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This document contains a lab manual for a Communication Systems laboratory course. It outlines 12 experiments involving various modulation and demodulation techniques including amplitude modulation, frequency modulation, pulse modulation, pulse code modulation, delta modulation, digital modulation, pre-emphasis/de-emphasis circuits, phase locked loops, line coding, error control coding, sampling, and frequency division multiplexing. For each experiment, it provides the aim, apparatus required, basic theory, circuit diagram where applicable, procedures to be followed, and space to record results.
This document contains a lab manual for a Communication Systems laboratory course. It outlines 12 experiments involving various modulation and demodulation techniques including amplitude modulation, frequency modulation, pulse modulation, pulse code modulation, delta modulation, digital modulation, pre-emphasis/de-emphasis circuits, phase locked loops, line coding, error control coding, sampling, and frequency division multiplexing. For each experiment, it provides the aim, apparatus required, basic theory, circuit diagram where applicable, procedures to be followed, and space to record results.
This document contains a lab manual for a Communication Systems laboratory course. It outlines 12 experiments involving various modulation and demodulation techniques including amplitude modulation, frequency modulation, pulse modulation, pulse code modulation, delta modulation, digital modulation, pre-emphasis/de-emphasis circuits, phase locked loops, line coding, error control coding, sampling, and frequency division multiplexing. For each experiment, it provides the aim, apparatus required, basic theory, circuit diagram where applicable, procedures to be followed, and space to record results.
DEPARTMENT OF ECE LAB MANUAL ACADEMIC YEAR(2013-2014)
SUBJECT CODE/NAME : EC2307-COMMUNICATION SYSTEM LABORATORY YEAR/SEM : III/V
PREPARED BY, Mrs. M.SUDHA , AP/ECE
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EC2307-COMMUNICATION SYSTEMS LABORATORY SYLLABUS
1. Amplitude modulation and demodulation 2. Frequency Modulation and Demodulation 3. Pulse Modulation-PAM/PPM/PWM 4. Pulse Code Modulation 5. Delta Modulation, Adaptive Delta Modulation 6. Digital modulation and Demodulation-ASK,FSK,QPSK,PSK 7. Designing ,Assembling and Testing of Pre-emphasis/De-emphasis circuits 8. PLL and Frequency Synthesizer 9. Line Coding 10. Error Control Coding using MATLAB 11. Sampling and Time Division Multiplexing 12. Frequency Division Multiplexing
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INDEX
Ex.No
Date
Name of the Experiment
Page.No
Marks
Signature
1.
Amplitude Modulation and Demodulation
2.
Frequency Modulation and Demodulation
3.
Pulse Modulation-PAM/PPM/PWM
4.
Pulse Code Modulation
5.
Delta Modulation, Adaptive Delta Modulation
6(A). 6(B).
Digital modulation and Demodulation- ASK,FSK,QPSK,PSK Modulation & Demodulation using MATLAB
7.
Designing ,Assembling and Testing of Pre-emphasis/De-emphasis circuits
8.
PLL and Frequency Synthesizer
9.
Line Coding
10.
Error Control Coding using MATLAB
11.
Sampling and Time Division Multiplexing
12.
Frequency Division Multiplexing using MATLAB
4
EX.NO: 1 AMPLITUDE MODULATION AND DEMODULATION DATE: TECHNIQUES AIM: To study the function of Amplitude modulation and demodulation techniques. APPARATUS REQUIRED: 1. ST2201&2202 TRAINER KIT 2. 2mm BANNANA CABLE 3. CRO 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 accordance to the modulating signal. 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 CIRCUIT DIAGRAM: PROCEDURE: 1. Ensure that the following initial conditions exist on the ST2201 board. a. Audio oscillator's amplitude pot in full clockwise position. b. Audio input select switch in INT position. c. Mode switch in SSB position. d. Output amplifier's gain pot in full clockwise position. e. TX output select switch in ANT position. f. Audio amplifier's volume pot in full counter-clockwise position. g. Speaker switch in ON position. h. On board antenna in vertical position, and fully extended.
2. Ensure that the following initial conditions exist on the ST2202 board. a. RX input select switch in ANT position. b. R.F amplifier's tuned circuit select switch in INT position. c. R.F amplifier's gain pot in full clockwise position. d. AGC switch in out position. e. Detector switch in product position. f. Audio amplifier's volume pot in fully counter clockwise position. g. Speaker switch in 'ON' position. h. Beat frequency oscillator switch in 'ON' position. i. On - board antenna in vertical position, and fully extended.
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4. Turn on power to the modules.
5. On the ST2201 module, examine the transmitter's output signal (TP13) 6.Turn ST2201's amplitude pot (in the audio oscillator block) to its full counter clockwise (minimum amplitude) position and note that amplitude of the monitored output signal from ST2201 (at TP13) drops to zero
7. On the ST2202 module, monitor the output of the IF amplifier 2 block (TP28) and turn the tuning dial until the amplitude of the monitored signal is at its greatest
8. On the ST2202 module, monitor the output of the product detector block (at TP37), together with the output of the audio amplifier block (TP39), triggering the scope with the later signal.
9. Turn the frequency pot in ST2201's audio oscillator block, throughout its range, noting that the frequency of the tone generated by ST2202 remains close to that generated by ST2201 for all pot positions.
10. With the receiver's tuning dial adjusted for correct demodulation, the transmitted signal is obtained.
TABULATION: Signal Amplitude Time Message signal
Carrier signal
AM signal
Demodulated signal
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MODEL GRAPH:
RESULT: Amplitude modulater and demodulater are constructed and its waveforms are analysed. Percentage Modulation = Vmax- Vmin/ Vmax+Vmin=__________________.
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EX.NO: 2 FREQUENCY MODULATION AND DEMODULATION DATE: TECHNIQUES AIM: 1. To generate frequency modulated wave . 2. To Demodulate the modulated wave using envelope detector. APPARATUS REQUIRED: 1. ST 2203 Trainer Kit. 2. CRO 3. Patch chords THEORY: In an AM System, the demodulator is designed to respond to changes in amplitude of the received signal but in a FM receiver the demodulator is only watching for changes in frequency and therefore ignores any changes in amplitude. Electrical noise thus has little or no effect on a FM communication system. The bandwidth of the FM signal is very wide compared with an AM transmission. Typical broadcast bandwidths are in the order of 250 KHz. This allows a much better sound quality, so signals like music sound significantly better if frequency modulation is being used. When an FM demodulator is receiving an FM signal, it follows the variations in frequency of the incoming signal and is said to lock on to the received at the same time. The receiver 'lock on' to the stronger of the two signals and ignores the other. This is called the 'capture effect' and it means that we can listen to an FM station on a radio without interference from other stations.
PROCEDURE: 1. Ensure that the following initial conditions exist on the ST2202 board. a. All Switched Faults in Off condition. b. Amplitude potentiometer (in mixer amplifier block) in fully clockwise position. c. VCO switch (in phase locked loop detector block) in Off position. 2. Make the connections as shown in figure 13. 3. Switch On the power. 4. Turn the audio oscillator block's amplitude potentiometer to its fully clockwise position, and examine the block's output TP1 on an Oscilloscope. This is the audio frequency sine wave, which will be used as our modulating signal. Note that the sine wave's frequency can be adjusted from about 300Hz to approximately 3.4 KHz, by adjusting the audio oscillator's frequency potentiometer. 5. Connect the output socket of the audio oscillator block to the audio input socket of the modulator circuits block. 6. Set the reactance / varactor switch to the varactor position. This switch selects the varactor modulator and also disables the reactance modulator to prevent any interference between the two circuits. 7. The output signal from the varactor modulator block appears at TP24 before being buffered and amplified by the mixer/amplifier block, any capacitive loading (e.g. due to 8
Oscilloscope probe) may slightly affect the modulators output frequency. In order to avoid this problem we monitor the buffered FM output signal the mixer / amplifier block at TP34. 8. Put the varactor modulator's carrier frequency potentiometer in its midway position, and then examine TP34. Note that it is a sine wave of approximately 1.2 Vpp, centered on 0V. This is our FM carrier, and it is un-modulated since the varactor modulators audio input signal has zero amplitude. 9. The amplitude of the FM carrier (at TP34) is adjustable by means of the mixer/amplifier block's amplitude potentiometer, from zero to its potentiometer level. Try turning this potentiometer slowly anticlockwise, and note that the amplitude of the FM signal can be reduced to zero. Return the amplitude potentiometer to its fully clockwise position. 10. Try varying the carrier frequency potentiometer and observe the effects. 11. Also, see the effects of varying the amplitude and frequency potentiometer in the audio oscillator block. 12. Turn the carrier frequency potentiometer in the varactor modulator block slowly clockwise and note that in addition to the carrier frequency increasing there is a decrease in the amount of frequency deviation that is present. 13. Return the carrier frequency potentiometer to its midway position, and monitor the audio input (at TP6) and the FM output (at TP34) triggering the Oscilloscope on the audio input signal. Turn the audio oscillator's amplitude potentiometer throughout its range of adjustment. . 9
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TABULATION : Signals Amplitude Time Frequency Message signal Carrier signal FM signal Demodulated signal
MODEL GRAPH:
RESULT: Frequency modulater and demodulater are constructed and its waveforms are analysed. 11
EX.NO: 3 PULSE MODULATION-PAM/PPM/PWM DATE: AIM: Study of Pulse Amplitude, Pulse Position and Pulse Width Modulation & Demodulation Technique. APPARATUS REQUIRED: 1. ST2110 with power supply cord 2. CRO with connecting probe 3. Connecting cords THEORY: PAM: Most digital modulation systems are based on pulse modulation. It involves variation of a pulse parameter in accordance with the instantaneous value of the information signal. This parameter can be amplitude, width, repetitive frequency etc. Depending upon the nature of parameter varied, various modulation systems are used. Pulse amplitude modulation, pulse width modulation, pulse code modulation are few modulation systems cropping up from the pulse modulation technique. In pulse amplitude modulation (PAM) the amplitude of the pulses are varied in accordance with the modulating signal. In true sense, pulse amplitude modulation is analog in nature but it forms the basis of most digital communication and modulation systems. The pulse modulation systems require analog information to be sampled at predetermined intervals of time. Sampling is a process of taking the instantaneous value of the analog information at a predetermined time interval. A sampled signal consists of a train of pulses, where each pulse corresponds to the amplitude of the signal at the corresponding sampling time. The signal sent to line is modulated in amplitude and hence the name Pulse Amplitude Modulation (PAM).
PPM: The Amplitude and width of the pulses is kept constant in this system, while the position of each pulse, in relation to the position of a recurrent reference pulse is varied by each instantaneous sampled value of the modulating wave. As mentioned in connection with pulse width modulation, pulse-position modulations has the advantage of requiring constant transmitter power output, but the disadvantages of depending on transmitter receiver is synchronization.
PWM: In pulse width modulation of pulse amplitude modulation is also often called PDM (pulse duration modulation) and less often, PLM (pulse length modulation). In this system, we have fixed amplitude and starting time of each pulse, but the width of each pulse is made proportional to the amplitude of the signal at that instant.
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PROCEDURE:
FIGURE:1 PULSE AMPLITUDE MODULATION & DEMODULATION : 1. Connect the circuit as shown in Figure 1. Keep the gain pot in AC amplifier block in anti clock wise position. 2. Switch On the power supply & oscilloscope. 3. Observe the outputs at TP (3 & 5) these are natural & flat top outputs respectively. 4. Observe the difference between the two outputs. 5. Vary the amplitude potentiometer and frequency change over switch & observe the effect on the two outputs. 6. Vary the frequency of pulse, by connecting the pulse input to the 4 frequencies available i.e. 8, 16, 32, 64 kHz in Pulse output block. 7. Switch On fault No. 1, 2, 3, 4 one by one & observe their effect on Pulse Amplitude Modulation output and try to locate them. 8. Monitor the output of AC amplifier. It should be a pure sine wave similar to input. 9. Vary the amplitude of input, the amplitude of output will vary. 10. Similarly connect the sample & hold & flat top outputs to low pass filter and see the demodulated waveform at the output of AC amplifier. 11. Switch On the switched faults No. 1, 2, 3, 4, 5 & 8 one by one and see their effects on output. 12. Switch Off the power supply. 13
FIGURE 2: PULSE POSITION MODULATION 1. Connect the circuit as shown in Figure 2 and also described below for clarity. a. Input of pulse position modulation blocks to sine wave output of FG block. 2. Switch On the power supply & oscilloscope. 3. Keep the oscilloscope at 0.5mS / div, time base speed and in X-5 mode, and observe the pulse position modulated waveform at the pulse position modulation block output. 4. Vary the amplitude of sine wave and observe the pulse position modulation, keep the amplitude preset in center. Here you can best observe the pulse modulation. 5. Switch On fault No. 1, 2, & 6 one by one & observe their effects in pulse position modulation output and try to locate them.
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FIGURE 3: PULSE POSITION DEMODULATION: 1. Connect the circuit as shown in Figure 3 and also described below for clarity. a. Sine wave of 1 KHz to input of PPM block. b. Output PPM block to input of low pass filter. c. Output of low pass filter to input of AC amplifier. d. Keep the gain potentiometer in amplifier block at maximum position. 2. Switch On the power supply & oscilloscope. 3. Observe the waveform at the TP12 output of low pass filter block. 4. Then observe the demodulated output at TP14 output of AC amplifier. 5. Switch On fault No. 1, 2, 6 & 8 one by one & observes their effect on demodulated waveform & tries to locate them.
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FIGURE 4: PULSE WIDTH MODULATION: 1. Connect the circuit as shown in Figure 4 and also described below for clarity. a. 1 KHz sine wave output of function generator block to modulation input of PWM block. b. 64 KHz square wave output to pulse input of PWM block. 2. Switch On the power supply & oscilloscope. 3. Observe the output of PWM block. 4. Vary the amplitude of sine wave and see its effect on pulse output. 5. Vary the sine wave frequency by switching the frequency selector switch to 2 KHz. 6. Also, change the frequency of the pulse by connecting the pulse input to different pulse frequencies viz. 8 KHz, 16 KHz, 32 KHz and see the variations in the PWM output. 7. Switch On fault No. 1, 2, & 5 one by one & observes their effect on PWM output and tries to locate them.
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FIGURE 5: PULSE WIDTH DEMODULATION: 1. Connect the circuit as shown in Figure 8.1 and also described below for clarity. a. 1 KHz sine wave output of function generator block to modulation input of PWM block. b. 64 KHz square wave output to pulse input. c. Output of PWM to input of low pass filter. d. Output of low pass filter to input of AC Amplifier. 2. Switch On the power supply & oscilloscope. 3. Observe the output of low pass filter and AC amplifier respectively to understand the demodulation of pulse width demodulation waveform in detail. 4. Vary the amplitude and frequency of sine wave and observe its effect on the demodulated waveform. 5. Now, connect the pulse input in the pulse width modulation block to the different frequencies available on board viz. 8, 16, 32 KHz and observe their demodulated waveforms. 6. Try varying the amplitude of sine wave signal; you will observe that the output signal varies similarly. 7. Switch On fault no, 1, 2, 5 & 8 one by one at a time. Observe their effects on final output and try to locate them.
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TABULATION : S.NO TYPE OF MODULATION AMPLITUDE TIME 1. INPUT SIGNAL s(t)
2. PAM (MODULATED)
PAM (DEMODULATED) 3. PPM (MODULATED)
PPM (DEMODULATED) 4. PWM (MODULATED)
PWM (DEMODULATED)
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MODEL GRAPH:
RESULT: Thus the Pulse amplitude, Pulse Position and Pulse width Modulation and Demodulation techniques have been determined and also graphs are plotted.
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EX.NO: 4 PULSE CODE MODULATION DATE: AIM: To study the pulse code modulation & demodulation technique. APPARATUS REQUIRED: 1. Power Supply 2. IC TL5501 3. DAC0805 4. Resistors 1k,5.6k,2.4k,5.1k(2 nos),10k,510 5. Capacitors-10F,0.1F(3 Nos),1nF. 6. AFO 7. CRO THEORY: Pulse Code Modulation technique involves following steps: (a) Sampling: The analog signal is sampled according to the nyquist criteria. The nyquist criteria states that for faithful reproduction of a band limited signal, the sampling rate must be at least twice the highest frequency component present in the signal. So sampling frequency 2 fm, where fm is maximum frequency component present in the signal Practically the sampling frequency is kept slightly more than the required rate. (b) Allocation of binary codes: Each binary word defines a particular narrow range of amplitude level. The sampled value is then approximated to the nearest amplitude level. The sample is then assigned a code corresponding to the amplitude level, which is then transmitted. This process is called quantization and it is generally carried out by the A/D Converter PROCEDURE: 1. Connections are made as per the circuit diagram. 2. Give the message signal as an input to the circuit by function generator. 3. Modulated signal output of pulse code can be observed through p7 of TL5501. 4. Demodulated output of the signal can be determined from the connections. 5. Draw the graph for the observed signals.
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CIRCUIT DIAGRAM:
TABULATION : S.NO SIGNAL AMPLITUDE TIME 1. MESSEGE SIGNAL
2. PCM SIGNAL
3. DEMODULATED SIGNAL
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MODEL GRAPH:
RESULT: Thus the Pulse Code Modulation and Demodulation technique have been studied.
AIM: To study delta modulation and adaptive delta modulation techniques APPARATUS REQUIRED: 1. ST2105 trainer kit 2. CRO 3. patch chords THEORY: DELTA MODULATION: Delta modulation is a system of digital modulation developed after pulse code modulation. In this system, at each sampling time, say the Kth sampling time, the difference between the sample value at sampling time K and the sample value at the previous sampling time (K-1) is encoded into just a single bit. i.e. at each sampling time we ask simple question.
ADAPTIVE DELTA MODULATION: Delta modulation system is unable to chase the rapidly changing information of the analog signal, which gives rise to distortion & hence poor quality reception. This is known as slope overloading phenomenon. The problem can be overcome by increasing the integrator gain (i.e. step-size). But using high step-size integrator would lead to a high quantization noise.
PROCEDURE: 1. Connect the mains supply to the Trainer 2. Make connection on the board as shown in the figure 1 3. Ensure that the clock frequency selector block switches A & B are in A = 0 and B = 0 position. 4. Ensure that integrator 1 block's switches are in following position: a) Gain control switch in left-hand position (towards switch A & B). b) Switches A & B in A=0 and B=0 positions. 5. Ensure that the switches in integrator 2 blocks are in following position: a) Gain control switch in right-hand position (towards switch A & B) b) Switches A & B are in A = 0 and B = 0 positions. 6. Switch 'ON' the trainer. 7. To ensure the correct operation, the I/P of Comparators (+) terminal is connect to DC source of OV & (-) terminal is connector to Integrator 1 O/P. O/P of Comparator is fed to the I/P of bistable CKT Transmission clock is connected to clock generator . 8. Connect Unipolar bipolar Connector O/P to the integrator.Insure the O/P at integrator 1 & it should be ensure that O/P of integrator is Triangular waveform & if it is not triangular then set the level control observe bistable also the O/P of Comparator & CKT on CRO & it should be square wave. The output from the transmitter's bistable circuit (TP14) will now be a stream of alternate '1' and '0', s' this is also the output of the delta modulator itself. The delta modulator is now said to be 'balanced' for correct operation. 23
9. Examine the signal at the output of integrator 2 (TP47) at the receiver. This should be a triangle wave, with step size equal to that of integrator 1, and ideally centre around 0 Volts. If there is any DC bias at the output of integrator 2, remove it by adjusting the receiver's level adjust preset (in the bistable & level changer circuit 2 block). This preset adjusts the relative amplitudes if the positive and negative output levels from the receiver's level changer circuit only when these levels are balanced will there be no offset at the output of integrator2. 10. Display the data of the transmitter's bistable (at TP14), together with the analog input at TP9 (again trigger on this signal), and note that the 250 Hz sine wave has effectively been encoded into a stream of data bits at the bistable's output, ready for transmission to the receiver. 11. For a full understanding of how the delta modulator is working, examine the output of the voltage comparator (TP11), the bistable's clock input (TP13), and the level changer's bipolar output (TP15) 12. Display the output of integrator 1 (TP17) and that of integrator 2 (TP47) on the scope. Note that the two signals are very similar in appearance, showing that the demodulator is working as expected. 13. Display the output of integrator 2 (TP47) together with the output of the receiver's low pass filter block (TP51). 14. The current system clock frequency is 32 KHz. This is set by the A, B switches in the clock frequency selector block, which are currently in the A= 0, B= 0 positions. While monitoring the same signals, increase the system clock frequency to 64 KHz, by putting the switches in the A = 0, B = 1 positions. 15. By changing the system clock frequency to first 128 KHz (clock frequency selector switches in A=l, B=0 positions), and then to 256 KHz (switches in A=l,B=1 positions), note the improvement in the low - pass filter's output signal (TP51). Once again, it may be necessary to adjust slightly the transmitter's level adjust preset, in order to obtain a stable oscilloscope trace. 16. Using a system clock frequency of 256 KHz (which gives a step size of approximately 60mV), compare the low pass filter's output. (TP51) with the original analog input (TP9). There should now be no noticeable difference between them, other than a slight delay. 17. While continuing to monitor the transmitter's analog input (TP9) and the receiver's low- pass filter output (TP51), disconnect the comparator's + input from the 250Hz sine wave output, and connect it the 500Hz, 1 KHz and 2 KHz outputs in turn. Note that, as the frequency of the analog signal increases, so the low pass filter's output becomes more distorted and reduced in amplitude.
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1. Connect the mains supply. 2. Connect the board as per figure 2 3. Ensure that the clock frequency selector switches A & B are in A=0 & B=0 position. 4. Ensure that the switches in TX. Integrator gain control block are in following positions. a) Gain control switch at the L.H.S. position. (towards switches A & B) b) Switches A & B in position A=0 & B=0. 5. Ensure that the switches in receiver's integrator gain control block are in following positions: a) Gain control switches at the R.H.S. position. (towards switches A & B) b) Switches A & B in Position A=0 & B=0. 6. Turn all the potentiometers of function, generator block namely 250Hz to 2 KHz to their fully clockwise positions. 7. Turn ON the supply. 8. As the gain control switch is towards A & B switches the gain setting is still manual, connect the voltage comparator's +ve input to 0V & check whether the modulator & demodulator are balanced for correct operation as in delta modulation experimentation. Change the clock frequency selector switches to the A=1, B=1, positions (256 KHz Clock Frequency) before continuing. 9. Disconnect the voltage comparators '+' input from 0V and reconnect it to the 2 KHz output from the function generator block. 10. Monitor the 2 KHz analog input at TP9 and the output of integrator 1 at TP17. 11. At the transmitter, move the slider of the gain control switch in the integrator 1 block to the right-hand position (towards the sockets labeled A, B). At the receiver, move the slider of the gain control switch in the integrator 2 blocks to the left-hand position (again towards the sockets labeled A, B). The gain of each integrator is now controlled by the outputs of the counter connected to it. 12. Once again examine the 2 KHz analog input at TP9 and the output of integrator 1 at TP17, noting that the" slope overloading problem has been eliminated, and that the integrator's output once again follows the analog input signal. Again, it may be necessary to adjust slightly the transmitter's level adjust preset, in order to obtain a stable trace of the integrator's output signal. 13. Compare the output of integrator 1 (TP17) with that of integrator 2 (TP47); noting that, as expected, both are identical in appearance. 14. Examine the output of the low pass filter (TP51) and the output of integrator 2 (TP47). The filter has removed the high-frequency components from the integrator's output signal, to leave goods, clean 2 KHz sine wave. 15. Compare the original 2 KHz analog input signal (at TP9) with the output signal from the receiver's low pass filter at TP47).. 16. Disconnect the voltage comparators '+' input from the 2 KHz function generator output, and reconnected it in turn to the 1 KHz, 500Hz and 250Hz outputs, noting in each case that the demodulators output signal is identical to the modulator's input signal, but delayed in time. 17. Examine also the test points in the adaptive control circuit 1 block (TP20-24), to ensure you have a complete understanding of how the adaptive delta modulator is operating.
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TABULATION : Amplitude time Frequency Input signal Integrator signal Modulated signal Demodulated signal
MODEL GRAPH:
RESULT: Thus the delta modulation and demodulation and adaptive delta modulation and demodulation is obtained and its corresponding graphs are drawn.
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EX.NO: 6(a) DIGITAL MODULATION AND DEMODULATION DATE: TECHNIQUES AIM: To study the function of ASK,PSK,FSK and QPSK modulation and demodulation
APPARATAUS REQUIRED: 1. ST2156 and ST2157 Trainers. 2. 2 mm Banana cable 3. Oscilloscope & Probes THEORY: In digital modulation, an analog carrier signal is modulated by a digital bit stream. Digital modulation methods can be considered as digital -to-analog conversion, and the corresponding demodulation or detection as analog -to-digital conversion. To be able to transmit the data over long distance, we have to modulate the signal that is varying phase, frequency or amplitude according to the digital data. At the receiver separate the signal and the digital information by the process of demodulation. A modulating carrier with a data stream is to change the amplitude of the carrier wave every time the data changes. This modulation technique is known as Amplitude Shift Keying.
PROCEDURE: 1.Connect the power supplies of ST2156 and ST2157 but do not turn on the power supplies until connections are made for this experiment. 2. Make the connections as shown in the Figure 1. 3. Switch 'ON' the power. 4. On ST2156, connect oscilloscope CH1 to Clock In and CH2 to Data In and observe the waveforms. 5. On ST2156, connect oscilloscope CH1 to NRZ (L) and CH2 to Output of modulator Circui t (l ) on ST2156 and observe the waveforms. 6. On ST2156, connect oscilloscope CH1 to NRZ (L) and CH2 to Output of comparator on ST2157 and observe the waveforms.
MODEL WAVEFORM:
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FREQUENCY SHIFT KEYING (FSK): 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. The time domain of an FSK modulated carrier is illustrated in the figures to the right. CIRCUIT DIAGRAM:
FIGURE 2: FREQUENCY SHIFT KEYING MODULATION & DEMODULATION
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PROCEDURE 1.Connect the power supplies of ST2156 and ST2157 but do not turn on the power supplies until connections are made for this experiment. 2. Make the connections as shown in the Figure 2. 3. Switch 'ON' the power. 4. On ST2156, connect oscilloscope CH1 to Clock In and CH2 to Data In and observe the waveforms. 5. On ST2156, connect oscilloscope CH1 to NRZ (L) and CH2 to Output of Summing Amplifier on ST2156 and observe the waveforms. 6.Adjust the amplitude of FSK waveform at Summing Amplifiers output on ST2156. 7. On ST2156, connect oscillscope CH1 to NRZ (L) and CH2 to Output of comparator on ST2157 and observe the waveforms. WAVEFORM:
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PHASE SHIFT KEYING (PSK) THEORY PSK is a digital modulation scheme which is analogues to phase modulation. In binary phase shift keying two output phases are possible for a single carrier frequency one out of phase represent logic 1 and logic 0. As the input digital binary signal change state the phase of output carrier shift two angles that are 180 o out of phase. In a PSK modulator the carrier input signal is multiplied by the digital data. The input carrier is multiplied by either a positives or negatives consequently the output signal is either +1sinwt or 1sinwt. The first represent a signal that is phase with the reference oscillator the latter a signal that is 180 out of phase with the reference oscillator. Each time a change in input logic condition will change the output phase consequently for PSK the output rate of change equal to the input rate range and widest output bandwidth occurs when the input binary data are alternating 1/0 sequence. The fundamental frequency of an alternate 1/0 bit sequence is equal to one half of the bit rate.
FIGURE :3 PHASE SHIFT KEYING MODULATION & DEMODULATION PROCEDURE: 1. Connect the power supplies of ST2156 and ST2157 but do not turn on the power supplies until connections are made for this experiment. 2. Make the connections as shown in the figure 3. 3. Switch 'ON' the power. 4. On ST215, connect oscilloscope CH1 to Clock In and CH2 to Data In and observe the waveforms. 33
5. On ST2156, connect oscilloscope CH1 to NRZ (L) and CH2 to Output of Modulator Circuit (l ) on ST2156 and observe the waveforms. 6. Adjust the Gain potentiometer of the Modulator Circuit (l ) on ST2156 to adjust the amplitude of PSK waveform at output of Modulator Circui t (l ) on ST2156. 7. Now on ST2157 connect oscilloscope CH1 to Input of PSK demodulator and connect CH2 one by one to output of double squaring circuit, output of PLL, output of Divide by four( 2) observe the wave forms. 8. On ST2157 connect oscilloscope CH1 to output of Phase adjust and CH2 to output of PSK demodulator and observe the waveforms. 9. Now connect oscilloscope CH1 to PSK output of PSK demodulator on ST2157 and connect CH2 Output of Low Pass Filter on ST2157 and observe the waveforms.
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QUADRATURE PHASE SHIFT KEYING (QPSK) THEORY: QPSK is another form of angle-modulated, constant-amplitude digital modulation. It is an M-ary encoding technique where M=4. with QPSK four output phases are possible for a single carrier frequency. Two bits (a dibit) are clocked into the bit splitter. After both bits have been serially inputted, they are simultaneously parallel outputted. One bit is directed to the I channel and the other to the Q channel. The I bit modulates a carrier that is in phase with the reference oscillator and the Q bit modulates a carrier that is 90 0 out of phase with the reference carrier. QPSK modulator is two BPSK modulators combined in parallel. The input QPSK signal is given to the I and Q product detectors and the carrier recovery circuit. The carrier recovery circuit produces the original transmit carrier oscillator signal. The recovered carrier must be frequency and phase coherent with the transmit reference carrier. The QPSK signal is demodulated in the I and Q product detectors, which generate the original I and Q data bits. The output of the product detectors are fed to the bit combining circuit, where they are converted from parallel I and Q data channels to a single binary output data stream.
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FIGURE :4 QPSK MODULATION & DEMODULATION PROCEDURE: 1.Connect the power supplies of ST2156 and ST2157 but do not turn on the power supplies until connections are made for this experiment. 2. Make the connections as shown in the Figure 4. 3. Switch 'ON' the power. 4. On ST2156, connect oscilloscope CH1 to Clock In and CH2 to Data In and observe the waveforms. 5. On ST2156, connect oscilloscope CH1 to Clock Output and CH2 one by one to Sine and Cosine output of 960 KHz and observe the waveforms. 6. On ST2156, connect oscilloscope CH1 to Data In and connect CH2 one by one to I Data and Q Data outputs and observe the waveforms. 7. Now connect oscilloscope CH1 to I Data output on ST2156 and connect CH2 one by one to Signal In, Carrier In and Output of modulator circuit (l ) on ST2156 and observe the waveforms. 8. Now connect oscilloscope CH1 to Q Data output on ST2156 and connect CH2 one by one to Signal In, Carrier In and Output of modulator circuit (ll) on ST2156 and observe the waveforms 9. Now connect oscilloscope CH1 to Data Out on ST2156 and CH2 to Output of Summing Amplifier on ST2156 and observe the waveforms. 10. Set Carrier frequency selection switch to 960 KHz on ST2157. 36
11. Now on ST2157 connect oscilloscope CH1 to Input of QPSK demodulator and connect CH2 one by one to output of double squaring circuit, output of PLL, output of Divide by four( 4) observe the wave forms. 12. On ST2157, connect oscilloscope CH1 to I output of QPSK demodulator and CH2 to Q output of QPSK demodulator and observe the waveforms. Set all toggle switch to 0, now vary the phase adjust potentiometer and observe its effects on the demodulated signal waveforms. 13. Connect oscilloscope CH1 to I output of QPSK demodulator on ST2157 then connect CH2 one by one to output of low pass filter, output of Comparator on ST2157 and observe the waveforms.
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Tabulation:
MODULATION
TECHNIQUES CLOCK SIGNAL DATA SIGNAL MODULATED OUTPUT DEMODULATED OUTPUT Amplitude (V) Time (sec) Amplitude (V) Time (sec) Amplitude (V) Time (sec) Amplitude (V) Time (sec)
ASK
FSK
PSK
QPSK
38
RESULT: Thus the ASK, FSK, PSK and QPSK modulation and demodulation process is obtained and its corresponding output is plotted.
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EX.NO: 6(B) DIGITAL MODULATION AND DEMODULATION TECHNIQUES USING DATE : MATLAB
AIM: To study the function of ASK,PSK,FSK and QPSK modulation and demodulation using MATLAB. SOFTWARE REQUIRED: 1. MATLAB 7 PROCEDURE: ASK USING MATLAB ALGORITHM : Initialization commands ASK modulation 1. Generate carrier signal. 2. Start FOR loop 3. Generate binary data, message signal(on-off form) 4. Generate ASK modulated signal. 5. Plot message signal and ASK modulated signal. 6. End FOR loop. 7. Plot the binary data and carrier. ASK demodulation 1. Start FOR loop 2. Perform correlation of ASK signal with carrier to get decision variable 3. Make decision to get demodulated binary data. If x>0, choose 1 else choose 0 4. Plot the demodulated binary data. PROGRAM %ASK Modulation clc; clear all; close all; %GENERATE CARRIER SIGNAL Tb=1; fc=10; t=0:Tb/100:1; 40
c=sqrt(2/Tb)*sin(2*pi*fc*t); %generate message signal N=8; m=rand(1,N); t1=0;t2=Tb for i=1:N t=[t1:.01:t2] if m(i)>0.5 m(i)=1; m_s=ones(1,length(t)); else m(i)=0; m_s=zeros(1,length(t)); end message(i,:)=m_s; %product of carrier and message ask_sig(i,:)=c.*m_s; t1=t1+(Tb+.01); t2=t2+(Tb+.01); %plot the message and ASK signal subplot(5,1,2);axis([0 N -2 2]);plot(t,message(i,:),'r'); title('message signal');xlabel('t--->');ylabel('m(t)');grid on hold on subplot(5,1,4);plot(t,ask_sig(i,:)); title('ASK signal');xlabel('t--->');ylabel('s(t)');grid on hold on end hold off %Plot the carrier signal and input binary data subplot(5,1,3);plot(t,c); title('carrier signal');xlabel('t--->');ylabel('c(t)');grid on subplot(5,1,1);stem(m); title('binary data bits');xlabel('n--->');ylabel('b(n)');grid on
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% ASK Demodulation t1=0;t2=Tb for i=1:N t=[t1:Tb/100:t2] %correlator x=sum(c.*ask_sig(i,:)); %decision device if x>0 demod(i)=1; else demod(i)=0; end t1=t1+(Tb+.01); t2=t2+(Tb+.01); end %plot demodulated binary data bits subplot(5,1,5);stem(demod); title('ASK demodulated signal'); xlabel('n--->');ylabel('b(n)');grid on
PSK USING MATLAB ALGORITHM Initialization commands PSK modulation 1. Generate carrier signal. 2. Start FOR loop 3. Generate binary data, message signal in polar form 4. Generate PSK modulated signal. 5. Plot message signal and PSK modulated signal. 6. End FOR loop. 7. Plot the binary data and carrier. PSK demodulation 1. Start FOR loop Perform correlation of PSK signal with carrier to get decision variable 42
2. Make decision to get demodulated binary data. If x>0, choose 1 else choose 0 3. Plot the demodulated binary data. PROGRAM % PSK modulation clc; clear all; close all; %GENERATE CARRIER SIGNAL Tb=1; t=0:Tb/100:Tb; fc=2; c=sqrt(2/Tb)*sin(2*pi*fc*t); %generate message signal N=8; m=rand(1,N); t1=0;t2=Tb for i=1:N t=[t1:.01:t2] if m(i)>0.5 m(i)=1; m_s=ones(1,length(t)); else m(i)=0; m_s=-1*ones(1,length(t)); end message(i,:)=m_s; %product of carrier and message signal bpsk_sig(i,:)=c.*m_s; %Plot the message and BPSK modulated signal subplot(5,1,2);axis([0 N -2 2]);plot(t,message(i,:),'r'); title('message signal(POLAR form)');xlabel('t--->');ylabel('m(t)'); grid on; hold on; subplot(5,1,4);plot(t,bpsk_sig(i,:)); title('BPSK signal');xlabel('t--->');ylabel('s(t)'); 43
grid on; hold on; t1=t1+1.01; t2=t2+1.01; end hold off %plot the input binary data and carrier signal subplot(5,1,1);stem(m); title('binary data bits');xlabel('n--->');ylabel('b(n)'); grid on; subplot(5,1,3);plot(t,c); title('carrier signal');xlabel('t--->');ylabel('c(t)'); grid on; % PSK Demodulation t1=0;t2=Tb for i=1:N t=[t1:.01:t2] %correlator x=sum(c.*bpsk_sig(i,:)); %decision device if x>0 demod(i)=1; else demod(i)=0; end t1=t1+1.01; t2=t2+1.01; end %plot the demodulated data bits subplot(5,1,5);stem(demod); title('demodulated data');xlabel('n--->');ylabel('b(n)'); grid on FSK USING MATLAB ALGORITHM Initialization commands FSK modulation 44
1. Generate two carriers signal. 2. Start FOR loop 3. Generate binary data, message signal and inverted message signal 4. Multiply carrier 1 with message signal and carrier 2 with inverted message signal 5. Perform addition to get the FSK modulated signal 6. Plot message signal and FSK modulated signal. 7. End FOR loop. 8. Plot the binary data and carriers. FSK demodulation 1. Start FOR loop 2. Perform correlation of FSK modulated signal with carrier 1 and carrier 2 to get two decision variables x1 and x2. 3. Make decisionon x = x1-x2 to get demodulated binary data. If x>0, choose 1 else choose 0. 4. Plot the demodulated binary data. PROGRAM % FSK Modulation clc; clear all; close all; %GENERATE CARRIER SIGNAL Tb=1; fc1=2;fc2=5; t=0:(Tb/100):Tb; c1=sqrt(2/Tb)*sin(2*pi*fc1*t); c2=sqrt(2/Tb)*sin(2*pi*fc2*t); %generate message signal N=8; m=rand(1,N); t1=0;t2=Tb for i=1:N t=[t1:(Tb/100):t2] if m(i)>0.5 m(i)=1; 45
m_s=ones(1,length(t)); invm_s=zeros(1,length(t)); else m(i)=0; m_s=zeros(1,length(t)); invm_s=ones(1,length(t)); end message(i,:)=m_s; %Multiplier fsk_sig1(i,:)=c1.*m_s; fsk_sig2(i,:)=c2.*invm_s; fsk=fsk_sig1+fsk_sig2; %plotting the message signal and the modulated signal subplot(3,2,2);axis([0 N -2 2]);plot(t,message(i,:),'r'); title('message signal');xlabel('t---->');ylabel('m(t)');grid on;hold on; subplot(3,2,5);plot(t,fsk(i,:)); title('FSK signal');xlabel('t---->');ylabel('s(t)');grid on;hold on; t1=t1+(Tb+.01); t2=t2+(Tb+.01); end hold off %Plotting binary data bits and carrier signal subplot(3,2,1);stem(m); title('binary data');xlabel('n---->'); ylabel('b(n)');grid on; subplot(3,2,3);plot(t,c1); title('carrier signal-1');xlabel('t---->');ylabel('c1(t)');grid on; subplot(3,2,4);plot(t,c2); title('carrier signal-2');xlabel('t---->');ylabel('c2(t)');grid on; % FSK Demodulation t1=0;t2=Tb for i=1:N t=[t1:(Tb/100):t2] %correlator x1=sum(c1.*fsk_sig1(i,:)); x2=sum(c2.*fsk_sig2(i,:)); 46
x=x1-x2; %decision device if x>0 demod(i)=1; else demod(i)=0; end t1=t1+(Tb+.01); t2=t2+(Tb+.01); end %Plotting the demodulated data bits subplot(3,2,6);stem(demod); title(' demodulated data');xlabel('n---->');ylabel('b(n)'); grid on;
QPSK USING MATLAB ALGORITHM Initialization commands QPSK modulation 1. Generate quadrature carriers. 2. Start FOR loop 3. Generate binary data, message signal(bipolar form) 4. Multiply carrier 1 with odd bits of message signal and carrier 2 with even bits of message signal 5. Perform addition of odd and even modulated signals to get the QPSK modulated signal 6. Plot QPSK modulated signal. 7. End FOR loop. 8. Plot the binary data and carriers. QPSK demodulation 1. Start FOR loop 2. Perform correlation of QPSK modulated signal with quadrature carriers to get two decision variables x1 and x2. 3. Make decision on x1 and x2 and multiplex to get demodulated binary data. If x1>0and x2>0, choose 11. If x1>0and x2<0, choose 10. If x1<0and x2>0, choose 01. If 47
x1<0and x2<0, choose 00. 4. End FOR loop 5. Plot demodulated data PROGRAM % QPSK Modulation clc; clear all; close all; %GENERATE QUADRATURE CARRIER SIGNAL Tb=1;t=0:(Tb/100):Tb;fc=1; c1=sqrt(2/Tb)*cos(2*pi*fc*t); c2=sqrt(2/Tb)*sin(2*pi*fc*t); %generate message signal N=8;m=rand(1,N); t1=0;t2=Tb for i=1:2:(N-1) t=[t1:(Tb/100):t2] if m(i)>0.5 m(i)=1; m_s=ones(1,length(t)); else m(i)=0; m_s=-1*ones(1,length(t)); end %odd bits modulated signal odd_sig(i,:)=c1.*m_s; if m(i+1)>0.5 m(i+1)=1; m_s=ones(1,length(t)); else m(i+1)=0; m_s=-1*ones(1,length(t)); end %even bits modulated signal 48
even_sig(i,:)=c2.*m_s; %qpsk signal qpsk=odd_sig+even_sig; %Plot the QPSK modulated signal subplot(3,2,4);plot(t,qpsk(i,:)); title('QPSK signal');xlabel('t---->');ylabel('s(t)');grid on; hold on; t1=t1+(Tb+.01); t2=t2+(Tb+.01); end hold off %Plot the binary data bits and carrier signal subplot(3,2,1);stem(m); title('binary data bits');xlabel('n---->');ylabel('b(n)');grid on; subplot(3,2,2);plot(t,c1); title('carrier signal-1');xlabel('t---->');ylabel('c1(t)');grid on; subplot(3,2,3);plot(t,c2); title('carrier signal-2');xlabel('t---->');ylabel('c2(t)');grid on; % QPSK Demodulation t1=0;t2=Tb for i=1:N-1 t=[t1:(Tb/100):t2] %correlator x1=sum(c1.*qpsk(i,:)); x2=sum(c2.*qpsk(i,:)); %decision device if (x1>0&&x2>0) demod(i)=1; demod(i+1)=1; elseif (x1>0&&x2<0) demod(i)=1; demod(i+1)=0; elseif (x1<0&&x2<0) demod(i)=0; demod(i+1)=0; elseif (x1<0&&x2>0) 49
demod(i)=0; demod(i+1)=1; end t1=t1+(Tb+.01); t2=t2+(Tb+.01); end subplot(3,2,5);stem(demod); title('qpsk demodulated bits');xlabel('n---->');ylabel('b(n)');grid on;
MODEL GRAPHS: ASK
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PSK:
FSK:
51
QPSK:
RESULT: The program for ASK, FSK, PSK and QPSK modulation and demodulation has been simulated in MATLAB and necessary graphs are plotted. 52
EX.NO: 7 DESIGNING, ASSEMBLING AND TESTING OF PRE-EMPHASIS/ DATE : DE-EMPHASIS CIRCUITS
AIM i) To observe the effects of pre-emphasis on given input signal. ii) To observe the effects of De-emphasis on given input signal.
APPARATUS REQUIRED NAME OF THE COMPONENT/EQUIPMENT SPECIFICATIONS/RANGE QUANTITY Transistor (BC 107) f T = 300 MHz P = 1W Ic(max) = 100 mA 1 Resistors 10 K, 7.5 K, 6.8 K
1 each Capacitors
10 nF 0.1 F
1 2 CRO
20MHZ 1 Function Generator 1MHZ 1
1 Regulated Power Supply 0-30V, 1A 1
THEORY In telecommunication, a pre-emphasis circuit is inserted in a system in order to increase the magnitude of one range of frequencies with respect to another. Pre-emphasis is usually employed in FM or phase modulation transmitters to equalize the modulating signal drive power in terms of deviation ratio. In high speed digital transmission, pre-emphasis is used to improve signal quality at the output of a data transmission. In transmitting signals at high data rates, the transmission medium may introduce distortions, so pre-emphasis is used to distort the transmitted signal to correct for this distortion. When done properly this produces a received signal which more closely resembles the original or desired signal, allowing the use of higher frequencies or producing fewer bit errors. In telecommunication, de-emphasis is the complement of pre-emphasis. It is designed to decrease, (within a band of 53
frequencies), the magnitude of some (usually higher) frequencies with respect to the magnitude of other (usually lower) frequencies in order to improve the overall signal-to-noise ratio by minimizing the adverse effects of such phenomena as attenuation differences. PROCEDURE
1. Connect the circuit as per circuit diagram as shown in Fig.1. 2. Apply the sinusoidal signal of amplitude 20mV as input signal to pre emphasis circuit. 3. Then by increasing the input signal frequency from 500Hz to 20KHz, observe the output voltage (vo) and calculate gain (20 log (vo/v). 4. Plot the graph between gain Vs frequency. 5. Repeat above steps 2 to 4 for de-emphasis circuit (shown in Fig.2). by applying the sinusoidal signal of 5V as input signal
CIRCUIT DIAGRAM Pre-emphasis circuit:
De-emphasis circuit
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TABLUATION Pre-emphasis V i =20mV Frequency(KHz) Vo(mV) Gain in dB(20 log Vo/Vi)
De-emphasis V i =5mV Frequency(KHz) Vo(mV) Gain in dB(20 log Vo/Vi)
MODEL GRAPH:
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RESULT:
The characteristics of Pre emphasis and De-emphasis circuits were studied
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EX.NO: 8 LINE CODING TECHNIQUES DATE : AIM: To study the different line coding techniques. APPARATUS REQUIRED: 1. ST2156 &ST2157 trainer kit. 2. CRO 3. 2mm banana cable. THEORY: Line coding consists of representing the digital signal to be transported, by an amplitude- and time-discrete signal that is optimally tuned for the specific properties of the physical channel (and of the receiving equipment). The waveform pattern of voltage or current used to represent the 1s and 0s of a digital signal on a transmission link is called line encoding. The common types of line encoding are unipolar, polar, bipolar and Manchester encoding. Line codes are used commonly in computer communication networks over short distances. Each of the various line formats has a particular advantage and disadvantage. It is not possible to select one, which will meet all needs. The format may be selected to meet one or more of the following criteria: Minimize transmission hardware Facilitate synchronization Ease error detection and correction Minimize spectral content Eliminate a dc component The Manchester code is quite popular. It is known as a self-clocking code because there is always a transition during the bit interval. Consequently, long strings of zeros or ones do not cause clocking problems.
PROCEDURE: Non return to zero- level (NRZ-L): Representation : +5V for data bit 1 and 0V for data bit 0. Bandwidth : Low bandwidth. DC Level : High DC component. Timing Information : No timing information (For long stream of 1s and 0s)
Waveforms of NRZ-L 57
Non return to zero- level (NRZ-M): Representation : Level transition for bit 1 and unchanged level for bit 0. Bandwidth : Low bandwidth. DC Level : High DC component. Timing Information : No timing information (For long stream of 0s)
Waveforms of NRZ-M Return to zero (RZ): Representation :0V for bit 0 and for bit 1, for half bit duration +5V and the rest of the bit duration is represented as 0V. Bandwidth : Twice as that required for the NRZ. DC Level : High DC component. Timing Information : No timing information (For long stream of 0s)
Waveforms of RZ-L Biphase (Manchester): Representation : For bit 1, +5V for first half bit time and 0V during the second half and for bit 0, 0V for first half bit time and +5V during the second half. Bandwidth : Twice as that required for the NRZ. DC Level : No DC component. Timing Information : Good clock recovery.
Waveforms of Manchester Biphase (Mark): 58
Representation : For any bit either 1 or 0, first half bit duration +5V or 0V and invert of first half during next half bit duration. Bit 0 Bit Pattern remains the same. Bit 1 Phase Reversal. Bandwidth : Twice as that required for the NRZ. DC Level : No DC component. Timing Information : Good clock recovery.
Waveforms of Mark Return to Bias (RB): Representation : During the first half a period, positive level for bit 1 and a negative level for bit 0 and during the second half bit time, both returns to the bias level. Bandwidth : Twice as that required for the NRZ. DC Level : The DC component depends on the string of 1s and 0s. Timing Information : Good clock recovery (Self clocking system).
Waveforms of RB Alternate Mark Inversion (AMI): Representation : Like RB encoding, the AMI always returns to the bias level during second half of the bit time interval and during the first half the transmitted level can be a positive, a negative or bias level, as for a bit 0 bias level and for a bit 1 either a positive level or negative level, the level being chose opposite to what it was used to represent the previous bit 1. Bandwidth : Twice as that required for the NRZ. DC Level : No DC component. Timing Information : No timing information (For long sequence of 0s). 59
Waveforms of AMI
TABULATION: S.NO SIGNALS AMPLITUDE TIME PERIOD 1. CLOCK SIGNAL
2. NRZ(L)
3. NRZ(M)
4. RZ
5. BIPHASE(MANCHESTER)
6. BIPHASE (MARK)
7. RB
8. AMI
60
ODEL GRAPH:
RESULT: Thus the different coding techniques were studied and observed for a given binary data, and their corresponding waveforms plotted. 61
EX.NO: 9 PLL AND FREQUENCY SYNTHESIZER DATE :
AIM: To study phase lock loop and its capture range, lock range and free running VCO APPARATUS REQUIRED: 1. Power Supply 2. LM565 3. Resistors -10K,680 (2 nos) 4. Capacitors -1F,0.1F,0.01F 5. CRO 6. AFO THEORY: 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 widely used in modern electronic devices, with output frequencies from a fraction of a hertz up to many gigahertz The LM565 and LM565C are general purpose phase locked loops containing a stable, highly linear voltage controlled oscillator for low distortion FM demodulation, and a double balanced phase detector with good carrier suppression. The VCO frequency is set with an external resistor and capacitor, and a tuning range of 10:1 can be obtained with the same capacitor. The characteristics of the closed loop systembandwidth, response speed, capture and pull in rangemay be adjusted over a wide range with an external resistor and capacitor. The loop may be broken between the VCO and the phase detector for insertion of a digital frequency divider to obtain frequency multiplication. 62
FEATURES: 1. 200 ppm /C Frequency Stability of the VCO 2. Power Supply Range of 5 to 12 Volts with 100 ppm/% Typical 3. 0.2% Linearity of Demodulated Output 4. Linear Triangle Wave with in Phase Zero Crossings Available 5. TTL and DTL Compatible Phase Detector Input and Square Wave Output 6. Adjustable Hold in Range from 1% to > 60%
APPLICATIONS:
1. Data and Tape synchronization 2. Modems 3. FSK Demodulation 4. FM Demodulation 5. Frequency Synthesizer 6. Tone Decoding 7. Frequency Multiplication and Division 8. SCA Demodulators 9. Telemetry Receivers 10. Signal Regeneration 11. Coherent Demodulators
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PROCEDURE: 1. Connect + 5V to pin 10 of LM 565. 2. Connect -5V to pin 1. 3. Connect 10k resistor from pin 8 to + 5V 4. Connect 0.01f capacitor from pin 9 to 5V 5. Short pin 4 to pin 5. 6. Without giving input measure(f O) free running frequency. 7. Connect pin 2 to oscillator or function generator through a 1f capacitor, adjust the amplitude aroung 2Vpp. 8. Connect 0.1 f capacitor between pin 7 and + 5V (C2) 9. Connect output to the second channel is of CRO. 10. Connect output to the second channel of the CRO. 11. By varying the frequency in different steps observe that of one frequency the wave form will be phase locked. 12. Change R-C components to shift VCO center frequency and see how lock range of the input
TABULATION: S.NO SIGNALS AMPLITUDE TIME 1. INPUT SIGNAL
2. DEMODULATED OUTPUT
3. VCO
RESULT: Thus the Phase Locked Loop have been determined.
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EX.NO: 10 ERROR CONTROL CODING USING MATLAB DATE :
AIM : To implement the error control linear and cyclic block codes and using MATLAB program. APPARATUS REQUIRED : PC with MATLAB software. THEORY : Its a sub class of linear block codes. Advantage of cyclic codes is that they are easy to encode. A binary code is to be cyclic code it exhibits two fundamental properties Linearity property Cyclic property In coding theory, a linear code is an error-correcting code for which any linear combination of code words is also a codeword. Linear codes are traditionally partitioned into block codes and convolutional codes, although Turbo codes can be seen as a hybrid of these two types. Linear codes allow for more efficient encoding and decoding algorithms than other codes. Linear codes are used in forward error correction and are applied in methods for transmitting symbols (e.g., bits) on a communications channel so that, if errors occur in the communication, some errors can be corrected or detected by the recipient of a message block. The code words in a linear block code are blocks of symbols which are encoded using more symbols than the original value to be sent. A linear code of length n transmits blocks containing n symbols. For example, the [7,4,3] Hamming code is a linear binary code which represents 4-bit messages using 7-bit code words. Two distinct code words differ in at least three bits. As a consequence, up to two errors per codeword can be detected and a single error can be corrected. This code contains 2 4 =16 code words. PROCEDURE : 1. Use the communication block set. 2. Perform the coding technique for the message that is generated randomly. 3. Similarly generate a noisy code signal randomly. 65
4. Perform the decoding operation. 5. Analyze the result and bit error rate is calculated. PROGRAM : LINEAR BLOCK CODES clc; clear all; close all; %Input Generator Matrix g=input('Enter The Generator Matrix'); disp('The order of Linear block code for given generator matrix is:'); [n,k]=size(transpose(g)) fori=1:2^k for j=k:-1:1 if rem(i-1,2^(-j+k+1))>=2^(-j+k) u(i,j)=1; else u(i,j)=0; end end end u disp('The possible codewords are:') c=rem(u*g,2) disp('The minimum hamming distance dmin for given block code is=') d_min=min(sum((c(2:2^k,:))')) disp('The error correction capability is= ') ec = (d_min-1)/2 %Code Word r=input('Enter the received code word:') p=[g(:,n-k+2:n)]; h=[transpose(p),eye(n-k)]; disp('Hamming code') ht=transpose(h) disp('syndrome decoding table'); 66
sundromematrix = ht errorpattern = eye(n) disp('Syndrome of a given codeword is:') s=rem(r*ht,2) fori=1:1:size(ht) if(ht(i,1:3)==s) r(i)=1-r(i); break; end end disp('The error is in bit:') i disp('The corrected codeword is:') r disp(' actual message bit is:') m=[r(1:k)]
disp(noise); newdata=decode(noise,n,k,'%cyclic,binary'); disp('newdata'); disp(newdata); [numerr,ratio]=biterr(newdata,data); disp('The bit error ratio is'); disp(ratio);
RESULT : Thus the program for error control cyclic and linear block code is implemented and the outputs are verified.
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EX.NO: 11 SAMPLING AND TIME DIVISION MULTIPLEXING DATE : AIM: To Study of sampling and reconstruction of signal and Time Division Multiplexing. APPARATUS REQUIRED: 1. ST 2101 & ST2153 Trainer kit 2. CRO 3. Patch Chords THEORY: SAMPLING: The signals we use in the real world, such as our voice, are called "analog" signals. To process these signals for digital communication, we need to convert analog signals to "digital" form. While an analog signal is continuous in both time and amplitude, a digital signal is discrete in both time and amplitude. To convert continuous time signal to discrete time signal, a process is used called as sampling. The value of the signal is measured at certain intervals in time. Each measurement is referred to as a sample.
TIME DIVISION MULTIPLEXING: Time division multiplexing is a technique of transmitting more than one information on the same channel. This means that several information signals can be transmitted over a single channel by sending samples from different information sources at different moments in time. This technique is known as time division multiplexing or TDM. TDM is widely used in digital communication systems to increase the efficiency of the transmitting medium.TDM can be achieved by electronically switching the samples such that they inter leave sequentially at correct instant in time without mutual interference.
PROCEDURE: Sampling & Reconstruction: Initial set up of trainer: Duty cycle selector switch position : Position 5 Sampling selector switch : Internal position 1. Connect the power cord to the trainer. Keep the power switch in Off position. 2. Connect 1 KHz Sine wave to signal Input. 3. Switch On the trainer's power supply & Oscilloscope. 4. Connect BNC connector to the CRO and to the trainers output port. 5. Select 320 KHz (Sampling frequency is 1/10th of the frequency indicated by the illuminated LED) sampling rate with the help of sampling frequency selector switch. 6. Observe 1 KHz sine wave (TP12) and Sample Output (TP37) on Oscilloscope. The display shows 1 KHz Sine wave being sampled at 32 KHz, so there are 32 samples for every cycle of the sine wave. (figure 1) 7. Connect the Sample output to Input of Fourth Order low pass Filter & observe reconstructed output on (TP46) with help of oscilloscope. The display shows the reconstructed original 1 KHz sine wave. (figure 2) 8. By successive presses of sampling Frequency Selector switch, change the sampling frequency to 2KHz, 4KHz, 8KHz, 16KHz and back to 32KHz (Sampling frequency is 1/10th 69
of the frequency indicated by the illuminated LED). Observe how SAMPLE output changes in each cases and how the lower sampling frequencies introduce distortion into the filters output waveform. This is due to the fact that the filter does not attenuate the unwanted frequency component significantly. Use of higher order filter would improve the output waveform. 9. So far, we have used sampling frequencies greater than twice the maximum input frequency.
FIGURE 1: Signal Sampling
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FIGURE 2: Signal Reconstruction Time Division Multiplexing: 1. Set up the following initial conditions on ST2153: a) Mode Switch in 320 KHz (FAST mode) position b) DC signal (I) & DC signal (II) Controls in function generator block fully clockwise. c) ~ 2 KHz and ~4 KHz control levels set to give 10Vpp. d) Pseudo - random sync code generator on/off switch in OFF Position. e) Error check code generator switch A & B in A=0 & B=0 position (OFF Mode) f) All switched faults off. 2. First, connect only the 2 KHz output to CH.I 71
3. Turn ON the power. Check that the PAM output of 2 KHz sine wave is available at TP17 of the ST2153. 4. Connect channel 1 of the oscilloscope to TP15 & channel 2 of the oscilloscope to TP17. Observe the timing & phase relation between the sampling signal TP15 & the sampled waveform at TP17. 5. Turn OFF the power supply. Now connect also the 4 KHz supply to CH.II. 6. Connect channel 1 of the oscilloscope to TP16 & channel 2 of the oscilloscope to TP17. 7. Observe & explain the timing relation between the signals at TP15, 7, 9, 16 & 17.
MODEL GRAPH:
Waveform at TP15
Waveform at TP16
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Waveform at TP 18when only one input signal is present
Waveform at TP17 when only one input is connected 73
Waveform at TP17 when only one input is connected TABULATION: (SAMPLING) S.NO SIGNAL AMPLITUDE TIME 1. INPUT SIGNAL
2. SAMPLED SIGNAL
3. RECONSTRUCTED SIGNAL
TABULATION: (TIME DIVISION MULTIPLEXING) S.NO SIGNAL AMPLITUDE TIME 1. AT TP 15
2. AT TP16
3. AT TP 17
4. AT TP18
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RESULT: Thus the signal have been sampled and reconstructed and also Time Division Multiplexing was studied and the graphs are plotted. 75
EX.NO: 12 FREQUENCY DIVISION MULTIPLEXING DATE : AIM: To determine the Frequency Division Multiplexing using MATLAB APPARATUS REQUIRED: 1. MATLAB Software/Simulink THEORY: Frequency-division multiplexing (FDM) is a technique by which the total bandwidth available in a communication medium is divided into a series of non-overlapping frequency sub-bands, each of which is used to carry a separate signal. This allows a single transmission medium such as the radio spectrum, a cable or optical fiber to be shared by many signals. The most natural example of frequency-division multiplexing is radio and television broadcasting, in which multiple radio signals at different frequencies pass through the air at the same time. At the source end, for each frequency channel, an electronic oscillator generates a carrier signal, a steady oscillating waveform at a single frequency such as a sine wave, that serves to "carry" information. The carrier is much higher in frequency than the data signal. The carrier signal and the incoming data signal (called the baseband signal) are applied to a modulator circuit. The modulator alters some aspect of the carrier signal, such as its amplitude, frequency, or phase, with the data signal, "piggybacking" the data on the carrier. Multiple modulated carriers at different frequencies are sent through the transmission medium, such as a cable or optical fiber. Each modulated carrier consists of a narrow band of frequencies, centered on the carrier frequency. The information from the data signal is carried in sidebands on either side of the carrier frequency. This band of frequencies is called the passband for the channel. As long as the carrier frequencies of separate channels are spaced far enough apart so that their passbands do not overlap, the separate signals will not interfere with one another. Thus the available bandwidth is divided into "slots" or channels, each of which can carry a data signal. At the destination end of the cable or fiber, for each channel, an electronic filter extracts the channel's signal from all the other channels. A local oscillator generates a signal at the channel's carrier frequency. The incoming signal and the local oscillator signal are applied to a demodulator circuit. This translates the data signal in the sidebands back to its original baseband frequency. An electronic filter removes the carrier frequency, and the data signal is output for use. Modern FDM systems often use sophisticated modulation methods that allow several data signals to be transmitted through each frequency channel. PROCEDURE: 1. Get the blocks from the MATLAB\Simulink tool 2. Determine the sine wave and the bandpass filters . 3. Obtain the output of FDM from the spectrum scope.
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BLOCK DIAGRAM;
RESULT: Thus the Frequency Division Multiplexing was determined using MATLAB