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Lab Manual
Faculty Name Dipika Sagne Designation Assistant Professor Subject/ Subject code DCOMM Semester/ Branch 6th Sem(DCOM) Lab/ Week 1 (2 Hours)
EXPERIMENT: - 1
OBJECTIVE:
To perform sampling and reconstruction of a signal and observe its waveform. EQUIPMENTS REQUIRED:
1. ST2101 with power supply cord 2. Oscilloscope with connecting probe 3. Connecting cords
THEORY:
In analog communication systems like AM, FM the instantaneous value of the Information signal is used to change certain parameter of the carrier signal. Pulse modulation systems differ from these systems in a way that they transmit a limited no. of discrete states of a signal at a predetermined time; sampling can be defined as measuring the value of an information signal at predetermined time intervals. The rate of which the signal is sampled is known as the sampling rate or sampling frequency. It is the major parameter, which decides the quality of the reproduced signal. If the signal is sampled quite frequently (whose limit is specified by Nyquist Criterion) then it can be reproduced exactly at the receiver with no distortion. Nyquist Criterion: The lowest sampling frequency that can be used without the side bands overlapping is twice the highest frequency component present in the information signal. If we reduce this sampling frequency even further, the side bands and the information signal will overlap and we cannot recover the information signal simply by low pass filtering. This phenomenon is known as fold-over distortion or aliasing.
Nyquist Criterion (Sampling Theorem): The Nyquist Criterion states that a continuous signal band limited to fm Hz can be completely represented by and reconstructed from the samples taken at a rate greater than or equal to 2fm samples/second. This minimum sampling frequency is known as NYQUIST RATE i.e. for faithful reproduction of information signal fs > 2fm. Effect of Sample and Sample Hold Output: If the pulse width of the carrier pulse train used in natural sampling is made very short compared to the pulse period, the natural PAM is referred to as instantaneous PAM. As it has been discussed, shorter pulse is desirous for allowing many signals to be included in TDM format but the pulse can be highly corrupted by noise due to lesser signal power. One way to maintain reasonable pulse energy is to hold the sample value until the next sample is taken. This Technique is formed as sample value until the next sample-and-hold techniques. Now, the area under the curve (which is equivalent to the signal power) is greater and so the filter output amplitude and quality of reproduced signal is improved. The hold facility can be provided by a capacitor when the switch connects the capacitor to PAM output it charges to the instantaneous value. Aliasing: If the signal is sampled at a rate lower than stated by Nyquist criterion, then there is an overlap between the information signal and the sidebands of the harmonics. Thus the higher and the lower frequency components get mixed and cause unwanted signals to appear at the demodulator output. This phenomenon is turned as aliasing or fold over distortion. The various reasons for aliasing and its prevention are as described.
A) Aliasing due to Under-Sampling If the signal is sampled at rate lower than 2fm then it causes aliasing. Let us assume a sinusoidal waveform of frequency fin which is being sampled at rate fs<2fm. In the figure 9 dots represents the sample points. The low-pass filter at demodulator effectively joins the sample causing an unwanted frequency component to appear at the output. This unwanted component has frequency equal to (fs-fm). B) Aliasing due to wide Band Signal The system is designed to take samples at frequency slightly greater than that stated by Nyquist rate. If higher frequencies are ever present in the information signal or it is affected by high frequency noise then the aliasing will occur. This does not generally happen in properly designed telephone network where speech channels are band-limited by filters before sampling. In control engineering and telemetry, however, out of band high frequencies either from source or due to
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DIGITAL COMMUNICATION LAB MANNUAL PROCEDURE: 1. Assemble all the required components to perform practical. 2. Connect a BNC to crocodile cable to CRO & m (t) (with in kit) to observe m (t). Note down amplitude & its frequency. 3. Now select a sampling frequency from sampling freq. selection circuit & observe it on CRO for amplitude. 4. Give both sampling clock & message signal as input to sampled output circuit, while keeping trigger at internal position switch. 5. Observe sampled o/p on CRO for amplitude & freq. 6. Connect a cable between sampled outputs to 4th filters input (LPF) for reconstructing signal. 7. Observe reconstructed signal output. 8. Repeat above procedure for further sampling frequency.
Amplitude
Frequency
Entity
Amplitude
Frequency
Entity
Amplitude
Frequency
2. What is the world wide standard sampling rate for speech signal?
EXPERIMENT: - 2
OBJECTIVE:
To generate a PCM signal and demodulate it.
EQUIPMENTS REQUIRED:
TDM Pulse Code Receiver Trainer (ST2104) TDM Pulse Code, Transmitter Trainer (ST2103), Patch cords, CRO etc.,
THEORY:
Pulse Code Modulation (PCM): In PCM System the amplitude of the sampled waveform at definite time intervals is represented as a binary code. The first three techniques of the above described systems are not truly digital but in fact are analog in nature. The very fact that the variation of a particular pulse parameter is continuous rather than being in the discrete steps makes the system analog in nature. As a result of this, the PAM signals are vulnerable to noise & dispersion of the pulse. The channel introduces noise on the signal from various sources. Also the receiver is not noise free. The pulses also suffer attenuation & dispersion as they pass through the channel. The primary line constants (L, C, G, & R) limit the velocity at which a particular frequency can travel. The result is different frequency travel at different velocities in the medium. Therefore some frequency component of the square wave arrives later as compared to the other. This causes widening of the pulse width. The phenomenon is called 'dispersion. The combined effect of attenuation, dispersion & noise is so large that the pulse is impaired & introduced at the receiver. Steps in Pulse Code Modulation: Sampling: The analog signal is sampled according to the Nyquist criteria. The Nyquist criteria states that for faithful reproduction of the band limited signal, the sampling rate must be at least twice the highest frequency component present in the signal. For audio signals the highest frequency component is 3.4 KHz. So,
Practically, the sampling frequency is kept slightly more than the required rate. In telephony the standard sampling rate is 8 KHz. Sample quantifies the instantaneous value of the analog signal point at sampling point to obtain pulse amplitude output. 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 as
The opposite effect is utilized at the receiver to undo the effect of compression, is termed as expanding. The two processes are combined are known as compounding this feature is not
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PROCEDURE: 1. Make the connection according to the circuit diagram. 2. Connect the audio frequency of 1 KHz, 2V signal to analog to digital converter. 3. Mode switches in fast position. 4. Pseudo - random sync code generator switched 'Off'. 5. Error check code selector switches A & B in A = 0 & B= 0 position ('Off' Mode). 6. Connect the PCM modulator output to CRO. 7. Observe output on CRO.
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DIGITAL COMMUNICATION LAB MANNUAL OBSERVATION TABLES: M(t) Amplitude: Amplitude Frequency Amplitude Frequency Frequency: Amplitude Frequency Amplitude Frequency
Pulse
Reconstructed M(t) Amplitude Frequency Amplitude Frequency Amplitude Frequency Amplitude Frequency
CONCLUSION:
QUESTIONS:
1. Which noise is occurs in PCM? 2. What is Quantization? 3. What is the advantage of PCM? 4. At which factor bandwidth of PCM depends?
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EXPERIMENT: - 3
OBJECTIVE:
Study and perform Delta Modulation and Demodulation. EQUIPMENTS REQUIRED:
Delta Adaptive and Delta Sigma Modulation-Demodulation Trainer (ST2105), CRO, patch cords
THEORY:
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. Has the signal amplitude increased or decreased since the last sample was taken? If signal amplitude has increased, then modulator's output is at logic level 1. If the signal amplitude has decreased, the modulator output is at logic level 0. Thus, the output from the modulator is a series of zeros and ones to indicate rise and fall of the waveform since the previous value. One way in which delta modulator and demodulator is assembled.
Delta Modulator: The analog signal which is to be encoded into digital data is applied to the positive input of the voltage comparator which compares it with the signal applied to its negative input from the integrator output (more about this signal in forth coming paragraph). The comparator's output is logic '0' or '1' depending on whether the input signal at positive terminal is lower or greater then the negative terminals input signal. The comparator's output is then latched into a D-flip-flop which is clocked by the transmitter clock. Thus, the output of D-flip-Flop is a latched 'l' or '0' synchronous with the transmitter clock edge. This binary data stream is transmitted to receiver and is also fed to the unipolar to bipolar converter. This block converts
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Block Diagram:
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Amplitude
Frequency
Transmitter Clk Data I/P Integrator O/P Data O/P CONCLUSION: QUESTIONS:
1. How analog signal can be encoded in to bits? 2. What is the advantage of DM over PCM? 3. Which types of noise occur in delta modulation? 4. Define adaptive delta modulation.
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EXPERIMENT: - 4
OBJECTIVE:
To study different types of digital data formats (RZ, NRZ and Manchester).
EQUIPMENTS REQUIRED:
8 bit Variable Binary Data Generator (ST2111), Data Formatting And Carrier Modulation Transmitter Trainer (ST2106), CRO, patch cords
THEORY:
Line Coding Basics: Transmission of serial data over any distance, be it a twisted pair, fiber optic link, coaxial cable, etc., requires maintenance of the data as it is transmitted through repeaters, echo chancellors and other electronically equipment. The data integrity must be maintained through data reconstruction, with proper timing, and retransmitted. Line codes were created to facilitate this maintenance. In selecting a particular line coding scheme some considerations must be made, as not all line codes adequately provide the all important synchronization between transmitter and receiver. Other considerations for line code selection are noise and interference levels, error detection and error checking, implementation requirements, and the available bandwidth. Unipolar Coding: The most basic transmission code is unipolar or unbalanced coding. In this scheme each discrete variable is transmitted with a different assigned level, 0V and for example +2.5V. But this holds a number of disadvantages: The average power is two times other bipolar codes The coded signal contains DC and low frequency components. When long strings of zeros are present, a DC or baseline wander occurs. This results in loss of timing and data because a receiver/repeater cannot optimally discriminate ones and zeros. Repeaters/receivers require a minimum pulse density for proper timing extraction. Long strings of ones or zeros contain no timing information and lead to timing jitter (when a clock recovery is used) and possible loss of synchronization. There is no provision for line error rate monitoring. Bipolar Coding: With bipolar, or also called balanced coding, the same data may be transmitted more efficiently achieving the same error distance with half the power. This coding is often referred to as Non-Return to Zero (NRZ) coding as the signal level is maintained for the duration of the signal interval. Although bipolar coding is more efficient than unipolar, it still lacks provisions for line error monitoring and is susceptible to DC wander and timing jitter. This coding scheme provides a number of features which: Eliminate DC Wander Minimize Timing Jitter Provide for Line Error Monitoring. This is accomplished by introducing controlled redundancy in the code through extra coding levels. Data Formatting: The symbols 0 and 1 in digital systems can be represented in various formats with different levels & waveforms. The selection of particular format for communication depends on the
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Return To Zero (RZ) Format : The RZ code provides a partial solution to overcome the receiver clock regeneration problem with NRZ (L) code. It is similar to NRZ (L) code, except that the information is contained in the first half of the bit, interval, while the level during the second half of each period is always 0 volts. The comparison of the two waveforms for a given data is shown in figure.
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BLOCK DIAGRAM:
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PROCEDURE: 1. Generate a clock signal having amplitude 5vp-p & freq. 240 kHz. 2. Using a kit, generate data signal. 3. Now pass the data signal & clock signal into another kit to generate NRZ L, RZ, and Manchester respectively on CRO. 4. Signal can be matched by seeing periodic repetition. 5. Unplugged the kits & CRO.
OBSERVATION TABLES:
Entity Pulse
Amplitude
Frequency
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CONCLUSION:
QUESTIONS: 1. Why line coding is required in digital communication? 2. What is the advantages of manchaster coding? 3. Compare RZ with NRZ coding scheme. 4. Define jitter.
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EXPERIMENT: - 5
OBJECTIVE:
Study of Amplitude Shift Keying Modulation & Demodulation Technique
EQUIPMENTS REQUIRED:
8 bit Variable Binary Data Generator, Data Formatting and Carrier Modulation Transmitter Trainer, Carrier Demodulation & Data Reformatting Receiver Trainer, CRO, patch cords.
THEORY:
Amplitude Shift Keying: The simplest method of modulating a carrier with a data stream is to change the amplitude of the carrier wave every time the data changes. This modulation technique is known amplitude shift keying. The simplest way of achieving amplitude shift keying is by switching On the carrier whenever the data bit is '1' & switching off. Whenever the data bit is '0' i.e. the transmitter outputs the carrier for a' 1 ' & totally suppresses the carrier for a '0'. This technique is known as On-Off keying figure 20 illustrates the amplitude shift keying for the given data stream. Thus, Data = 1 carrier transmitted Data = 0 carrier suppressed The ASK waveform is generated by a balanced modulator circuit, also known as a linear multiplier. As the name suggests, the device multiplies the instantaneous signal at its two inputs. The output voltage being product of the two input voltages at any instance of time. One of the input is AC coupled 'carrier' wave of high frequency. Generally, the carrier wave is a sine wave since any other waveform would increase the bandwidth, without providing any advantages. The other input which is the information signal to be transmitted, is DC coupled. It is known as modulating signal.
Amplitude Shift Keying: The data stream applied is unipolar i.e. 0 volts at logic '0' & + 5 Volts at logic '1'. The output of balanced modulator is a sine wave, unchanged in phase when a data bit
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The method to demodulate the ASK modulation results in a great simplicity at the receiver. The method to demodulate the ASK waveform is to rectify it, pass it through the filter & 'Square Up' the resulting waveform. The output is the original data stream. Figure shows the functional blocks required in order to demodulate the ASK waveform at receiver.
ASK Demodulator
BLOCK DIAGRAM:25
PROCEDURE:
1. Make the connection according to the circuit diagram. 2. Connect Binary Data Generator to the ASK modulator with desired data pattern output to CRO. 3. Connect ASK modulator output on CRO. 4. Now demodulate the ASK modulator output at receiver side. 5. Find the transmitted data pattern on CRO
Data at Receiver
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QUESTIONS:
1. Give the application of ASK. 2. List out the disadvantages of ASK. 3. Define symbol rate.
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EXPERIMENT: - 6
OBJECTIVE:
Study of Frequency Shift Keying Modulation & Demodulation Technique
EQUIPMENTS REQUIRED:
8 bit Variable Binary Data Generator, Data Formatting and Carrier Modulation Transmitter Trainer, Carrier Demodulation & Data Reformatting Receiver Trainer, CRO, patch cords.
THEORY:
Frequency Shift Keying: In frequency shift keying, the carrier frequency is shifted in steps (i.e. from one frequency to another) corresponding to the digital modulation signal. If the higher frequency is used to represent a data '1' & lower frequency a data '0', the resulting Frequency shift keying waveform appears as shown in figure. Thus
Frequency Shift Keying Modulator : On a closer look at the FSK waveform, it is apparent that it can be represented as the sum of two ASK waveforms.
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FSK Demodulator: The demodulation of FSK waveform can be carried out by a phase locked loop. As known, the phase locked loop tries to 'lock' to the input frequency. It achieves this by generating corresponding output voltage to be fed to the voltage controlled oscillator, if any frequency deviation at its input is encountered. Thus the PLL detector follows the frequency changes & generates proportional output voltage. The output voltage from PLL contains the carrier components. Therefore the signal is passed through the low pass filter to remove them. The resulting wave is rounded to be used for digital data processing. Also, the amplitude level may be very low due to channel attenuation. The signal is 'Shaped Up' by feeding it to the voltage comparator. The functional block diagram of FSK demodulator is shown in the following figure.
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Since the amplitude change in FSK waveform does not matter, this modulation technique is very reliable even in noisy & fading channels. But there is always a price to be paid to gain that advantage. The price in this case is widening of the required bandwidth. The bandwidth increase depends upon the two carrier frequencies used & the digital data rate. Also, for a given data, the higher the frequencies & the more they differ from each other, the wider the required bandwidth. The bandwidth required is at least doubled than that in the ASK modulation. This means that lesser number of communication channels for given band of frequencies.
BLOCK DIAGRAM:-
PROCEDURE:
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OBSERVATION TABLE:Signal Data Carrier 1 Carrier 2 FSK Data at Receiver Amplitude Frequency
CONCLUSION:
QUESTIONS:
1. What is FSK?
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