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DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING, YCCE

DIGITAL COMMUNICATION LAB MANNUAL

Lab Manual
Faculty Name Dipika Sagne Designation Assistant Professor Subject/ Subject code DCOMM Semester/ Branch 6th Sem(DCOM) Lab/ Week 1 (2 Hours)

Purpose of the laboratory

The main goal of this laboratory is to give you a practical idea about different modulation and coding schemes important for digital communication.

DIGITAL COMMUNICATION LAB MANNUAL

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.

DIGITAL COMMUNICATION LAB MANNUAL

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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noise pick-up can be present. In this case band-limiting filters, generally known as anti-aliasing filters are usually installed prior to sampling to prevent aliasing. As a principle, the system is designed to sample at rate higher than the rate to take into account the equipment tolerances, ageing and filter response. C) Aliasing due to Filter Roll-off Roll-off is a term applied to the cut-off gradient of a filter. No filter is ideal and therefore frequencies above the nominal cut-off frequency may still have significant amplitudes at a filters o/p. If proper sampling rate and appropriate filter response is not chosen, aliasing will occur. D) Aliasing due to Noise If very small duty cycle is used in sample-and-hold circuit aliasing may occur if the signal has been affected by noise. High frequency noises generally mix with the high frequency component of the signal and hence causes undesirable frequency components to be present at the output. Low pass filters The PAM system the message is recovered by a low pass filter. The type of filter used is very important, as the signal above the cut-off frequency would affect the recovered signal if they are not attenuated sufficiently.

DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING, YCCE

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

DIGITAL COMMUNICATION LAB MANNUAL CONCLUSION: QUESTIONS:

1. What is the use of sampling theorem?

2. What is the world wide standard sampling rate for speech signal?

DIGITAL COMMUNICATION LAB MANNUAL

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

DIGITAL COMMUNICATION LAB MANNUAL

Quantization & it is generally carried out by the A/D converter. There are two important problems associated with quantization. a. Quantization noise : As we have seen the signal is approximated to the nearest level (step). Since the levels are discrete where as the signal is continuous, the discrepancy creeps in. The difference between the analog signal value & its approximated one (quantized one) is random & unpredictable. This is a sort of unwanted, unpredictable, random signal which accompanies the information signal and is termed as 'Quantization noise'. Quantization noise can be reduced by increasing the number of levels, hence reducing the approximation. But it can never be eliminated. Increasing the number of levels to reduce quantization noise has the effect of increasing the number of bits. But nothing comes without price. Increasing the number of bits to represent a sample increases the system's bandwidth requirement. b. Finite sampling time of A/D converter : Another problem associated with quantization is that the A/D Converter requires finite time to convert the analog information to digital data. The A/D Converter requires that the value at its input, remain unchanged till the conversion is complete. But in practice, the duration of sampled pulse is much smaller than the A/D converter's sampling time. This problem can be overcome by using a sample & hold circuit prior to A/D converter output. The sample & hold circuitry holds the sample value till the next sample. The encoding method described above is called as uniform encoding i.e. the quantization levels are uniform for all the amplitude range. But this method of encoding has disadvantages of its own. The quantization noise plays havoc with the low level signals because the % approximation compared to the signal amplitude is very high. This causes a great amount of distortion at the receiver for low level signals. Also the quieter part of music or speech could become severely distorted & would make them unpleasant to listen. To overcome this problem, a non-uniform encoding scheme is used. Here the quantization levels are clear together for low level than they are for the high levels. This has an effect of compression on the extreme ends of the signal. The input/output characteristics for compression signal passed through a comparator network 'prior to compression (See figure 1). This process is called compression.

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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provided on trainer but you should be aware of its existence. Some error correcting codes & synchronization can also be transmitted along with the information signal. At receiver, the data is decoded by the D/A converter; the recovered samples are filtered & reconstructed to provide the original waveform. Various channels can be multiplexed in time domain i.e. the information data from various sources are sequentially transmitted over the same transmission medium e.g Let us assume a 3 channel PCM system. The system samples 0-2 samples sequentially providing 3 samples to be converted to 3 "n" bit words. These three n bit words forms the basis of a frame. The frame contains these three n bit words also contains some synchronization & reference positioning information.

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Figure: Message and Quantized Signal

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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DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING, YCCE

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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DIGITAL COMMUNICATION LAB MANNUAL

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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DIGITAL COMMUNICATION LAB MANNUAL

logic '0' to voltage level of + 4V and logic 'l' to voltage level - 4V. The Bipolar output is applied to the integrator whose output is as follows: a. Rising linear ramp signal when - 4V is applied to it, (corresponding to binary 1) b. Falling linear ramp signal when + 4V is applied to it (corresponding to binary 0).The integrator output is then connected to the negative terminal of voltage comparator, thus completing the modulator circuit. Let us understand the working of modulator circuit with the analog input waveform applied as below:

Block Diagram:

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2. Make connection on the board as shown in the figure 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 left-hand position (towards switch A & B) b) Switches A & B are in A = 0 and B = 0 positions. 6. Connect the DM modulator output to CRO. 7. Connect the DM modulator output to receiver side & observe the output on CRO.

OBSERVATION TABLES: Entity

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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DIGITAL COMMUNICATION LAB MANNUAL

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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DIGITAL COMMUNICATION LAB MANNUAL

system bandwidth, systems ability to pass DC level information, error checking facility, ease of clock regeneration & synchronizations at receiver, system complexity & cost etc. The most widely used formats of data representation are given below. These are also available on ST2106 trainer. Every data format has specific advantages & disadvantages associated with them. Non - Return To Zero (Level) NRZ (L) : It is the simplest form of data representation. The NRZ (L) waveform simply goes low for one bit time to represent a data '0' & high for one bit time to represent a data '1'. Thus the signal alternates only when there is a data change.

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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DIGITAL COMMUNICATION LAB MANNUAL

Biphase (Manchester) Coding: The encoding rules for biphase (Manchester) code are as follows. A data '0' is encoded as a low level during first half of the bit time and a high level during the second half. A data '1' is encoded as a high level during first half of the bit time and a low level during the second half. Thus string of l's or 0's as well as any mixture of them will not pass any synchronization problem in receiver. Figure shows the biphase (Manchester) waveform for a given data stream.

BLOCK DIAGRAM:

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DIGITAL COMMUNICATION LAB MANNUAL

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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DIGITAL COMMUNICATION LAB MANNUAL Data NRZ RZ Manchester

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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DIGITAL COMMUNICATION LAB MANNUAL

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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DIGITAL COMMUNICATION LAB MANNUAL

l' is applied to it. In this case the carrier is multiplied with a positive constant voltage when the data bit '0' is applied, the carrier is multiplied by 0 volts, giving rise to 0 volt signal at modulator's output. The ASK modulation result in a great simplicity at the receiver.

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.

BLOCK DIAGRAM:25

DIGITAL COMMUNICATION LAB MANNUAL

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

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DIGITAL COMMUNICATION LAB MANNUAL CONCLUSION:

QUESTIONS:
1. Give the application of ASK. 2. List out the disadvantages of ASK. 3. Define symbol rate.

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DIGITAL COMMUNICATION LAB MANNUAL

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

Data = 0 low frequency

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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DIGITAL COMMUNICATION LAB MANNUAL

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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DIGITAL COMMUNICATION LAB MANNUAL

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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DIGITAL COMMUNICATION LAB MANNUAL

1. Make the connection according to the circuit diagram. 2. Connect Binary Data Generator to the FSK modulator with desired data pattern output to CRO. 3. Connect FSK modulator output on CRO. 4. Now demodulate the FSK modulator output at receiver side. 5. Find the transmitted data pattern on CRO

OBSERVATION TABLE:Signal Data Carrier 1 Carrier 2 FSK Data at Receiver Amplitude Frequency

CONCLUSION:

QUESTIONS:
1. What is FSK?

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DIGITAL COMMUNICATION LAB MANNUAL

Ms. Dipika S. Sagne Subject In-Charge, ET Department YCCE Dr. P.L.Zade Head of Department, ET Department YCCE

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