Analog Communications Lab Manual_asrao

Analog Communications Lab Manual

DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING

LABORATORY MANUAL

ANALOG COMMUNICATIONS (III B.Tech., - I – Sem.)

Prepared by

A.SANYASI RAO, Assoc. Prof. S.SRINIVAS, Asst. Prof

BALAJI INSTITUTE OF ENGINEERING & SCIENCES Laknepally, Narsampet, Warangal Dept. of ECE

Balaji Institute of Engineering & Sciences


LIST OF EXPERIMENTS 1. Amplitude modulation and demodulation. 2. DSB-SC Modulator & Detector. 3. SSB-SC Modulator & Detector (Phase shift method). 4. Frequency modulation and demodulation. 5. Study of spectrum analyzer and analysis of AM & FM signals. 6. Pre – emphasis & De – emphasis. 7. Time division multiplexing & de-multiplexing. 8. Frequency division multiplexing & de-multiplexing. 9. Verification of sampling theorem. 10. Pulse Amplitude Modulation & De-modulation. 11. Pulse Width Modulation & De-modulation. 12. Pulse Position Modulation & De-modulation. 13. Frequency Synthesizer. 14. AGC Characteristics. 15. PLL as FM Demodulator.

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LIST OF EXPERIMENTS

Cycle1 1. Amplitude modulation and demodulation. 2. DSB-SC Modulator & Detector. 3. SSB-SC Modulator & Detector (Phase shift method). 4. Frequency modulation and demodulation. 5. Pre – emphasis & De – emphasis. 6. Frequency Synthesizer.

Cycle2 7. Time division multiplexing & de-multiplexing. 8. Verification of sampling theorem. 9. Pulse Amplitude Modulation & De-modulation. 10. Pulse Width Modulation & De-modulation. 11. Pulse Position Modulation & De-modulation. 12. PLL as FM Demodulator.

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Analog Communications Hardware Experiments

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1. AMPLITUDE MODULATION AND DEMODULATION Objective 1. To observe the process of amplitude modulation and demodulation and calculate depth of modulation. 2. To study the process of over modulation.

Equipment 1. Amplitude modulation and demodulation trainer kit 2. CRO with probes 3. Patch cords THEORY: Amplitude Modulation is defined as a process in which the amplitude of the carrier wave c(t) is varied linearly with the instantaneous amplitude of the message signal m(t).The standard form of an amplitude modulated (AM) wave is defined by S(t) = Ac [1 + Ka m(t)] Cos(2пfct) Where Ka is constant called the amplitude sensitivity of the modulator. The demodulation circuit is used to recover the message signal from the incoming AM wave at the receiver. An envelope detector is a simple and yet highly effective device that is well suited for the demodulation of AM wave, for which the percentage modulation is less than 100%.Ideally, an envelope detector produces an output signal that follows the envelop of the input signal wave form exactly; hence, the name. Some version of this circuit is used in almost all commercial AM radio receivers. The Modulation Index is defined as

m = (Vmax + Vmin) / (Vmax –

Vmin) Where Vmax and Vmin are the maximum and minimum amplitudes of the modulated wave.

Block Diagram

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Procedure Modulation 1. Switch on the trainer kit. 2. Measure the output voltages of regulated power supply +12V, -12V. 3. Observe outputs of RF and AF signal generators using CRO. Set RF voltage approximately 10Vpp of 100 KHz frequency and AF voltage as 300 mVpp of 2KHz frequency. 4. Now connect AF and RF signals to the respective inputs of modulator as shown in figure. Initially set both the signals at zero level. 5. Observe both modulating and modulated signals simultaneously in CRO. 6. Adjust RF signal amplitude to have modulator output at 10 Vpp by keeping AF signal at zero level. 7. Now vary the amplitude of AF signal and observe the minimum and maximum voltages of the modulated signal for different values of modulating voltages and calculate the percentage of modulation.

Modulation Index 

Vmax  Vmin Vmax  Vmin

8. Observe 100% modulation and over modulation by varying amplitude of AF signal.

Demodulation 1. Now connect modulator output to demodulator input pin. 2. Observe demodulated signal at output of demodulator using oscilloscope. 3. Compare it with the original AF signal.

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Calculations & Observations


Vmax  Vmin Vmax  Vmin

Modulating Signal Generator

Amplitude =

Time Period =

Frequency =

Time Period =

Frequency =

Time Period =

Frequency =

Carrier Signal Generator:

Amplitude =

Demodulated Output:

Amplitude =

Tabular Form: Modulating signal amplitude (V)

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Vmax

Vmin

Modulation index V  Vmin ma  max Vmax  Vmin



Model Wave Forms

Precautions 1. Avoid loose connections. 2. Avoid parallax error while taking observations.

Result Dept. of ECE



2. DSB – SC MODULATOR AND DETECTOR Objective 1. To observe the generation and detection processes of DSB-SC signal using Balanced Modulator. 2. To observe the modulated and demodulated outputs.

Equipment 1. Balanced modulator and demodulator trainer kit 2. CRO with probes 3. Patch cards

Theory Balanced modulator is used for generating DSB-SC signal. A balanced modulator consists of two standard amplitude modulators arranged in a balanced configuration so as to suppress the carrier wave. The two modulators are identical except the reversal of sign of the modulating signal applied to them. 1. RF Generator: Colpitts oscillator using FET is used here to generate RF signal of approximately 100 KHz Frequency to use as carrier signal in this experiment. Adjustments for Amplitude and Frequency are provided in panel for ease of operation. 2. AF Generator: Low Frequency signal of approximately 5 KHz is generated using OP-AMP based Wein Bridge Oscillator. IC TL 084 is used as an active component; TL 084 is FET input general purpose quad OP-AMP integrated circuit. One of the OP-AMP has been used as amplifier to improve signal level. Facility is provided to change output voltage. 3. Regulated Power Supply: This consists of bridge rectifier, capacitor filters and three terminal regulators to provide required dc Voltage in the circuit i.e. +12v, -8v @ 150 ma each. 4. Modulator: The IC MC 1496 is used as Modulator in this experiment. MC 1496 is a monolithic integrated circuit Balanced modulator/Demodulator, is versatile and can be used up to 200 MHz.

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5. Multiplier: A balanced modulator is essentially a multiplier. The output of the MC 1496 balanced modulator is proportional to the product of the two input signals. If you apply the same sinusoidal signal to both inputs of a ballooned modulator, the output will be the square of the input signal AM-DSB/SC: If you use two sinusoidal signals with deferent frequencies at the two inputs of a balanced modulator (multiplier) you can produce AMDSB/SC modulation. This is generally accomplished using a high- frequency “carrier” sinusoid and a lower frequency “modulation” waveform (such as an audio signal from microphone).

Circuit Diagram

Procedure 1. Switch on the trainer kit. 2. Measure the output voltages of regulated power supply +12V, -12V & -8V. 3. Observe the output of RF generator using CRO. Set carrier wave in RF generator to 100 KHz frequency and 300 mV peak to peak. 4. Observe the output of AF generator using CRO. Set modulating wave in AF generator to 5 KHz frequency and 100 mV peak to peak.

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5. Connect the respective AF and RF outputs to the Balanced modulator input pins and observe the DSB-SC output. (Here you can clearly observe the phase reversal at zero crossings). 6. Connect the Balanced modulator output and RF generator output to the Synchronous detector and observe the demodulator output. 7. Compare demodulated output with original AF signal.

Observations Message Signal: Amplitude =

Time Period =

Frequency =

Time Period =

Frequency =

Time Period =

Frequency

Carrier Signal: Amplitude = Demodulated Signal: Amplitude =

Model Wave Forms

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Precautions 1. Avoid loose connections 2. Avoid parallax error while taking observations.

Result

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3. SSB – SC MODULATOR AND DETECTOR Aim To observe process of single side band signal generation using phase shift method and to demodulate the same using synchronous detector.

Equipment 1. SSB Trainer kit 2. CRO with probes 3. Frequency counter 4. Patch cords

Theory An SSB signal is produced by passing the DSB signal through a highly selective band pass filter. This filter selects either the upper or the lower sideband. Hence transmission bandwidth can be cut by half if one sideband is entirely suppressed. This leads to single side band modulation (SSB). In SSB modulation bandwidth saving is accompanied by a considerable increase in equipment complexity. Single Sideband Suppressed Carrier (SSB-SC) modulation was the basis for all long distance telephone communications up until the last decade. It was called "L carrier." It consisted of groups of telephone conversations modulated on upper and/or lower sidebands of contiguous suppressed carriers. The groupings and sideband orientations (USB, LSB) supported hundreds and thousands of individual telephone conversations. Due to the nature of-SSB, in order to properly recover the fidelity of the original audio, a pilot carrier was distributed to all locations (from a single very stable frequency source), such that, the phase relationship of the demodulated (product detection) audio to the original modulated audio was maintained. Also, SSB was used by the U.S. Air force's Strategic Air Command (SAC) to insure reliable communications between their nuclear bombers and NORAD. In fact, before satellite communications SSB-was the only reliable form of communications with the bombers. The main reason-SSB-is superior to-AM,-and most other forms of modulation are:

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(1) Since the carrier is not transmitted in SSB, there is a reduction by 50% of the transmitted power. In AM out of 100% modulation: 67% of the power is comprised of the carrier; with the remaining 33% power in both sidebands. (2) Because in SSB, only one sideband is transmitted, there is a further reduction by 50% in transmitted power. (3) Finally, because only one sideband is received, the receiver's needed bandwidth is reduced by one half--thus effectively reducing the required power by the transmitter another 50%

Block Diagram:

Procedure Modulation: 1. Study the circuit operation of SSB system thoroughly. 2. Observe the Output of the RF generator using CRO. There are 2 outputs from the RF generator one is direct output and another is 90º phase shift of the direct output. Adjust the RF signal frequency to 100 KHz and amplitude to 0.2 Vp-p (potentiometer is provided to vary the output amplitude). 3. Observe the output of the AF generator using CRO. There are 2 outputs from the AF generator. One is direct output and another is 90º phase shift to the direct output. A switch is provided to adjust the required frequency of any one of 2 KHz, 4 KHz or 6 KHz. Set the

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AF generator frequency to 2 KHz and amplitude of 10VP-P. AGC Potentiometer is provided to adjust the gain of the oscillator (to set the output to good shape). 4. Connect the RF generator direct output (0o) and AF generator indirect (90º) output to the balanced modulator (A) and similarly RF generator indirect output (90º) and AF generator direct output (0º) to the balance modulator (B). 5. Observe the outputs of both the balanced modulators simultaneously using dual trace oscilloscope and adjust the balance control until you get the output waveforms properly. 6. To get SSB Lower Side Band (LSB) signal, connect the two balanced modulators outputs to subtractor. 7. Measure and record the LSB signal frequency using frequency counter. 8. Calculate theoretical frequency of LSB and compare it with the practical value, LSB = RF - AF 9. To get SSB Upper Side Band (USB) signal, connect the both outputs of balanced modulators to the summer. 10. Measure and record the USB signal frequency using frequency counter. 11. Calculate the theoretical value of USB frequency and compare it with the practical value, USB = RF + AF

Demodulation

1. Connect SSB-SC signal from the summer or subtractor to the SSB signal input pin of the demodulator. Also connect RF direct output to the RF input pin of the demodulator. 2. Observe the demodulator output using CRO. 3. The output has to be the replica of the modulating signal. Compare the demodulated output with the modulating signal.

Calculations & Observations a. Theoretical frequency of LSB = RF – AF b. Theoretical frequency of USB = RF + AF RF Generator (Carrier Waveform): Amplitude =

Time Period =

Frequency =

AF Generator (Message Waveform):

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Amplitude =

Time Period =

Subtractor Output (fc - fm)

=

Adder Output (fc + fm)

=

Frequency =

LSB Output Amplitude =

Time Period =

Frequency =

Time Period =

Frequency =

Time Period =

Frequency =

USB Output Amplitude = Demodulated Output: Amplitude =

Model Graphs

DSB-Sc output

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SSB – SC SIGNAL

Precautions 1. Avoid loose connections. 2. Avoid parallax error while taking observations. RESULT

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4. FREQUENCY MODULATION AND DEMODULATION Objective To observe the process of frequency modulation and demodulation and to calculate the modulation index (or depth of modulation).

Equipment 1. Frequency modulation and demodulation trainer kit 2. CRO with probes 3. Patch cords

Theory The process, in which the frequency of the carrier is varied in accordance with the instantaneous amplitude of the modulating signal, is called “Frequency Modulation”. The FM signal is expressed as

Where Ac is amplitude of the carrier signal, fc is the carrier frequency β is the modulation index of the FM wave. Modulator has been developed using XR-2206 integrated circuit. The IC XR-2206 is a monolithic Function generator; the output waveforms can be both amplitude and frequency modulated by an external voltage. Frequency of operation can be selected externally over a range of 0.01 MHz. The circuit is ideally suited for communications, instrumentations and function generator applications requiring sinusoidal tone, AM, FM or FSK generation. In this experiment, IC XC-2206 is connected to generate sine wave, which is used as a carrier signal. The amplitude of carrier signal is 5vPP of 100 KHz frequencies. Demodulator had been developed using LM4565 integrated circuit. The IC LM565 is a general-purpose phase locked loop containing a stable, highly linear voltage controlled oscillator for low distortion FM demodulation. The VCO free running frequency f0 is adjusted to the centre frequency of input frequency modulated signal i.e. carrier frequency which is of 100 KHz. When FM signal is connected to the demodulator input, the deviation in the input signal (FM signal) frequency which creates a DC error voltage at output of the phase comparator which is proportional to the change of frequency ζf. This error voltage pulls the VCO to the new point. This error voltage will be the demodulated version of the frequency modulated input signal.

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Circuit Diagram Frequency Modulator

Frequency Demodulator

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Procedure Modulation 1. Switch on the trainer kit. 2. Measure the output voltages of regulated power supply circuit +15 V, -15 V, +5V and -5 V. 3. Observe the output of AF generator using CRO. Set the AF signal frequency to any one of 500 Hz or 5 KHz and amplitude to 20 VP-P. 4. Observe the carrier signal (fc) at the modulator output pin, which is approximately 2.5 VPP

of 100 KHz frequency. (Keep the frequency deviation potentiometer at minimum

position). 5. Connect AF signal of 500 Hz frequency to the AF input pin of the modulator. 6. Observe the frequency modulated wave at modulator output pin by varying frequency deviation potentiometer. 7. Now set the deviation potentiometer to middle position and observe the modulated signal. Calculate the maximum frequency deviation using

S  fc  fa where

(or ) fb 

1 Tmin

S  fb  fc and

fa 

1 Tmax

8. Calculate the modulation index using the formula mf 

S fm

where f m is the mod ulating signal frequency

9. Now set the AF signal to 5 KHz and repeat steps 5, 6, 7 and 8.

Demodulation

1. Connect the modulator output to the demodulator input pin. 2. Observe demodulated signal at the output pin of demodulator and compare it with the original AF signal.

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Calculations a) Modulation index mf 

S fm

where f m is the mod ulating signal frequency

b) Frequency deviation, S

S  fc  fa where

(or ) fb 

1 Tmin

S  fb  fc and

fa 

1 Tmax

Modulating Signal Generator: Amplitude =

Time Period =

Frequency =

Time Period =

Frequency =

Time Period =

Frequency =

Carrier Signal Generator: Amplitude = Demodulated Output: Amplitude =

Observations Modulating signal frequency(fm)

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Modulating signal amplitude (V)

Tmin

Tmax

Frequency deviation (S)

Modulation index S mf  fm



Model Wave Forms

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Result

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5. PRE-EMPHASIS AND DE-EMPHASIS Objective To observe the process of Pre-emphasis and De-emphasis. To calculate the gain of Preemphasis and de-emphasis and draw the corresponding frequency response curves.

Equipment 1. Pre-emphasis and De-emphasis trainer kit. 2. CRO with probes. 3. Patch cards.

Theory Pre-emphasis: Frequencies contain in human speech mostly occupy the region from 100 to 10,000 Hz, but most of the power is contained in the region of 500 Hz for men and 800 Hz for women. Common voice characteristics emit low frequencies higher in amplitude than higher frequencies. The problem is that in FM system the noise output of the receiver increases linearly with the frequency, which means that the signal to noise ratio becomes poorer as the modulating frequency increases. Also, noise can make radio reception less readable and unpleasant. This noise is greatest in frequencies above 3KHz.The high frequency noise causes interference to the already weak high frequency voice. To reduce the effect of this noise and ensure an even power spread of audio frequencies, Pre emphasis is used at the Transmitter side. A preemphasis network in the transmitter accentuates the audio frequencies above 3 KHz, so providing the higher average deviation across the voice spectrum, thus improving the signal to noise ratio. The preemphasis is obtained by using the simple audio filter; even simple RC filter will do the job. The preemphasis circuit produces higher output at higher frequencies because the capacitive reactance is decreased as the frequency increases. De-emphasis: The problem in FM broadcasting is that noise and hiss tend to be more noticeable, especially when receiving the weaker stations. To reduce this effect, the treble response of the audio signal is artificially boosted prior to transmission. This is known as pre-emphasis

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At the receiver side a corresponding filter or “de-emphasis” circuit is required to reduce the treble response to correct level. Since most noise and hiss tend to be at the higher frequencies, the de-emphasis removes a lot of this. Pre-emphasis and de-emphasis thus allow an improved signal to noise ratio to be achieved while maintaining the frequency response of the original audio signal. The de-emphasis stage is used after the detector stage.

Circuit Diagram

Procedure Pre-emphasis 1. Connect the circuit as shown in the Circuit Diagram. 2. Select frequency range 1-10 KHz in the function generator. 3. In the function generator using frequency pot keep frequency at 200 Hz and using amplitude pot keep amplitude at 10Vpp. 4. Connect the output of function generator to the Pre-emphasis circuit input pin. 5. Observe output voltage at the Pre-emphasis circuit output pin. 6. Vary the frequency in steps of 500Hz and note down the output voltage at the output pin of Pre-emphasis circuit. 7. Plot the graph of gain v/s input frequency on graph paper. Dept. of ECE



8. From the response you can easily understand that using the Pre-emphasis Circuit we can increase the amplitude of modulating signal at higher frequencies thus improving the Signal to Noise ratio at higher frequencies.

De-emphasis: 1. Connect the circuit as shown in the Circuit Diagram. 2. Select frequency range 1-10 KHz in the function generator. 3. In the function generator using frequency pot keep frequency at 200 Hz and using amplitude pot keep amplitude at 10Vpp. 4. Connect the output of function generator to the De-emphasis circuit input pin. 5. Observe output voltage at the De-emphasis circuit output pin. 6. Vary the frequency in steps of 500Hz and note down the output voltage at the output pin of De-emphasis circuit. 7. Plot the graph of gain v/s input frequency on graph paper.

Tabular Form: Pre-emphasis

Input Frequency, f (Hz)

Output voltage (V)

 f  Gain (dB)  20 log 1     f1 

Output voltage (v)

 f  Gain (dB)  20 log 1     f1 

2

De-emphasis

Input frequency(Hz)

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2



Calculations f1 

1 2RC

Where RC is the time constant and is equal to 75 µsec f1=2122 Hz.

Model Graphs


Result

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6.TIME DIVISION MULTIPLEXING & DEMULTIPLEXING Objective To study Time Division Multiplexing and Demultiplexing, using Pulse Amplitude Modulation and Demodulation and to reconstruct the signals at the Receiver, using Filters. The Transmitter Clock and the Channel Identification Information is linked directly to the Receiver.

Equipments 1. Experiment kit DCL-02. 2. Connecting Chords. 3. Power supply. 4. 20 MHz Dual Trace Oscilloscope.

Note: KEEP ALL THE SWITCH FAULTS IN OFF POSITION. Procedure 1. Refer to the Block Diagram (Fig. 1) & Carry out the following connections and switch settings. 2. Connect power supply in proper polarity to the kit DCL-02 & switch it on. 3. Connect 250Hz, 500Hz, 1KHz, and 2KHz sine wave signals from the Function Generator to the multiplexer inputs channel CH0, CH1, CH2, CH3 by means of the connecting chords provided. 4. Connect the multiplexer output TXD of the transmitter section to the Demultiplexer input RXD of the receiver section. 5. Connect the output of the receiver section CH0, CH1, CH2, CH3 to the IN0, IN1, IN2, IN3 of the filter section. 6. Connect the sampling clock TX CLK and Channel Identification Clock TXSYNC of the transmitter section to the corresponding RX CLK and RX SYNC of the receiver section respectively. 7. Set the amplitude of the input sine wave as desired. 8. Take observations as mentioned below.

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Observations: Observe the following waveforms on oscilloscope and plot it on the paper. a. Input Channel CH0, CH1, CH2, CH3. b. Channel Selection Signal. c. TX CLK and RX CLK. d. Channel Identification Signal TX SYNC and RX SYNC. e. Multiplexer Output TXD. f. Demultiplexer Input RXD. g. Demultiplexer output CH0, CH1, CH2, and CH3. h. Reconstructed signal OUT 0, OUT 1, OUT 2, OUT 3.

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Precautions: 1. Make sure that all the switch faults are in the OFF position initially. 2. Note the readings without parallax errors. 3. Make sure that all the LEDs of the power supply glow

Result

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7.FREQUENCY DIVISON MULTIPLEXING

Objective To construct the frequency division multiplexing and demultiplexing circuit and to verify its operation.

Equipment 1. CRO 2. CRO probes 3. Function generator

4.

Regulated Power Supply

Theory When several communications channels are between the two same point’s significant economics may be realized by sending all the messages on one transmission facility a process called multiplexing.

Applications of multiplexing range from the vital, if prosaic, telephone networks to the glamour of FM stereo and space probe telemetry system. There are two basic multiplexing techniques 1. Frequency Division Multiplexing (FDM) 2. Time Division Multiplexing (TDM)

The principle of the frequency division multiplexing is that several input messages individually modulate the subcarriers fc1, fc2,etc.after passing through LPFs to limit the message bandwidth. We show the subcarrier modulation as SSB, and it often is; but any of the CW modulation techniques could be employed or a Mixture of them. The modulated signals are then summoned to produce the baseband signal with the spectrumXb9f), the designation “baseband” is used here to indicate that the final carrier modulation has not yet taken place.

The major practical problem of FDM is cross talks, the unwanted coupling of one message into another. Intelligible cross talk arises primarily because of non linearity’s in the system, which causes message signal to appear as modulation on subcarrier. Consequently, standard practice calls for negative feedback to minimize amplifier non linearity in FDM systems Dept. of ECE



Circuit Diagram

Procedure 1. Connections are given as per the Circuit Diagram. 2. The FSK signals are obtained with two different frequency pair with two different FSK generators. 3. The 2 signals are fed to op-amp which performs adder operation. 4. The filter is designed in such a way that low frequency signal is passed through the HPF. 5. Fixed signal is obtained will be equal to the one signal obtained from FSK modulator.

Observations Signals

Amplitude (V)

Time (ms)

Input 1 Input 2 Modulated Input Demodulated Output 1 Demodulated Output 2

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Result

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8. VERIFICATION OF SAMPLING THEOREM

Objective a) To study different types of signal samplings and its reconstruction. 1) Natural Sampling, 2) Sample and Hold, 3) Flat top sampling. b) To study the effect of different sampling frequencies on the reconstructed signal. c) To study the effect of 2nd Order and 4th Order Low Pass Butterworth Filters on the reconstruction of the signal.

Equipment 1. Experiment kit DCL –01. 2. Connecting Chords. 3. DC Power supply. 4. 20 MHz Dual Trace Oscilloscope.

Note: KEEP ALL THE SWITCH FAULTS (EXCEPT SWITCH 1) IN OFF POSITION. Procedure (For Objective-a) 1) Natural Sampling And Its Reconstruction. 1. Refer to the Block Diagram (Fig. 1.1) & Carry out the following connections and switch settings. 2. Connect power supply in proper polarity to the kit DCL-01 & switch it on. 3. Connect the 1KHz, 5Vpp Sine wave signal, generated onboard, to the BUF IN post of the BUFFER and BUF OUT post of the BUFFER to the IN post of the Natural Sampling block by means of the Connecting chords provided. 4. Connect the sampling frequency clock in the internal mode INT CLK using switch (SW4). 5. Using clock selector switch (S1) select 8 KHz sampling frequency. 6. Using switch SW2 select 50% duty cycle. 7. Connect the OUT post of the Natural sampling block to the input IN1 post of the 2nd Order Low Pass Butterworth Filter and take necessary observation as mentioned below. (Fig. 1.4) 8. Repeat the procedure for the 2 KHz sine wave signal as input.

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Observations: Observe the following waveforms in order for every setting and plot it on the paper. (i) Analog Input waveform. (ii). Sampling frequency waveform. (iii). Natural sampling signal and its corresponding reconstructed output of 2nd order Low Pass Butterworth Filter. 2) Sample And Hold And Its Reconstruction. 1. Refer to the Block Diagram (Fig. 1.2) & Carry out the following connections and switch settings. 2. Connect power supply in proper polarity to the kit DCL-01 & switch it on. 3. Connect the 1KHz, 5Vpp Sine wave signal, generated onboard, to the BUF IN post of the BUFFER and the BUF OUT post of the BUFFER to the IN post of the Sample and Hold Block by means of the Connecting chords provided. 4. Connect the sampling frequency clock in the internal mode INT CLK using switch (SW4). 5. Using clock selector switch SW1 select 8 KHz sampling frequency. 6. Using switch SW2 select 50% duty cycle. 7. Connect the OUT post of the Sample and Hold block to the input IN 1 post of the 2nd Order Low Pass Butterworth Filter and take necessary observation as mentioned below. (Fig. 1.5) 8. Repeat the procedure for the 2 KHz sine wave signal as input.

Observations Observe the following waveforms in order for every setting and plot it on the paper. (i)1 KHz Analog Input waveform. (ii) Sampling frequency waveform. (iii) Sample and hold signal and its corresponding reconstructed output of 2nd order Low Pass Butterworth Filter. 3) Flat Top Sampling And Its Reconstruction: 1. Refer to the Block Diagram (Fig. 1.3) & Carry out the following connections and switch settings. 2. Connect power supply in proper polarity to the kit DCL-01 & switch it on.

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3. Connect the 1 KHz, 5Vpp Sine wave signal, generated onboard to the BUF IN post of the Buffer and the BUF OUT post of the Buffer to the IN post of the Flat Top Sampling block by means of the Connecting chords provided. 4. Connect the sampling frequency clock in the internal mode INT CLK using switch (SW4). 5. Using clock selector switch S1 select 8 KHz sampling frequency. 6. Using switch SW2 select 50% duty cycle. 7. Connect the OUT post of the Flat top sampling block to the input IN 1 of the 2nd Order Low Pass Butterworth Filter and take necessary observation as mentioned below. (Fig. 1.6) 8. Repeat the procedure for the 2 KHz sine wave signal as input.

Observations Observe the following waveforms in order for every setting and plot it on the paper. (i) Analog Input waveform (ii) Sampling frequency waveform (iii) Flat Top signal and its corresponding reconstructed output of 2nd order Low Pass Butterworth Filter

Procedure: (for objective-b) 1. Refer to the Block Diagram (Fig. 2.1) & Carry out the following connections and switch settings. 2. Connect power supply in proper polarity to the kit DCL-01 & switch it on. 3. Connect the 1 KHz, 5Vpp Sine wave signal, generated onboard, to the BUF IN post of the BUFFER and the BUF OUT post of the BUFFER to the IN post of Natural sampling block by means of the Connecting chords provided. 4. Connect the sampling frequency signal in the internal mode INT CLK using Switch (SW4). 5. Using switch SW2 select 50% duty cycle. 6. Connect the Sampled Output OUT to the input of the IN 1 post of 2nd Order Low Pass Butterworth Filter. 7. Using clock selector switch S1, select desired sampling frequency. The sampling frequency selected is indicated by the LED. 8. Take observation as mentioned below for various sampling frequencies: 64 KHz, 32 KHz, 16 KHz, 8 KHz, 4 KHz, and 2 KHz. Dept. of ECE



9. Similarly repeat the procedure for sample & hold circuit and flat top sample circuit. 10. Also observe waveforms by applying onboard 2 KHz sine wave signal.

Observations Observe the following waveforms in order for every setting and plot it on the paper. (i) Analog Input waveform. (ii) Sampling frequency (2KHz) waveform. (iii)Natural Sampled Signal and corresponding reconstructed output of 2nd order Low Pass Butterworth Filter. (Fig. 2.2) (iv) Sample and Hold signal output and corresponding reconstructed output of 2nd order Low Pass Butterworth Filter. (Fig. 2.3) (v) Flat top sample signal output and corresponding reconstructed output of 2nd order Low Pass Butterworth Filter. (Fig. 2.4) (vi) Sampling frequency (8 KHz) waveform. (vii) Natural Sampled Signal and corresponding reconstructed output of 2nd order Low Pass Butterworth Filter. (Fig. 2.5) (viii) Sample and Hold signal output and corresponding reconstructed output of 2ndorder Low Pass Butterworth Filter. (Fig. 2.6) (ix) Flat top sample signal output and corresponding reconstructed output of 2nd order Low Pass Butterworth Filter. (Fig. 2.7)

Procedure: (for objective-c) 1. Refer to the Block Diagram (Fig. 4.1) & Carry out the following connections and switch settings. 2. Connect power supply in proper polarity to the kit DCL-01 & switch it on. 3. Connect the 1KHz, 5Vpp Sine wave signal, generated on board, to the BUF IN post of the BUFFER. 4. Connect the BUF OUT post to the IN post of the Natural Sampling block, by means of the Connecting chords provided. 5. Connect the sampling frequency signal in the internal mode INT CLK using switch SW4. 6. Using clock selector switch S1, select desired sampling frequency. The corresponding red LED indicates the selected sampling frequency.

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7. Connect the Sampled Output post OUT to the input post IN 1 of the 2nd Order Low Pass Butterworth Filter. 8. Using switch SW2 select 50% duty cycle. 9. Take necessary observation as mentioned below. Similarly Connect the Sampled Output post OUT to the input post IN 2 of the 4th order Low Pass Butterworth Filter and take necessary observation. 10. Similarly repeat the procedure for sample & hold circuit and flat top sample circuit. 11. Also observe waveforms by applying onboard 2KHz sine wave signal.

Observations Observe the following waveforms in order for every setting and plot it on the paper. (i) Analog Input waveform. (ii) Sampling frequency. (iii) Natural Sampled Signal and its corresponding reconstructed output of 2nd order Low Pass Butterworth Filter. (Fig. 4.2) (iv) Natural Sampled Signal and its corresponding reconstructed output of 4th order Low Pass Butterworth Filter. (Fig. 4.2) (v) Sampled and Hold Signal and its corresponding reconstructed output of 2nd order Low Pass Butterworth Filter. (Fig. 4.3) (vi) Sampled and Hold Signal and its corresponding reconstructed output of 4th order Low Pass Butterworth Filter. (Fig. 4.3) (vii) Flat Top Sampled signal and its corresponding reconstructed output of 2ndorder Low Pass Butterworth Filter. (Fig. 4.4) (viii) Flat Top Sampled signal and its corresponding reconstructed output of 4th order Low Pass Butterworth Filter. (Fig. 4.4)

Precautions 1. Make sure that all the switch faults are in the OFF position initially. 2. Note the readings without parallax errors. 3. Make sure that all the LEDs of the power supply glow

Result

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9. PULSE AMPLITUDE MODULATION & DEMODULATION Objective To conduct study Pulse Amplitude Modulation & Demodulation.

Equipments 1. Experiment kit DCL-08. 2. Connecting Chords. 3. Power supply. 4. 20 MHz Dual trace oscilloscope.

Note: KEEP THE SWITCH FAULTS IN OFF POSITION. Procedure 1. Refer to the block diagram (Fig. 1) and carry out the following connections and switch settings. 2. Connect the Power Supply with proper polarity to the kit DCL-08 and switch it on. 3. Select 16 KHz sampling frequency by jumper JP1. 4. Connect the 1 KHz, 2Vp-p sine wave signal generated onboard to PAM IN Post. 5. Observe the Pulse Amplitude Modulation output at PAM OUT Post. 6. Short the following posts with the Connecting chords provided as shown in block diagram. PAM OUT and AMP IN. AMP OUT and FIL IN. 7. Keep the amplifier gain control potentiometer P5 to maximum completely clockwise. 8. Observe the Pulse Amplitude Demodulated signal at FIL OUT, which is same as the input signal. 9. Repeat the experiment for different input signal and sampling frequencies.


Result:

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10. PULSE WIDTH MODULATION & DEMODULATION

Objective To conduct an experiment on Pulse Width Modulation and Demodulation and observe the process of width variation of the pulse in accordance with message signal variations.

Equipments 1. Experiment kit DCL-08. 2. Connecting Chords. 3. Power supply. 4. 20 MHz Dual trace oscilloscope. Note: KEEP THE SWITCH FAULTS IN OFF POSITION.

Procedure: 1. Refer to the block diagram (Fig. 2) and carry out the following connections and switch settings. 2. Connect the Power Supply with proper polarity to the kit DCL-08 and switch it on. 3. Put jumper JP3 to 2nd position. 4. Select 1KHZ 1v-pp sine wave signal generated onboard. 5. Connect this signal to PWM/PPM IN. 6. Observe the Pulse Width Modulated output at PWM OUT post. Note that since the sampling frequency is high, only blurred band in waveform will be observed due to persistence of vision. In absence of input signal only square wave of fundamental frequency and fixed on time will be observed and no width variation are present. To observe the variation in pulse width, apply 1-30Hz sine wave signal to PWM/PPM IN post. Vary the frequency from 1-30 Hz. 7. Short the following posts with the Connecting chords provided as shown in block diagram for demodulation section. PWM OUT and BUF IN. BUF OUT and PWM DMOD IN. DMOD OUT and FIL IN. 8. Observe the Pulse Width Demodulated output at FIL OUT. 9. Repeat the experiment for different input signal and different sampling clocks with the help of jumper JP3. Dept. of ECE



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Result:

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11. PULSE POSITION MODULATION & DEMODULATION Objective To conduct an experiment on Pulse Position Modulation and Demodulation and observe the process of position variation of the pulse in accordance with message signal variations.

Equipments 1. Experiment kit DCL-08. 2. Connecting Chords. 3. Power supply. 4. 20 MHz Dual trace oscilloscope. Note: KEEP THE SWITCH FAULTS IN OFF POSITION.

Procedure 1. Refer to the block diagram (Fig. 3) and carry out the following connections and switch settings. 2. Connect the Power Supply with proper polarity to the kit DCL-08 and switch it on. 3. Put jumper JP3 to 2nd position. 4. Select 1KHZ,1v-pp sine wave signal generated onboard. 5. Connect the selected signal to the PWM/PPM IN. 6. Observe the Pulse Position Modulated output at PPM OUT post with shifted position on time scale. Please note amplitude and width of pulse are same and there is shift in position which is proportional to input Analog signal. 7. To observe the variation in pulse positions, apply 1-30Hz sine wave signal to PWM/PPM IN post vary the frequency from 1-30 Hz and observe the signal on oscilloscope in dual for posts PPM OUT and PWM OUT simultaneously. 8. Then short the following posts with the link provided as shown in block diagram for Demodulation section. PPM OUT and BUFIN. BUFOUT and PPM DMOD IN. DMOD OUT and FIL IN. 9. Observe the Pulse Position Demodulated signal at FIL OUT. 10. Repeat the experiment at different input signal and different sampling frequencies.

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Result:

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12. FREQUENCY SYNTHESIZER Objective To observe the operation of frequency synthesizer using PLL.

Equipment 1. Frequency Synthesizer trainer kit. 2. CRO dual trace with probes 3. Digital frequency counter or Multimeter

Theory: Phase locked loop: PLL stands for ‘Phase locked loop’ and it is basically a closed loop frequency control system, which functioning is based on phase sensitive detection of phase difference between the input and output signals of controlled oscillator. Before the input is applied the PLL is in free running state. Once the input frequency is applied the VCO frequency starts change and phase locked loop is said to be in captured mode. The VCO frequency continues to change until it equals the input frequency and PLL is then in the phase locked state. When phase locked the loop tracks any change in the input frequency through its repetitive action. Frequency synthesizer: The frequency divider is inserted between the VCO and the phase comparator. Since the output of the divider is locked to the input frequency fin, VCO is running at multiple of the input frequency. The desired amount of multiplication can be obtained by selecting a proper divide by N network. Where N is an integer. For example fout = 5 fin a divide by N=10, 2 network is needed as shown in block diagram. This function performed by a 4 bit binary counter 7490 configured as a divide by 10, 2 circuit. In this circuit transistor Q1 used as a driver stage to increase the driving capability of LM565 as shown in above figure. To verify the operation of the circuit, we must determine the input frequency rangeand then adjust the free running frequency Fout of VCO by means of R1 (between 10th and 8th pin) and C1 (9th pin), so that the output frequency of the 7490 driver is midway within the predetermined input frequency range. The output of the VCO now should be 5Fin.

Free running frequency (f0):

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Where there is no input signal applied, it is in free running mode. f0 

0.3 Rt Ct

Where Rt is the timing resistor and Ct is the timing capacitor

Block Diagram

Circuit Diagram

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Procedure 1. Switch on the trainer kit and observe the output of the square wave generator using oscilloscope and measure the minimum and maximum frequency range. Adjust the amplitude to 4V and frequency to the (Fin) 1 KHz 2. Connect the Square wave to input of PLL and short 4th and 5th of PLL. Adjust the Output frequency is five times of input frequency by using timing resistor Rt. Measure the timing resistor Rt value by using Multimeter. Verify the Fout by using below formula. f out 

0.3 Rt Ct

Where Rt is the timing resistor and Ct is timing capacitor of value 0.01µf.

Input frequency, fin

Practical fout

Theoretical fout

3. Connect 4th pin of PLL (fout) to the driver stage and 5th pin of PLL connected to 11th pin of decade counter 7490. Output can be taken at the 11th pin of the decade counter 7490. It should be divided by 10 times of the fout For Example: fout = 5KHz, Decade counter output is 500Hz 4. Output can be taken at the 12th pin of decade counter 7490. It should be divide by 2 times of the fout For Example: fout = 5KHz, Decade counter output is 2.5KHz

fout

Divided 10 output

Divided 2 output

5. Repeat the same Procedure for the fin = 2KHz.

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Model Graph


Result

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13. AGC CHARACTERISTICS Objective To Study the Automatic gain control characteristics by using Amplitude Modulation Technique.

Equipment 1. AGC Trainer Kit 2. CRO with probes. 3. Patch cards 4. Connecting wires

Theory AGC was implemented in first radios for the reason of fading propagation (defined as slow variations in the amplitude of the received signals) which required continuing adjustments in the receiver’s gain in order to maintain a relative constant output signal. Such situation led to the design of circuits, which primary ideal function was to maintain a constant signal level at the output, regardless of the signal’s variations at the input of the system. Now AGC circuits can be found in any device or system where wide amplitude variations in the output signal could lead to a lost of information or to an unacceptable performance of the system. Automatic Gain Control (AGC) circuits are employed in many systems where the amplitude of an incoming signal can vary over a wide dynamic range. The role of the AGC circuit is to provide relatively constant output amplitude so that circuits following the AGC circuit require less dynamic range. If the signal level changes are much slower than the information rate contained in the signal, then an AGC circuit can be used to provide a signal with a well defined average level to downstream circuits. In most system applications, the time to adjust the gain in response to an input amplitude change should remain constant, independent of the input amplitude level and hence gain setting of the amplifier. The large dynamic range of signals that must be handled by most receivers requires gain adjustment to prevent overload or IM of the stages and to adjust the demodulator input level for optimum operation. A simple method of gain control would involve the use of a variable attenuator between the input and the first active stage. Such an attenuator, however, would decrease the signal level, but it would also reduce the S/N of any but the weakest acceptable signal. Gain control is generally distributed over a number of stages, so that the gain in later stages (the IF amplifiers) is

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reduced first, and the gain in earlier stages (RF and first IF) is reduced only for signal levels sufficiently high to assure a large S/N.

Block Diagram

Procedure 1. Connect the trainer to the mains and switch on the power supply. 2. Observe outputs of RF and AF signal generator using CRO, note that RF voltage is approximately 50mv p-p of 455KHZ frequency and AF voltage is 5v p-p of 1KHZ frequency. 3. Now vary the amplitude of AF signal and observe the AM wave at output, note the % of modulation for different values of AF signal.


VMax  VMin VMax  VMin

4. Now adjust the modulation index to 30 % by varying the amplitudes of RF and AF signals simultaneously. 5. Connect AM output to the input of AGC and also to the CRO channel-1. 6. Connect AGC link to the feedback network through 0A79 diode 7. Now connect CRO channel-2 at output. The detected audio signal of 1 KHz will be observed. 8. Calculate the voltage gain by measuring the amplitude of output signal (Vo) waveform, using formula A=Vo/Vi. 9. Now disconnect the AGC link vary detected. The output will be distorted when AGC link removed i.e there is no AGC action. Calculate Voltage Gain A=Vo/Vi. 10. This explains AGC effect in Radio circuit.

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Observations and Calculations Modulating Signal Generator: Amplitude =

Time Period =

Frequency =

Time Period =

Frequency =

Time Period =

Frequency =

Carrier Signal Generator: Amplitude = Demodulated Output: Amplitude =

Tabular Form Modulating signal amplitude (V)

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Vmax

Vmin

Modulation index V  Vmin ma  max Vmax  Vmin



Model Wave Forms

Detected Output with AGC

Detected Output without AGE


Result

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14. PHASE LOCKED LOOP Objective To observe the characteristics of the phase locked loop using the IC 565 and to obtain the free running frequency, Lock frequency and capture frequency.

Equipment 1. PLL using 565 trainer kit. 2. CRO with Probe 3. Connecting wires 4. Multimeter

Theory The phase locked loop detector is another demodulator that employs a phase comparator circuit. It is a very good demodulator and has an advantage that it is available as a self-contained integrated circuit, so no setting is required. You just plug it in and it works. For these reasons, it is often used in commercial broadcast receivers. It has very low of distortion.

The overall action of the circuit may, at first, seem rather pointless. As we can see in figure, there is a voltage-controlled oscillator (VCO). The DC output voltage from the output of the low pass filter controls the frequency of this oscillator. Now, this DC voltage keeps the oscillator running at the same frequency as the original input signal but 90° out of phase. The question often arises why we would want the oscillator to run at the same frequency and 90° out of phase. And if we did, then why not just add a phase shifting circuit at the input to give the 90° phase shift? The answer can be got by imagining what happens when the input frequency changes – as it would with an FM signal.

If the input frequency increases and decreases, the VCO frequency is made to follow it. To do this, the input control voltage must increase and decrease. These changes in DC voltage level form the demodulated signal. The AM signal then passes through a signal buffer to prevent any loading effect from disturbing the VCO and then through an audio amplifier if necessary. The Frequency response is highly linear.

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Circuit Diagram

Procedure Free Running Frequency 1. Connect the ckt as shown in diagram.The VCO output TP4 to TP5. 2. Switch on the power supply. 3. Observe the free running frequency at TP4 on the oscilloscope with out given the input. (Observe the fout for different values of Rt by varying the Pot P1) 4. Switch off the power supply. 5. Calculate the free running frequency fout by using the formulae Fout = 1.2 / (4RtCt) Where Rt and Ct are external resistor and capacitor and Rt=R1+P1. Make sure that the practical and theoretical values are exactly same. Lock Frequency 1. Connect the ckt as shown in the diagram 2. 2. The Square wave of amplitude 3V from the square wave generator TP1 to TP2. 3. The VCO Output TP4 to TP5. 4. Switch on the power supply.

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5. Observe the output at TP4 by varying the frequency of square wave whose range is 350 hz to 10 Khz. At some frequency of the input, the output frequency of VCO at TP4 is same as the input frequency. Note down the frequency. Let it be f1. 6. Continue varying the input frequency and its some frequency the output frequency different that of the input frequency. Let that frequency f2, this range of frequency over which output frequency is equal to the input frequency is the Lock range and is given as FL= f2 – f1 Hz. 7. Switch off the power supply. 8. Now, calculate the fl by using the formulae i. Fl = (8 fout/v) hz Capture Frequency 1. Connect the ckt as shown in the diagram 1. 2. The square wave of amplitude 3V from square generator TP1 to TP2. 3. The VCO output TP4 to TP5 4. Switch on the power supply. 5. Observe the output of TP4 by varying the input frequency . At some frequency output signal will be in the same phase as input signal. Note this frequency as fmin. 6. Continue varying the frequency of the input signal, at some frequency the output signal phase changes and this frequency be fmax. The ranges of frequencies over which the PLL acquires Phase Lock is capture frequency and is given fl=fmax – fmin. 7. Switch off the power supply.

Calculations and Observations


Result

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Analog communications MATLAB Programs Experiments

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1. Generation of Amplitude Modulation Aim To perform the generation of Amplitude Modulation using MATLAB. Apparatus Required a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Program clc; close all; clear all; fc=100000; % Carrier frequency fs=1000000; % Sampling frequency f=1000; % Tone modulation m=0.5; % Modulation index A=1/m; % Amplitude of carrier signal opt=-A; t=0:1/fs:((2/f)-(1/fs)); % Gives exact two cycles of modulating signal x=cos(2*pi*f*t); y=modulate(x,fc,fs,'amdsb-tc',opt); subplot(2,2,1); plot(x); title('modulating signal-time domain') % Modulating signal subplot(2,2,2); plot(y); title('DSB-C signal-time domain, m=0.5') % Under modulation m=1.0;

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opt=-1/m; y=modulate(x,fc,fs,'amdsb-tc',opt); subplot(2,2,3); plot(y); title('DSB-C signal-time domain, m=1.0') % Critical modulation m=1.2; opt=-1/m; y=modulate(x,fc,fs,'amdsb-tc',opt); subplot(2,2,4); plot(y); title('DSB-C signal-time domain, m=1.2') % Over modulation

OUTPUT:

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2. SPECTRUM OF AM Aim To perform the generation of Amplitude Modulation spectrum using MATLAB. Apparatus Required a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Program %Amplitude modulation ----Single Tone Modulation %Carrier Amplitude Ac=1; %Carrier frequency Fc=0.4; %baseband frequency Fm=0.05; %sampling Fs=10; %%undermodulation mu=0.5; t=0:1/Fs:200; mt=cos(2*pi*Fm*t); st=Ac*(1+mu*mt).*cos(2*pi*Fc*t); subplot(2,1,1); plot(t,st,t,Ac*(mu*mt+ones(1,length(mt))),'r'); title('\mu=0.5 undermodulation£ºAc(1+\mucos(2\pi0.05t))cos(2\pi0.4t)'); xlabel('time (s)');ylabel('amplitude'); st_fft=fft(st); st_fft=fftshift(st_fft); Dept. of ECE



st_fft_fre=Fs/2*linspace(-1,1,length(st_fft)); % st_fft=abs(st_fft(1:length(st_fft)/2+1)); % st_fft_fre=[0:length(st_fft)-1]*Fs/length(st_fft)/2; subplot(2,1,2); plot(st_fft_fre,abs(st_fft)); title('spectrum£ºside frequency amplitude=carrier frequency amplitude*\mu/2'); xlabel('Frequency (Hz)');axis([-1 1 0 1000*Ac+100]);

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3. Generation of Amplitude DSB-SC Modulation Aim To perform the generation of Amplitude DSB-SC Modulation using MATLAB. Apparatus Required a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Program clc; close all; clear all; fc=2000; % Carrier frequency fs=10000; % Sampling frequency f=200; % Single tone modulation t=0:1/fs:((2/f)-(1/fs)); x=cos(2*pi*f*t); y1 = modulate(x,fc,fs,'amdsb-sc'); f1 = abs(fft(y1, 1024)); f1 = [f1(514:1024) f1(1:513)]; f=(-511*fs/1024):(fs/1024):(512*fs/1024); subplot(1,2,1); plot(f,f1); title('DSB-SC spetrum for single tone modulation, f=200,fc=2000,fs=10000'); xlabel('FREQUENCY IN Hz'); ylabel('AMPLITUDE'); f1=200; f2=400; % Two tone modulation

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x2=cos(2*pi*f1*t)+cos(2*pi*f2*t) y2 = modulate(x2,fc,fs,'amdsb-sc'); f2 = abs(fft(y2, 1024)); f2 = [f2(514:1024) f2(1:513)]; subplot(1,2,2); plot(f,f2); title('DSB-SC spetrum for two tone modulation, f=200 & 400,fc=2000,fs=10000'); xlabel('FREQUENCY IN Hz'); ylabel('AMPLITUDE');

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DSB-SC spetrum for two tone modulation, f=200 & 400,fc=2000,fs=10000 30

AMPLITUDE

AMPLITUDE

DSB-SC spetrum for single tone modulation, f=200,fc=2000,fs=10000 30

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4. Generation of Amplitude SSB-SC Modulation Aim To perform the generation of Amplitude SSB-SC Modulation using MATLAB. Apparatus Required: a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Program: clc; close all; clear all; fc=2000; % Carrier frequency fs=10000; % Sampling frequency f=200; % Single tone modulation t=0:1/fs:((2/f)-(1/fs)); x=cos(2*pi*f*t); y1 = modulate(x,fc,fs,'amssb'); f1 = abs(fft(y1, 1024)); f1 = [f1(514:1024) f1(1:513)]; f=(-511*fs/1024):(fs/1024):(512*fs/1024); subplot(1,2,1); plot(f,f1); title('SSB spetrum for single tone modulation, f=200,fc=2000,fs=10000'); xlabel('FREQUENCY IN Hz'); ylabel('AMPLITUDE'); f1=200; f2=400; % Two tone modulation x2=cos(2*pi*f1*t)+cos(2*pi*f2*t)

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y2 = modulate(x2,fc,fs,'amssb'); f2 = abs(fft(y2, 1024)); f2 = [f2(514:1024) f2(1:513)]; f=(-511*fs/1024):(fs/1024):(512*fs/1024); subplot(1,2,2); plot(f,f2); title('SSB spetrum for two tone modulation, f=200 & 400 ,fc=2000,fs=10000'); xlabel('FREQUENCY IN Hz'); ylabel('AMPLITUDE');

SSB spetrum for single tone modulation, f=200,fc=2000,fs=10000

SSB spetrum for two tone modulation, f=200 & 400 ,fc=2000,fs=10000

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AMPLITUDE

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0 -5000 -4000 -3000 -2000 -1000 0 1000 2000 3000 4000 5000 FREQUENCY IN Hz



5. Generation of Frequency Modulation Aim To perform the generation of Frequency Modulation using MATLAB. Apparatus Required: a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Program: %The frequency modulation(FM)waveform in time and frequency domain. %fm=250HZ,fc=5KHZ,Vm=1V,Vc=1V,m=10,t=0:0.00001:0.09999 % setting vc=1; vm=1; fm=250; fc=5000; m=10; % x-axis:Time(second) t=0:0.00001:0.09999; f=0:10:99990; % y-axis:Voltage(volt) wc=2*pi*fc; wm=2*pi*fm; sc_t=vc*cos(wc*t); sm_t=vm*cos(wm*t); %kf=1000; s_fm=vc*cos((wc*t)+10*sin(wm*t)); vf=abs(fft(s_fm,10^4))/5000;

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% Plot figure in time domain figure; plot(t,s_fm); hold on; plot(t,sm_t,'r'); axis([0 0.01 -1.5 1.5]); xlabel('time(second)'),ylabel('amplitude'); title('FM time-domain'); grid on;

% Plot figure in frequency domain figure; plot(f,vf); axis([ 0 10^4 0 0.4]); xlabel('frequency'), ylabel('amplitude'); title('FM frequency-domain'); grid on;

%Plot modulating signal figure; plot(t,sm_t); axis([0 0.1 -1.5 1.5]); title('FM modulating signal');

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FM modulating signal 1.5

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0.5

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FM frequency-domain 0.4

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amplitude

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0

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6. Verification of Sampling Theorem Aim To perform and verification of sampling theorem using MATLAB. Apparatus Required a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Program clc; close all; clear all; % over sampling : fs > 2fm fm=100; fs=600; t=0:1/fs:((10/fm)-(1/fs)); x=sin(2*pi*fm*t); fx=fft(x,64); xr=ifft(fx,64); f=(-31*fs/64):(fs/64):(32*fs/64); fx=[fx(34:64) fx(1:33)]; subplot(231); stem(x); title('sampled signal , fm=100,fs=600'); subplot(232); stem(f,abs(fx)); axis([-300 300 0 30]);

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title('frequency spectrum,fm=100,fs=600'); subplot(233); stem(x); title('recovered signal,fm=100,fs=600'); %under sampling : fs < 2fm fm=400; x=sin(2*pi*fm*t); fx=fft(x,64); xr=ifft(fx,64); fx=[fx(34:64) fx(1:33)]; subplot(234); stem(x); title('sampled signal,fm=400,fs=600'); subplot(235); stem(f,abs(fx)); axis([-300 300 0 30]); title('frequency spectrum,fm=400,fs=600'); subplot(236); stem(x); title('recovered signal, fm=400,fs=600');

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sampled signal , fm=100,fs=600

frequency spectrum,fm=100,fs=600

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recovered signal,fm=100,fs=600 1

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-1

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7. Time Division Multiplexing Aim To perform the generation of Time Division Multiplexing using MATLAB. Apparatus Required: a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Program: clc; close all; clear all; % Signal generation x=0:.5:4*pi; % siganal taken upto 4pi sig1=8*sin(x); % generate 1st sinusoidal signal l=length(sig1); sig2=8*triang(l); % Generate 2nd traingular Sigal % Display of Both Signal subplot(2,3,1); plot(sig1); title('Sinusoidal Signal'); ylabel('Amplitude--->'); xlabel('Time--->'); subplot(2,3,2); plot(sig2); title('Triangular Signal'); ylabel('Amplitude--->');

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xlabel('Time--->'); % Display of Both Sampled Signal subplot(2,3,3); stem(sig1); title('Sampled Sinusoidal Signal'); ylabel('Amplitude--->'); xlabel('Time--->'); subplot(2,3,4); stem(sig2); title('Sampled Triangular Signal'); ylabel('Amplitude--->'); xlabel('Time--->'); l1=length(sig1); l2=length(sig2); for i=1:l1 sig(1,i)=sig1(i); % Making Both row vector to a matrix sig(2,i)=sig2(i); end % TDM of both quantize signal tdmsig=reshape(sig,1,2*l1); % Display of TDM Signal figure stem(tdmsig); title('TDM Signal'); ylabel('Amplitude--->'); xlabel('Time--->');

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% Demultiplexing of TDM Signal demux=reshape(tdmsig,2,l1); for i=1:l1 sig3(i)=demux(1,i); % Converting The matrix into row vectors sig4(i)=demux(2,i); end % display of demultiplexed signal figure subplot(2,1,1) plot(sig3); title('Recovered Sinusoidal Signal'); ylabel('Amplitude--->'); xlabel('Time--->'); subplot(2,1,2) plot(sig4); title('Recovered Triangular Signal'); ylabel('Amplitude--->'); xlabel('Time--->');

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Recovered Sinusoidal Signal 10

Amplitude--->

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-5

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Recovered Triangular Signal 8

Amplitude--->

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15 Time--->

Triangular Signal

Sampled Sinusoidal Signal 10

5

6

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-5

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Amplitude--->

8

Amplitude--->

Amplitude--->

Sinusoidal Signal 10

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30

Sampled Triangular Signal 8

Amplitude--->

6

4

2

0

0

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5

10

15 Time--->

20



TDM Signal 8 6

Amplitude--->

4 2 0 -2 -4 -6 -8

0

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30 Time--->

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Recovered Sinusoidal Signal 10

Amplitude--->

5 0 -5 -10

0

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10

15 20 Time---> Recovered Triangular Signal

25

30

0

5

10

25

30

Amplitude--->

8 6 4 2 0

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15 Time--->

20



8. Pulse Width Modulation & Demodulation Aim To perform Pulse Width Modulation & Demodulation using MATLAB. Apparatus Required a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Program clc; close all; clear all; fc=1000; % Carrier frequency fs=10000; % Sampling frequency f1=200; % Single tone modulation t=0:1/fs:((2/f1)-(1/fs)); x1=0.4*cos(2*pi*f1*t)+0.5; %single tone message to be [0,1] y1=modulate(x1,fc,fs,'pwm'); subplot(3,1,1); plot(x1); title(' single tone message, f1=200,fs=10000'); subplot(3,1,2); plot(y1); axis([0 500 -0.2 1.2]); title('pwm, one cycle of f1, fc=1000,f1=200'); x1_recov=demod(y1,fc,fs,'pwm'); subplot(3,1,3);

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plot(x1_recov); title('time domain recovered,single tone, f1=200');

single tone message, f1=200,fs=10000 1 0.5 0

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pwm, one cycle of f1, fc=1000,f1=200 1 0.5 0 0

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time domain recovered,single tone, f1=200 1 0.5 0

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0

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9. Pulse Position Modulation & Demodulation Aim To perform Pulse Position Modulation & Demodulation using MATLAB. Apparatus Required: a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Program: clc; close all; clear all; fc=1000; % Carrier frequency fs=10000; % Sampling frequency f1=200; % Single tone modulation t=0:1/fs:((2/f1)-(1/fs)); x1=0.4*cos(2*pi*f1*t)+0.5; %single tone message to be [0,1] y1=modulate(x1,fc,fs,'ppm'); subplot(2,2,1); plot(x1); title(' single tone message, f1=200,fs=10000'); subplot(2,2,2); plot(y1); axis([0 500 -0.2 1.2]); title('ppm, one cycle of f1, fc=1000,f1=200'); x1_recov=demod(y1,fc,fs,'ppm'); subplot(2,2,3); plot(x1_recov);

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title('time domain recovered,single tone, f1=200');

single tone message, f1=200,fs=10000 1

ppm, one cycle of f1, fc=1000,f1=200 1

0.5

0.5

0 0

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100 200 300 400 500

time domain recovered,single tone, f1=200 1

0.5

0

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0

50

100



10. Pulse Amplitude Modulation & Demodulation Aim To perform Pulse Position Modulation & Demodulation using MATLAB. Apparatus Required: a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Program: % over sampling : fs > 2fm fm=100; fs=600; t=0:1/fs:((10/fm)-(1/fs)); x=sin(2*pi*fm*t); fx=fft(x,64); xr=ifft(fx,64); f=(-31*fs/64):(fs/64):(32*fs/64); fx=[fx(34:64) fx(1:33)]; subplot(231); stem(x); title('sampled signal , fm=100,fs=600'); subplot(232); stem(f,abs(fx)); axis([-300 300 0 30]); title('frequency spectrum,fm=100,fs=600'); subplot(233); stem(x);

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title('recovered signal,fm=100,fs=600'); %under sampling : fs < 2fm fm=400; x=sin(2*pi*fm*t); fx=fft(x,64); xr=ifft(fx,64); fx=[fx(34:64) fx(1:33)]; subplot(234); stem(x); title('sampled signal,fm=400,fs=600'); subplot(235); stem(f,abs(fx)); axis([-300 300 0 30]); title('frequency spectrum,fm=400,fs=600'); subplot(236); stem(x); title('recovered signal, fm=400,fs=600');

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sampled signal , fm=100,fs=600


1

recovered signal,fm=100,fs=600

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sampled signal,fm=400,fs=600

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recovered signal, fm=400,fs=600

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-0.5 5

-1

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0

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-1

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Analog Communications MATLAB Simulink Experiments

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1. Generation of Amplitude Modulation Aim To perform the generation of Amplitude Modulation using MATLAB. Apparatus Required a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Simulink model

Procedure 1. Switch on the computer and click on the MATLAB icon. 2. Go to start at the bottom of the command window, then select “Simulink” then go to library browser and drag it into creating file. (or) Once you open the Matlab then click on the Simulink icon . Go to file and select new and then select model. You will get a new window. 3. Arrange the functional blocks as shown in Simulink model. 4. Assign required parameters to each functional block. 5. Observe the outputs on scope.

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Output 100% Modulation

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Under Modulation

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Over Modulation:

Result

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2. SPECTRUM OF AM Aim To perform the generation of Amplitude Modulation spectrum using MATLAB. Apparatus Required a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Simulink Model


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Output

Result

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3. AM-DSB-SC Modulation & Demodulation Aim To perform the generation of Amplitude DSB-SC Modulation using MATLAB. Apparatus Required a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Simulink Model


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Result

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4. AM-SSB-SC Modulation & Demodulation Aim To perform the generation of Amplitude SSB-SC Modulation using MATLAB. Apparatus Required: a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Simulink Model


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Output

Result

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5. Generation of Frequency Modulation Aim To perform the generation of Frequency Modulation using MATLAB. Apparatus Required: a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Simulink Model


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Result

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6. Verification of Sampling Theorem Aim To perform and verification of sampling theorem using MATLAB. Apparatus Required a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Simulink Model


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Output

Result

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7. Time Division Multiplexing Aim To perform the generation of Time Division Multiplexing using MATLAB. Apparatus Required: a) Hardware Tools: Computer system b) Software Tool: MATLAB 7.0 (Signal Processing Tool) Simulink Model


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Output

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Result

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Analog Communications Lab Manual_asrao

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