulates a carrier that is 90° out of phase or in quadrature with
the reference carrier (hence, the name “Q” for “quadrature” channel).

Program:
QPSK:
%XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
%XXXX QPSK Modulation and Demodulation without consideration of noise XXXXX
%XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX

clc;
clear all;
close all;
data=[0 1 0 1 1 1 0 0 1 1]; % information

%Number_of_bit=1024;
%data=randint(Number_of_bit,1);

figure(1)
stem(data, 'linewidth',3), grid on;
title(' Information before Transmiting ');
axis([ 0 11 0 1.5]);

data_NZR=2*data-1; % Data Represented at NZR form for QPSK modulation

s_p_data=reshape(data_NZR,2,length(data)/2); % S/P convertion of data

br=10.^6; %Let us transmission bit rate 1000000

f=br; % minimum carrier frequency
T=1/br; % bit duration
t=T/99:T/99:T; % Time vector for one bit information

% XXXXXXXXXXXXXXXXXXXXXXX QPSK modulatio XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX

y=[];
y_in=[];
y_qd=[];
for(i=1:length(data)/2)
y1=s_p_data(1,i)*cos(2*pi*f*t); % inphase component
y2=s_p_data(2,i)*sin(2*pi*f*t) ;% Quadrature component
y_in=[y_in y1]; % inphase signal vector
y_qd=[y_qd y2]; %quadrature signal vector
y=[y y1+y2]; % modulated signal vector
end
Tx_sig=y; % transmitting signal after modulation
tt=T/99:T/99:(T*length(data))/2;

figure(2)

subplot(3,1,1);
plot(tt,y_in,'linewidth',3), grid on;
title(' wave form for inphase component in QPSK modulation ');
xlabel('time(sec)');
ylabel(' amplitude(volt0');

subplot(3,1,2);
plot(tt,y_qd,'linewidth',3), grid on;
title(' wave form for Quadrature component in QPSK modulation ');
xlabel('time(sec)');
ylabel(' amplitude(volt0');

subplot(3,1,3);
plot(tt,Tx_sig,'r','linewidth',3), grid on;
title('QPSK modulated signal (sum of inphase and Quadrature phase signal)');
xlabel('time(sec)');
ylabel(' amplitude(volt0');

% XXXXXXXXXXXXXXXXXXXXXXXXXXXX QPSK demodulation XXXXXXXXXXXXXXXXXXXXXXXXXX

Rx_data=[];
Rx_sig=Tx_sig; % Received signal
for(i=1:1:length(data)/2)

%%XXXXXX inphase coherent dector XXXXXXX

 Z_in=Rx_sig((i-1)*length(t)+1:i*length(t)).*cos(2*pi*f*t);
% above line indicat multiplication of received & inphase carred signal

Z_in_intg=(trapz(t,Z_in))*(2/T);% integration using trapizodial rull

if(Z_in_intg>0) % Decession Maker
Rx_in_data=1;
else
Rx_in_data=0;
end

%%XXXXXX Quadrature coherent dector XXXXXX

Z_qd=Rx_sig((i-1)*length(t)+1:i*length(t)).*sin(2*pi*f*t);
%above line indicat multiplication ofreceived & Quadphase carred signal

Z_qd_intg=(trapz(t,Z_qd))*(2/T);%integration using trapizodial rull

if (Z_qd_intg>0)% Decession Maker
Rx_qd_data=1;
else
Rx_qd_data=0;
end

Rx_data=[Rx_data Rx_in_data Rx_qd_data]; % Received Data vector

end

figure(3)
stem(Rx_data,'linewidth',3)
title('Information after Receiveing ');
axis([ 0 11 0 1.5]), grid on;

% XXXXXXXXXXXXXXXXXXXXXXXXX end of program XXXXXXXXXXXXXXXXXXXXXXXXXX

DPSK:

N = 10^4 % number of bits or symbols

rand('state',100); % initializing the rand() function
randn('state',200); % initializing the randn() function
ip = rand(1,N)>0.5; % generating 0,1 with equal probability
ipD = mod(filter(1,[1 -1],ip),2); % %differential encoding y[n]=y[n-1]+x[n]
s = 2*ipD-1; % BPSK modulation 0 -> -1; 1 -> 0
n = 1/sqrt(2)*[randn(1,N) + j*randn(1,N)]; % white gaussian noise, 0dB
variance
Eb_N0_dB = [-3:10]; % multiple Eb/N0 values
for ii = 1:length(Eb_N0_dB)
y = s + 10^(-Eb_N0_dB(ii)/20)*n; % additive white gaussian noise
ipDHat_coh = real(y) > 0; % coherent demodulation
ipHat_coh = mod(filter([1 -1],1,ipDHat_coh),2); %differential decoding
nErr_dbpsk_coh(ii) = size(find([ip - ipHat_coh]),2); % counting the number of
errors 10
end
simBer_dbpsk_coh = nErr_dbpsk_coh/N;
theoryBer_dbpsk_coh = erfc(sqrt(10.^(Eb_N0_dB/10))).*(1 -
.5*erfc(sqrt(10.^(Eb_N0_dB/10))));
close all
figure
semilogy(Eb_N0_dB,theoryBer_dbpsk_coh,'b.-');
hold on
semilogy(Eb_N0_dB,simBer_dbpsk_coh,'mx-');
axis([-2 10 10^-6 0.5])
grid on
legend('theory', 'simulation');
xlabel('Eb/No, dB')
ylabel('Bit Error Rate')
title('Bit error probability curve for coherent demodulation of DBPSK')

QAM:
M = 16; % M determine the size of signal constellation
k = log2(M); % k determine the number of bits per symbol
n = 3e4; % n determine the number of bits to process
nSyms = n/k; % nSyms represent the number of symbols
hMod = modem.qammod(M); % We will create a 16-QAM modulator
hMod.InputType = 'Bit'; % Then accept bits as inputs
hMod.SymbolOrder = 'Gray'; % accept bits as inputs
hDemod = modem.qamdemod(hMod); %
x = randi([0 1],n,1); % Random binary data stream
tx = modulate(hMod,x);
EbNo = 0:10; % In dB
SNR = EbNo + 10*log10(k);
rx = zeros(nSyms,length(SNR));
bit_error_rate = zeros(length(SNR),1);
for i=1:length(SNR)
rx(:,i) = awgn(tx,SNR(i),'measured');
end
rx_demod = demodulate(hDemod,rx);
for i=1:length(SNR)
[~,bit_error_rate(i)] = biterr(x,rx_demod(:,i));
end
theoryBer = 3/(2*k)*erfc(sqrt(0.1*k*(10.^(EbNo/10))));
figure;
semilogy(EbNo,theoryBer,'-',EbNo, bit_error_rate, '^-');
grid on;
legend('Theory BER', 'simulation');
xlabel('Eb/No, dB');
ylabel('Bit Error Rate');
title('Bit error probability curve for 16-QAM modulation');

Result:
Thus the experiment was verified successfully.
 12. A. IMPLEMENTATION OF LINEAR BLOCK CODES
AIM:
Construct a (7, 4) linear block code whose generator matrix is given by,
1000111
G= 0100110
Determine all code words and the minimum weight of the code.
SOFTWARE REQUIRED:
MATLAB 7.0 software
THEORY:
Linear block codes:
Its one of the error control coding. Linear codes means that sum of any two code vector
gives another code vector. Also it is a systematic code. Block codes in which the message
bits are transmitted in unaltered form are called systematic code.
Consider an (n, k) linear block code in which „k‟ is a message bit, „n‟ is block length and
b=n-k is a parity check bit.
bo,b1,b2,b2…………………….bn-k-1 mo,m1,m2.m3……………………….mk-
1
Structure of code word
Message Vector m= [mo, m1, m2 …………mk-1]
Parity check vector b= [ b0, b1, b2,……….bn-k-1]
Code vector X= [Xo, X1, X2………….Xn-1 ]
b=m x P
Define the k by n generator matrix G= {P: Ik}
Define the (n-k) by k sub matrix H= [ Ik : P^T]
Parity check vector b= m x P
Code vector X= Message vector check vector

ALGORITHM:
• From the given (n, k) block code assign the values of „k‟ nothing
but Number of message bit.
• Assign the given generator matrix.
• Compute the check vector, then arrange code vector by
combining Message and check vector.
• Find the weight of the code that is by finding minimum hamming
weight Of the Code which is nothing but number of non zero bits in a
code Vector.
• Find the minimum weight of the code, from that we can understand that

clc;
clear;
k=4;
for i=1:2^k
for j=k:-1:1
if rem((i-1),2^(-j+k+1))>=2^(-j+k)
u(i,j)=1;
else
u(i,j)=0
end
echo off;
end
end
echo on;
G=[1 0 0 0 1 1 1;
0 1 0 0 1 1 0;
0 0 1 0 1 0 1;
0 0 0 1 0 1 1]
c=rem(u*G,2);
disp(c);
w_min=min(sum((c(2:2^k,:))'));
disp(w_min);

RESULT:
All possible code vector and weight of the given linear block code is found. From the values
of dmin=3 the given linear block code is found that is hamming code.
 13. B. IMPLEMENTATION OF CYCLIC CODE GENERATION

AIM:
To simulate the generates Matrix, Code word, Parity check Matrix and error syndrome for
a (7, 4) cyclic code using MATLAB.

APPARATUS REQUIRED:
1. Personal computer.
2. MATLAB software.

THEORY:
Error control coding is the processor of adding redundant list to the information bits, So on to
simulate two level objectives at the receiver. Error detection and correction. A block code is
linear if any linear combination of its code words a code is cyclic, if any cyclic shift of a code
and is also a code word. They are usually denoted by (n, k) in which the first position of k bits
is always identical to the message sequence to the transmitted. The block length is denoted by
n.

%ENCODING
clc;
n=7; % CODE LENGTH
k=4; % NUMBER OF MESSAGE BITS
disp('MESSAGE'); % RANDOM MESSAGE GENERATION
m=randint(2,k,[0,1]);
disp(m);
disp('POLYNOMIAL'); % GENERATOR POLYNOMIAL
pol=cyclpoly(n,k,'min');
disp(pol);
disp('CODE VECTOR'); % CODE VECTOR GENERATION
code=encode(m,n,k,'CYCLIC/FMT',pol);
disp(code);
disp('ERROR'); % RANDOM ERROR GENERATION
e=randerr(2,n,[1 0;0.8 0.2]);
disp(e);
disp('RECEIVED MATRIX'); % RECEIVED MATRIX
r=rem(plus(code,e),2);
disp(r);
[newmsg err cc]=decode(r,n,k,'CYCLIC'); % DECODING OF RECEIVED MESSAGE
disp('DECODED RECEIVED VECTOR');
disp(cc);
disp('DECODED MESSAGE');
disp(newmsg);

RESULT:
Thus the simulation for cyclic code is done using MATLAB.