EEC-659.doc

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Experiment 1 Objective:- Draw the AM waveforms for less than 100% and with greater than 100% modulation. Assume that the modulating signal is a pure sine wave. Apparatus Required:- Amplitude Modulator -Demodulator Trainer kit,CRO,CRO Probes,Patch cords etc. Theory:- Amplitude modulation (AM) is a technique used in electronic communication, most commonly for transmitting information via a radio carrier wave. AM works by varying the strength of the transmitted signal in relation to the information being sent. For example, changes in signal strength may be used to specify the sounds to be reproduced by a loudspeaker, or the light intensity of television pixels. Contrast this with frequency modulation, in which the frequency is varied, and phase modulation, in which the phase is varied. In the mid-1870s, a form of amplitude modulation —initially called "undulatory currents"—was the first method to successfully produce quality audio over telephone lines. Beginning with Reginald Fessenden's audio demonstrations in 1906, it was also the original method used for audio radio transmissions, and remains in use today by many forms of communication—"AM" is often used to refer to the medium wave broadcast band . When an amplitude modulated signal is created, the amplitude of the signal is varied in line with the variations in intensity of the sound wave. In this way the overall amplitude or envelope of the carrier is modulated to carry the audio signal. Here the envelope of the carrier can be seen to change in line with the modulating signal.

Transcript of EEC-659.doc

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Experiment 1

Objective:- Draw the AM waveforms for less than 100% and with greater than 100% modulation. Assume that the modulating signal is a pure sine wave.

Apparatus Required:- Amplitude Modulator -Demodulator Trainer kit,CRO,CRO Probes,Patch cords etc.

Theory:-

Amplitude modulation (AM) is a technique used in electronic communication, most commonly for transmitting information via a radio carrier wave. AM works by varying the strength of the transmitted signal in relation to the information being sent. For example, changes in signal strength may be used to specify the sounds to be reproduced by a loudspeaker, or the light intensity of television pixels. Contrast this with frequency modulation, in which the frequency is varied, and phase modulation, in which the phase is varied. In the mid-1870s, a form of amplitude modulation—initially called "undulatory currents"—was the first method to successfully produce quality audio over telephone lines. Beginning with Reginald Fessenden's audio demonstrations in 1906, it was also the original method used for audio radio transmissions, and remains in use today by many forms of communication—"AM" is often used to refer to the medium wave broadcast band .

When an amplitude modulated signal is created, the amplitude of the signal is varied in line with the variations in intensity of the sound wave. In this way the overall amplitude or envelope of the carrier is modulated to carry the audio signal. Here the envelope of the carrier can be seen to change in line with the modulating signal.

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Forms of Amplitude modulation

In radio communication, a continuous wave radio-frequency signal (a sinusoidal carrier wave) has its amplitude modulated by an audio waveform before transmission. The audio waveform modifies the amplitude of the carrier wave and determines the envelope of the waveform. In the frequency domain, amplitude modulation produces a signal with power concentrated at the carrier frequency and two adjacent sidebands. Each sideband is equal in bandwidth to that of the modulating signal, and is a mirror image of the other. Amplitude modulation resulting in two sidebands and a carrier is called "double-sideband amplitude modulation" (DSB-AM). Amplitude modulation is inefficient in power usage; at least two-thirds of the power is concentrated in the carrier signal, which carries no useful information (beyond the fact that a signal is present).

To increase transmitter efficiency, the carrier may be suppressed. This produces a reduced-carrier transmission, or DSB "double-sideband suppressed-carrier" (DSB-SC) signal. A suppressed-carrier AM signal is three times more power-efficient than AM. If the carrier is only partially suppressed, a double-sideband reduced-carrier (DSBRC) signal results. For reception, a local oscillator will typically restore the suppressed carrier so the signal can be demodulated with a product detector.

Amplitude modulation advantages & disadvantages

Like any other system of modulation, amplitude modulation has several advantages and disadvantages.

Advantages Disadvantages

It is simple to implement It can be demodulated using a circuit

consisting of very few components

AM receivers are very cheap as no specialised components are needed.

An amplitude modulation signal is not efficient in terms of its power usage

It is not efficient in terms of its use of bandwidth, requiring a bandwidth equal to twice that of the highest audio frequency

An amplitude modulation signal is prone to high levels of noise because most noise is amplitude based and obviously AM detectors are sensitive to it.

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Procedure:-

1. Turn on power to the Amplitude Modulator -Demodulator Trainer kit.

2. Provide the kit with a sine wave from Function Generator as modulating signal.

3. Provide the kit with another high frequency sine wave from Function Generator as carrier signal.

4.Connect the output of kit to the input of CRO.

5.Trace the waveform of AM signal from CRO.

Observation Table:-

Result:- The amplitude modulated waveform is generated and determine the depth of Modulation is calculated as___________.

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Experiment 2

Objective:- Generate DSB-SC signal and calculate the percent power saving for a DSB-SC signal for the percent modulation of 100% .

Apparatus Required:-ST2201 Kit,CRO,CRO Probes,Patch cords etc.

Theory:-

Double-sideband suppressed-carrier transmission (DSB-SC): transmission in which

(a) frequencies produced by amplitude modulation are symmetrically spaced above and below the carrier frequency and

(b) the carrier level is reduced to the lowest practical level, ideally completely suppressed.

In the double-sideband suppressed-carrier transmission (DSB-SC) modulation, unlike AM, the wave carrier is not transmitted; thus, a great percentage of power that is dedicated to it is distributed between the sidebands, which implies an increase of the cover in DSB-SC, compared to AM, for the same power used. This is used for RDS (Radio Data System).This is basically an amplitude modulation wave without the carrier therefore reducing power wastage, giving it a 100% efficiency rate. This is an increase compared to normal AM transmission (DSB), which has a maximum efficiency of 33.33%, since efficiency is defined as:

E = (Sideband Power)/(Sideband Power + Carrier Power).

Generation

DSB-SC is generated by a mixer. This consists of a message signal combined with the frequency carrier.

Procedure:-

1.Ensure that the mode switch of the ST2201 kit is in DSB position.

2.Turn on power to the ST2201 board.

3.Connect the output of the balanced modulator on the kit to the input of CRO.

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4.Trace the waveform of DSB-SC signal from CRO.

Waveform:-

Result:- The DSB-SC signal waveform is generated.

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Experiment 3

Objective:- Generate SSB signal and calculate the percent power saving for a SSB signal if the AM wave is modulated to a depth of 100% .

Apparatus Required:- ST2201 Kit,CRO,CRO Probes,Patch cords etc.

Theory:-In radio communications, single-sideband modulation (SSB) or single-sideband suppressed-carrier (SSB-SC) is a refinement of amplitude modulation that more efficiently uses transmitter power and bandwidth. Amplitude modulation produces an output signal that has twice the bandwidth of the original baseband signal. Single-sideband modulation avoids this bandwidth doubling, and the power wasted on a carrier, at the cost of increased device complexity and more difficult tuning at the receiver.

In an AM-modulated radio signal, a base signal, called the carrier, is continuously broadcast. The two modulating signals are called the sidebands. Any audio that one hear on an AM broadcast station is from the two sidebands. When the radio station is not transmitting any sound, one can still hear that a signal is present; that is the carrier. These two modulating (audio) sidebands are located on either side of the carrier signal--one just above the other just below. As a result, the sideband located just above the carrier frequency is called the upper sideband and that which is located just below the carrier frequency is called the lower sideband.

Procedure:-

1.Ensure that the mode switch of the ST2201 kit is in SSB position.

2.Turn on power to the ST2201 board.

3.Connect the output of the balanced modulator on the kit to the input of CRO.

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4.Trace the waveform of DSB-SC signal from CRO.

Waveform:-

Result:- The SSB signal waveform is generated.

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Experiment 4

Objective:- To observe diagonal peak clipping effect by demodulating amplitude modulated waveform using envelope detector.

Apparatus Required:- Amplitude Modulator -Demodulator Trainer kit,CRO,CRO Probes,Patch cords etc.

Theory:-

Demodulation is the act of extracting the original information-bearing signal from a modulated carrier wave. A demodulator is an electronic circuit (or computer program in a software-defined radio) that is used to recover the information content from the modulated carrier wave. These terms are traditionally used in connection with radio receivers, but many other systems use many kinds of demodulators. Another common one is in a modem, which is a contraction of the terms modulator/demodulator.

An envelope detector is an electronic circuit that takes a high-frequency signal as input and provides an output which is the envelope of the original signal. The capacitor in the circuit stores up charge on the rising edge, and releases it slowly through the resistor when the signal falls. The diode in series rectifies the incoming signal, allowing current flow only when the positive input terminal is at a higher potential than the negative input terminal.

Simple envelope detector

Most practical envelope detectors use either half-wave or full-wave rectification of the signal to convert the AC audio input into a pulsed DC signal. Filtering is then used to smooth the final result. This filtering is rarely perfect and some "ripple" is likely to remain on the envelope follower output, particularly for low frequency inputs such as notes from a bass guitar. More filtering gives a smoother result, but decreases the responsiveness; thus, real-world designs must be optimized for the application.

Procedure:-

1. Turn on power to the Amplitude Modulator -Demodulator Trainer kit.

2. Provide the kit with a sine wave from Function Generator as modulating signal.

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3. Provide the kit with another high frequency sine wave from Function Generator as carrier signal.

4.Connect the output of AM Modulator to the input of AM Demodulator on the kit.

5. Connect the output of AM DeModulator on the kit to the input of CRO.

5.Trace the waveform of demodulated AM signal from CRO.

Waveform:-

Result:- The Amplitude modulated signal is demodulated and the waveform of demodulated AM wave is generated.

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Experiment 5

Objective:- To generate frequency modulated waveform using voltage controlled oscillator with a 6 MHz carrier ,frequency modulated by a 7 KHz sine wave.

Apparatus Required:- FM using VCO kit,CRO,CRO Probes,Patch cords etc.

Theory:-

In telecommunications and signal processing, frequency modulation (FM) conveys information over a carrier wave by varying its instantaneous frequency. FM is widely used for broadcasting music and speech, two-way radio systems, magnetic tape-recording systems and some video-transmission systems. In radio systems, frequency modulation with sufficient bandwidth provides an advantage in cancelling naturally-occurring noise.The voltage controlled oscillator (VCO) is a device whose frequency changes linearly with an input voltage. It is used to perform direct frequency modulation on signals. VCO has a center frequency fc

and the input (control) voltage m(t) modulates the instantaneous frequency around this center frequency.

Procedure:-

1. Turn on power to the FM using VCO Trainer kit.

2. Provide the VCO input of the kit with a sine wave from Function Generator.

3.Connect the VCO output of the kit to the input of CRO.

4.Connect the ground terminals of kit, function generator and CRO together.

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5.Trace the waveform of FM signal from CRO.

Waveform:-

Result:- The FM signal waveform is generated using VCO.

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Experiment 6

Objective:- To detect FM signal using Phase Locked Loop and find the bandwidth of FM signal if fc is 100 MHz , Δf is 75 KHz and fm is 15 KHz

Apparatus Required:- ST2203 Kit,CRO,CRO Probes,Function Generator, Patch cords etc.

Theory:-

In an FM signal, the instantaneous frequency varies in accordance with the modulating signal. For a sinusoidal modulating signal, the frequency deviation in an FM signal is sinusoidal, and it is proportional to the modulating, amplitude. The changes in the instantaneous frequency of the carrier signal occur with respect to the previously attained value of the carrier frequency. Thus a PLL can be used to demodulate an FM signal.Suppose the center frequency of the FM signal is fc, and it lies within the hold-in range of PLL the VCO is locked to fc, by applying an demodulated carrier at the input of the phase detector. When VCO is locked to fc, the error signal is zero, and therefore, the control signal that changes the VCO frequency is also equal to zero. If an FM signal is applied to the phase detector, there will be a difference in the phases of the VCO output and the input FM signal. The control signal is produced in proportion to the phase difference at an instance of time. This control voltage will modify the VCO frequency, which is again compared with the incoming frequency. In this way, the current incoming frequency is compared with the previously attained value of the VCO frequency, which is the previously attained frequency of the FM Signal.

The VCO, therefore, tries to track the instantaneous frequency of the applied FM signal. The control signal is produced in proportion to the difference between the VCO frequency and the instantaneous frequency of FM signal. In other words, the control signal so produced is proportional to the frequency deviation in the FM signal. Since the frequency deviation si proportional to the modulating signal, the control signal appearing at the output of LPF is the modulating signal. Therefore, the FM signal is demodulated by PLL.

A phase-locked loop or phase lock loop (PLL) is a control system that generates an output signal whose phase is related to the phase of an input "reference" signal. It is an electronic circuit consisting of a variable frequency oscillator and a phase detector. This circuit compares the phase of the input signal with the phase of the signal derived from its output oscillator and adjusts the frequency of its oscillator to keep the phases matched. The signal from the phase detector is used to control the oscillator in a feedback loop.

Frequency is the time derivative of phase. Keeping the input and output phase in lock step implies keeping the input and output frequencies in lock step. Consequently, a phase-locked loop can track an input frequency, or it can generate a frequency that is a multiple of the input frequency. The former property is used for demodulation, and the latter property is used for indirect frequency synthesis.

Phase-locked loops are widely employed in radio, telecommunications, computers and other electronic applications. They can be used to recover a signal from a noisy communication channel, generate stable frequencies at a multiple of an input frequency (frequency synthesis), or distribute clock timing pulses in digital logic designs such as microprocessors. Since a single integrated circuit can provide a complete phase-locked-loop building block, the technique is

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widely used in modern electronic devices, with output frequencies from a fraction of a hertz up to many gigahertz.

IC 565 PLL

A PLL is also available as an integrated circuit IC. IC 565 PLL can be used for FM detection.

Figure (a) shows the circuit diagram of an FM detector using 565 PLL.

Internal Block Diagram of IC 565

The internal block diagram shows that IC 565 PLL consists of phase detector, VCO, and amplifier. The amplifier also functions as the low pass filter. This is a 14 pin dual in line package.

In figure (b), notice that PLL IC consists of two power supply pins marked ±Vcc. The positive terminal of the Vcc is connected to pin number 10, and the negative (ground) terminal of Vcc is connected to pin number 1. The output signal to the phase detector is applied to pin numbers 2 and 3. The VCO output is applied to the phase detector through pin number 5. The output of the phase detector is internally connected to the amplifier (low-pass filter).

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The output or the phase detector is low-pass filtered and amplified by the amplifier stage. The Output or the amplifier is the control voltage that is applied to VCO to force it to track the incoming frequency. The control voltage is also available at pin number 7. This is the output signal. In the ease of FM demodulator, the signal at pin number 7 is the modulating signal. The amplifier also generates an output at pin number 6for reference purposes.

The VCO gets its control voltage internally from the amplifier and its output at pin number 4. The VCO output should be given to the phase detector through pin number 5. It is customary to short pin numbers 4 and 5 so that the VCO output is applied directly to the phase detector. The external resistor and capacitor can set the free-running frequency of the VCO. The resistor and capacitor are called the timing resistor and the timing capacitor. The timing resistor is connected at pin number 8, and the timing capacitor is connected at pin number 9.

Pin numbers 11, 12, 13, and 14 are not connected because they do not have any internal circuitry with them. These are marked as NC in Figure (b). These pins are there because they are connected to IC.

Procedure:-

1. Turn on power to the ST2203 kit.

2. Connect the Audio output of the Audio Oscillator on the kit to the input of Varactor modulator on the kit.

2. Connect the FM output of the kit to the input of PLL Detector on the kit.

3. Connect the output of PLL Detector on the kit to the input of Low Pass Filter Amplifier.

4. Trace the waveform of demodulated FM signal from output of PLL detector on CRO.

Result:- The FM signal waveform is detected using PLL and original sine wave is obtained as output.

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Experiment 7

Objective:- To generate PAM, PWM and PPM waveforms and for PAM find the transmission bandwidth BT if the width of each pulse is τ = 0.1 Ts and sampling frequency fs is 8 KHz.

Apparatus Required:- PAM, PWM and PPM Modulation and Demodulation Trainer (ST2110), CRO,Patch cords etc.

Theory:-

Pulse amplitude modulation is a pulse modulation system in 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.

Pulse-amplitude modulation, acronym PAM, is a form of signal modulation where the message information is encoded in the amplitude of a series of signal pulses. It is an analog pulse modulation scheme in which the amplitude of train of carrier pulse are varied according to the sample value of the message signal.The signal is sampled at regular intervals and each sample is made proportional to the magnitude of the signal at the instant of sampling. These sampled pulses may then be sent either directly by a channel to the receiving end or may be made to modulated using a carrier wave before transmission.

Pulse-width modulation (PWM), or pulse-duration modulation (PDM),In this, the amplitude and starting time of each pulse is kept fixed but the width of each pulse is made proportional to the amplitude of the signal at that instant. The main advantage of PWM is that power loss in the switching devices is very low.

Pulse-position modulation (PPM):-The amplitude and width of the pulses are kept constant in this system, while the position of each pulse in relation to the position of recurrent pulse is varied by each instantaneous sampled value of the modulating wave. It is primarily useful for optical communications systems, where there tends to be little or no multipath interference.

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Procedure:-

1. Turn on power to the ST2110 kit.

2. Make the Connections according to the block diagram.

3. Connect Pulse Generator clock to the PAM/PPM/PWM modulator.

4. Connect the audio frequency of 2 KHz,2V to modulator.

5. Connect the modulator output to CRO.

6. Observe output on CRO.

Result:- The PAM, PWM and PPM waveforms are generated.

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Experiment 8

Objective:- To find the nature of PSK modulated waveform if the data bit sequence is 1 0 1 1 1 0 1 0.

Apparatus Required:- ST2206 Kit,CRO,CRO Probes,Patch cords etc.

Theory:-

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).

Procedure:-

1. Turn on power to the ST2106 kit.

2. Provide the Carrier input of Carrier Modulation Circuit on the kit with a sine wave from Function Generator.

3. Provide the Modulation input of Carrier Modulation Circuit on the kit with a Clock.

4.Connect the output of Carrier Modulation Circuit on the kit to the input of CRO.

5.Trace the waveform of PSK signal from CRO.

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Result:- The PSK waveform is generated.

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Experiment 9

Objective:- To generate FSK modulated waveform and calculate the bandwidth of an FSK system in which the transmission takes place at 4000 bits per second rate and the frequency difference between the two carriers is 3000 Hz.

Apparatus Required:- FSK Kit, CRO,CRO Probes, Patch cords etc.

Theory:-Frequency-shift keying (FSK) is a frequency modulation scheme in which digital information is transmitted through discrete frequency changes of a carrier wave. The simplest FSK is binary FSK (BFSK). BFSK uses a pair of discrete frequencies to transmit binary (0s and 1s) information.With this scheme, the "1" is called the mark frequency and the "0" is called the space frequency.

Frequency-shift keying (FSK) is a method of transmitting digital signals. The two binary states, logic 0 (low) and 1 (high), are each represented by an analog waveform. Logic 0 is represented by a wave at a specific frequency, and logic 1 is represented by a wave at a different frequency. A modem converts the binary data from a computer to FSK for transmission over telephone lines, cables, optical fiber, or wireless media. The modem also converts incoming FSK signals to digital low and high states, which the computer can "understand."

Procedure:-

1. Make the Connections according to the block diagram.

2. Ensure that the mode 2 of the ST2106 kit is switched to Down position.

3. Observe the FSK output on CRO.

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Result:- The FSK waveform is generated.

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Experiment 10

Objective:- To observe the effect of sampling rate & the width of the sampling pulses by generating sampled signal through a Sample-Hold circuit and reconstruct the sampled signal.

Apparatus Required:- ST2101 Kit,CRO,CRO Probes,Patch cords etc.

Theory:-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 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.

The continuous signal is sampled using an analog-to-digital converter (ADC), a device with various physical limitations. This results in deviations from the theoretically perfect reconstruction, collectively referred to as distortion.

In electronics, a sample and hold (S/H, also "follow-and-hold" circuit is an analog device that samples (captures, grabs) the voltage of a continuously varying analog signal and holds (locks, freezes) its value at a constant level for a specified minimal period of time. Sample and hold circuits and related peak detectors are the elementary analog memory devices. They are typically used in analog-to-digital converters to eliminate variations in input signal that can corrupt the conversion process.

A typical sample and hold circuit stores electric charge in a capacitor and contains at least one fast FET switch and at least one operational amplifier. To sample the input signal the switch connects the capacitor to the output of a buffer amplifier. The buffer amplifier charges or discharges the capacitor so that the voltage across the capacitor is practically equal, or proportional to, input voltage. In hold mode the switch disconnects the capacitor from the buffer. The capacitor is invariably discharged by its own leakage currents and useful load currents,

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which makes the circuit inherently volatile, but the loss of voltage (voltage drop) within a specified hold time remains within an acceptable error margin.

In the context of LCD screens, it is used to describe when a screen samples the input signal, and the frame is held there without redrawing it. This does not allow the eye to refresh and leads to blurring during motion sequences, also the transition is visible between frames because the backlight is constantly illuminated, adding to display motion blur.

Sampled Signal

Sample and Hold Signal

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Various types of distortion can occur, including:

Aliasing. A precondition of the sampling theorem is that the signal be bandlimited. However, in practice, no time-limited signal can be bandlimited. 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 bandlimited. However, by designing a sampler with an appropriate guard band, it is possible to obtain output that is as accurate as necessary.

Integration effect or aperture effect. This results from the fact that the sample is obtained as a time average within a sampling region, rather than just being equal to the signal value at the sampling instant. The integration effect is readily noticeable in photography when the exposure is too long and creates a blur in the image. An ideal camera would have an exposure time of zero. In a capacitor-based sample and hold circuit, the integration effect is introduced because the capacitor cannot instantly change voltage thus requiring the sample to have non-zero width.

Jitter or deviation from the precise sample timing intervals. Noise, including thermal sensor noise, analog circuit noise, etc. Slew rate limit error, caused by an inability for an ADC output value to change

sufficiently rapidly. Quantization as a consequence of the finite precision of words that represent the

converted values. Error due to other non-linear effects of the mapping of input voltage to converted output

value (in addition to the effects of quantization).

Procedure:-

1. Turn on power to the ST2101 kit.

2. Connect 1 KHz sine wave of Sampling Frequency Selection Circuit on the kit to Ext.Sampling Pulse input of Sampling circuit on the kit.

3. Provide the sine wave from Function Generator to Signal Input of Sampling circuit on the kit.

4. Connect the ground terminal of Function Generator & CRO to ground terminal of the Kit.

5.Connect the sample output of Sampling circuit on the kit to CRO.

6.Trace the waveform of Sampled signal from CRO.

7. Connect the Sample & Hold output of Sampling circuit on the kit to CRO.

8.Trace the waveform of Sampled & Hold signal from CRO.

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Result:- The sampled and sample and hold signal through a Sample-Hold circuit is generated.