EXPERIMENT NO:1

SIGNAL SAMPLING AND RECONSTRUCTION

AIM:-
Study of Sampling Process and Signal Reconstruction

APPARATUS REQUIRED:-
1. ST2101 with power supply cord.
2. Oscilloscope with connecting probe.
3. Connecting cords.
THEORY:-
The signals we use in the real world, such as our voice, are called "analog" signals. To process these signals for
digital communication, we need to convert analog signals to "digital" form. While an analog signal is continuous
in both time and amplitude, a digital signal is discrete in both time and amplitude. To convert continuous time
signal to discrete time signal, a process is used called as sampling. The value of the signal is measured at certain
intervals in time. Each measurement is referred to as a sample.

Principle of sampling:-
Consider an analogue signal x(t) that can be viewed as a continuous function of time, as shown in figure. We
can represent this signal as a discrete time signal by using values of x(t) at intervals of nTs to form x(nTs) as
shown in figure . We are “grabbing" points from the function x(t) at regular intervals of time, Ts, called the
sampling period.

Aliasing:
A precondition of the sampling theorem is that the signal to be band limited. However, in practice, no time-
limited signal can be band limited. Since signals of interest are almost always time-limited (e.g., at most
spanning the lifetime of the sampling device in question), it follows that they are not band limited. However,
by designing a sampler with an appropriate guard band, it is possible to obtain output that is as accurate as
necessary. Aliasing is the presence of unwanted components in the reconstructed signal. These components
were not present when the original signal was sampled. In addition, some of the frequencies in the original
signal may be lost in the reconstructed signal. Aliasing occurs because signal frequencies can overlap if the
sampling frequency is too low. As a result, the higher frequency components roll into the reconstructed signal
and cause distortion of the signal Frequencies "fold" around half the sampling frequency. This type of signal
distortion is called aliasing. We only sample the signal at intervals. We don't know what happened between the
samples. A crude example is to consider a 'glitch' that happened to fall between adjacent samples. Since we
don't measure it, we have no way of knowing the glitch was there at all.
In a less obvious case, we might have signal components that are varying rapidly in between samples. Again,
we could not track these rapid inter-sample variations. We must sample fast enough to see the most rapid
changes in the signal. Sometimes we may have some a prior knowledge of the signal, or be able to make some
assumptions about how the signal behaves in between samples.

CIRCUIT DIAGRAM:-

Signal Sampling
SIGNAL RECONSTRUCTION:-

Signal Reconstruction

PROCEDURE:-
A. Set up for Sampling and reconstruction of signal. Initial set up of trainer: Duty cycle selectors switch
position: Position 5.
Sampling selector switch: Internal position.
1. Connect the power cord to the trainer. Keep the power switch in ‘Off’ position.
2. Connect 1 KHz Sine wave to signal Input.
3. Switch ‘On’ the trainer's power supply & Oscilloscope.
4. Connect BNC connector to the CRO and to the trainer’s output port.

5. Select 320 KHz (Sampling frequency is 1/10th of the frequency indicated by the illuminated LED)
sampling rate with the help of sampling frequency selector switch.
6. Observe 1 KHz sine wave (TP12) and Sample Output (TP37) on Oscilloscope. The display shows 1
KHz Sine wave being sampled at 32 KHz, so there are 32 samples for every cycle of the sine wave.
 7. Connect the Sample output to Input of Fourth Order low pass Filter & observe reconstructed output on
(TP46) with help of oscilloscope. The display shows the reconstructed original 1 KHz sine wave.
8. By successive presses of sampling Frequency Selector switch, change the sampling frequency to
2KHz, 4KHz, 8KHz, 16KHz and back to 32KHz (Sampling frequency is 1/10th of the frequency
indicated by the illuminated LED). Observe how SAMPLE output changes in each cases and how the
lower sampling frequencies introduce distortion into the filter’s output waveform. This is due to the
fact that the filter does not attenuate the unwanted frequency component significantly. Use of higher
order filter would improve the output waveform.
9. So far, we have used sampling frequencies greater than twice the maximum input frequency.

CONCLUSION:-
As the sampling frequency increases the output of sample port has more number of samples of applied input
signal.
B. Setup of Nyquist criteria and aliasing:-
Initial set up of trainer: Duty cycle selector switch position: Position 5
Sampling selector switch: Internal position.
1. Keep the power switch in ‘Off’ position.
2. Connect 2 volts peak, 2 KHz sine wave from 600 ohms output of the Function Generator to the signal
Input of the trainer.
3. Switch ‘On’ the trainer's power supply & Oscilloscope.

4. Connect BNC connector to the CRO and to the trainer’s output port.
5. Select 320 KHz (Sampling frequency is 1/10th of the frequency indicated by the Illuminated LED)
sampling rate with the help of sampling frequency selector Switch.
6. Connect the sample output to fourth order low pass filter & observe the output (TP46) on oscilloscope.
Observe the two waveforms (applied input signal & filter output) which are similar but the second
waveform (filter output) is lagging in phase. This is as expected from filters phase/ frequency response.

7. Decrease the sampling rate from 32 KHz to 2 KHz. Observe the distorted waveform at filter's output
(TP46). This is due to the fact that we under-sampled the input waveform overlooking the Nyquist
criteria and thus the output was distorted even though the signal lies below the cut-off frequency (3.4
KHz) of the filter. This explains the phenomena of Aliasing.

RESULT:-
As the input sampling frequency is smaller than the applied input signal then the output is distorted means the
original signal cannot be reconstructed.
 EXPERIMENT.NO:2
GENERATION OF AMPLITUDE MODULATION AND DEMODULATION

AIM: To study the function of Amplitude Modulation & Demodulation (under modulation, perfect modulation & over
modulation) and also to calculate the modulation index, efficiency,

APPARATUS REQUIRED:-

1. ACL-01 Amplitude & ACL-02 Amplitude with power supply cord.

2. CRO with connecting probe.
3. Connecting cords.

THEORY:-
Most digital modulation systems are based on pulse modulation. It involves variation of a pulse
parameter in accordance with the instantaneous value of the information signal. This parameter can be
amplitude, width, repetitive frequency etc. Depending upon the nature of parameter varied, various
modulation systems are used. Pulse amplitude modulation, pulse width modulation, pulse code
modulation are few modulation systems cropping up from the pulse modulation technique. In pulse
amplitude modulation (PAM) the amplitude of the pulses are varied in accordance with the modulating
signal. In true sense, pulse amplitude modulation is analog in nature but it forms the basis of most
digital communication and modulation systems. The pulse modulation systems require analog
information to be sampled at predetermined intervals of time. Sampling is a process of taking the
instantaneous value of the analog information at a predetermined time interval. A sampled signal
consists of a train of pulses, where each pulse corresponds to the amplitude of the signal at the
corresponding sampling time. The signal sent to line is modulated in amplitude and hence the name
Pulse Amplitude Modulation (PAM).
Natural sampling:- In the analogue-to-digital conversion process an analogue waveform is sampled
to form a series of pulses whose amplitude is the amplitude of the sampled waveform atb the time the
sample was taken. In natural sampling the pulse amplitude takes the shape of the
analogue waveform for the period of the sampling pulse as shown in figure.

BLOCK DIAGRAM OF AM TRANSMITTER:

 BLOCK DIAGRAM OF AM RECEIVER:

EXPECTED WAVE FORMS

PROCEDURE:-
1. Connect the circuit as shown in Figure.
a. Output of sine wave to modulation signal input in PAM block keeping the switch in 1 KHz position.
b. 8 KHz pulse output to pulse input.
2. Switch ‘On’ the power supply & oscilloscope.

3. Observe the outputs at TP(3 & 5) these are natural & flat top outputs respectively.
4. Observe the difference between the two outputs.
5. Vary the amplitude potentiometer and frequency change over switch & observe the effect on the two
outputs.
6. Vary the frequency of pulse, by connecting the pulse input to the 4 frequencies available i.e. 8, 16,
32, 64 kHz in Pulse output block.
OBSERVATIONS:

MODULATION:

MODULATIONS Vc Vm Vmax Vmin ( 𝐕𝐦𝐚𝐱− 𝐕𝐦𝐢𝐧) 𝐕𝐦

µ=( 𝐕𝐦𝐚𝐱+ 𝐕𝐦𝐢𝐧) µ= 𝐕𝐜
Under modulation

Perfect modulation

Over modulation

DEMODULATION:

Modulating signal frequency Demodulated signal frequency

RESULT :
The amplitude modulated wave is observed for different modulation indexes.In amplitude modulation by
increasing the message amplitude we observed different modulationindexes such as under modulation
(µ<1), over modulation(µ>1) and exact modulation(µ=1).
 EXPERIMENT NO: 3
FREQUENCY MODULATION & DEMODULATION
AIM:

To study the functioning of frequency modulation & demodulation and to calculate the
modulation index.

APPARATUS:

1. ACL-03 Frequency modulation & ACL-04 demodulation trainer kit.

2. C.R.O (20MHz)
3. Function generator (1MHz).
4. Connecting wires

BLOCK DIAGRAM:
Frequency modulator & De modulator

THEORY:
FM is a system in which the amplitude of the modulated carrier is kept constant, while its
frequency and rate of change are varied by the modulating signal.
By the definition of FM, the amount by which the carrier frequency is varied from its
unmodulated value, called the deviation, is made proportional to the instantaneous amplitude of the
modulating voltage. The rate at which this frequency variation changes or takes place is equal to
the modulating frequency.
FM is that form of angle modulation in which the instantaneous frequency fi(t) is varied
linearly with the message signal m(t), as
fi(t) =fC+kf m(t)
 The term fc represents the frequency of the unmodulated carrier, and the constant Kf
represents the frequency sensitivity of the modulator expressed in Hertz per volt.
Unlike AM, the spectrum of an FM signal is not related in a simple manner to that of modulating
signal, rather its analysis is much more difficult than that of an AM signal

PROCEDURE:
1. Switch on the experimental board.
2. Observe the FM modulator output without any modulator input which is the carrier signal
and note down its frequency and amplitude.
3. Connect modulating signal to FM modulator input and observe modulating signal and FM
output on two channels of the CRO simultaneously.
4. Adjust the amplitude of the modulating signal until we get less distorted FM output.

OBSERVATIONS: MODULATION

Vm F1 F2 Frequency Modulating Bandwidth

deviation index 2(β+1)fm
Δf= F1-F2 𝚫𝐟
β= 𝐅𝐦

DEMODULATION

Modulating Signal Demodulating signal

frequency frequency
EXPECTED WAVE FORM:

RESULT :
The Frequency modulated wave is observed for different modulation indexes.In frequency modulation
by increasing the message amplitude we observed different modulationindexes such as under
modulation (µ<1), over modulation(µ>1) and exact modulation(µ=1).
 EXPERIMENT NO:4
PRE-EMPHASIS AND DE-EMPHASIS

AIM:

To Construct and Verify Pre-emphasis and De-emphasis and Plot the Waveforms.

APPARATUS REQUIRED:

1. Resistors (10 K-2, 47K, 75K, 1K)

2. Capacitors (22μF, 0.1μ-2,)
3. Transistor BC107
4 Function generators
5. CRO
6. Connecting Wires
7. RPS (15V)

THEORY:

The noise has a effect on the higher modulating 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 de-emphasis

CIRCUIT DIAGRAM:

Pre-Emphasis:
 De-Emphasis:
PROCEDURE:

1. Connect the circuit as per circuit diagram as shown in Fig.

2. Apply the sinusoidal signal of amplitude 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 20 log (vo/vi).
4. Plot the graph between gain Vs frequency.
5. Repeat above steps 2 to 4 for de-emphasis circuit (shown in Fig.2). by applying the
sinusoidal signal of 5V as input signal.

TABULAR FORM:
PRE-EMPHASIS:
S.NO. Frequency (Hz) I/P Voltage O/P Voltage Gain in dB
Vi Vo 20log(Vo /V i)
DE-EMPHASIS:

S.NO. Frequency (Hz) I/P Voltage O/P Voltage Gain in dB

Vi Vo 20log(Vo /V i)

MODEL GRAPH

RESULT:-
Thus the pre-emphasis and de-emphasis characteristics are studied
 EXPERIMENT NO: 5

PULSE AMPLITUDE MODULATION (PAM) &

PULSE POSITION MODULATION

AIM:
To study Pulse Amplitude modulation and demodulation process with Relevant
waveforms.

APPARATUS:-
1. Pulse amplitude modulation & demodulation Trainer Kit.
2. Dual trace CRO.
3. Patch chords.
4. PC with windows(95/98/XP/NT/2000)
5. MATLAB Software with communication toolbox

BLOCK DIAGRAM: PULSE AMPLITUDE MODULATION

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 syncing signals.
At the receiving end, the original waveforms may be reconstituted from the
information regarding the samples.

The pulse amplitude modulation is the simplest form of the pulse modulation.
PAM is a pulse modulation system is which the signal is sampled at regular intervals, and
each sample is made proportional to the amplitude of the signal at the instant of sampling.
The pulses are then sent by either wire or cables are used to modulated carrier.
The two types of PAM are i) Double polarity PAM, and ii) the single polarity
PAM,
in which a fixed dc level is added to the signal to ensure that the pulses are always
positive. Instantaneous PAM sampling occurs if the pulses used in the modulator are
infinitely short.
Natural PAM sampling occurs when finite-width pulses are used in the
modulator, but the tops of the pulses are forced to follow the modulating waveform.
Flat-topped sampling is a system quite often used because of the ease of
generating the modulated wave.
PAM signals are very rarely used for transmission purposes directly. The
reason for this lies in the fact that the modulating information is contained in the
amplitude factor of the pulses, which cused frequently as an intermediate step in
other pulsemodulating methods, especially where time- division multiplexing is
used.

PROCEDURE:

1. Switch “ON” the experimental kit.

2. Observe the clock generator output & modulation signal outputs.
3. Connect clock generator output to the clock input point of
PWM modulatorAnd observe the same clock on channel of a
dual trace CRO.
4. Trigger the CRO with respect to CH 1.

5. Apply a variable DC voltage of 8 to 12 volts from any external regulated

Power supply.
6. Observe the PWM output on CH 2.
7. If we observe the PWM output, it’s width varies according to the Modulating
voltage.
8. A variable amplitude modulating signal is given to
observe how the PWMare varying for AC modulating
voltages.
9. In this case we have to trigger the CRO with respect to modulating voltage.

EXPECTED WAVE FORMS:

BLOCK DIAGRAM:

PPM MODULATION
EXPECTED WAVE FORMS

RESULT:

The PAM AND PPM wave forms are generated and plotted.
 EXPERIMENT NO:6
GENERATION OF ASK AND FSK
AIM:

To Generate ASK and FSK

APPARATUS REQUIRED:

ASK kit, connecting probes, CRO.

THEORY:

The binary ASK was one of the earliest forms of digital modulation used in wireless
telegraphy. this simplest form of the digital modulation is no longer used widely in digital
communication. Nevertheless it serves as a useful model which helps in understanding certain
concepts. In an ASK system binary symbol 1 is represented by transmitting a sinusoidal carrier
wave of fixed amplitude Ac and fixed frequency Fc for the bit duration Tb seconds whereas
binary symbol is represented by switching of the carrier for Tb seconds. This signal can be
generated simply by tuning the carrier of sinusoidal oscillator on end of for the prescribed
period by the modulating pulse trend. For this reason, the scheme is also known as on-off
scheme.

CIRCUIT DIAGRAM:-

PROCEDURE:

FOR ASK MODULATION:

1. Connect the output of carrier wave generator (carrier output) to carrier input of ASK
modulator & connect the modulating output to modulating input through petchcords.
2. Connect the CRO across output of ASK modulator.
3. Switch on the CRO as well as instrument.
4. Observe the output wave shape on CRO.
 5. Change the amplitude & frequency of modulating signal & observe corresponding
effect on ASK modulated wave. set the output shape (Ask modulated) through offset
potentiometer provided on the front panel.

PROCEDURE
FOR ASK DEMODULATION:
 Connect the output of ASK modulator to the input of demodulator through petchcord.

 Connect CRO probe across output of demodulator.

 Observe the demodulated (square wave) shape on CRO. We will observe that the
demodulated output is of same frequency as the modulated signal with little distortion.

PRECAUTIONS:

1. Check the cont unity of the connecting probes.

2. Handle the CRO properly.

EXPECTED WAVE FORMS:

 (a) Study of Frequency Shift Keying.

APPARATUS REQUIRED:-

Data generator, FSK modulation kit, DSO and connecting leads.

THEORY:-

FSK is one of the basic modulation techniques for the transmission of digital data .If the
frequency of the sinusoidal carrier is switched depending upon the input digital signal , then it
is known as frequency shift keying. As the amplitude remains constant in FSK, so the effect
of non-linear ties, noise interference is minimum on digital detection. So FSK is preferred over
ASK.
Frequency shift keying consists of shifting of frequency of carrier from a mask frequency to a
space frequency according to the base band digital signal Frequency shift keying is identical
to modulating an FM carrier with a binary digital signal In an FSK system, two sinusoidal
carrier waves of the same amplitude Ac but different frequencies fc1 and fc2 are used to
represent binary symbols 1and 0 respectively. It can be easily verified that binary FSK
waveform is a superposition of two binary ASK waveforms, one with a frequency fc1 and
other with a frequency fc2. No discrete components appear in the signal spectrum of FSK
signal. The main advantage of FSK lies in its easy hardware implementation.

Generation of FSK signal:-

The PSK signal can be generated by applying the incoming binary data to a frequency
modulator. To the other input a sinusoidal carrier wave of constant amplitude Ac and
frequency fc is applied. As the modulating voltages changes from one level to another, the
frequency modulator output changes its frequency in the corresponding fashion.

Detection of FSK signal:-

FSK can be demodulated by using coherent and non-coherent detector. The detector based on
coherent detection requires phase and timing synchronization. Non coherent detection can be
done by using envelop detector.
RESULT:-

The ASK and FSK output is obtained on DSO.

 EXPERIMENT NO:7

GENERATION OF PSK AND QPSK

AIM :

TO GENERATE AND PLOT OF PSK AND QPSK

APPARATUS REQUIRED:-

PSK Kit , connecting probes, CRO.

THEORY:-
Digital communication became important with the expansion of the use of the computers and
data processing and have continue to develop in to a major industry providing the
interconnection of computer peripherals and transmission of data between distance sides. PSK
is a relatively new system in which carrier may be phase shifted by 90° for a mark and by
minus 90° for a space. PSK has a number of similarities to FSK in many aspects, as in FSK,
frequency of the carrier is shifted according to modulating square wave.

CIRCUIT DIAGRAM:-
PROCEDURE:-
FOR PSK MODULATION:-
 Connect the carrier Output of the carrier wave generator (IC 8038) to carrier input of
PSK modulator through patch cords. Also connect any data output from the data
outputs of the data generator to the data input of the PSK modulator.
 Connect the channel 1 of CRO across output of PSK. And channel 2 across data input
on a dual trace oscilloscope.

 Switch on the instrument using ON/OFF toggle switch provided on the front panel.
 Observe the output wave shape on CRO.
 Change the data inputs and observe the PSK output on CRO.

FOR PSK DEMODULATION:-

 Connect the output of PSK to the demodulator input through patch cord.

 Connect the carrier to the carrier input of the PSK demodulator.

 Connect the channel 1 of CRO across output of demodulator.
 Observe the demodulated output on CRO (it will be same as data input applied in the
modulated in the modulator section).

PRECAUTIONS:-
1. Check the cont unity of the connecting probes.
2. Handle the CRO properly.
PSK MODULATION:
PSK DEMODULATION:
QPSK MODULATOR:
QPSK DEMODULATOR:

RESULT:
PSK AND QPSK are successfully generated.
 EXPERIMENT NO:8
GENERATION OF PN SEQUENCES AND DIRECT SEQUENCE SPREAD
SPECTRUM

AIM:
To Generate PN SEQUENCES AND DIRECT SEQUENCE SPREAD SPECTRUM.

PN Sequence Generation

Theory:

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

In the figure, the output xk is binary (0 or 1), and so are the constants hj, j=0,1,…n, and 
denotes the XOR operation. This means that xk is given by:

xk = h1 xk-1……

hnxk-n Since xkxk=0, it follows from the above that:

xk  h1 xk-1……hnxk-n =0
or

h(D)x(D) =
0

where h(D) = 1  h1D…… hnDn and D denotes a unit delay (xkDj=xk-j for any j). Note
that the “1” in the polynomial does not correspond to a tap.

It is not difficult to see that the output xk will be periodic. However, the dependence of the
length of the period (which we would like to be as large as possible) on the constants hj,
j=1,2,…n is not obvious. We can see that the "state" (xk-1 …. xk-n) can assume at most 2 n
values. We note the following:
 If the state of the shift register is all zero at any time, it remains so for all time. We need to ensure that
this never happens (we start with a non-zero value).
 If the state ever remains unchanged from one clock cycle to the next, it remains the same forever.
 The sequence must be periodic (since there are at most 2n -1 states).
 Since all the all zero state is not allowed, the period of the output sequence can be at most 2 n-1. A feed-
back shift register that generates a sequence of this period is said to be of maximal length.

How do we design the feed-back shift register (i.e., hj) so that it is maximal length,
keeping the hardware (XORs) minimum? The answer to this question involves the
divisibility of 1 Dm by h(D) for m<2n-1, and need not concern us here. The following table
lists the tap-points (points where hj is 1) for registers of various sizes. The first column lists
the order n of the register, and the second column lists the tap points or h(D) in octal
notation. For example, for a register of size 12, the table lists 10123 in octal or 10000010
n=2 7 14 42103 26 400000107
3 13 15 100003 27 1000000047
4 23 16 210013 28 2000000011
5 45 17 400011 29 4000000005
6 103 18 1000201 30 10040000007
7 211 19 2000047 31 20000000011
8 435 20 4000011 32 40020000007
9 1021 21 10000005 33 -
10 2011 22 20000003 34 -
11 4005 23 40000041 35 -
12 10123 24 100000207 36 -
13 20033 25 200000011

Minimal weight polynomials of order 2 to 32 are listed in the above table. Each entry in
the table is an octal number for each n in the first column, which when converted to binary specifies
the coefficients of h(D). The most significant (leftmost) bit h n is 1, and so is the least-significant
(right-most) bit h0. . the tap points for larger length shift registers in various sites on the internet
(use this hyperlink for example)

If you look at any n-length segment of the output, you will find all possible sequences with
the exception of the all zero sequence. However, when you look at a smaller segment, you will
come across all possible sequences. Intuitively therefore, when the register length is large, the
sequence is approximately "white". Consider for example the case when n=2 and h1=h2=1 implying
that both points are tapped so that xk=xk-1xk-2. Starting with xk-1=0 and xk-2=1 gives the following
sequence of shift register contents: (0,1),(1,0),(1,1),(0,1),… Notice that periodicity is three since 22-
1=3 and this happens to be the tapping to get a maximal length sequence. Note that the shift-register
contents are shifted versions of each other and it makes no difference which register output is
considered the output.

The output of the PN sequence generator is purely deterministic – given the state of the
generator, the output is uniquely determined for all time. With the zero level mapped to a “-1” to
make it an antipodal sequences, the autocorrelation of the maximal-length PN sequence is periodic,
and its value in one period is
-1/M except at one location where it is 1 (M is 2 n-1). This sequence can be filtered to generate
bandlimited Gaussian-like noise.
 SPREAD SPECTRUM – DSSS MODULATION & DEMODULATION

INTRODUCTION:

Recall that when a sinusoidal carrier is DSBSC modulated by a message, the two signals are
multiplied together. Recall also that the resulting DSBSC signal consists of two sets of sidebands
but no carrier.

When the DSBSC signal is demodulated using product detection, both sidebands re multiplied
with a local carrier that must be synchronized to the transmitter’s carrier that is , it has the same
frequency and phase. Doing so produces two messages that are in phase with each other and so
add to form a single bigger message.

Direct sequence spread spectrum is a variation of DSBSC modulation scheme with a pulse train
for the carrier instead of a simple sinewave. This may sound radical until you remember that pulse
trans are actually made up of a theoretically infinite number of sinewaves. That being the cse,
spread spectrum is really the DSBSC modulation of a theoretically infinite number of sinusoidal
carrier signals. The result is a theoretically infinite number of pairs of tiny sidebands about a
suppressed carrier.

In practice, not all of these sidebands have any energy of significance. However, the fact that the
message information is distributed across so many of them makes spread spectrum signalsdifficult
to deliberately interfere with “jam”. To do so , you have to upset a significant number of the
sidebands which is difficult considering their number .

Spread spectrum signals are demodulated in the same way as DSBSC signals using a product
detector. Importantly, the product detectors local carrier signal must contain all the sinewaves
that make up transmitter’s pulse train at the same frequency and phase. If this is not done, the tiny
demodulated signals will be at the wrong frequency and phase and so they wont add up to
reproduce the original message. Instead, they’ll produce a garbage signal that looks like noise.

The only way to obtain the right number of sinewaves at the right frequency and phase at the
receiver is to use a pulse train with an identical sequence to that used by the transmitter.
Moreover, it must be synchronized. This issue gives spread spectrum another of its advantages
over other modulation schemes. The transmitted signal is effectively encrypted.

Of course, with trial and error its possible for an unauthorized person to guess the correct PN
sequence to use for their receiver. However, this can be made difficult by making the sequence
longer before it repeats itself. Longer sequences can produce more combinations of unique codes
which would take longer to guess using a trial and error approach. To illustrate this point , an 8 bit
code has 256 combinations while a 20 bit code has 1048575 combinations. A 256 bit code has
1.1579*1077 combinations.
Increasing the sequence’s chip-length has another advantage. To explain, the total energy in a
spread spectrum signal is distributed between all of the tiny DSBSC that make it up. A
mathematical technique called Fourier Analysis shows that the greater the number of chips in a
sequence before repeating, the greater the number of sinewaves of significance needed to make
it.

That being the case, using more chips in the transmitter’s PN sequence products more DSBSC
signals and so the signal’s total energy is distributed more thinly between them. This in turn means
that the individual signals are many and extremely small. Infact , if the PN sequence is long
enough, all of these DSBSC signals are smaller than the background electrical noise that’s always
present in free space. This fact gives spread spectrum yet another important advantage. The signal
is difficult to detect.

Spread spectrum finds use in several digital applications including: CDMA mobile phone
technology, cordless phones. The global positioning system and two of the 805.11 Wi-fi standards

Generate a DSSS signal by implementing its mathematical model. You’ll then use a product
detector (with a stolen carrier) to reproduce the message. Once done, you’ll examine the
importance of using the correct PN sequence for the local carrier and difficulty of jamming DSSS
signals.

Equipment

1. Emona Telecoms Trainer 101 (plus power pack

2. Dual channel 20 MHz oscilloscope
3. Two Emona Telecoms Trainer 101 oscilloscope leads
4. Assorted Emona Telecoms Trainer 101 patch leads.

Procedure:
Part A

As DSSS is basically just DSBSC with a pulse train for the carrier instead of a simple sinusoid, it
can be generated by implementing the mathematical model for DSBSC.

1. Gather a set of the equipment listed on the previous page.

2. Set up the scope per the instructions in experiment 1.
a. The Trigger source control is set to the CHI position
b. The Moe control is set to the CHI position.
3. Set the scope’s Trigger source coupling control the t he HF REJ position.
4. Locate the sequence Generator module and set its dip-switches to 00
 a. To do this, push both switches up.
5. Connect the set-up shown in Figure 1 below.
a. Note : Insert the black plugs of the oscilloscope leads into a ground (GND)
socket.

The set up iN FIG 1 can be represented by the block diagram in Fig 2 below. It multiplies the 2
kHz sinewave message with a PN sequence modeled by the sequence Generator’s 32 bit pulse
train output.
 6. Adjust the scope’s Time base control to view two or more cycles of the Master signals
modules 2 kHz sine output.
7. Set the scope’s Mode control to the DUAL position to view the DSSS signal out of the
Multiplier Module as well as the message signal.
8. Adjust the scope’s Vertical Attenuations controls to the appropriate settings for the
signals.
9. Draw the two waveforms to scale in the space provided on the next page leaving room to
draw a third waveform.
a. Tip: Draw the message signal in the upper third of the graph and DSSS signal in
the middle third.
10. Use the scope’s channel 1 Vertical position control to overlay the message with the DSSS
signal’s envelope’s and compare them.

Part B – Generating a DSSS signal using speech.

So far , this experiment has generated a DSSS signal using a sinewave for the message. The next
part of the experiment lets you see what a DSSS signal looks like when modulated by speech.

11. Disconnect the plugs to the Master signals module’s 2kHz sine output.
12. Connect them to the speech module’s output as shown in fig 3 below.
a. Remember Dotted lines sow leads already in place.

13. Set the scope’s time base control to the 2ms/div position.
14. Talk, sing or hum while watching the scope’s display.
Part C
Using the product detector to recover the message.

15. Return the scope’s Time base control to its original position.
16. Locate the Tunable low pass filter module and set its Gain control to about the middle of
its travel.
17. Turn the tunable low pass filter modules cut-off frequency adjust control fully anti-
clockwise.
18. Disconnect the plugs to the speech module’s output and modify the set-up as shown in
Fig 4 below

19. Slowly turn the Tunable low-pass Filter module’s cut – off Frequency control clockwise
while watching the scope’s display and stop when it’s at about half its travel.
 20. Draw the demodulated DSSS signal to scale in the space that you left on the graph paper.

Recall that the message can only be recovered by the product detector if an identical PNsequence
to the DSSS modulator’s carrier is used. The next part of the experiment demonstrates. This.

21. Modify the setup as shown in Fig 7 below to make the demodulator’s local carrier a
different PN sequence to the transmitter’s carrier.

22. Compare the message with the product detector’s new output.

Part D
DSSS and deliberate interference ( Jamming)

Interferences occurs when an unwanted electrical signal gets added to the transmitted signal and
changes it enough to change the recovered message. Electrical noise is a significant source of
unintentional interference.

However, sometimes noise is deliberately added to the transmitted signal for the purpose of
interfering or Jamming it. The next part of the experiment models deliberate interference to show
how spread spectrum signals are highly resistant to it.

23. Move the patch lead from the sequence Generator’s Y output back to its X output.
a. Note : The product detector should now be recovering the message again.
24. Locate the VCO module and set its Range control the HI position.
25. Set the VCO modules Frequency Adjust control to about the middle of its travel.
26. Locate the Adder Module and turn its g control fully anti clockwise.
 27. Set the Adder module’s G control to about the middle of its travel.
28. Modify the setup as shown in Fig 8 below.

This modification forces the VCO module’s output to sweep continuously through a wide range
of Frequencies.

29. compare the two signals. Notice that the DSSS signal with interference is very distorted
but the recovered message is only mildly affected.
An even more sophisticated approach to jamming involves using jamming signals at once to increase the
chances of upsetting the transmitted signal. The next part of the experiment let’s you see how spread
spectrum handles this.

30. Modify the setup shown in Fig 11 below. This modification uses the Noise generatormodule
to model a jamming signal that consists of thousands of frequencies.
31. Compare the two signal. Notice that the DSSS signal with interference is distorted but the
recovered message is only mildly affected.
32. Increase the strength of the broadband jamming signal by connecting the Adder.
33. Compare the DSSS signal and the recovered message.
34. Increase the strength of the broadband jamming signal even more by connecting theAdder
module’s B input to the Noise Generator module’s 0dB output.
35. Compare the two signals. Notice how distorted DSSS signal is but how little therecovered
message is affected.
36. Modify the Setup as shown.

Result:

Thus the Experiment was performed successfully.

 EXPERIMENT NO:9
SIMULATION OF ANALOG MODULATION SCHEMES IN MATLAB

AIM:
To simulate ANALOG MODULATION SCHEMES IN MATLAB

APPARATUS REQUIRED:
1. Amplitude Modulation and Demodulation Trainer
2. Function Generator
3. Oscilloscope
4. Connecting Wires

THEORY:
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 modulation signal.

Amplitude modulation is defined as the process in which is the amplitude of the carrier wave is varied about
a means values linearly with the base band signal. The envelope of the modulating wave has the same shape
as the base band signal provided the following two requirements are satisfied

(1) The carrier frequency fc must be much greater then 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

PROCEDURE:
1 Switch on the trainer and check the O/P of carrier generator on oscilloscope.
2. Connect 1 KHz with 2 Volts A.F signal at AF I/P to the modulator circuit.
3. Connect the carrier signal at carrier I/P of modulator circuit.
4. Observe the modulator output signal at AM O/P by making necessary changes in A.F. Signal.
5. Vary the modulating frequency and amplitude and observe the effects on the modulated waveform.
6. The depth of modulation can be varied using the variable knob (potentiometer) provided at A.F. input.
7. The percentage of modulation or modulation factor can be calculated using the following formulas.
Fig:- (1)Circuit Diagram For modulation
t = [0:2*Fs+1]'/Fs;
Fc = 10; % Carrier frequency
x = sin(2*pi*2*t); % message signal
Ac=1;
% compute spectra of am
xam=ammod(x,Fc,Fs,0,Ac);
zam = fft(xam);
zam = abs(zam(1:length(zam)/2+1));
frqam = [0:length(zam)-1]*Fs/length(zam)/2;
% compute spectra of dsbsc
ydouble = ammod(x,Fc,Fs, 3.14,0);
zdouble = fft(ydouble);
zdouble = abs(zdouble(1:length(zdouble)/2+1));
frqdouble = [0:length(zdouble)-1]*Fs/length(zdouble)/2;
% compute spectra of ssb
ysingle = ssbmod(x,Fc,Fs,0,'upper');
zsingle = fft(ysingle);

zsingle = abs(zsingle(1:length(zsingle)/2+1));
frqsingle = [0:length(zsingle)-1]*Fs/length(zsingle)/2;
% Plot spectrums of am dsbsc and ssb
figure;
subplot(3,1,1); plot(frqam,zam); title('Spectrum of am signal'); subplot(3,1,2); plot(frqdouble,zdouble);
title('Spectrum of double-sideband signal');subplot(3,1,3); plot(frqsingle,zsingle); title('Spectrum of
single-sideband
RESULT:
The simulation of ANALOG MODULATION SCHEMES IN MATLAB ARE GENERATED
 EXPERIMENT NO:10
SIMULATION OF ANALOG DEMODULATION SCHEMES IN MATLAB
AIM:
To simulate analog demodulation schemes in matlab

THEORY :

Amplitude modulation (AM) is defined as a process in which the amplitude of the carrier wave c(t)
is varied about a mean value, linearly with the base band signal m(t).

An AM wave may thus be described, in its most general form, as a function of time as follows.

S(t)=A [1+Kam(t)] Cos (2πfct)

The amplitude of Kam(t) is always less than unity, that is |Kam(t)| <1 for all t. It ensures that the
function 1+Kam(t) is always positive. When the amplitude sensitivity Ka of the modulator is large
enough to make |Kam(t)| >1 for any t, the carrier wave becomes over modulated, resulting in carrier
phase reversals. whenever the factor 1+Kam(t) crosses zero.

The absolute maximum value of Kam(t) multiplied by 100 is referred to as the percentage
modulation.

PROCEDURE:

1. Connect the AC Adapter to the mains and the other side to the Experimental Trainer.

Switch „ON‟ the power.

2. Observe the carrier and modulating waveforms and note their frequencies.

(Carrier frequency is around 100 KHz and amplitude is variable from 0 -8Vp-p,

modulating signal is 1KHz).

3. Connect the carrier and modulating signals to the modulator circuit.

4. Observe the amplitude modulated wave.

5. Connect Carrier I/P to ground and apply a 2V peak to peak AF Signal input to (modulating I/P)
and adjust P1 in anti-clock wise position to get minimum A.C output.

6. Connect modulating I/P to ground and apply a 3V peak to peak carrier signal to carrier I/P and
adjust P2 in clock wise direction to get minimum A.C ouyput..
7. Connect modulating input &carrier input to ground and adjust P3 for zero D.C output.

8. Make modulating i/p 2 Vpp and carrier i/p 3 Vpp peak to peak and adjust
potentiometer P4 formaximum output.

9. Calculate maximum and minimum points on the modulated envelope on a CRO

and calculatethe depth of modulation.

10. Observe that by varying the modulating voltage, the depth of modulation varies.

EXPECTED WAVE FORMS:

Matlab code:
fs=8000;
f
m
=
2
0
;
f
c
=
5
0
0
;
A
m
=
1
;
A
c
=
1
;
t=[0:.1*fs]/fs;
m=Am*cos(
2*pi*fm*t);
c=Ac*cos(2
*pi*fc*t);
ka=0.5;
u=ka*Am;
s1=Ac*(1+u*cos(2*pi*fm*t)).*cos(2
*pi*fc*t);subplot(4,3,1:3);
plot(t,m);
title('Modulating or Message
signal(fm=20Hz)');subplot(4,3,4:6);
plot(t,c);
title('Carrier
signal(fc=500Hz)');
subplot(4,3,7);
plot(t,s1);
title( 'Under Modulated
signal(ka.Am=0.5)');Am=2;
ka=0.5;
u=ka*Am;
s2=Ac*(1+u*cos(2*pi*fm*t)).*cos(2
*pi*fc*t);subplot(4,3,8);
plot(t,s2);
title( 'Exact Modulated
signal(ka.Am=1)');Am=5;
ka=0.5;
u=ka*Am;
s3=Ac*(1+u*cos(2*pi*fm*t)).*cos(2
*pi*fc*t);subplot(4,3,9);
plot(t,s3);
title('Over Modulated
signal(ka.Am=2.5)');r1= s1.*c;
[b a] = butter(1,0.01);
mr1=
filter(b
,a,r1);
subplo
t(4,3,1
0);
p
l
o
t
(
t
,
m
r
1
)
;
r
2
=
s
2
.
*
c
;
[b a] = butter(1,0.01);
mr2=
filter(b
,a,r2);
subplo
t(4,3,1
1);
plot(t,mr2);

r3= s3.*c;