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EXPERIMENT NO 1

OBJECT: - To study sample and hold circuit (made using op amp). APPARATUS REQUIRED Op amp -1(IC 741),capacitor -1(16pf)Resistor-2(10 K) Function generator-1(for input)CRO1(for output)E-MOSFET -1(N- channel)Connecting wires THEORY Signal can be defined as a single value function dependent on in dependent variable. A signal is any time-varying or spatial-varying quantity. It can be broadly classified as: A) Continuous time signal which is defined continuously in its time domain and B) Discrete time signal which is defined only at certain time instants. Continuous time signals can be converted into discrete time signal by sampling and original signal are obtained from these samples only. In signal processing, sampling is the reduction of a continuous signal to a discrete signal. A common example is the conversion of a sound wave (a continuous-time signal) to a sequence of samples (a discrete-time signal).A sample refers to a value or set of values at a point in time and/or space. A sampler is a subsystem or operation that extracts samples from a continuous signal. A theoretical ideal sampler produces samples equivalent to the instantaneous value of the continuous signal at the desired points. Natural sampling In the analogue-to-digital conversion process, an analogue waveform is sampled to form a series of pulses whose amplitude is that of the sampled waveform at 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. Flat Top Sampling In analog to digital conversion process, when an analog waveform is sampled, after that the continuous analogue waveform is converted into a series of pulses whose amplitude is equal to the amplitude of the analog signal at the start of the sampling process. Since the sampled pulses have uniform amplitude, the process is called flat top sampling. Alternatively, in a process called natural sampling, the amplitude of the sampled pulse is allowed to vary with the amplitude of the analogue waveform as it changes during the sampling period. Sample and hold circuit Sample and hold circuit is used to sample the input signal applied and holds on to its last value until the input is sampled again. They are mainly used in digital interfacing and communication such as analog to digital and pulse modulation systems.

In the following circuit, the E-MOSFET works as a switch which is controlled by the sample and hold control voltage ( Vs ) and the capacitor (C) serves as a storage element. To obtain close approximation of the input waveform, the frequency of sample and hold control voltage must be significantly higher than that of input. WORKING The analog signal Vin to be sampled is applied and sample and hold control voltage is applied to the gate of E-MOSFET. During the positive portion of Vs, the E-MOSFET conducts and acts as a closed switch. It allows the input voltage to charge the capacitor C. On the other hand, when Vs is zero, and the E-MOSFET is off and acts as an open switch. The only path for discharging of C is through the op-amp. The time period Ts of sample and hold control voltage Vs during which the voltage across the capacitor is equal to the input voltage is called as the sample period. The period T of Vs during which the voltage across the capacitor is constant is called as the hold period. NOTE: In critical applications a high speed op-amp is helpful. If possible, then a low leakage capacitor must be used.
R 3 1k 0 KyA e= 3 4
7 1 5

V C C 1V 5 V C C Q 1 2 65 N69 7 6 V 1 1 0Vm 2 r s 6 H 0 z 0 0 XG F1 C 1 1p 6F R 1 1k 0
2 4

5% 0

U 1
6

71 4

2 R 2 1k 0 0

V E E

VE E -5 1V

XC S1
G T A B C D

Circuit diagram

Waveform after passing capacitor

Waveform after passing op amp

Result : we have studied and performed the sample and hold circuit .
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EXPERIMENT NO. - 2
OBJECT- Study of PAM generation and detector and observe characteristics of both single and dual pulse amplitude modulation.

COMPONENT REQUIRED- 9V D.C. at 100 mA, IC regulated power supply internally connected, Variable frequency sampling pulse generator,Sine wave audio frequency modulating signal generator,PAM modulator circuit based on operational amplifier,PAM demodulator circuit based on a point connected diode and operational amplifier,-The unit is operative on 230V 10% at 50Hz A.C. Mains, THEORYThis process by which an information of lower frequency is made a distinguishable part of a higher frequency is known as MODULATION. The low frequency information as called the MOODULATING SIGNAL whereas the high frequency is called the carrier wave. Very often the modulating signal is a complex waveform consisting of many frequencies with varying amplitude and phases. The carrier wave is, on the other hand. A signal frequency wave of constant amplitude. The reverse process by which information is separated out from a modulated wave is known as detection or DEMODULATION In pulse amplitude modulation, the amplitude of individual pulses in the pulse train is varied from its default value in accordance with the instantaneous amplitude of the modulating signal at sampling intervals. The width and position of the pulses is kept constant. The PAM transmitter design is very simple since the very act of sampling the modulating signal at regular intervals produces pulse amplitude modulation. Main advantages of PAM are simple transmitter and receiver designs. PAM is used to carry information as well as to generate other pulse modulations.

Fig. (1)- Pulse amplitude modulation Audio frequency signal may travel long distance through overhead wires or underground cables. It has been found that the message carrying capacity of the transmission lines
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increases if each individual message signal is translated from the audio frequency to a high frequency or what is commonly known as a carrier frequency. By suitably selecting the carrier frequency the resulting signal may also be made to transmit itself through space without the use of transmission lines. TYPES OF MODULATION:Modulation can be basically of two types: (1)Continuous wave(a)amplitude (2) pulse(a) Pulse time- pulse time and pulse duration (b) angle- frequency and phase pulse (b) pulse code amplitude

CONTINUOUS WAVE MODULATIOMN Where the carrier wave is a pure sinusoidal waveform. There are various sub types under this heading viz. Amplitude Modulation, Frequncy Modulation. PULSE MODULATION Where the carrier wave is in the form of pulses with constant pulse repetition frequency. This can be further divided into the following- Pulse Amplitude Modulation (PAM). Pulse code Modulation (PCM) ,Pulse Position Modulation(PPM) and Pulse Duration Modulation (PDM) In Pulse Modulation the single frequency continuous wave carrier is replaced by a series of pulses having a fixes duration and repetition frequency. Parameters of the pulses are now varied to give various types of Pulse Modulation. PULSE AMPLITUDE MODULATION (PAM) This training is concerned with this type of pulse modulation; therefore lets look at it in further detail. Here the amplitude of the pulse is varied according to the modulating signal. This corresponds to amplitude modulation of the continuous wave. The PAM waveform can be two types. DUAL POLARITY PAM If we sample the modulating signal as shown in fig.2(a) by dual polarity sampling pulses as shown in fig.2(b) then resulting PAM waveform is also dual polarity types as shown in fig.2(c) SINGLE POLARITY PAM If we sample the modulating single whose DC level has been raised to a positive value say +V1 from 0 volts, as shown in fig.3(a) and sample it with singal polarity pulses as shown in fig.3(b) Then the resulting waveform as shown in fig.3(c) is also a single polarity type. In this training board we produce single polarity PAM.

Fig.4(a) shows the frequency spectrum of the sampling pulses and fig4(b). showsPAM waveform. The frequency spectrum of PAM signal shows that the modulating frequency fm and various sidebands fcfm maintain their individuality, that is, fm does not spread into the lower sideband region around fc or fc+fm does not spread into the lower sideband region around 2fc if: fmfc-fm or fc2fm.

DEMODULATION OF PAM From the discussion we can conclude that the modulating signal can be extracted from the PAM simply by a diode and a low pass filter with its cut of frequency at Fm.

Fig.5 shown the effect of the frequency of the sampling pulses on the recovered modulating signal after demodulation. The recorved signal of fig.5(a) due to high frequency sampling pulses used has less error than that of fig,5(b) which uses low frequency sampling pulses. BLOCK DIAGRAM OF PAM MODUATOR AND DEMODULATOR :-

WORKING:The block diagram of pulse amplitude modulator based on operational amplifier (ic-741) is show on in fig. we know that the train of pulses corresponding to the samples of each signal are modulated in amplitude in accordance with the signal itself. Accordingly, the scheme of sampling is called pulse amplitude modulation. In this, the modulating signal is feds to pin-3 of op-amp. through sine wave generator. The sampling pulses are feds to pin-6 of op-amp. 10k resistor.+9v supply is feds to pin-7 through 100f capacitor. The pin-2 of 741 connected to two resistor of 2k2 , 4k7 & four diodes (1n4148). The pin-2 provides trigger pulse to 741 through diode and resistor. The o/p of 741 depends on the amplitude of the external trigger pulse applied to this pin. Thus, we obtained PAM waveform as shown in fig-2. When we feds the Pam waveform to the i/p of demodulator, we obtained modulating signal. The block diagram of Pam demodulator is shown in fig. The Pam demodulator is low pass filter which passes only low frequency modulating signal. Fig. 5 shows the effects of the frequency of the frequency of sampling pulses on the recovered modulating signal after demodulation. The fig. 5(a) due to high frequency sampling pulses used has less error then of fig. 5(b) which uses low frequency sampling pulses.

RESULT:Hence, we study and perform pulse amplitude modulation and demodulation

EXPERIMENT No:-3
OBJECT:-Study of P.W.M. (Pulse Width Modulation) using timer I.C. 555 and Demodulation using Op-Amp I.C. 741. COMPONENT REQUIRED: Pulse Train(carrier) 50 Hz A.C. sinusoidal modulating signal. Pulse Width Modulator based on 555 Timer. Pulse Width Demodulator based on Operational Amplifier(741). ON/OFF Switch, Fuse. Connecting wires. Some other electronic component (Capacitors and Resistors). Cathode Ray Oscilloscope (C.R.O.) 20 MHz.

THEORY:MODULATION:Baseband signals produced by various information sources are not always suitable for direct transmission over a given channel. These signals are usually further modified to facilitate transmission. This conversion process is known as MODULATION. In this process, the baseband signal is used to modify some parameter of a high-frequency carrier signal. A carrier is a sinusoid of high frequency. BASICALLY MODULATION IS OF TWO TYPES:1. CONTINOUS WAVE MODULATION :- It is of three types Amplitude Modulation(AM) Frequency Modulation(FM) Phase Modulation(PM)

2. DISCRET MODULATION :- In this type of modulation pulse train is used as a carrier signal. Again it is of Four types: Pulse Amplitude Modulation(PAM)
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Pulse position Modulation(PPM) Pulse Width Modulation(PWM) Pulse Code Modulation(PCM)

PULSE WIDTH MODULATION:Pulse Width Modulation (PWM), also known as Pulse Duration Modulation (PDM), is a digital modulation technique where by the width of the pulse carrier is made to vary in accordance with the modulation voltage. The PWM waveform has fixed amplitude and starting time of each pulse, but the width of each pulse is proportional to the amplitude of the modulating signal at that instant. Thus the pulses corresponding to positive peaks of a modulating signal shall be wider as compare to the pulses corresponding to negative going point. This variation is shown in the fig 1-

Fig-1, Wave form of pulse width modulation WORKING:PWM can be obtained by using a monostable multivibrator circuit. The starting time of the pulses is controlled by the trigger pulses i.e. pulse carrier waveform and the modulating signal controls the width of the pulses. We are using timer I.C. 555.

Internal working of the timer I.C. 555 :-

Fig 2 shows the functional block diagram of the timer I.C. 555 and external components to form a monostable multivibrator. Resistors R1, R2 and R3 form a voltagereference potential divider. There are two voltage-comparator OP-amp, one R-S Flip-Flop, a low power complementary output stage and a transistor T1.

I.C.

FIG-2, Timer 555

2/3 Vcc appears at the junction of the upper two resistors of the potential divider i.e. R1, R2. This voltage is fed to one input of the upper voltage-comparator Op-amp. The output of two comparator controls the R-S Flip-Flop which in turn controls the states of the complementary output stage and the slave transistor. The state of the flip-flop can also influenced by signals applied to the Pin 5 i.e. RESET terminal. When the circuit is in quiescent state the Pin 2 i.e. TRIGGER terminal is held high via resistor R4. Under this condition T1 is driven to saturation and forms a short circuit across external timing capacitor Ct and the Pin 3 output terminal of the I.C. is driven to LOW state. The monostable action is initiated as soon as a negative going trigger pulse is applied at Pin 2. When this pulse falls below the 1/3 Vcc reference value of built in potential divider the output of the lower stage comparator op-amp, changes state and causes the R-S Flip-Flop to switch over. As the FlipFlop switches over it cuts off T1 and drives the Pin 3 output of the IC 555 to the HIGH state. As T1 cuts off it removes the short circuit across capacitor Ct, so Ct starts to charge up towards the apply voltage Vcc. As the voltage across Ct reaches 2/3 Vcc the upper voltage comparator changes state and causes the R-S Flip-Flop to its original state i.e. turning on T1, rapidly discharging Ct and putting the output at LOW state. The delay time of the circuit is given by t = 1.1RtCt where t is the time in seconds during which the output at pin 3 is high. The timing period of the circuit is independent of the supply voltage that can however be varied by applying a voltage at Pin 5 is CONTROL VOLTAGE terminal of the IC 555.

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If we apply a continuously varying voltage at Pin 5 then the width of the output pulses vary in accordance with this control voltage, hence a pulse width modulated output results and the circuit functions as a Pulse Width Modulator. The trigger or the Carrier Pulses are generated by the astable multivibrator built around another timer IC 555. the modulating signal to be fed to Pin 5 of the PW Modulator is obtained from the secondary of the stepdown transformer and is 50 Hz sinusoidal AC waveform.

Fig-3, CIRCUIT DIAGRAM FOR PULSE WIDTH MODULATOR DEMODULATION:Demodulation is the process of extracting the original information from a modulated carrier wave. A demodulator is an electronic circuit used to recover the information content from the modulated carrier wave. The process of integration is employed in to demodulate a P.W. signal for recover the modulating signal. In this circuit we are using an op-amp integrator built around IC 741. In this current is summed over a period of time and the resultant voltage generated is the integral of that current as a function of time. In this process we are measuring the area under each pulse, and for this we are using integration method. And to integrate the wave we use I.C. 741 as an integrator. The 100K resistor across 0.1uF capacitor is added to provide DC stability. Fig- 4, shows the demodulated waveform

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Fig -5

P.W.M. Demodulator using Op.Amp.

RESULT:Hence we study and perform the Pulse Width Modulation ( P.W.M.) and demodulation and obtained the waveforms.

EXPERIMENT NO 4:

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Object: Conduct an experiment to generate pulse position modulated signal and also design a circuit to demodulate the obtained pulse position modulated signal and plot the relevant waveforms. Components Required: Connecting wires, CRO, CRO probes, function generator etc. Theory: In Pulse Position Modulation, both the pulse amplitude and pulse duration are held constant but the position of the pulse is varied in proportional to the sampled values of the message signal. Pulse modulation is of four distinct types as shown in figure:-

Modulation

Continuous wave

Puls e Pulse Pulse amplitude time Phase e Pulse code

Amplitude

Angle

Frequency

Pulse position

Pulse width

Pulse time modulation is a class of signaling techniques that encodes the sample values of an analog signal on to the time axis of a digital signal and it is analogous to angle modulation techniques. The two main types of PTM (Pulse Time Modulation) are PWM and PPM. In PPM the analog sample value determines the position of a narrow pulse relative to the clocking time. In PPM rise time of pulse decides the channel bandwidth. Pulse position modulation has the advantage over pulse amplitude modulation (PAM) and pulse width modulation (PWM) in that it has a higher noise immunity since all the receiver needs to do is detect the presence of a pulse at the correct time; the duration and amplitude of the pulse are not important. 555 Timer in monostable operation:Another popular application for the 555 timer is the monostable mode (one shot) which requires only two external components, Ra and C in figure below. Time period is determined by the following formula:T=1.1 X Ra C.

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Astable Multivibrator by two transistors:An astable multivibrator is also known as a free-running multivibrator. It is called freerunning because it alternates between two different output voltage levels during the time it is on. The output remains at each voltage level for a definite period of time. If we looked at this output on an oscilloscope, we would see continuous square or rectangular waveforms. The astable multivibrator has two outputs, but NO inputs. The figure shows astable multivibrator. The astable multivibrator is said to oscillate. To understand why the astable multivibrator oscillates, assume that transistor Q1 saturates and transistor Q2 cuts off when the circuit is energized. This situation is shown in figure. We assume Q1 saturates and Q2 is in cutoff because the circuit is symmetrical; that is, R1 = R4, R2 = R3, C1 = C2, and Q1 = Q2. It is impossible to tell which transistor will actually conduct when the circuit is energized. For this reason, either of the transistors may be assumed to conduct for circuit analysis purposes.

Astable multivibrator (Q1 saturated)

Astable multivibrator (Q2 saturated)

Essentially, all the current in the circuit flows through Q1; Q1 offers almost no resistance to current flow. Notice that capacitor C1 is charging. Since Q1 offers almost no resistance in its

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saturated state, the rate of charge of C1 depends only on the time constant of R2 and C1 (recall that TC = RC). Notice that the right-hand side of capacitor C1 is connected to the base of transistor Q2, which is now at cutoff. The right-hand side of capacitor C1 is becoming increasingly negative. If the base of Q2 becomes sufficiently negative, Q2 will conduct. After a certain period of time, the base of Q2 will become sufficiently negative to cause Q2 to change states from cutoff to conduction. The time necessary for Q2 to become saturated is determined by the time constant R2C1. The next state is shown in figure. The negative voltage accumulated on the right side on capacitor C1 has caused Q2 to conduct. Now the following sequence of events takes place almost instantaneously. Q2 starts conducting and quickly saturates, and the voltage at output 2 changes from approximately -VCC to approximately 0 volts. This change in voltage is coupled through C2 to the base of Q1, forcing Q1 to cutoff. Now Q1 is in cutoff and Q2 is in saturation. This is the circuit situation shown in figure . Notice that figure is the mirror image of figure In figure the left side of capacitor C2 becomes more negative at a rate determined by the time constant R3C2. As the left side of C2 becomes more negative, the base of Q1 also becomes more negative. When the base of Q1 becomes negative enough to allow Q1 to conduct, Q1 will again go into saturation. The resulting change in voltage at output 1 will cause Q2 to return to the cutoff state. Look at the output waveform from transistor Q2, as shown in figure. The output voltage (from either output of the multivibrator) alternates from approximately 0 volts to approximately -VCC, remaining in each state for a definite period of time. The time may range from a microsecond to as much as a second or two. In some applications, the time period of higher voltage (-VCC) and the time period of lower voltage (0 volts) will be equal. Other applications require differing higher- and lower-voltage times. For example, timing and gating circuits often have different pulse widths as shown in figure. Working:Modulation: A 555 timer IC is used in monostable mode for the pulse position modulation which is triggered by pulse train (carrier) and modulating signal is provided on control voltage i. e. pin#5 of 555 timer IC. So when this monostable multivibrator triggers, the output generates. So actually the position is decided by pulse train (triggering). Now the output of the monostable multivibrator is provided to differentiator circuit. Here RC network is working as a differentiator circuit and the diode OA-79 is used to clip off the positive half cycle of the wave form. So finally we get pulse position modulated waveform at TP3.

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Circuit Diagram:- Pulse Position Modulation & Demodulation

Demodulation: Two transistor astable multivibrator is used for the demodulation of pulse position modulated wave. Pulse position modulated signal is provided to the base of first transistor after filtering by capacitor and on the other hand pulse train is provided at the base of second transistor. This astable multivibrator works by charging and discharging of capacitors which are situated on the collector of the transistors. First this PPM wave is converted into pulse width modulated wave and then the original signal. The output of the astable multivibrator is spike and this spike it sends to the integrator circuit. The integrator works on the principal of by calculating the area under the curve and then draws the wave form. Then finally we get our original message signal at the output of integrator. Result: Thus we have studied the pulse position modulation and demodulation techniques and traced the waveforms also we have learnt how pulse train takes part in the formation of pulse position modulation and its demodulation.

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EXPERIMENT NO. 5
Object : TO STUDY OF DELTA MODULATION AND DEMODULATION USING COMPARATOR AND OBSERVE EFFECTS OF SLOPE OVERLOAD. Features 1. 5v D.C. at 100mA IC integrated power supply internally connected. 2. 12 D.C. at 100mA IC regulated power supply internally connected. 3. Quad op-amps IC. 4. Two Up/Down counter IC. 5. Digital to Analog converter (DAC) IC. 6. Quad two input NAND gate IC. 7. Timers IC. 8. Comparator IC. 9. Potentiometer for varying amplitude of modulating signal. 10. Mains ON/OFF switch, fuse and jewel light. Introduction In radio transmission, it is necessary to send audio signal from a broad casting station over great distance to a receiver. This communication of audio signal does not employ any wire and its sometimes called wireless. The audio signal cannot be sent directly over the air for appreciable distance. Even if the audio signal is converted into electrical signal, the latter cannot be sent very far without employing very large amount of power. The energy of a wave is directly proportional to its frequency. At audio frequency, the signal power is quite small and radiation is not practicable. The radiation of electrical energy is practicable only at high frequencies e.g. Above 20 kHz. The high frequency signals cannot be sent thousands of miles even with comparatively small power. Therefore if audio is to be transmitted properly, some means must be devised which permit transmission is to occur at high frequency while it simultaneously allows the carrying of audio signal. This is achieved by imposing electrical audio signal on high frequency carrier. The resultant waves are known as modulated wave or radio waves and the process is called modulation. At radio receiver, the audio signal is extracted from the modulated wave

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called demodulation. The signal is then amplified and reproduced into sound by the loudspeaker. Modulation A high frequency carrier wave is used to carry the audio signal which is done by changing some characteristic of carrier wave in accordance with the signal. Under such condition, the audio signal will be contained in the resultant wave. The process is called modulation and defined as The process of changing some characteristic of a carrier wave in accordance with the intensity of the signal is known as modulation. Modulation means to change. In some characteristic of a carrier wave is change in accordance with the intensity of the signal. The resultant wave is called modulated wave or radio wave and contains the audio signal. There for modulation permits the transmission to occur at high frequency while it simultaneously allows the carrying of the audio signal.

Delta Modulation and Demodulation Delta modulation is a differential Pulse code Modulation, in which the difference signal between two successive samples is encoded into a single bit code. Reason to use Delta Modulation We have observed in PCM that it transmits all the bits which are used to code a sample. Hence, signaling rate and transmission channel bandwidth are quite large in PCM.To overcome this problem, Delta modulation is used.

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Theory Delta modulation is also known as Linear Delta Modulator. In Delta Modulation, we have used the Comparator, Level Shifter, Up and Down Counter, DAC (Digital to Analog Converter). The signal m (t) is the analog input signal. While r (t) is a reconstructed signal which is same as the quantized input signal with 1 bit delay. The signal r (t) tries to follows the input signal m (t) with one bit period delay.

The process is encoding as follows. The comparator compares the input signal m (t) and r (t). If m (t)>r (t) a logic 1 is generated at the output of the comparator otherwise a logic 0 is generated. The value of logic 1 or logic 0 turned as (t) is held for the bit duration by the sample and hold current to generate the delta modulated output. This output So (t) is fed to the 8 bit binary up/down counter to control its count direction. A logic 1at the mode control input increases the count value by one and a logic 0 decrements he count values by one. All the 8 outputs of the counter are given to DAC to reconstruct the original signal. In essence the counter & Decoder form the Delta Modulator in the feedback loop of the comparator. Thus, if the input signal is higher than the reconstructed signal the counter increments at each step so as to enable the DAC output to reach to the input signal values. Similarly, if the input signal m (t) is lower than the reconstructed signal r (t), the

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counter decrements at each step, and the DAC output gets reduced to reach a value to that of m (t). The block diagram of Delta Modulator is shown above. It works on the same way as it was in the feedback loop of the Delta modulator. The received Delta modulated signal So (t) is given to the mode control input of the up/down counter. All the 8 bit outputs are connected to an 8 bit DAC which gives a quantized analog signal (stepped waveform). A low pass filter is used to smooth out the steps. A buffer amplifier provides the necessary drive capability to the output signal. Thus, the digital delta modulated data is demodulated and reconstructed into analog signal. Although, this process of Delta Modulation and Demodulation is a simple and cost effective method of coding, there will be poor approximation at starting buildup.

WAVEFORMS

LJ

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Advantages In Delta Modulation, it can transmit only one bit for one sample, therefore the signaling rate and bandwidth are quite large. Drawbacks of Delta Modulation It has two major drawbacks: 1) Slope overload distortion 2) Granular or idle noise

1) Slope Overload Distortion

This distortion arises because of large dynamic range of the input signal. The rate of rise of input signal x (t) is so high that the staircase signal cannot approximate it, the steep size becomes too small for staircase signal to follow the step segment of x (t). Hence, there is a large error between the staircase approximated signal and the original input signal x (t).This error or noise is known as slope overload distortion. To reduce this error, the step size must be increased when slope of signal x (t) is high.

Granular and slope overloaded distortion in delta modulation

2) Granular or Idle Noise

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Granular or idle noise occurs when the step size is too large compared to small variation in the input signal. This means that for very small variation in the input signal, the staircase signal is changed by large amount because of large step size. The error between the input and approximated signal is called granular noise. The solution of this problem is to make step size small. RESULT We have studied and performed Delta modulation & Demodulation and observe the effects of slope overload.

EXPERIMENT NO. 6
OBJECT - To study and perform the frequency modulation. COMPONENT REQUIREDResistor (R1 = 220K,R2 = 4,7K,R3 = R4 = 10K,R5 = 82 Ohm,R = 150Ohm 1/2W x2 *,VR1 = 22K trimmer) Capacitor (C1 = C2 = 4,7uF 25V electrolytic,C3 = C13 = 4,7nF ceramic,C4 = C14 = 1nF ceramic,C5 = C6 = 470pF ceramic,C7 = 11pF ceramic,C8 = 3-10pF trimmer C9 = C12 = 7-35pF trimmer,C10 = C11 = 10-60pF trimmer,C15 = 4-20pF trimmer,C16 = 22nF ceramic *) Inductor (L1 = 4 turns of silver coated wire at 5,5mm diameter,L2 = 6 turns of silver coated wire at 5,5mm diameter,L3 = 3 turns of silver coated wire at 5,5mm diameter,L4 = printed on PCB,L5 = 5 turns of silver coated wire at 7,5mm diameter) RFC1=RFC2=RFC3= VK200 RFC tsok Transistor (TR1 = TR2 = 2N2219 NPN,TR3 = 2N3553 NPN,TR4 = BC547/BC548 NPN) Diode (D1 MIC = crystal microphone Basic System The basic communications system has: Transmitter: The sub-system that takes the information signal and processes it prior to transmission. The transmitter modulates the information onto a carrier signal, amplifies the signal and broadcasts it over the channel.
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1N4148

diode

*)

Channel:
The medium which transports the modulated signal to the receiver. Air acts as the

channel for broadcasts like radio. May also be a wiring system like cable TV or the Internet. Receiver:
The sub-system that takes in the transmitted signal from the channel and processes it

to retrieve the information signal.


The receiver must be able to discriminate the signal from other signals which may

using the same channel (called tuning), amplify the signal for processing and demodulate (remove the carrier) to retrieve the information. It also then processes the information for reception (for example, broadcast on a loudspeaker).

Modulation The information signal can rarely be transmitted as is, it must be processed. In order to use electromagnetic transmission, it must first be converted from audio into an electric signal. The conversion is accomplished by a transducer. After conversion it is used to modulate a carrier signal. A carrier signal is used for two reasons: To reduce the wavelength for efficient transmission and reception (the optimum antenna size is or of a wavelength). A typical audio frequency of 3000 Hz will have a wavelength of 100 km and would need an effective antenna length of 25 km. By comparison, a typical carrier for FM is 100 MHz, with a wavelength of 3 m, and could use an antenna only 80 cm long. 2. To allow simultaneous use of the same channel, called multiplexing. Each unique signal can be assigned a different carrier frequency (like radio stations) and still share the same channel. The phone company actually invented modulation to allow phone conversations to be transmitted over common lines.
1.

The process of modulation means to systematically use the information signal (what you want to transmit) to vary some parameter of the carrier signal. The carrier signal is usually just a simple, single-frequency sinusoid (varies in time like a sine wave). The basic sine wave goes like V (t) = V o sin (2 t + f ) where the parameters are defined below:

V (t) the voltage of the signal as a function of time.

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Vo the amplitude of the signal (represents the maximum value achieved each cycle) f the frequency of oscillation, the number of cycles per second (also known as Hertz

= 1 cycle per second) the phase of the signal, representing the starting point of the cycle. To modulate the signal just means to systematically vary one of the three parameters of the signal: amplitude, frequency or phase. Therefore, the type of modulation may be categorized as either 1 .CONTINOUS WAVE MODULATION 2. PULSE MODULATION Where continuous wave modulation is of two types: 1 .Amplitude modulation 2.Angle modulation

Angle modulation is of 2 types : 1. Frequency modulation 2.Phase modulation FM Frequency modulation uses the information signal, V m(t) to vary the carrier frequency within some small range about its original value. Here are the three signals in mathematical form:

Information: V m(t) Carrier: V c(t) = V co sin ( 2 f c t + FM: VFM (t) = V co sin (2 f c + ( f/V

mo

) V m (t) + t

We have replaced the carrier frequency term, with a time-varying frequency. We have also introduced a new term: the frequency deviation. f, In this form, we should be able to see that the carrier frequency term: f c + (f/V mo) V m (t) now varies between the extremes of f c - f and f c + f. The interpretation of f becomes clear: it is the farthest away from the original frequency that the FM signal can be. Sometimes it is referred to as the "swing" in the frequency. The carrier swing = 2 * frequency deviation We can also define a modulation index for FM, analogous to AM:
m=

frequency deviation/modulation where fm is the modulating frequency used. Here is a simple FM signal:

frequency

= f/f

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Here, the carrier is at 30 Hz, and the modulating frequency is 5 Hz. The modulation index is about 3, making the peak frequency deviation about 15 Hz. That means the frequency vary somewhere between 15 and 45 Hz. How fast the cycle is completed is a function of the modulating frequency. FM Performance Bandwidth As we have already shown, the bandwidth of a FM signal may be predicted using: BW = 2( f + f m ) Where fm is the maximum modulating frequency used. FM radio has a significantly larger bandwidth than AM radio, but the FM radio band is also larger. The combination keeps the number of available channels about the same. The bandwidth of an FM signal has a more complicated dependency than in the AM case (recall, the bandwidth of AM signals depend only on the maximum modulation frequency). In FM, both the modulation index and the modulating frequency affect the bandwidth. As the information is made stronger, the bandwidth also grows. Efficiency The efficiency of a signal is the power in the side-bands as a fraction of the total. In FM signals, because of the considerable side-bands produced, the efficiency is generally high. Recall that conventional AM is limited to about 33 % efficiency to prevent distortion in the receiver when the modulation index was greater than 1. FM has no analogous problem. The side-band structure is fairly complicated, but it is safe to say that the efficiency is generally improved by making the modulation index larger (as it should be). But if we make the modulation index larger, so make the bandwidth larger (unlike AM) which has its disadvantages.
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Noise

The modulation index is normally limited to a value between 1 and 5, depending on the application.

FM systems are far better at rejecting noise than AM systems. Noise generally is spread uniformly across the spectrum (the so-called white noise, meaning wide spectrum). The amplitude of the noise varies randomly at these frequencies. The change in amplitude can actually modulate the signal and be picked up in the AM system. As a result, AM systems are very sensitive to random noise. An example might be ignition system noise in your car. Special filters need to be installed to keep the interference out of your car radio. FM systems are inherently immune to random noise. In order for the noise to interfere, it would have to modulate the frequency somehow. But the noise is distributed uniformly in frequency and varies mostly in amplitude. As a result, there is virtually no interference picked up in the FM receiver. FM is sometimes called "static free, " referring to its superior immunity to random noise. FM TRANSMITTER

Working of fm transmitter The transmitted signal is Frequency Modulated (FM) which means that the carriers amplitude stays constant and its frequency varies according to the amplitude variations of the audio signal. When the input signals amplitude increases (i.e. during the positive half cycles) the frequency of the carrier increases too, on the other hand when the input signal decreases in amplitude (negative half-cycle or no signal) the carrier frequency decreases accordingly. In figure 1 you can see a graphic representation of Frequency Modulation as it would appear on an oscilloscope screen, together with the modulating AF signal. The output frequency the transmitter is adjustable from 88 to 108 MHz which is the FM band that is used for radio broadcasting. The circuit as we have already mentioned consists of four stages. Three RF stages and one audio preamplifier for the modulation. The first RF stage is an oscillator and is
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built around TR1. The frequency of the oscillator is controlled by the LC network L1-C15. C7 is there to ensure that the circuit continues oscillating and C8 adjusts the coupling between the oscillator and the next RF stage which is an amplifier. This is built around TR2 which operates in class C and is tuned by means of L2 and C9. The last RF stage is also an amplifier built around TR3 which operates in class C the input of which is tuned by means of C10 and L4. From the output of this last stage which is tuned by means of L3-C12 is taken the output signal which through the tuned circuit L5-C11 goes to the aerial. The circuit of the preamplifier is very simple and is built around TR4. The input sensitivity of the stage is adjustable in order to make it possible to use the transmitter with different input signals and depends upon the setting of VR1. As it is the transmitter can be modulated directly with a piezoelectric microphone, a small cassette recorder etc. Result Thus we have studied and performed FM modulation EXPERIMENT NO 7

Object: - To generate and demodulate Frequency shift keyed (FSK) signal. Components Required: - Frequency Shift Keyed Practical kit, Cathode Ray Oscilloscope (CRO), Connecting Wires. Circuit Diagrams:Modulator:-

Demodulator:-

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THEORY: Modulation:- Modulation is a process in which the three basic characteristics (frequency, phase and amplitude) of the carrier signal changes according to the instantaneous value of the modulating signal. The only difference into the analog and the digital modulation is the carrier wave in the analog modulation is in analog form and in the digital modulation is in digital form. The types of analog communication are: 1. Amplitude modulation 2. Frequency modulation 3. Phase modulation The types of digital modulation are:

1. Amplitude shift keying modulation 2. Frequency shift keying modulation 3. Phase shift keying modulation Advantage of digital communication:-

1. Digital communication is more rugged then analog communication because it can withstand channel noise and distortion is within limits. 2. We can use regenerative repeaters in the digital communication. The distance over which an analog message can be transmitted is limited by transmitted power. 3. Digital hardware implementation is flexible and permits the use of microprocessor, miniprocessor, digital switching and large scale integrated circuit.

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4. Digital signal can be coded to yield extremely low error rate and highly fedility and as well as privacy. 5. Digital signal is inherently more efficient then analog in realizing the exchange of SNR and bandwidth. 6. Digital signal storage is relatively easy and inexpensive. Introduction to Generation of FSK: A frequency shift keyed transmitter has its frequency shifted by the message. Although there could be more than two frequencies involved in an FSK signal, in this Experiment the message will be a binary bit stream, and so only two frequencies will be involved. The word keyed suggests that the message is of the on-off (mark-space) variety, such as one (historically) generated by a morse key, or more likely in the present context, a binary sequence. The output from such a generator is illustrated in Figure 1 below.

Conceptually, and in fact, the transmitter could consist of two oscillators (on frequencies f1 and f2), with only one being connected to the output at any one time. This is shown in block diagram form in Figure 2 below.

Unless there are special relationships between the two oscillator frequencies and the bit clock there will be abrupt phase discontinuities of the output waveform during transitions of the message. Bandwidth: - Practice is for the tones f1 and f2 to bear special inter-relationships, and to be integer multiples of the bit rate. This leads to the possibility of continuous phase, which offers advantages, especially with respect to bandwidth control. Alternatively the frequency of a single oscillator (VCO) can be switched between two values, thus guaranteeing continuous phase - CPFSK.

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The continuous phase advantage of the VCO is not accompanied by an ability to ensure that f1 and f2 are integer multiples of the bit rate. This would be difficult (impossible ?) to implement with a VCO. FSK signals can be generated at baseband, and transmitted over telephone lines (for example). In this case, both f1 and f2 (of Figure 2) would be audio frequencies. Alternatively, this signal could be translated to a higher frequency. Yet again, it may be generated directly at carrier frequencies. Demodulation: - There are different methods of demodulating FSK. A natural classification is into synchronous (coherent) or asynchronous (noncoherent).Representative demodulators of these two types are the following: Asynchronous Demodulator: A close look at the waveform of Figure 1 reveals that it is the sum of two amplitude shift keyed (ASK) signals.

The receiver of Figure 3 takes advantage of this. The FSK signal has been separated into two parts by bandpass filters (BPF) tuned to the mark and space frequencies.

The output from each BPF looks like an Amplitude Shift Keyed (ASK) signal. These can be demodulated asynchronously, using the envelope. The decision circuit, to which the outputs of the envelope detectors are presented, selects the output which is the most likely one of the two inputs. It also re-shapes the waveform from a bandlimited to a rectangular form. This is, in effect, a two channel receiver. The bandwidth of each is dependent on the message bit rate. There will be a minimum frequency separation required of the two tones. Synchronous Demodulator: In the block diagram of Figure 4 two local carriers, on each of the two frequencies of the binary FSK signal, are used in two synchronous demodulators. A decision circuit examines the two outputs, and decides which is the most likely.

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This is, in effect, a two channel receiver. The bandwidth of each is dependent on the message bit rate. There will be a minimum frequency separation required of the two tones. This demodulator is more complex than most asynchronous demodulators. Phase Locked Loop: - A phase locked loop is a well known method of demodulating an FM signal. It is thus capable of demodulating an FSK signal. It is shown, in block diagram form, in Figure 5 below.

The control signal, which forces the lock, is a bandlimited copy of the message sequence. Depending upon the bandwidth of the loop integrator, a separate LPF will probably be required (as shown) to recover the message. Demodulator: - An example of this is the demodulator of Figure 3. The demodulator requires two bandpass (BPF) filters, tuned to the mark and space frequencies. Suitable filters exist as sub-systems in the bit clock regen module. To prepare the filters it is necessary to set the on-board switch SW1. Put the left hand toggle up, and right hand toggle down. This tunes BPF1 to 2.083 kHz, and BPF2 anywhere in the range 1 < fo < 5 kHz, depending on the VCO (the filter centre frequency will be 1/50 of the VCO frequency). If you do not have extra utilities and tuneable LPF modules, then complete just one arm of the demodulator. Alignment requires the BPFs to be tuned to the mark and space frequencies. The first is already done (2.083 kHz is already pre-set with SW1); the other is set with the VCO (already pre-set with SW2).

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Note that the specified bit rate is, by TIMS standards, rather low. The average oscilloscope display can be a little flickery. Use a short sequence, and the SYNC signal from the sequence generator to ext. trig. Result : thus we have performed the experiment frequency shift keying.

EXPERIMENT NO. 8 OBJECT: To generate and demodulate amplitude shift keyed (ASK) signal.

Components Required: - Amplitude Shift Keyed Practical kit, Cathode Ray Oscilloscope

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(CRO), Connecting Wires. Circuit Diagrams:Modulator:-

Demodulator:-

THEORY: Modulation:- Modulation is a process in which the three basic characteristics (frequency, phase and amplitude) of the carrier signal changes according to the instantaneous value of the modulating signal. The only difference into the analog and the digital modulation is the carrier wave in the analog modulation is in analog form and in the digital modulation is in digital form. Introduction to Generation of ASK: Amplitude Shift Keying [ASK] in the context of digital communications is a modulation process, which imparts to a sinusoid two or more discrete amplitude levels. These are related to the number of levels adopted by the digital message.
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For a binary message sequence there are two levels, one of which is typically zero. Thus the modulated waveform consists of bursts of a sinusoid. Figure 1 illustrates a binary ASK signal (lower), together with the binary sequence which Initiated it (upper). Neither signal has been bandlimited.

Figure 1: an ASK signal (below) and the message (above) There are sharp discontinuities shown at the transition points. These result in the signal having an unnecessarily wide bandwidth. Bandlimiting is generally introduced before transmission, in which case these discontinuities would be rounded off. The bandlimiting may be applied to the digital message, or the modulated signal itself. The data rate is often made a sub-multiple of the carrier frequency. This has been done in the waveform of Figure 1. One of the disadvantages of ASK, compared with FSK and PSK, for example, is that it has not got a constant envelope. This makes its processing (eg, power amplification) more difficult, since linearity becomes an important factor. However, it does make for ease of demodulation with an envelope detector. Introduction to Bandwidth Modification: As already indicated, the sharp discontinuities in the ASK waveform of Figure 1 imply a wide bandwidth. A significant reduction can be accepted before errors at the receiver increase unacceptably. This can be brought about by bandlimiting (pulse shaping) the message Before Modulation, or band limiting the ASK signal itself After Generation.

Figure 2: ASK generation method Figure 3 shows the signals present in a model of Figure 2, where the message has been bandlimited. The shape, after bandlimiting, depends naturally enough upon the amplitude and phase characteristics of the bandlimiting filter.

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Figure 3: original TTL message (lower), bandlimited message (center), and ASK (above) Introduction to Demodulation:It is apparent from Figures 1 and 4 that the ASK signal has a well defined envelope. Thus it is enabling to demodulation by an envelope detector. With bandlimiting of the transmitted ASK neither of these demodulation methods (envelope detection or synchronous demodulation) would recover the original binary sequence; instead, their outputs would be a bandlimited version. Thus further processing by some sort of decision-making circuitry for example - would be necessary. Thus demodulation is a two-stage process: 1. Recovery of the bandlimited bit stream 2. Regeneration of the binary bit stream

Figure 4: the two stages of the demodulation process Modelling an ASK Generator:It is possible to model the rather basic generator shown in fig. 2 the switch can be modelled by one half of a DUAL ANALOG SWITCH module. Being an analog switch, the carrier frequency would need to be in the audio range. The TTL output from the SEQUENCE GENERATOR is connected directly to the CONTROL input of the DUAL ANALOG SWITCH. For a synchronous carrier and message use the 8.333 kHz TTL sample clock (filtered by a TUNEABLE LPF) and the 2.083 kHz sinusoidal message from the MASTER SIGNALS module. If you need the TUNEABLE LPF for bandlimiting of the ASK, use the sinusoidal output from an AUDIO OSCILLATOR as the carrier. For a synchronized message as above, tune the oscillator close to 8.333 kHz, and lock it there with the sample clock connected to its SYNCH input.

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Demodulation of an ASK signal:Having a very definite envelope, an envelope detector can be used as the first step in recovering the original sequence. Further processing can be employed to regenerate the true binary waveform. The output from the above demodulators will not be a copy of the binary sequence TTL waveform. Bandlimiting will have shaped it, as (for example) illustrated in Figure 3. If the ASK has been bandlimited before or during transmission (or even by the receiver itself) then the recovered message, in the demodulator, will need restoration (cleaning up) to its original bi-polar format. Some sort of decision device is then required to regenerate the original binary sequence. This could be done with a COMPARATOR. Working: BIT CLOCK GENERATOR The bit clock generator is designed around the popular timer ic 555 operated in astable mode. The 1M potentiometer in conjunction with .004 micro farad condenser used in the timing circuit facilitates the freq to be set & at any chosen value from 200 Hz to 15KHz. 8 BIT WORD GENERATOR The 8 bit parallel load serial shift IC-74165 is used to generate the required word pattern. A set of 8 switches are used to set 1 & pattern. The bit pattern set by the switches is parallel loaded by controlling the logic level at pin 1. The serial shift clock is given at pin 2. The 8- it data set by the switches and loaded with the resistor parallel is now shifted serially and circulated repetitively thus 8- bit work pattern is generated cyclically which is used as modulating signal in the ASK modulator ASK MODULATOR A 4052 multiplexer is used as an ASK modulator. This is used as a 2 to 1 multiplexer. for one input carrier is given directly , and for the second input the is given by a resistive attenuator of 2:1 ratio control signal to the multiplexer is as the data signal, according to which the ASK modulator modulates. ASK DEMODULATOR A diode detector and a low pass filter of 3.4 KHz frequency is used to demodulate the ASK signal. The output of the low pass filter is given to the op-amp comparator the o/p is original data transmitted to the ASK modulator. RESULT: We have successfully study and perform the Amplitude Shift Keying (ASK) Generator and the Demodulator.

EXPERIMENT NO 9
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Object : To study phase shift keying modulator and demodulator. Apparatus required: ic : op-amp 741,4051,TL084,EXOR,transistor BC107,diode(1N4148),register(47k,22k,10k,1k),capacitor(470pf, 0.01uf) Theory: Modulation :modulation is the process of varying one or more properties of high frequency periodic waveform, called the carrier signal, with respect to a modulating signal. In analog modulation, the modulation is applied continuously in response to the analog information signal. Generally two types of modulation is used in communication: 1. analog modulation 2. digital modulation 1. Analog modulation Amplitude modulation (AM) (here the amplitude of the carrier signal is varied in accordance to the instantaneous amplitude of the modulating signal) Angle modulation 2. Digital modulation : In digital modulation, an analog carrier signal is modulated by a digital bit stream. Digital modulation methods can be considered as digital-to-analog conversion, and the corresponding demodulation or detection as analog-to-digital conversion. There are three major classes of digital modulation techniques used for transmission of digitally represented data:

Amplitude-shift keying (ASK) Frequency-shift keying (FSK) Phase-shift keying (PSK)

PSK, phase shift keying enables data to be carried on a radio communications signal in a more efficient manner than Frequency Shift Keying, FSK, and some other forms of modulation. Phase Shift Keying( PSK) In phase shift keying there are defined states or points that are used for signalling the data bits. The basic form of phase shift keying is known as Binary Phase Shift Keying (BPSK) or it is occasionally called Phase Reversal Keying (PRK). A digital signal alternating between +1 and -1 (or 1 and 0) will create phase reversals, i.e. 180 degree phase shifts as the data shifts state.

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Phase-shift keying (PSK) is a digital modulation scheme that conveys data by changing, or modulating, the phase of a reference signal (the carrier wave). Any digital modulation scheme uses a finite number of distinct signals to represent digital data. PSK uses a finite number of phases, each assigned a unique pattern of binary digits. Usually, each phase encodes an equal number of bits. Each pattern of bits forms the symbol that is represented by the particular phase. The demodulator, which is designed specifically for the symbol-set used by the modulator, determines the phase of the received signal and maps it back to the symbol it represents, thus recovering the original data. This requires the receiver to be able to compare the phase of the received signal to a reference signal such a system is termed coherent (and referred to as CPSK) There are several different types of phase shift key (PSK) modulators.

Two-phase (2 PSK) Four-phase (4 PSK) Eight-phase (8 PSK) Sixteen-phase (16 PSK) Sixteen-quadrature amplitude (16 QAM)

Two-Phase Shift Key Modulation In this modulator the carrier assumes one of two phases. A logic 1 produces no phase change and a logic 0 produces a 180 phase change. The output waveform for this modulator is shown below.

Applications 1. The wireless LAN standard, IEEE 802.11b-1999, uses a variety of different PSKs depending on the data-rate required
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2. It is appropriate for low-cost passive transmitters, and is used in RFID standards such as ISO/IEC 14443 which has been adopted for biometric passports, credit cards such as American Express's ExpressPay, and many other applications. 3. Bluetooth 2 will use / 4-DQPSK at its lower rate (2 Mbit/s) and 8-DPSK at its higher rate (3 Mbit/s) when the link between the two devices is sufficiently robust. 4. Bluetooth 1 modulates with Gaussian minimum-shift keying, a binary scheme, so either modulation choice in version 2 will yield a higher data-rate. 5. A similar technology, IEEE 802.15.4 (the wireless standard used by ZigBee) also relies on PSK. IEEE 802.15.4 allows the use of two frequency bands: 868915 MHz using BPSK and at 2.4 GHz using OQPSK.

Modulator

RESULT:

thus we have studied the phase shift keying and the results verified.

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EXPERIMENT NO. 10 Aim: -TO STUDY PULSE CODE MODULATION AND DEMODULATION Apparatus required: - Sampler, Quantizer, A/D converter, D/A converter, parallel to serial and serial to parallel converter, low band pass filter. Modulation: - It is the process of varying one or more properties of high frequency periodic waveform, called the carrier signal, with respect to a modulating signal. The three key parameters of a periodic waveform are its amplitude (strength), its phase ("timing") and its frequency ("pitch"), all of which can be modified in accordance with a low frequency signal to obtain the modulated signal. Typically a high-frequency sinusoid waveform is used as carrier signal, but a square wave pulse train may also occur. In telecommunications, modulation is the process of conveying a message signal, for example a digital bit stream or an analog audio signal, inside another signal that can be physically transmitted. Modulation of a sine waveform is used in view to transform a baseband message signal to a pass band signal, for example a radio-frequency signal (RF signal). Modulating a sine wave carrier makes it possible to keep the frequency content of the transferred signal as close as possible to the centre frequency (typically the carrier frequency) of the pass band. Modulation can broadly divide according to the carrier used:Analog modulation- carrier is analog sine wave of high frequency
a. Amplitude modulation (AM), in which the voltage applied to the carrier is varied

over time
b. Frequency modulation (FM), in which the frequency of the carrier waveform is

varied in small but meaningful amounts


c. Phase modulation (PM)-In which the natural flow of the alternating current

waveform is delayed temporarily Digital modulation- pulses of trains are used of high frequency. Pulse modulation-

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PAM (pulse amplitude modulation) PWM (pulse width modulation) PPM (pulse position modulation) PCM (pulse code modulation) PULSE-CODE MODULATION (PCM):- It is the heart of technology in communications in todays digital world. PCM is a method of converting an analog into digital signals. Information in an analog form cannot be processed by digital computers so it's necessary to convert them into digital form. PCM is a term which was formed during the development of digital audio transmission standards. Digital data can be transported robustly over long distances unlike the analog data and can be interleaved with other digital data so various combinations of transmission channels can be used. In the text which follows this term will apply to encoding technique which means digitalization of analog information in general. PCM doesn`t mean any specific kind of compression, it only implies PAM (pulse amplitude modulation) - quantization by amplitude and quantization by time which means digitalization of the analog signal. The range of values which the signal can achieve (quantization range) is divided into segments; each segment has a segment representative of the quantization level which lies in the middle of the segment. To every quantization segment (and quantization level) one and unique code word (stream of bits) is assigned. The value that a signal has in certain time is called a sample. The process of taking samples is called quantization by time. After quantization by time, it is necessary to conduct quantization by amplitude. Quantization by amplitude means that according to the amplitude of sample one quantization segment is chosen (every quantization segment contains an interval of amplitudes) and then record segments code word. To conclude, PCM encoded signal is nothing more than stream of bits.

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PCM BLOCK DIAGRAM Why PCM? The stream of pulses and non-pulse streams of 1s and 0s are not easily affected by interference and noise. Even in the presence of noise, the presence or absence of a pulse can be easily determined. Since PCM is digital, a more general reason would be that digital signals are easy to process by cheap standard techniques. This makes it easier to implement complicated communication systems such as telephone. One disadvantage of PCM is that the signal accuracy is reduced because of the quantizing of the samples. The practical implementation of PCM makes use of other processes. The processes are carried out in the order in which they appear below:

Filtering Sampling Quantizing Encoding

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FILTERING The filtering stage removes frequencies above the highest signal frequency. These frequencies if not removed, may cause problems when the signal is going through the stage of sampling. SAMPLING SAMPLING is the reduction of a continuous signal to a discrete signal. A common example is the conversion of a sound wave (a continuous-time signal) to a sequence of samples (a discrete-time signal). A sample refers to a value or set of values at a point in time and/or space. A sampler is a subsystem or operation that extracts samples from a continuous signal. A theoretical ideal sampler produces samples equivalent to the instantaneous value of the continuous signal at the desired points.

Sampling circuit

Signal sampling representation


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SAMPLING THEOREM The sampling theorem states that for a limited bandwidth (band-limited) signal with maximum frequency fmax, the equally spaced sampling frequency fs must be greater than twice of the maximum frequency fmax, i.e., fs > 2fmax In order to have the signal be uniquely reconstructed without aliasing. The frequency 2fmax is called the Nyquist sampling rate. Half of this value, fmax, is sometimes called the Nyquist frequency.

Spectrum

of

Sampled

Signal

Sampled at greater than the Nyquist rate Advantage of sampling is time division multiplexing:

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QUANTIZATION QUANTIZATION is the process of allocating levels to the infinite range of amplitudes of sample values of the analog signal. It is the process of approximating ("mapping") a continuous range of values (or a very large set of possible discrete values) by a relatively small ("finite") set of ("values which can still take on continuous range") discrete symbols or integer values. Sampling transforms a continuous-time signal into a discrete-time signal or sequence. The samples of the sequence can assume arbitrary values. However, in a digital implementation, real numbers have to be represented using a finite number of bits and the discrete-time sequence has therefore to be represented as a digital sequence. This can be achieved via quantization. Quantization is a non linear and irreversible operation that maps a given amplitude x(n) at time t=nT into a value xn, that belongs to a finite set of values. Linear Quantization In this experiment we deal only with linear quantization where the finite sets of values to choose from are uniformly spaced. More specifically, assume we are allowed only B bits to represent the amplitude of a sample. This means that for each sample, we should map its amplitude to only one of 2B possible levels. Now if the input signal is known to have a range R (i.e., the amplitudes of its samples can only lie between -R /2 and +R /2), then the spacing between these levels is equal to =R/2B. Once we have identified or defined the finite set of reconstruction levels, we can then choose a rule that allows us to map an amplitude to a certain reconstruction level. There are several ways to achieve this mapping and in this experiment we study three of them. The optimal way to map from the input value to the reconstruction level is to round the input value to the closest reconstruction level. Another way is to shorten or truncate an input value to the closest level that is smaller than it. A third way is to apply sign magnitude truncation, which maps an input value to the closest level whose absolute value is smaller than its absolute value.

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Assume, for example, that the reconstruction levels are {-3,-1, 1, 3}. The amplitude 2.9 will be mapped to the level 3 if rounding is used, to the level 1 if truncation is used, and to the level 1 again if sign magnitude truncation is used. On the other hand, the amplitude -2.9 will be mapped to the level -3 if rounding is used, to the level -3 again if truncation is used, and to the level -1 if sign magnitude truncation is used.

Quantized signal ENCODING The last process which is encoding. In this process each step level is assigned a number. The numbers start with zero at the lowest level. These assigned numbers are then expressed in binary form (in terms of 0s and 1s). This will be the last part of the conversion and the PCM signal will be transmitted/sent.

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DEMODULATION: To produce output from the sampled data, the procedure of modulation is applied in reverse. After each sampling period has passed, the next value is read and a signal is shifted to the new value. As a result of these transitions, the signal will have a significant amount of highfrequency energy. To smooth out the signal and remove these undesirable aliasing frequencies, the signal would be passed through analog filters that suppress energy outside the expected frequency range (that is, greater than the Nyquist frequency fs / 2).

Demodulation Block Diagram The electronics involved in producing an accurate analog signal from the discrete data are similar to those used for generating the digital signal. These devices are DACs (digital-toanalog converters), and operate similarly to ADCs. They produce on their output a voltage or current (depending on type) that represents the value presented on their inputs. This output would then generally be filtered and amplified for use.

RESULT: We have successfully performed the PCM of the signal and its demodulation.

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