School of Engineering

Laboratory Manual

DIGITAL COMMUNICATION Lab EC-303

DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING

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DIGITAL COMMUNICATION LAB
This laboratory will commence with ‘Orientations’ to familiarize students with surroundings or circumstances includes following:
A. Familiarization with digital modulation and demodulation techniques B. Familiarization with sampling theorem & line coding used in digital communication system

S. N. 1 2 3 4 5 6 7 8 9 10

Title of the Experiment
Sampling and reconstruction of an analog signal. To generate a PAM, PWM and PPM signal and demodulate them. To generate TDM-PAM signal and demodulate it. To generate PCM signal and demodulate it. Delta modulation & demodulation. To study different types of digital data formats (RZ, NRZ and Manchester) ASK modulation and demodulation. FSK modulation and demodulation. BPSK modulation and demodulation. To study quadrature amplitude modulation & demodulation.

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LIST OF EXPERIMENTS
NAME OF THE STUDENT………………………………………. ENROLL. NO…………………………

INDEX S. N. Name of the experiment Sampling and reconstruction of an analog signal. To generate a PAM, PWM and PPM signal and demodulate them. To generate demodulate it. TDM-PAM signal and Page no. Date of allotment Date of conduction Sign /grade

1 2 3 4 5 6 7 8 9 10

To generate PCM signal and demodulate it. Delta modulation & demodulation. To study different types of digital data formats (RZ, NRZ and Manchester) ASK modulation and demodulation. FSK modulation and demodulation.

BPSK modulation and demodulation.

To study quadrature amplitude modulation & demodulation.

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Experiment No. 1 AIM: Sampling and reconstruction of an analog signal.

APPARATUS/COMPONENTS REQUIRED: Signal Sampling and Reconstruction Trainer kit (ME 776), CRO and patch 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. 4

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 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 filter’s 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 o/p.

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. 5

Note down amplitude & its frequency. Connect a BNC to crocodile cable to CRO & m (t) (with in kit) to observe m (t). 6 . 5. Give both sampling clock & message signal as input to sampled output circuit. while keeping trigger at internal position switch. selection circuit & observe it on CRO for amplitude. 8. Observe sampled o/p on CRO for amplitude & freq. Repeat above procedure for further sampling ferq. Now select a sampling frequency from sampling freq. Observe reconstructed signal output. 2. 3.Block Diagram: Sampling frequency & selector circuit Sampling cicuit Sample o/p Sample & hold o/p Message signal 1 KHz Analog Sampling 2nd/4th order low pass filter Sample output Reconstructed o/p Reconstruction Procedure:1. Assemble all the required components to perform practical. 4. Connect a cable between sampled outputs to 4th filter’s input (LPF) for reconstructing signal. 6. 7.

Observation Table Entity M(t) clk Sample o/p M’(t) Entity M(t) clk Sample o/p M’(t) Amplitude Frequency 320khz Amplitude Frequency 80khz Entity M(t) clk Sample o/p M’(t) Amplitude Frequency 20khz CONCLUSION:- 7 .

Questions: 1. What do you mean by natural sampling? 8 . What is the use of sampling theorem? 2. What is the world wide standard sampling rate for speech signal? 3. What do you mean by over sampling? 4. What is aliasing effect in sampled signal? 5.

equal in amplitude to the signal voltage at each instant. the signal to be converted to Pulse Amplitude Modulation is fed to one input of an AND gate. The output of the gate then consists of pulses at the sampling rate. and since Pulse Amplitude Modulation does not utilize constant amplitude pulses. When it is used. is illustrated in Figure. PPM. to ensure that the pulses are always positive. the ability to use constant amplitude pulses is a major advantage of pulse modulation. The pulses are then passed through a pulse shaping network. PWM and PPM signal and demodulate them. Patch cords. It is very easy to generate and demodulate pulse amplitude modulation. which is self-explanatory and single polarity pulse amplitude modulation. Pulses at the sampling frequency are applied to the other input of the AND gate to open it during the wanted time intervals. the pulses frequency modulates the carrier. APPARATUS/COMPONENTS REQUIRED: PAM. It forms an excellent introduction to pulse modulation in general. the simplest form of pulse modulation. As shown in Figure. Theory: PAM: Pulse amplitude modulation. which gives them flat tops. CRO etc. and each sample is made proportional to the amplitude of the signal at the instant of sampling.Experiment No. In a generator. PWM Modulation & Demodulation Trainer (ST2110). Pulse amplitude modulation is a pulse modulation system in which the signal is sampled at regular intervals. As will be seen shortly. 9 . in which a fixed DC level is added to the signal. 2 Aim: To generate a PAM. it is infrequently used. The two types are double polarity pulse amplitude modulation.

Connect pulse generator clock to the PAM modulator. Connect the modulator output to CRO. Connect the audio frequency of 2 KHz. Observe output on CRO. 2V to modulator. 5. 10 . 3. 2. Observe output on CRO. 7. Now connect modulator output to low pass filter for demodulation. 6. 4. Make the connection according to the block diagram.PROCEDURE:1.

Block Diagram 11 .

as shown in Figure. but the width of each pulse is made proportional to the amplitude of the signal at that instant. Pulse width modulation. 7. 5. 6. Make the connection according to the circuit diagram. 12 . Observe output on CRO. 2. Switch ON the power supply. 4. PLM (pulse length modulation). In this system. we have fixed amplitude and starting time of each pulse. Observe output on CRO. 2V to modulator. Connect the audio frequency of 2 KHz. (a) Signal (b) PWM (width variations exaggerated) PROCEDURE:1. Now connect modulator output to low pass filter for demodulation. Connect the modulator output to CRO.PWM: In pulse width modulation of pulse amplitude modulation is also often called PDM (pulse duration modulation) and less often. 3.

Block Diagram. 13 .

pulse-position modulations has The advantage of requiring constant transmitter power output. 2V to modulator. 6. Observe output on CRO. PROCEDURE:1. Make the connection according to the circuit diagram. 2. 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. Now connect modulator output to low pass filter for demodulation. As Mentioned in connection with pulse width modulation. Observe output on CRO.PPM:-The Amplitude and width of the pulses is kept constant in this system. Connect the audio frequency of 2 KHz. 4. 5. Connect the PPM modulator output to CRO. but the disadvantages of Depending on transmitter receiver is synchronization. 3. CONCLUSION:- 14 .

Block Diagram 15 .

Observation Table A) PAM Entity M(t) clk Sample o/p Sample & hold Flat top M’(t) Amplitude Frequency B) PWM Entity M(t) pulse PWM M’(t) Amplitude Frequency C) PPM Entity M(t) pulse PPM M’(t) Amplitude Frequency 16 .

How many types of pulse time modulation? 2. What is the merit of flat top sampling? 4. Which Multivibrator is used for PPM De-modulator? 8. Which multivibrator is used for PWM? 6. What is the disadvantage of PWM? 5.Questions 1. At which factor the band-width of PPM depends? 17 . What is advantage of PPM? 7. How many types of pulse amplitude modulation? 3.

18 .7 KHz. the modulated signals occupy different frequency ranges in the frequency spectrum. A separation between two modulated signals in frequency band reduces call interference and also allows for the gradual roll-off gradient of the filters.7 KHz.g. Patch cords. 3 Aim: To generate a TDM-PAM signal and demodulate it. Let us suppose two information signals occupy a frequency range of 300 . Frequency division multiplexing (FDM) 2. Only lower side band is transmitted. Theory: Multiplexing: Multiplexing is the process of combining signals from different information sources so that they can be transmitted over a common channel. the modulated signal occupies a frequencies band of 60.Experiment No. The pass band is chosen so as to extract the information from one channel. APPARATUS/COMPONENTS REQUIRED: TDM Pulse Amplitude Modulation-Demodulation Trainer (ST2102). As it can be seen from above example. At receiver. Multiplexing is advantageous in cases where it is impracticable and uneconomical to provide separate links for the different information sources. the modulated signal occupies a frequency band of 64. The first signal modulates a 64 KHz carrier. Time division multiplexing (TDM) Frequency Division Multiplexing: Frequency division multiplexing is the process of combining several information channels by shifting their signals to different frequency groups within the frequency spectrum so that they can all be transmitted over a common transmission channel. The second signal modulates a 68 KHz carrier. The information signals are shifted in different frequency groups by making them modulate carrier signals at different frequencies e.6 KHz to 63. Hence they can be transmitted over the same channel. filters having different pass band frequency range are used to separate the various information signals. The price that has to be paid to acquire this advantage is in the form of increased system complexity and bandwidth.3400Hz speech signal). The two most commonly used methods of multiplexing are 1.6 KHz to 67. CRO etc.

It must start at the same time as the transmitting switch and it must establish electrical contact with the same channel no. The samples from the other sources can be placed within these time intervals. message 3 and then again message 1 so on. Frequency/ rate of operation at transmitter. Sample identification. provided you are aware of the existence of circuit delays and setting times. the receiver is said to be in synchronization with transmitter. If these two conditions are met. Theoretically large number of samples can be multiplexed in time domain. This increases the transmitter and receiver complexity and cost. the information from source 1 will be received by some other channel which is not intending to accept the information from that particular channel. The fact utilized in TDM technique is that there are large intervals between the message samples. is a must. If constraint one is not met. To establish synchronization. It can be anticipated from above process that the receiver switch has to follow two constraints: 1. (See figure 6) 19 . in time domain so that they can be transmitted over the same channel. as that of the transmitter. The functioning of TDM switch is complex. but its practical implementation becomes harder and harder as the time interval between consecutive samples decreases. Figure 4 Each signal is sampled over one sampling interval and transmitted one after the other along a common channel. the receiver needs to know: a. It must rotate at the same rate as the transmitter switch. Thus part of message 1 is transmitted first followed by part of message 2. If constraint two is not met. the operations of transmitter switch are controlled by the transmitter timing logic. one at transmitter and the other at receiver. (See figure 4) The two wipers rotate and establish electrical contact with one channel at a time. But its understanding is easy.Time Division Multiplexing : Time division multiplexing is the process of combining the samples from different information signals. Thus every sample is separated from other in time domain. the samples of different sources would get mixed at the receiver. In ST2102. The time division multiplexing system can be simulated by two rotating switches. Practical Aspects of Time Division Multiplexing : In time division multiplexing the correct operations of transmitter switch which creates samples. 2. b.

Divider output b. The following table summarizes the switch operation for various inputs to the decoder. 20 . Observe that the output at each of these test points is a train of pulses at frequency 16 KHz and with pulse duration set by the duty cycle selector switch.Figure 5 switch depends upon the decoder output provided in transmitter timing logic.10. The decoder's output depends upon two quantities: a.9. The decoder's output can be obtained at TP7. Decoder Enable pulse train which is provided by the duty cycle control signal. 8.

Figure 6 Divider Output Transmitter Sampling Switches 21 .

the duty cycle output is made to lag by 5% of one channel time slot. The groups of four time slots are termed as a frame. Make following connections with 4mm banana to banana connectors: a. 2KHz fully clockwise. d. Hence the duration for which the switch remains closed also varies. Transmitter Clock to Receiver Clock c. But the operation takes places only when the decoder is enabled. 250Hz.e. The Duty cycle Selector switch is presently in position 5 i. 2. The duration of a particular switch closing is decided by the duty cycle control switch whose output drives the enable input of the decoder. It causes sampling error. The enable signal is active low and it is supplied from the duty cycle control switch (TP4) the switch is closed when a low signal is applied to it. The signal at TP42 shows the reconstructed ~250Hz sine wave which was transmitted at CH0. To overcome this problem. Display the Receiver's Low Pass Filter's input (TP41) & output (TP42) simultaneously on the oscilloscope. 500 Hz to CH 1 input socket of Transmitter block. The problem arises when the decoder enable signal appears while divider output is changing. 46. 4. 22 . 1 KHz to CH 2 input socket of Transmitter block. observe the Transmitter Output signal (TP20) & the Receiver’s CH 0 Low Pass Filter's input (TP41) 7. 2 KHz to CH 3 input socket of Transmitter block.Divider Output MSB TP3 0 0 1 1 LSB TP2 0 1 0 1 Transmitter Sampling Switches CH0 TP7 Closed Open Open Open CH0 TP8 Open Closed Open Open CH2 TP9 Open Open Closed Open CH3 TP10 Open Open Open Closed The decoder output is decided by the divider output. Turn all the potentiometers in Function Generator block viz Sync Level. Transmitter Output to Receiver Input b. Procedure: 1. On ST2102 four channels are multiplexed. the output width driving the decoder's enable input varies. 8. This will help to achieve a stable waveform. because the binary code at the divider output may initially correspond to a different switch. Use Transmitter's CH0 Input for external triggering of oscilloscope. With the help of oscilloscope. set the duty cycle control switch in position '5'. The period allocated for transmitting one sample is called as a time slot. Make the following connections with banana connectors: a. 3. the duration of each sample is 50% of the timeslot allotted to each channel. For different setting of the duty cycle switch. Observe the Transmitter Output (TP20) along with CH0 input (TP11) for reference with the aid of oscilloscope. 5. This allows the divider output to settle to a constant level before the enable signal arrives. Similarly view the outputs of all Receiver Low Pass Filters at TP44. b. Figure 7 illustrates the timing of the two signals when The duty cycle control is set to "9". 1 KHz. 250 Hz to CH 0 input socket of Transmitter block. 48. c. Transmitter CH0 to Receiver CH 0 6. The two control signals to the decoder create a problem.

m3(t)m4(t)} Connect the message signal to CRO & then obtain TDM output & observe it. 23 . Assemble all required apparatus. 4.4 {m1(t). 2. observe it. 5.3. Give PAM-TDM o/p to demodulator.Block diagram: Procedure:1. Simultaneously do the same for message signal 1. 3. Connect the message signal with CRO.2.m2(t).

Observation Table:Signal m1(t) M2(t) M3(t) M4(t) clk TDM o/p m’1(t) m’2(t) m’3(t) m’4(t) amplitude frequency CONCLUSION:- 24 .

Define synchronous TDM.Questions: 1. What is frame in TDM 4. 3. 2. List out the application of TDM 25 . Define asynchronous TDM.

The difference between the analog signal value & its approximated one(quantized one) is random & unpredictable. But nothing comes without price. The combined effect of attenuation. G.4 KHz. In telephony the standard sampling rate is 8 KHz. random signal which accompanies the information signal and is termed as 'Quantization noise'. The sampled value is then approximated to the nearest amplitude level. The pulses also suffer attenuation & dispersion as they pass through the channel. As a result of this. Steps in Pulse Code Modulation: Sampling : The analog signal is sampled according to the Nyquist criteria. Increasing the number of levels to reduce quantization noise has the effect of increasing the number of bits. Quantization noise : As we have seen the signal is approximated to the nearest level (step). the sampling frequency is kept slightly more than the required rate. The result is different frequency travel at different velocities in the medium. Sampling Frequency ≥ 2 fm ≥ 2 x 3. So. The primary line constants (L. C. But it can never be eliminated.Experiment No. Also the receiver is not noise free. Since the levels are discrete where as the signal is continuous. the discrepancy creeps in. Quantization noise can be reduced by increasing the number of levels. APPARATUS/COMPONENTS REQUIRED: TDM Pulse Code Receiver Trainer (ST2104) TDM Pulse Code Transmitter Trainer (ST2103). Therefore some frequency component of the square wave arrives later as compared to the other. Increasing the number of bits to represent a sample increases the system's bandwidth requirement 26 . the PAM signals are vulnerable to noise & dispersion of the pulse. This process is called as Quantization & it is generally carried out by the A/D converter. which is then transmitted. & R) limit the velocity at which a particular frequency can travel. a. 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. This causes widening of the pulse width. Patch cords. The Nyquist criteria states that for faithful reproduction of the band limited signal. hence reducing the approximation. 4 Aim:.To generate a PCM signal and demodulate it. There are two important problems associated with quantization. the sampling rate must be at least twice the highest frequency component present in the signal. The channel introduces noise on the signal from various sources. The sample is then assigned a code corresponding to the amplitude level. This is a sort of unwanted. dispersion & noise is so large that the pulse is impaired & introduced at the receiver. The phenomenon is called 'dispersion’. For audio signals the highest frequency component is 3. unpredictable. Sample quantifies the instantaneous value of the analog signal point at sampling point to obtain pulse amplitude output.8 KHz Practically. 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.4 KHz ≥ 6. Allocation of Binary Codes : Each binary word defines a particular narrow range of amplitude level.

To overcome this problem. the quantization levels are uniform for all the amplitude range. This process is called compression. The encoding method described above is called as uniform encoding i. This problem can be overcome by using a sample & hold circuit prior to A/D converter output.b. The input/output characteristics for compression signal passed through a comparator network 'prior to compression (See figure 1). the duration of sampled pulse is much smaller than the A/D converter's sampling time. This has an effect of compression on the extreme ends of the signal. The quantization noise plays havoc with the low level signals because the % approximation compared to the signal amplitude is very high. An input output characteristic providing compression 27 . But in practice. remain unchanged till the conversion is complete. But this method of encoding has disadvantages of its own. 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. 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. Here the quantization levels are clear together for low level than they are for the high levels. Refer page 18 & 19 A/D conversion for details. a non-uniform encoding scheme is used. The sample & hold circuitry holds the sample value till the next sample.e. The A/D Converter requires that the value at its input.

The opposite effect is utilized at the receiver to undo the effect of compression. the recovered samples are filtered & reconstructed to provide the original waveform.e. The system samples 0-2 samples sequentially providing 3 samples to be converted to 3 "n" bit words. At receiver. the data is decoded by the D/A converter. Various channels can be multiplexed in time domain i. These three n bit words forms the basis of a frame. The two processes are combined are known as compounding this feature is not provided on trainer but you should be aware of its existence. the information data from various sources are sequentially transmitted over the same transmission medium e. is termed as expanding. Some error correcting codes & synchronization can also be transmitted along with the information signal. 28 . The frame contains these three n bit words also contains some synchronization & reference positioning information.g Let us assume a 3 channel PCM system.

Error check code selector switches A & B in A = 0 & B= 0 position ('Off' Mode). Connect the audio frequency of 1 KHz. Pseudo . 7. Make the connection according to the circuit diagram.random sync code generator switched 'Off'. 6. Mode switch in fast position.PROCEDURE:1. 29 . 2V signal to analog to digital converter. Connect the PCM modulator output to CRO. 4. 3. 2. 5. Observe output on CRO.

Block diagram: : 30 .

Observation Table:Signal m(t) Clk PAM o/p PCM o/p PAM’ m’(t) Amplitude Frequency CONCLUSION:- 31 .

which noise is occurs in PCM? 2. At which factor bandwidth of PCM depends? 32 .Questions: 1. what is Quantization? 3. what is the advantage of PCM? 4.

at each sampling time we ask simple question. at each sampling time. Falling linear ramp signal when + 4V is applied to it (corresponding to binary 0). APPARATUS/COMPONENTS REQUIRED: Delta Adaptive And Delta Sigma Modulation-Demodulation Trainer (ST2105). Delta Modulator: The analog signal which is to be encoded into digital data is applied to the +ve input of the voltage comparator which compares it with the signal applied to its -ve input from the integrator output (more about this signal in forth coming paragraph). 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. Let us 33 .The integrator output is then connected to the -ve terminal of voltage comparator.Experiment No. (corresponding to binary 1) b. This block converts logic '0' to voltage level of + 4V and logic 'l' to voltage level . CRO . Thus. In this system. If the signal amplitude has decreased. thus completing the modulator circuit. One way in which delta modulator and demodulator is assembled. I. then modulator's output is at logic level 1.4V is applied to it. The Bipolar output is applied to the integrator whose output is as follows: a. Thus. This binary data stream is transmitted to receiver and is also fed to the unipolar to bipolar converter. The comparator's output is logic '0' or '1' depending on whether the input signal at +ve terminal is lower or greater then the -ve terminals input signal. the output from the modulator is a series of zeros and ones to indicate rise and fall of the waveform since the previous value. the modulator output is at logic level 0. the output of D-flip-Flop is a latched 'l' or '0' synchronous with the transmitter clock edge. patch cords Theory: Delta modulation is a system of digital modulation developed after pulse code modulation. Rising linear ramp signal when . The comparator's output is then latched into a D-flip-flop which is clocked by the transmitter clock.e. 5 Aim: Delta modulation & demodulation.4V. say the Kth sampling time. Has the signal amplitude increased or decreased since the last sample was taken? If signal amplitude has increased.

understand the working of modulator circuit with the analog input waveform pplied as below: Technique of Delta Modulation 34 .

Block Diagram:- 35 .

Observation table:Signal m(t) Transmitter Clk Data i/p Integrator o/p Data o/p Amplitude Frequency CONCLUSION:- 36 . Connect the DM modulator output to receiver side & observe the output on CRO. Ensure that integrator 1 block's switches are in following position: a) Gain control switch in left-hand position (towards switch A & B). 4. 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. 7. Connect the DM modulator output to CRO. Ensure that the clock frequency selector block switches A & B are in A = 0 and B = 0 position. b) Switches A & B in A=0 and B=0 positions. Make connection on the board as shown in the figure 3. 5.Procedure : 1. 6. Connect the mains supply 2.

Define adaptive delta modulation.Questions: 1. What is the advantage of DM over PCM? 3. How analog signal can be encoded in to bits? 2. Which types of noise occur in delta modulation? 4. 37 .

be it a twisted pair. and is susceptible to DC wander and timing jitter. Although bipolar coding is more efficient than unipolar. The data integrity must be maintained through data reconstruction. Bipolar Coding: With bipolar. 0V and for example +2. Line codes were created to facilitate this maintenance. the same data may be transmitted more efficiently achieving the same error distance with half the power. In this scheme each discrete variable is transmitted with a different assigned level.To study different types of digital data formats (RZ. and the available bandwidth. 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. and retransmitted. with proper timing. Unipolar Coding: The most basic transmission code is unipolar or unbalanced coding. fiber optic link. Repeaters/receivers require a minimum pulse density for proper timing extraction. implementation requirements. This coding is often referred to as NonReturn to Zero (NRZ) coding as the signal level is maintained for the duration of the signal interval. Other considerations for line code selection are noise and interference levels. a DC or baseline wander occurs. echo chancellors and other electronically equipment. etc. 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. Data Formatting And Carrier Modulation Transmitter Trainer (ST2106). patch cords Theory:Line Coding Basics: Transmission of serial data over any distance. 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. CRO. This results in loss of timing and data because a receiver/repeater cannot optimally discriminate ones and zeros. or also called balanced coding. When long strings of zeros are present. NRZ.Experiment 6 Aim: . 38 . it still lacks provisions for line error monitoring. requires maintenance of the data as it is transmitted through repeaters.5V.. error detection and error checking. and Manchester) APPARATUS/COMPONENTS REQUIRED: 8 bit Variable Binary Data Generator (ST2111). 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. There is no provision for line error rate monitoring. coaxial cable.

Data Formatting: The symbols ‘0’ and ‘1’ in digital systems can be represented in various formats with different levels & waveforms. except that the information is contained in the first half of the bit. Every data format has specific advantages & disadvantages associated with them.Return To Zero (Level) NRZ (L) : It is the simplest form of data representation. interval. system’s ability to pass DC level information. These are also available on ST2106 trainer. The selection of particular format for communication depends on the system bandwidth. It is similar to NRZ (L) code. A data '0' is encoded as a low level during first half of the bit time and a high level during the second half. while the level during the second half of each period is always 0 volts. 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. ease of clock regeneration & synchronizations at receiver. The most widely used formats of data representation are given below. error checking facility. NRZ (L) Encoding Return To Zero (RZ) Format : The RZ code provides a partial solution to overcome the receiver clock regeneration problem with NRZ (L) code. RZ Format Biphase (Manchester) Coding: The encoding rules for biphase (Manchester) code are as follows. Non . A data '1' is encoded as a high level 39 . system complexity & cost etc. The comparison of the two waveforms for a given data is shown in figure.

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. Biphase (Manchester) Format Block diagram:- 40 . Figure shows the biphase (Manchester) waveform for a given data stream.

generate data signal. 4. 5. Using a kit. RZ. Now pass the data signal & clock signal into another kit to generate NRZ L. 3. Unplugged the kits & CRO. Generate a clock signal having amplitude 5vp-p & freq. Observation Table:Amplitude CLK Data NRZ RZ Manchester frequency 240 Khz CONCLUSION:- 41 . and Manchester respectively on CRO. Signal can be matched by seeing periodic repetition. 240 kHz. 2.Procedure:1.

What do you mean by dc wandering? 42 . What is the advantages of manchaster coding? 3. 5 . Compare RZ with NRZ coding scheme.Questions: 1. 4. Why line coding is required in digital communication? 2. Define jitter.

Data = 1 carrier transmitted Data = 0 carrier suppressed The ASK waveform is generated by a balanced modulator circuit.e. 0 volts at logic '0' & + 5 Volts at logic '1'. also known as a linear multiplier. This modulation technique is known amplitude shift keying. The output voltage being product of the two input voltages at any instance of time. The ASK modulation result in a great simplicity at the receiver. This technique is known as ‘On-Off’ keying figure 20 illustrates the amplitude shift keying for the given data stream. The method to demodulate the ASK waveform is to rectify it. APPARATUS/COMPONENTS REQUIRED: 8 bit Variable Binary Data Generator (ST2111). ASK modulation Amplitude Shift Keying: The data stream applied is unipolar i.Experiment 7 Aim: . The other input which is the information signal to be transmitted. Thus. the carrier is multiplied by 0 volts. giving rise to 0 volt signal at modulator's output. It is known as modulating signal. In this case the carrier is multiplied with a positive constant voltage when the data bit '0' is applied. unchanged in phase when a data bit ‘l' is applied to it. pass it through the filter & 43 . CRO . Generally. Whenever the data bit is '0' i. the device multiplies the instantaneous signal at its two inputs. 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.ASK modulation & demodulation. As the name suggests. The simplest way of achieving amplitude shift keying is by switching ‘On’ the carrier whenever the data bit is '1' & switching off. The method to demodulate the ASK modulation results in a great simplicity at the receiver. Carrier Demodulation & Data Reformatting Receiver Trainer(ST2107). is DC coupled.e. the transmitter outputs the carrier for a' 1 ' & totally suppresses the carrier for a '0'. without providing any advantages. the carrier wave is a sine wave since any other waveform Would increase the bandwidth. Data Formatting And Carrier Modulation Transmitter Trainer (ST2106). The output of balanced modulator is a sine wave. One of the input is AC coupled 'carrier' wave of high frequency.

Now demodulate the ASK modulator output at receiver side. 2. Make the connection according to the circuit diagram. Find the transmitted data pattern on CRO Observation Table:Signal Carrier Data ASK 1s 0s Data at receiver 44 Amplitude Frequency . Connect Binary Data Generator to the ASK modulator with desired data pattern output to CRO. 4. Figure shows the functional blocks required in order to demodulate the ASK waveform at receiver.'Square Up' the resulting waveform. ASK Demodulator ASK Modulation PROCEDURE: 1. 5. Connect ASK modulator output on CRO. The output is the original data stream. 3.

Block Diagram:- CONCLUSION:- 45 .

List out the disadvantages of ASK 3. What is the bandwidth of BFSK? 46 . 2. Give the application of ASK. 4.Questions: 1. Define symbol rate.

the carrier frequency is shifted in steps (i. If the higher frequency is used to represent a data '1' & lower frequency a data '0'. Thus Data = 1 high frequency Data = 0 low frequency FSK Waveform On a closer look at the FSK waveform. 47 . CRO.FSK modulation & demodulation. the resulting Frequency shift keying waveform appears as shown in figure.Experiment 8 Aim: . it can be seen that it can be represented as the sum of two ASK waveforms. from one frequency to another) corresponding to the digital modulation signal. patch cords Theory:Frequency Shift Keying: In frequency shift keying.e. Carrier Demodulation & Data Reformatting Receiver Trainer (ST2107). Data Formatting and Carrier Modulation Transmitter Trainer (ST2106). APPARATUS/COMPONENTS REQUIRED: 8 bit Variable Binary Data Generator (ST2111).

Also. Thus the PLL detector follows the frequency changes & generates proportional output voltage. the wider the required bandwidth. The output voltage from PLL contains the carrier components. As known. This means that lesser number of communication channels for given band of frequencies. the higher the frequencies & the more they differ from each other. this modulation technique is very reliable even in noisy & fading channels. Figure shows the functional blocks involved in FSK demodulation.FSK Waveform from ASK Waveforms FSK Modulator: The demodulation of FSK waveform can be carried out by a phase locked loop. 48 . It achieves this by generating corresponding output voltage to be fed to the voltage controlled oscillator. Also. The bandwidth required is at least doubled than that in the ASK modulation. the amplitude level may be very low due to channel attenuation. if any frequency deviation at its input is encountered. for a given data. But there is always a price to be paid to gain that advantage. the phase locked loop tries to 'lock' to the input frequency. The signal is 'Squared Up' by feeding it to the voltage comparator. FSK Demodulator: Since the amplitude change in FSK waveform does not matter. The price in this case is widening of the required bandwidth. Therefore the signal is passed through the low pass filter to remove them. The bandwidth increase depends upon the two carrier frequencies used & the digital data rate. The resulting wave is too rounded to be used for digital data processing.

Connect Binary Data Generator to the FSK modulator with desired data pattern output to CRO. 3. 2. 4. Find the transmitted data pattern on CRO Observation Table: Signal Data Carrier 1 Carrier 2 FSK Data at receiver Amplitude Frequency 49 . Now demodulate the FSK modulator output at receiver side.Block Diagram: PROCEDURE: 1. Make the connection according to the circuit diagram. Connect FSK modulator output on CRO. 5.

What is BFSK? 3. What is the difference between FM and FSK? 4. What is the bandwidth of BFSK? 5. 50 .CONCLUSION:- Questions: 1. Why FSK is preferred over ASK? 2. What is the disadvantage of BFSK? .

equal positive and negative voltage level. Sometimes known as quaternary or quadriphase PSK or 4-PSK.e. it is easier to see it as two independently modulated quadrature carriers. Although QPSK can be viewed as a quaternary modulation. PSK modulator is similar to ASK modulator both used balanced modulator to multiply the carrier with balanced modulator signal. With this interpretation. 225º. With four phases. Analysis shows that this may be used either to double the data rate compared to a BPSK system while maintaining the bandwidth of the signal or to maintain the data-rate of BPSK but halve the bandwidth needed. the output of modulator is a sine wave which is switched out of phase by 180 from the carrier input. 01. 185º. QPSK can encode two bits per symbol. The four possible combinations at bib it code are 0º. CRO. Data Formatting And Carrier Modulation Transmitter Trainer (ST2106). relative to the phase at the original unmodulated carrier QPSK offers an advantage over PSK is a no carrier that how each phase represents a two bit code rather than a single bit. 10.Experiment 9 Aim: . Carrier Demodulation & Data Reformatting Receiver Trainer (ST2107). When the modulating input is positive the out put at modulator is a line wave in phase with the carrier input whereas for positive voltage level. and 315º lagging. This means that either we can charge phase per sec. APPARATUS/COMPONENTS REQUIRED: 8 bit Variable Binary Data Generator (ST2111). The digital signal with applied to Modulation input for PSK generation is bipolar i. shown in the diagram with Gray coding to minimize the BER — twice the rate of BPSK. patch cords Theory:PSK: . and 11 each code represents either a phase of 45º.in QPSK each pair at consecutive data bit is treated as a two bit code which is switch the phase of the carrier sine wave between one at four phase 90º apart. or the same amount of data can be transmitted with Constellation diagram for QPSK with Gray coding. QPSK uses four points on the constellation diagram. the even (or odd) bits are used to modulate the inphase component of the carrier. while the odd (or even) bits are used to modulate the quadrature-phase component of the carrier. Quadrature Phase-shift Keying (QPSK) QPSK:. equispaced around a circle. Each adjacent symbol only differs by one bit. BPSK is used on both carriers and they can be independently demodulated 51 .PSK involves the phase change at the carrier sine wave between 0 to 180 in accordance with the data stream to be transmitted.PSK modulation & demodulation.

Connect BPSK modulator output on CRO. Now. 2. Connect Binary Data Generator to the BPSK modulator with desired data pattern output to CRO. 3. demodulate the BPSK modulator output at receiver side. Make the connection according to the circuit diagram. 5. 4. Find the transmitted data pattern on CRO Observation Table:Signal Amplitude Carrier Data PSK 1s 0s Data Frequency 52 .BLOCK DIAGRAM:- PROCEDURE:1.

53 . What is the difference between QPSK and BPSK? 4. What is the disadvantage of PSK? 2. Compare bandwidth of BFSK and BPSK. What is the advantage of PSK? 3. What is DPSK? 5.CONCLUSION:- Questions: 1.

983. 10 Aim: To study QAM modulation and demodulation APPARATUS/COMPONENTS REQUIRED: 8 bit Variable Binary Data Generator (ST2111). In this way. "C"): the first 2 ("I" and "Q") determine the phase of the output signal. 8-QAM Demodulator : The 8-QAM demodulator mounted on the module uses the 4-PSK demodulator to detect the signals "I" and "Q".Experiment No. patch cords Theory:Quadrature Amplitude Modulation (QAM) The QAM is a digital modulation where the information is contained into the phase as well as the amplitude of the transmitted carrier. The modulated signal can take 4 different phases and 2 different amplitudes. 8-QAM : In the 8-QAM the data are divided into groups of 3 bits (Tribit). N-QAM : At the moment we reach to a data subdivision into groups of 9 bits. CRO . "Q". each "modulation interval" depends on the state of 3 data bits ("I". defined as the ratio between Fb and Bw. BELL 209) and digital radio transmission. The 8-QAM signal can be seen as 4-PSK signal whose amplitude can take 2 different values. Applications in modems for high speed data transmission (ITU-TV22bis. V34. The main aspects characterizing the QAM are: a. the last two the phase. one of which varies the amplitude of the carrier. which can take 16 different states. 16-QAM : In the 16-QAM the data are divided into groups of 4 bits (Quadbit). V32bis. Possibility of error higher than the PSK called Fb the bit transmission speed and "n" the number of bits considered for the modulation. The transmission efficiency. the minimum spectrum Bw of the modulated signal is equal to Ft/n c. V33. while the signal "C" is obtained detecting the amplitude of the positive values of the signal "I".2. This amplitude can take 2 positive and two negative values. as function of the value of the signal "c" in transmission. V34bis. while the block diagram of the modulator mounted on the module is shown in Fig. It needs circuits of high complexity b.. Quadrature Amplitude Modulation and Demodulation Trainer (ST2112) . obtaining constellations with 512 modulation points. is equal to "n" Modulator QAM : The functional diagram of a 8-QAM modulator is shown in Fig. the third ("C") the amplitude. V32. V29. for a total of 8 different states. The 16 possible combinations change amplitude and phase of the carrier. If the 54 . The demodulator "c" detects which of the two levels is present in the coming signal.

TP21 and TP22. if the value is the lowest you obtain the value "0". The block diagram of the 8-QAM demodulator is shown in Fig.level is the highest you obtain the value "1 ". The demodulator includes the following circuits: a. A data clock extraction circuit and three data re-timing circuits. A circuit discriminating the amplitude of the signal "I". Demodulator QAM : 55 . This enables to obtain the signal "C" d. "Q" and "C" are supplied across the outputs TP20. Two 2-PSK demodulators (indicated on the diagram as I-DEM and Q DEM) two low pass filters. The signals "I". The regenerator of the carriers at 0° and 90° (the same of the 4-PSK demodulator) b. c.

b. TP21. you can observe Clock & Data which you have set. To observe I switch & Q switch in the demodulator section. Now press SW8 which is reset switch then press SW4 which is start. Now to observe QAM modulated signal with respect to data. 4. TP20. connect channel 1 of oscilloscope to the test point TP12 you will observe squarer frequency. SW9 should be in the OFF mode. TP22 ( if you have logic analyzer you can observe I. 9. 3. Connect Test point TP6 on Channel 1 & TP7 on Channel 2 of Oscilloscope. SW7. SW5. TP2. 11. Switch ON all the DIP switches on SW3.38 Procedure : 1. first bit is I bit then second bit is Q bit then third bit is C bit. 10. You can add noise by using DIP switch SW9 (001 / 010 / 111). TP21. connect Channel 1 to TP1 & Channel 2 to TP9. TP5. To observe I . As there are 24 bits data available on the trainer so. 7. TP3. To observe input data . To observe the demodulator section. 12. SW3. Set I.Fig. you will observe 1 KHz sine & cosine wave. encoded data & decoded data you have to connect logic analyzer to test points TP1. Q & C simultaneously) 13. For example: SW5=11000110 SW6=01011000 SW7=01100010 5. TP4. 6. connect channel 1 of oscilloscope to TP16 & channel 2 of the oscilloscope to TP17. Q & C Channel data with the help of DIP switch SW5.TP22. output data . Q & C demodulated signal connect oscilloscope to TP20. TP23. In this experiment you have to use I bit & Q bit & C bit so you can select combination according to your requirement. SW7. Ensure the following initial conditions on ST2112 trainer : a. 8. 56 . Power supply should be OFF. Now connect Channel 1 of Oscilloscope to TP2 & Channel 2 to TP1. SW6. TP24 etc. SW6.

Waveforms : a. QPSK & QAM modulated signal : 57 . Clock & Data : b.

List out advantages of QAM 3. Give the application of QAM 58 . Define QAM 2.CONCLUSION:- Questions: 1.