Acom Lab Manual
Acom Lab Manual
Acom Lab Manual
LAB Manual
Contents
LAB MANUAL
INTRODUCTION TO COMMUNICATION ENGINEERING I LABORATORY ............................................ 3 EXP 1: GETTING FAMILIAR WITH THE LABORATORY EQUIPMENT....................................................... 7 EXP 2: SIMULATION OF COMMUNICATION SYSTEMS USING MATLAB ............................................. 11 EXP 3: REPRESENTATION OF SIGNALS & SYSTEMS ............................................................................ 15 EXP 4: SPEECH SIGNALS .................................................................................................................... 19 EXP 5: DSBSC MODULATION & DEMODULATION ............................................................................. 23 EXP 6: AM AND QAM ....................................................................................................................... 27 EXP 7: FM MODULATION.................................................................................................................. 31 EXP 8: FM DEMODULATION ............................................................................................................. 35 EXP 9: PCM ENCODING..................................................................................................................... 39 EXP 10: PCM DECODING ................................................................................................................... 43 EXP 11: LINE CODING ....................................................................................................................... 47 EXP 12: DIGITAL MODULATION: FSK................................................................................................. 51 APPENDIX A: LABORATORY REGULATIONS AND SAFETY RULES ....................................................... 55 APPENDIX B: SAMPLE REPORT ......................................................................................................... 56
1. to allow you to perform experiments that demonstrate the theory of signals and communication systems that are discussed in course, 2. to introduce you to some of the electronic blocks that make up communication systems (which may not be discussed in the lecture course because of time limitations) , and 3. to familiarize you with proper laboratory procedure, including precise recordkeeping, logical troubleshooting, safety, and learning about the capabilities and limitations of your equipment.
Introduction
This document contains the laboratory experiments to accompany the course EE 370 Communications Engineering I, offered by Electrical Engineering Department, KFUPM. The document contains twelve experiments, four on basic and general background, four on analog modulation, and four on digital modulation. The four basic experiments cover introduction to the laboratory equipment, simulation of communication systems using MATLAB, time- and frequency-domain representation of signals, and processing of speech signals. The analog modulation part covers the generation and detection of Double-Side Band Suppressed Carrier (DSBSC) modulation, Double-Side Band With Carrier (also known as AM) modulation, Quadrature Amplitude Modulation (QAM), and Frequency Modulation. The digital modulation experiments include PCM encoding and decoding, line codes and digital carrier modulation (ASK and FSK). Each experiment, whenever applicable, contains the following sections: Objectives: where the expected achievements by the end of the experiment are stated. Introduction: where the theory of the subject is reviewed. The introduction is kept brief, assuming the student has covered the material in detail in class, or can refer to his textbook for further reading. System Modules: where the main new modules to be used in the experiment are described. Lab Work: leading the student on how to run the experiment. The lab work is organized in parts in order to have a clear and integrated structure of the work.
Kousa & Muqaibel | Introduction to Communication Engineering I Laboratory
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Post-Lab Work: extra questions and tasks for the student to carry after the lab, and include in the lab report.
In addition to the experiment write up, a Lab Sheet has been prepared for every experiment. The Lab Sheet is a working document, designed to help students record all lab activities (measurements, observations, answers to questions in the lab manual, ). The student must have his instructor sign the sheet before he leaves. The material in the sheet shall be utilized in writing the report. Plots from the PC-based oscilloscope and spectrum analyzer may be saved on a storage media (or student file-box if network is available) to reproduce them later in the report. The lab sheets for the 12 experiments are collected in one booklet separate from this document. A set of Laboratory Regulations and Safety Rules are attached in Appendix A. All students have to observe them carefully. MATLAB will be frequently invoked as part of the post-lab work, mainly in the form of designing a simulation counterpart for the experimental work. Such exercise will improve the student programming skills, and acquaint him with the most frequently-encountered functions and techniques for simulating communication systems. It is the sole responsibility of the student to learn the basics of MATLAB. Every student should submit a report on each experiment. The report must be selfcontained, and can be read independent from the lab manual. All axes in all graphs should be clearly labeled. If there is more than one trace in the plot, they should be clearly labeled. A sample report is attached in Appendix B.
Troubleshooting
Things will not always go as expected; this is the nature of the learning process. While testing a communication block, if the output signal is not what you expect, don't just try things at random, i.e replacing wires, rotating knobs, and toggling switches, hoping to get lucky. Rather, think before you do anything. If you do so you will avoid wasting time going down dead-end streets.
Kousa & Muqaibel | Introduction to Communication Engineering I Laboratory
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Be logical and systematic. First, look for obvious errors that are easy to fix. Is your measuring device correctly set and connected? Are you looking at the proper scale? Is the power supply set for the correct voltage? Is the signal generator correctly set and connected? And so on. Next, check for obvious misconnections or broken connections, at least in simple circuits.
As you work through your circuit, use your lab sheet to record tests and changes that you make as you go along; don't rely on your memory for what you have tried. Identify some test points in the system at which you know what the signal should be, and work your way backwards from the output through the test points until you find a good signal. Now you have a section of the system to focus your efforts on. Here is where a little thought about laying out your board before connecting it up will pay off; if your system looks like a jungle, it is going to be very hard to troubleshoot, but if it is well organized and if the wires are short, it is going to make your job a lot easier.
Final remark: if you do discover a bad module or wire, do not just throw it back in the box. Tell your instructor or the lab technician about it.
Neatness
When you have finished for the day, return all modules to their proper storage bins, return all test leads and probes to their storage racks, return all equipment to its correct location, and clean up the lab station. If appropriate switch off the unneeded equipments. We hope you an enjoyable learning experience!
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Objectives
Learn the various components and conventions of the lab equipment from TIMS. Use the data sheets to learn about the operation, parameters and limitations of system modules. Explore the features and capabilities of the PC-based oscilloscope and spectrum analyzer. Perform basic modeling using TIMS.
TIMS Overview
Throughout the course, we will be using the laboratory equipment 301C PC-based from TIMS to complement and demonstrate the theoretical part of the course. We will devote this experiment to introduce the equipment and get familiar with its usages. TIMS is a telecommunications modeling system that models block diagrams representing telecommunications systems. Physically, TIMS is a dual rack system; the upper rack accepts up to 12 plug-in cards, or modules; the lower rack houses a number of fixed modules, as well as the system power supply.
Plug-in Modules
Fixed Modules
Figure 1: TIMS 301-C System Unit The modules are simple electronic circuits, which serve as basic communications building blocks. Each module, fixed or plug-in, has a specific function; functions fall into three categories: 1. Signal Generation - oscillators, variable DC, etc 2. Signal Processing - multipliers, filters, etc 3. Signal Measurement - frequency counter, PC-based instrument inputs.
Kousa & Muqaibel | Exp 1: Getting Familiar with the Laboratory Equipment
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Some of those modules are classified as basic modules while others are advanced modules. The fixed modules are all basic. They include: BUFFER AMPLIFIERS, FREQUENCY AND EVENT COUNTER, HEADPHONE AMPLIFIER, MASTER SIGNALS, TRUNK PANEL, VARIABLE DC and PCBASED INSTRUMENT INPUT. The list of available plug-in modules is shown in the table below. #
1 2 3 4 5 6 7 8 9 10 11
Module
AUDIO OSCILLATOR ADDER DUAL ANALOG SWITCH MULTIPLIER PHASE SHIFTER QUADRATURE PHASE SHIFTER SEQUENCE GENERATOR UTILITIES TUNEABLE LOW PASS FILTER 1 TUNEABLE LOW PASS FILTER 2 TWIN PULSE GENERATOR
Type
Basic Basic Basic Basic Basic Basic Basic Basic Basic Basic Basic
#
12 13 14 15 16 17 18 19 20 21 22
Module
VOLTAGE CONTROL OSCILLATOR 1 VOLTAGE CONTROL OSCILLATOR 2 60KHz LOW PASS FILTER QUADRATURE UTILITIES LINE CODE ENCODER LINE CODE DECODER 100KHz CHANNEL FILTER PCM ENCODER PCM DECODER BIT CLOCK GENERATOR SPEECH MODULE
Type
Basic Basic Basic Advanced Advanced Advanced Advanced Advanced Advanced Advanced Advanced
A data sheet for each module describing its input(s), output(s), configurable parameters and function can be found in the User Manuals (Basic and Advanced) available in the lab bench drawers. A soft copy is also available on all laboratory computers desktop. All TIMS modules conform to the following conventions: Signal interconnections are made via front panel sockets Sockets on the left hand side are for module inputs. Sockets on the right hand side are for module outputs. Yellow sockets are for analog signals. Red sockets are for digital signals. Analog signals are held near the level of 4V p-p. Digital signals are TTL level, 0 to 5 V. The green socket is the system Ground. Any plug-in module may be placed in any of the 12 positions of the upper rack. All modules use the back plane bus to obtain power supply. The modules can be plugged-in or removed without turning off the power.
It is important to note that: The plug-in modules are not firmly locked in the rack, and need to be held in position while interconnecting leads are removed. When removing the leads, hold them from their solid heads and DO NOT PULL them from the flexible segment, in order not to damage the wires. There are 22 plug-in modules. Make sure you leave them in sequence in the storage shelves.
Kousa & Muqaibel | Exp 1: Getting Familiar with the Laboratory Equipment
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The name of this module is not intuitive. We will instead refer to it as DISPLAY INTERFACE module.
Kousa & Muqaibel | Exp 1: Getting Familiar with the Laboratory Equipment
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Lab Work
1. Read the data sheet of the ADDER in the TIMS Manuals-Basic Modules. Which of the following equations can be implemented using the ADDER and which cannot? Write your answers in the Lab Sheet. -2 cos(2 2x106t) - 1.5 cos(2 2x105t); -1.3 cos(2 2x104t)x(t) 0.5 sin(2 2x103t); -2.5 cos(2 2x104t)x(t) 10.5 sin(2 2x103t); 1.3 cos(2 2x104t)x(t) + 0.5 sin(2 2x103t). 2. Use the FREQUENCY COUNTER module to verify the frequencies of the following four signals from the MASTER SIGNALS: 100 kHz sine, 8.3 kHz Clock, 2 kHz TTL and 2 kHz sinusoid. Note down the values. Warning: The FREQUENCY COUNETR module accepts TTL and analog inputs. ONLY ONE OF THEM SHOULD BE CONNECTED AT A TIME, otherwise you may get erroneous measurement. 3. Connect the previous four signals of the MASTER SIGNALS module to the four inputs of the DISPLAY INTERFACE. Use the switches to display them on the oscilloscope (picoscope). Measure the amplitude of each signal and note them down in the Lab Sheet. 4. Use the VARIABLE DC, BUFFER AMPLIFIERS and ADDER modules to generate the signal 3cos(2x2x103t)+6 V. Draw the modules and show the connections. Let your instructor verify the waveform. 5. Observe and plot the spectra of each of the four signals of the MASTER SIGNALS module. a. Do the spectra plots coincide with your expectations? Explain. b. How far is the noise level below the signal level? 6. Using a 2 kHz sinusoidal signal on one channel and 8.33 kHz digital signal on the other channel, familiarize yourself with the picoscope by exploring the following features. Feature Switch between oscilloscope and frequency analyzer on the same view Display one or both channels on the same view (window) Separate the two channels on the same view so that they are nonoverlapping (do it manually and auto) Change the setting of the axes. Take a snap shot or continuous scan Zoom in a specific segment of the graph Display measurements of DC value, frequency, period, Use horizontal and vertical markers Set the oscilloscope on external triggering Create time view and spectrum view and save them
Kousa & Muqaibel | Exp 1: Getting Familiar with the Laboratory Equipment
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Objectives:
The main objective of this session is to learn the basic tools and concepts for simulating communication systems using MATLAB.
Introduction
MATLAB is a user-friendly, widely used software for numerical computations. MATLAB is vector-oriented, that is, it mainly deals with vectors (or matrices). It is assumed that you have used MATLAB before, and you can do simple operations, as well as create and run *.m files. Some useful tutorials can be found on the EE 370 course/lab website. If you need help on how to start working on MATLAB, we advise you to read Matlab Primer available in the internet. Our focus in this session will be on using MATLAB for simulating communication systems. Instead of going in the traditional approach of explaining items individually, we will work through one complete example, and introduce the application as we go.
Case Study:
Write a MATLAB program to simulate the following system
m(t)
g(t)
Full-Wave Rectifier
y(t)
z(t)
c(t) where m(t) = exp(-100|t|) ; c(t) = cos(2 103t) m-File: % Define the time interval
ts=0.00001; t= -0.1:ts:0.1; % Define the functions m(t) and c(t) m=exp(-100*abs(t)); c=cos(2*pi*1000*t);
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Discussion
% Define the time interval This is usually the first step in any simulation. There are three parameters to define: the beginning of the interval, the step size, the end of the interval. The beginning and end of the
Kousa & Muqaibel | Exp 2: Simulation of Communication Systems Using MATLAB
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interval are intuitive; for periodic signals you want to cover 3-5 periods; for non-periodic signals, you usually want to cover the non-zero part of the signal. The selection of the step size is crucial for the accuracy of the simulation. You need enough sample points to represent the signal. Usually, the step size is taken to be of the order of one hundredth of the smallest period in the program (Or, the sampling frequency fs = 1/ts should be 100 times the frequency of the signal). In our example, since we are having c(t) of frequency 1000 Hz, we selected fs = 100000, or ts = 0.00001. % Define the functions m(t) and c(t) This is a straightforward step. The function abs stands for | |, while pi=. Note that the signals m and c are now vectors of the same size as t. % Perform the multiplication This is also a straightforward step. However note the dot after m. Why this is necessary here? What would happen if you remove the dot? % Perform full-wave rectification This is again a straightforward step, provided you recognize that full-wave rectification is mathematically equivalent to taking the absolute value. % Create the LPF This operation is frequently encountered in simulating communication systems. A LPF is defined by one parameter, the cutoff frequency. A filter in MATLAB is represented by its transfer function. The transfer function is in general in the form of the division of two polynomials. The filter is completely defined by the coefficients of the polynomial at the numerator and the polynomial at the denominator. These are the vectors a and b respectively in the program. There are many realizations for designing filters. One common realization is Butterworth, which is the one used here, hence the function name butter. The butter function has two arguments. The first argument is the order of the filter. The larger the order the sharper the filter (closer to ideal), but more processing is required. For most of our applications an order of 3-5 should be sufficient. The second argument is a coefficient related to the cutoff frequency. Without going into the details of the derivation, to design a LPF filter of cutoff frequency W, the argument should be set to 2*W*ts, where ts is the time step size of the program. For more details about the command butter , type: >> help butter ; in the MATLAB prompt How many arguments would a BPF require? What are they?
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In the previous step we have only created the filter. To apply the filter to a given signal, we use the function filter. This function has three parameters: the coefficients of the filter a and b, and the vector to be filtered. Note that although we think of the filter operation in frequency domain, the filter function operates on a time-domain vector. The output should as well be taken as a time-domain vector. % Finding the FT of the signals The Fourier Transform of signals can be found in MATLAB using the function fft. It can be used with a single argument, which is the time-domain vector. The fft function yields only the positive side of the spectrum. To get the double-sided spectrum, augment fft by fftshift. Finally, if you are only interested in the amplitude spectrum, augment all by the function abs. The resulting frequency-domain vector will have the same size as the size of the input time-domain vector. % Creating the vector for the frequency axis To plot the frequency spectrum as a function of frequency, you need to create the frequency axis. The available range of frequencies depends on ts, and is given by the relation: f=[-length(t)/2:length(t)/2-1]/(length(t)*ts); % Plotting We leave this step to the student to explore. Use the help command to read about plot subplot, figure, legend, xlabel, ylabel, title and axis commands
Lab Work
1. Create and run the m-file above, and produce Figure (1) and (2). 2. Change m(t) to 2+ sin(2 1000t) and c(t) to cos(2 104t) and the cutoff frequency of the filter to 2 kHz. Redo part 1.
Post-Lab Work
1. Include the m-file and the figures for the work you did in the lab in your report. 2. Using MATLAB, add the signals m(t) = exp(-100|t|) and c(t) = cos(2 103t) then separate them by means of filtering only (LPF and BPF). Provide the m-file and a plot of the sum in time and frequency, and of each of the recovered signals in time and frequency.
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Objective:
verify experimentally the relation between frequency and time domain representation of signals. observe some of none idealities related to noise floor and harmonics. measure the transfer function of a given system (filter) using narrow pulses.
Introduction
A signal is a function that symbolizes a physical variable of interest. Signals can be represented in time or frequency domains (Remember this is only representation). The two representations are related by Fourier Transformation. In this experiment we are going to examine some Fourier Transform properties, namely: Property
Fourier transform of sinusoids Linearity Modulation Time Scaling
1 1 () + 2 2 () ()(0 ) g(at)
Time cos 0
Frequency [( 0 ) + ( + 0 )] 1 1 () + 2 2 ()
Fill in the missing blocks in the table (See Lab Sheet). A system, on the other hand, is a combination and interconnection of several components to perform a desired task. Systems can be characterized by their impulse responses in time domain, or transfer functions in frequency domain. For a subclass of systems, the linear systems, the impulse response (or transfer function) provides a very convenient and straightforward relation between the input and out of the system. One type of system that is frequently-encountered in communications is the filter. A filter is a frequency-selective device that allows a certain frequency band to pass (with high gain), and blocks other bands. Depending on which band it passes, a filter can be classified as low pass (LPF), band pass (BPF) or high pass (HPF).
Lab Work
There are three parts in this experiment. In part I, we verify some of Fourier transform properties. In the second part, we study the effect of filtering on periodic signals. The last part is devoted to identify unknown systems by measuring their impulse response and transfer function.
Kousa & Muqaibel | Exp 3: Representation of Signals & Systems
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To conduct the experiment, the following modules are needed: TUNABLE LPF, AUDIO OSCILLATOR, TWIN PULSE GENERATOR, 100 kHz CHANNEL FILTER.
+
y(t) cos (2 f t) g
m(t)
z(t)
Draw the equivalent Modules and show their interconnection. 2. Using the frequency counter, set the AUDIO OSCILLATOR module to produce a 5-kHz sinusoidal signal and connect it to the system as x(t). 3. Set y(t) as a sinusoidal signal of frequency 2 kHz from the MASTER SIGNALS module. 4. Set f in the above block to 100 kHz. 5. Set g to zero (full counter clockwise) and G to maximum. 6. Obtain the plot of m(t) from both the spectrum analyzer and the scope and compare with your theoretical expectations. Comment on the noise level and harmonics. 7. Vary the frequency of x(t) and observe the impact on both frequency and time domain. Describe what you observe in light of the time scaling property. 8. Re-adjust the frequency of x(t) to 5 kHz and increase g gradually. Observe the change in m(t) on the spectrum analyzer and the oscilloscope. When g is maximum, obtain plots of m(t) waveform and spectrum. What is the property we are trying to prove? 9. Plot the waveform and the spectrum of z(t). 10. Zoom the spectrum of z(t) around 100 kHz and observe its contents. 11. Compare the spectrum of m(t) and z(t) and comment on the modulation property.
1. Apply a square wave signal with frequency of 2 kHz to the TUNABLE LPF module. Set the TUNE and GAIN knobs on the module to maximum (full clockwise), and set the toggle switch to WIDE. Observe the input and the out of the filter in both time and frequency domain. Are they similar? Why?
Kousa & Muqaibel | Exp 3: Representation of Signals & Systems
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2. Turn the TUNE knob to minimum (full counter clockwise). Observing the output, gradually increase the cutoff frequency to allow one harmonic, then two harmonics, then three, and so on. 3. Obtain time and frequency plots for three cases (one harmonic, two harmonics, max filter bandwidth). Adjust the axes to zoom-in the important data and get clear plots. 4. Explain the effect of the filter cutoff frequency on the output waveform.
Post-Lab Work
Use MATLAB to plot and compare the transfer function of (1) Butterworth LPF of cutoff frequency 1 kHz, and order 1, 3, 5. (2) Butterworth BPF of cutoff frequencies 5 kHz and 8 kHz, order 1, 3, 5. You can generate an impulse using the function rectpuls(t,W). You can make W as narrow as time step size ts in order to get an excellent approximation of an impulse.
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Objectives
Understand the features and characteristics of speech signals. Get acquainted with the SPEECH module from TIMS. Perform simple processes on speech signals (filtering, frequency translation), and examine their effect on the sound.
Introduction
Speech is the most frequently encountered message in communication systems. Throughout the lab work we will use a real speech message, whenever appropriate. In order to be prepared, we devote this experiment to study the basic characteristics of speech signals and get acquainted with the SPEECH module of TIMS. We generate speech, or voice in general, by virtue of the vibration of our vocal cords. The sounds we produce are composed of many harmonics, or pitches. Typically, the significant part of human voice occupies the range from 300 Hz 3 kHz. This can be seen from the spectrum of the voice signal. The low-end of the spectrum represents the low-pitch sounds, while the high-end of the spectrum represents the high or sharp pitches. We can hear sounds over a much wider frequency range than the ones we produce. These sounds are called audible signals. A healthy human being can hear frequencies up to 15-20 kHz. This is another proof that we are created to hear more than we talk! The following plug-in modules will be needed for this experiment: SPEECH, TUNABLE LPF, MULTIPLIER, in addition to external signal generator.
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Figure 1: Speech Module Channels 1 and 2 can each record up to 32 seconds of speech from the common MICrophone input. To record speech or other sounds on either channel, set the front panel switch to RECORD and speak clearly into the microphone. The length of your message may vary from a few seconds up to 32 seconds. As soon as you have finished your message, set the switch to the PLAY position. The recorded content will automatically repeat upon switching to PLAY. Note that the length of the recorded message will only be the length of time the switch was in the RECORD position. A third non-recordable channel, LIVE, is also provided where the sound at the MICrophone is continuously output as an electrical signal. A pair of headphones is provided to allow the user to listen to the recorded messages by patching any one of the SPEECH modules outputs to the HEADPHONE AMPLIFIER in the TIMS System Unit. WARNING: DO not put the headphone on if you are not having sound yet. A sudden high-volume sound may harm your inner ear.
Lab Work
This experiment consists of four parts. We start by measuring the audible range of our hearing system! Then, in part II, we record few different voice signals and observe the variations in their spectra. In part III, we examine the effect of filtering on the sound quality. Finally we listen to the effect of slight frequency translation, or modulation.
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4. Increase the frequency of the input gradually. Note the change in the sound. Record the range (lowest frequency and highest frequency) over which you can hear the sound. 5. Let your partner do the same. Who of you has a wider hearing range?
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Post-Lab Work
Using MATLAB: Record a speech message for 5 seconds. (use wavrecord command) Play the message to confirm the recording. (use wavplay command). You can use the pause and disp command to help you control beginning of recording. Calculate and plot the spectrum of the message. Translate the spectrum by 5 kHz. Plot the translated spectrum. Submit the m file and the plots.
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Objectives:
demonstrate the modulation and demodulation process of DSBSC. realize the real-life difficulties and challenges in designing coherent demodulators. examine the implications of the lack of perfect coherence on the recovered signal, and distinguish the different forms of distortion.
Introduction
Double Side Band Suppress Carrier (DSBSC) is one type of Amplitude Modulation. The modulation process is straightforward: the message is multiplied by a high-frequency carrier. The modulated signal occupies double the bandwidth of the baseband signal. Recovering the message signal from the demodulated signal is performed coherently. That is, the demodulated signal is multiplied by a high-frequency sinusoid in perfect synchronization (in phase and frequency) with the incoming carrier. This requirement poses a challenge on the design of the demodulator circuit, as it would then require a part for carrier-recovery. Failing to accomplish perfect synchronization will result in phase mismatch or frequency mismatch, leading to some form of distortion in the recovered signal. Multiplying the modulated signal with a local carrier will produce a baseband signal as well as a signal modulated at double the carrier frequency. Therefore, a LPF is needed at the far end of the demodulator to recover the baseband signal. The following plug-in modules will be needed to run this experiment: AUDIO OSCILLATOR, QUADRATURE UTILITIES, TUNABLE LPF, PHASE SHIFTER and VCO.
Lab Work
This experiment consists of four parts. In Part I we generate the DSBSC signal using singletone message signal. In Part II we demodulate the signal, assuming perfect synchronization of incoming and local carriers. We also examine the effect of improper filtering. In part III and IV, we examine the effect of phase and frequency mismatch, respectively.
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VCO mode. Use the FREQUENCY COUNTER to measure the frequency. Make sure nothing is connected to the TTL input of the counter). 2. Observe, simultaneously, the original signal and the recovered signal, in time and frequency. Describe the effect of frequency mismatch. 3. Try to eliminate the frequency mismatch by fine tuning either oscillator. 4. Replace the 2 kHz message with a speech signal. Increase and decrease the frequency mismatch and describe the effect on the sound quality.
Post-Lab Work
1. You noticed that there is a wide range for the design of the LPF cutoff in order to recover the demodulated signal. Is there a particular value you prefer? Why? 2. You have seen that constant phase mismatch results in no shape distortion of the signal, only magnitude reduction. Prove this mathematically. 3. If the phase mismatch was not constant but time-varying, how would it affect the sound? 4. Implement DSBSC modulation and demodulation (perfectly coherent) in MATLAB for the signal and carrier frequencies used in the experiment. Submit the *.m file, plot of modulated signal and recovered signal in time and frequency. What is the difference between spectra plots in MATLAB simulation and the ones observed on the picoscope.
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Objectives
Demonstrate the modulation and demodulation process of AM signals. Implement the modulation and demodulation of Quadrature Amplitude Modulation (QAM). Examine the sensitivity of QAM to phase errors.
Introduction
AM is the term given to Double Side Band with Carrier modulation. The main advantage of AM over DSBSC is the simplicity of the demodulator circuit: the envelope detector. For that to work, the message signal has to be always positive. Therefore, the message is DC shifted before modulation. This gives rise to the model s(t) = [m(t)+A]cos(2fct) (1)
For the envelope of s(t) to be a true representation of m(t), A > mp where mp is the max negative value of the message. The ratio mp/A defines the modulation index, which varies from 0 to 1. The demodulation of AM signal can be achieved by a simple circuit of a diode, a resistor and a capacitor. The simplicity of the demodulator is the main attraction of AM modulation. However, this is done at the cost of less power efficiency. In terms of bandwidth, AM requires the same band for transmission as DSBSC. Quadrature Amplitude Modulation, on the other hand, allows for twice the bandwidth efficiency of DSBSC or AM. In this scheme two messages are modulated with carries that have the same frequency but at quadrature to each other (i.e. have a phase shift of 900). The two modulated signals are added and transmitted over the same channel. It is easy to show that each of the messages can be perfectly recovered with coherent demodulation at the receiver. The whole system is summarized in Figure 1. Such a scheme, however, is very sensitive to phase errors. Any error will result in one message leaking to the other. The following plug-in modules will be needed to run this experiment: AUDIO OSCILLATOR, ADDER, MULTIPLIER, QUADRATURE UTILITIES, PHASE SHIFTER and UTILITIES.
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m1
X
Cos(ct)
X
cos(ct)
LPF
m1
Sin(ct)
sin(ct)
m2
X
QUADRATURE modulator branch
LPF
m2
Lab Work
This experiment consists of three parts. In Part I, we generate and demodulate an AM signal using single-tone message as well as a real speech signal. In part II, we implement a QAM modulator and demodulator. In part III, we study the effect of phase error on the operation of QAM.
(Tune the DC VARIABLE and the gains of the ADDER G and g until you get the desired settings. Make sure the MULTIPLIER coupling is set to DC) Hint: modulation index = (A-B)/(A+B) (2)
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[EE370 COMMUNICATIONS ENGINEERING] 2. Plot the waveform and the spectrum of the AM signal
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3. Demodulate the signal using the envelope detector (DIODE+ LPF circuit on the UTILITIES module). 4. Plot the demodulated signal. Is it perfectly recovered? 5. Feed the demodulated signal to the HEADHPHONE AMPLIFIER module and plot the signal at the output of the module. Remember that this module has a built in LPF. Is the recovery better? Why? Now we want to examine if the level of improvement achieved in step 5 compared to step 4 is significant to a speech signal. 6. Replace the single-tone message by a speech signal. 7. Listen to the signal with and without the LPF of the HEADPHONE AMPLIFIER module (use the toggle switch of the LPF SELECT). Do you notice any difference in quality? What is the lesson learned?
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3. Vary the phase shift and keep listening while demodulating message 2. What do you notice?
Post-Lab Work
1. Prove Equation (2). 2. Show mathematically that when the carries are not perfectly in quadrature with each other, the two messages leak in one another at the demodulator output. 3. Implement Part I of the experiment (AM Modulation and Demodulation) in MATLAB. Submit the *.m file, and the plots of modulated signal and recovered signal in time and frequency.
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Exp 7: FM Modulation
[EE370 COMMUNICATIONS ENGINEERING]
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Objective:
generate FM signals using VCO. understand the modulator sensitivity and linear range of operation. examine the factors affecting the shape of the spectrum and the bandwidth of FM signals.
Introduction
A simple and direct method of generating an FM signal is by the use of a voltage controlled oscillator -VCO. The frequency of such an oscillator can be varied by the magnitude of an input (control) voltage. The block diagram of VCO-FM generator is shown in Figure 1(a). Figure 1(b) shows a snap shot of an FM signal, together with the message from which it was derived 2.Note particularly that there are no amplitude variations - the envelope of an FM waveform is a constant. For the VCO to work as a frequency modulator, it has to manifest a linear relation between the magnitude of the input signal and the output oscillation. Large signal amplitude may take the system out of its linear range of operation. Therefore a careful design of the deviation sensitivity of the VCO is required to ensure linear operation over the full range of input signal amplitudes. Unlike Amplitude modulation, the bandwidth of FM signals is not determined by the message bandwidth only, but also by message (maximum) amplitude and deviation sensitivity. The product of the last two factors yields frequency deviation. The bandwidth of FM signal can be approximated by (Carsons rule): Bandwidth of FM signal = 2 x (message bandwidth + frequency deviation)
In this figure the frequency deviation is comparable to the carrier for the objective of visualizing the frequency change. In real signals, the frequency deviation is very small compared to the carrier.
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Lab Work
This experiment has four parts. The first part studies sensitivity and the range of linear operation of the voltage controlled oscillator (VCO). In preparation for FM generation in the third part, part II addresses designing the frequency deviation ratio for the modulator. Spectrum analysis and bandwidth estimation will be the subject of the last part. The following modules are needed to complete the experiment: AUDIO OSCILLATOR, VCO.
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Post-Lab Work
We would like to verify our results in Part IV: Spectrum Analysis and Bandwidth Estimation using MATLAB: 1. Use MATLAB to generate an FM signal y(t), let the message signal be x(t)=2cos(2000(2)t). Use a carrier frequency of fc=100 kHz and design the sensitivity factor to get a frequency deviation of 10 kHz. Hint: use the fmmod command available from the Communication ToolboxTM. 2. Plot x(t) and y(t) . 3. Plot the magnitude spectrum for y(t). 4. Change the message frequency to 4 kHz, observe the spectrum and quantify the effect on the bandwidth 5. Change the frequency deviation to 15 kHz (keep message frequency at 2 kHz), observe the spectrum and quantify the effect on the bandwidth 6. Compare Simulation results with the experimental ones.
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Exp 8: FM Demodulation
[EE370 COMMUNICATIONS ENGINEERING]
LAB MANUAL
Objective:
to implement the phase locked loop (PLL) for FM demodulation. to implement frequency discriminator method for demodulating FM.
Introduction
The block diagram of a phase locked loop (PLL) is shown in Figure 1. The principle of operation is simple. Suppose there is an unmodulated carrier at the input. If the VCO was tuned precisely to the frequency of the incoming carrier, 0, then the instantaneous output would be a DC voltage, of magnitude depending on the phase difference between the output of the VCO and the incoming carrier. Now suppose the that the incoming carrier started to drift slowly in frequency. Depending upon which way it drifts, the output voltage will vary accordingly. If the incoming carrier is frequency modulated by a message, the output of the PLL will follow the message.
Figure 1: the PLL FM can be demodulated as well by using a differentiator or a frequency discriminator. Frequency discrimination can be achieved by applying the FM signal to the linear part (transition region) of a BPF as depicted in Figure 2. The output of the discriminator is both FM and AM modulated. The message can be recovered by applying the discriminator output to an envelope detector followed by LPF. The BPF of the 100 kHz CHANNEL FILTERS module has close-to-linear pattern in the band 8090 kHz.
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The following modules are needed to complete the experiment: AUDIO OSCILLATOR, ADDER, MULTIPLIER, UTILITIES, 100 kHz CHANNEL FILTERS, VCO (2 modules, one for the modulator and the second is used for the PLL demodulator).
Lab Work
Part I: FM Demodulation Using PLL
1. Reconstruct the FM modulator as in the previous experiment (FM Modulation). Let the message be 2 kHz from the AUDIO OSCILLATOR, the carrier 100 kHz from VCO, and the modulator VCO GAIN about 25%. 2. Model the PLL demodulator illustrated in Figure 1. For the filter use RC LPF provided in the UTILITIES Module. In the MULTIPLIER module set the toggle switch to AC. Draw the corresponding module diagram. 3. Set the VCO in the demodulator to 100 kHz. Set the GAIN control to mid-range position. 4. Connect the output of the modulator to the input of the demodulator. 5. The PLL may or may not lock on to the incoming FM signal. Tune the GAIN (and if necessary the center frequency) of the PLL-VCO until you obtain lock. Examine the output of the PLL VCO and compare it with the original message. 6. Replace the message from the AUDIO OSCILLATOR by a speech signal and make sure that you can hear the recorded message correctly. Study the effect of varying the frequency f0 and GAIN of the PLL-VCO on the quality of the received speech.
Post-Lab Work:
1) Show mathematically the operation of the frequency discriminator. 2) Comment on the pros and cons of the two demodulation techniques covered in this experiment.
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3) Use the FM signal generated in the LAST experiment post-lab work and write a MATLAB code to demodulate the signal. Plot the transmitted and recovered signal and note down your observations.
Useful MATLAB Function: fmdemod command available from the Communication
ToolboxTM
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Objectives:
Recognize the various processes of PCM encoding. Realize the structure of the PCM stream. Understand the operation of linear and non-linear quantizers.
Introduction
Pulse Code Modulation (PCM) is a method of converting an analog signal into a digital signal (A/D conversion). This is achieved by a PCM encoder via three operations in sequence: sampling, quantization and coding. A step-by-step description of the operations of a standard PCM encoder is as follows: 1. The encoder is driven by a TTL clock. 2. The input analog message is sampled periodically. The sample rate is determined by the external clock and the number of bits per frame (fs= CLK/n+1). 3. Each sample amplitude is compared with a finite set of amplitude levels, called quantization levels. These are distributed within the range V volts. 4. Each sample is assigned a digital (binary) codeword representing the number associated with the quantizing level which is closest to the sample amplitude. The number of bits n in the digital codeword and the number of quantizing levels L are related by the equation L= 2n. 5. The codeword is assembled into a time frame, together with other bits as may be required. In many commercial systems, a single extra bit is added in the least significant bit position. This is alternately a 0 or a 1. These bits are used by subsequent decoders for frame synchronization. The frame is transmitted serially. A typical operation that takes place when performing A/D conversion of speech signals is companding. Companding stands for signal compression at the encoder and expansion at the decoder. In the encoder, compression makes the quantizing levels for small input amplitudes closer than those for large amplitudes, in a logarithmic proportion. At the decoder the reverse action is performed to restore the original amplitude distribution. Companding is particularly advantageous when the message has high peak-to-average amplitude characteristic, as in speech signals. The following modules will be used: PCM ENCODER and AUDIO OSCILLATOR.
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designed for the range 2.0 volts, therefore the input message amplitude must be held within this range. The technical details of the module are described in the TIMS Advanced Modules User Manual. We go briefly over each of the input and output connections which will be used in this experiment. DIGITISING SCHEME SELECT: a three-position toggle switch which selects the 4-bit or 7-bit linear encoding scheme; or the 4-bit companding scheme. FS: frame synchronization, a signal indicating the end of each data frame. Vin: the analog signal to be encoded. PCM DATA: the output data stream. CLK: a TTL input serves as the MASTER CLOCK for the module. Clock rate of this module must be 10 kHz or less (manufacture limitation). For this experiment we will use the 8.333 kHz TTL signal from the MASTER SIGNALS module.
Figure 1: The front panel of the PCM ENCODER Each binary word is arranged in a time frame. The time frame contains eight slots of equal length, and is eight clock periods long. The slots, from first to last, are numbered 7 through 0. These slots contain the bits of a binary word. The least significant bit (LSB) is contained in slot 0. See Figure 2.
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The LSB consists of alternating ones and zeros. These are placed (embedded) in the frame by the encoder itself, and cannot be modified by the user. They are used by subsequent decoders to determine the location of each frame in the data stream, and its length. The remaining seven slots are available for the bits of the binary codeword. Thus the system is capable of a resolution of a maximum of seven bits (128 levels). This resolution, for purposes of experiment, can be reduced to four bits (by front panel toggle switch). The 4-bit mode uses only five of the available eight slots - one for the embedded frame synchronization bits, and the remaining four for the binary codeword (in slots 4, 3, 2, and 1). See Figure 3. 7- bit mode 4-bit mode b7 0 b6 0 b5 b4 b3 0 b4 b3 Code word F R A M b2 b2 E b1 b1 b0 b0
FS
Lab Work
The experiment consists of four parts. In Part I, we set up the module and observe the time frame structure. In Part II, we identify the quantization levels by examining PCM of a DC input using 4-bit linear quantizer. A DC input ensures completely stable oscilloscope displays, and enables easy identification of the quantizing levels. In Part III, we switch to 4-bit companding mode and observe the difference in quantizer behavior to the linear mode. Finally, in Part IV, we consider a more meaningful input; an AC signal.
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2. Feed a DC voltage level from the VARIABLE DC module to Vin of the PCM ENCODER. Turn the knob of the variable DC voltage to the least (negative) value (full counterclockwise). 3. Display on the oscilloscope both the FS signal and the PCM DATA output. Describe the binary sequence of the PCM DATA for three consecutive frames. Give reasons for such sequence. 4. Vary the DC voltage slowly back and forth over its complete range, and note how the data pattern changes in discrete jumps. 5. Adjust Vin to its maximum negative value. Record the DC voltage. You should be getting all zeros for the 4-bit binary number for this DC setting. 6. Gradually increase the amplitude of the DC input signal until you notice a change to the PCM output. Record the binary sequence of the new digital word, and the input amplitude at which the change occurred. 7. Continue this process over the full range of the DC supply. 8. Sketch the input-output characteristics of the quantizer. This is a staircase plot, with the input voltage on the x-axis and the output level (labeled as a binary sequence) on the y-axis. 9. Connect the corners of the stair steps with the best fit. Is the curve linear?
Post-Lab Work
Write a MATLAB code to implement uniform quantization. The quantizer should accept a sampled signal and generate the quantized one. The dynamic range of the quantizer is -V to +V. The number of bits should be controllable. Demonstrate your program by two examples.
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Objective:
implement PCM decoding and understand its operation. appreciate the importance of data structure and synchronization. evaluate the effect of companding.
Introduction
In the previous experiment, the PCM encoder was examined. To complete the picture, the PCM decoder will be investigated in this experiment. Upon reception of a PCM sequence, the PCM decoder: 1. extracts a frame synchronization signal FS from the data itself (from the embedded alternate ones and zeros in the LSB position), or uses an FS signal borrowed from the transmitter . 2. extracts the binary number, which is the coded (and quantized) amplitude of the sample. 3. identifies the quantization level which this number represents. 4. generates a voltage proportional to this amplitude level. 5. presents this voltage to the output Vout. The voltage appears at Vout for the duration of the sampling period. Note that, it is not possible to recover a distortionless message from these samples. They are flat top, rather than natural samples. The decoder itself has introduced no distortion of the received signal, but the signal from the PCM encoder is already an inexact version of the signal at the input of the encoder. Message reconstruction can be improved by low pass filtering. To complete the experiment the following modules are needed: PCM DECODER, PCM
ENCODER, SPEECH MODULE, TUNABLE LPF.
PCM Modules
In this experiment we will use the PCM ENCODER and DECODER modules. The decoder is driven by an external clock, borrowed from, and so synchronized to, that of the encoder. Frame synchronization may be achieved either by extracting the FS signal from the embedded information in the received data, or by borrowing it externally from the encoder. A toggle switch (FS SELECT) in the PCM DECODER allows switching between the two modes. Though an external signal can be used for demonstrating the PCM operation, the PCM encoder is equipped with test periodic signals (SYNC MESSAGE) of frequencies which are fraction of the clock rate. These signals are synchronized with the clock and, therefore,
Kousa & Muqaibel | Exp 10: PCM Decoding
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provide improved triggering. The frequency of these signals can be set from a dip switch SW2 on the board of the PCM ENCODER. Note that these signals are not pure sinusoids. You can read more about the PCM DECODER module in the TIMS Advanced Modules User Manual.
Lab work
This experiment consists of four parts. In part I, we set up the PCM encoder and decoder. In parts II to IV, we examine quantization, signal recovery, and companding, respectively.
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2. 3. 4. 5. 6. 7. 8.
Connect an external triangular signal of amplitude 4 V (peak-to-peak). Adjust the frequency to 20 Hz. Do not forget to connect the ground of the source to the GND of TIMS. 3 Set the Encoder/DECODER to 4-bit LINEAR. Slow down the oscilloscope sweep speed to 10 ms/div. Observe the decoder output signal. Plot both the input to the PCM ENCODER and the output of the PCM DECODER. Measure the delay between the input and the output. How many samples are taken in one period of the input signal? Using the Picoscope >tools>math channels>A-B, obtain a plot of the error (output input). Explain how the delay affects the error. Change the coding scheme from 4-bit to 7-bit. Redo step 7.
Message reconstruction may be improved using careful low pass filtering. 9. Connect the output of the PCM DECODER to a TUNABLE LPF. Now observe the original signal (input to the PCM ENCODER) together with the output of the filter and start adjusting the cutoff frequency. 10. Using the FREQUENCY COUNTER, find the cutoff frequency of the TUNABLE LPF for best possible recovery. (Recall that the cutoff frequency = frequency counter readying/100). Plot the input and output waveforms. 11. Adjust the cutoff frequency of the TUNABLE LPF to 50Hz, then 500Hz. Plot the waveforms and comment on the type of distortion in the output signal.
(Note: if no external source is available, the SYNC MESSAGE of the PCM ENCODER can be used after setting its peak-to-peak voltage between -2 and +2 using a BUFFER AMPLIFIER. The frequency of the SYNCH MESSAGE is determined by the SW2 on board, Dip switches on board SW2: set left hand toggle UP, right hand toggle DOWN use 65 Hz).
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to the case of 4-bit LINEAR in both modules. First guess the output, and then verify your guess experimentally.
Post-Lab Work
When testing the overall PCM encoding decoding system, we examined a DC signal with both 4-bit and 7 bit-linear quantizers (Part II). Describe the behavior if 4 bitnonlinear (compand) was used? Using MATLAB, consider a sinusoidal signal with amplitude of V and frequency 1 kHz, sampled at 100 kHz. With the help of the Quantizer you designed in the post-lab work of the previous experiment, plot the quantized output in two cases: 4-bit and 8-bit quantizers. Compare the input with the output and plot the error in the two cases. Comment.
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Objective:
get familiar with definitions and properties of commonly used line codes. evaluate the spectrum content for different line codes. demonstrate the advantage of differential encoding through signal inversion test. measure the delay associated with practical digital systems.
Introduction
Digital data can be transmitted by various Line codes. Line codes are waveform patterns of voltage or current used to represent the 1s and 0s. Each line code has its advantages and disadvantages. Among other desirable properties, a line code is preferred to have the following: Bandwidth efficiency; the possibility of transmitting at a higher rate than other schemes over the same bandwidth. Power efficiency: For a given bandwidth and quality, the transmitted power should be as small as possible. No DC component; this allows AC coupling (capacitor or transformer) between stages (as in telephone lines). Spectrum shaping; this is important in telephone line applications, for example, where the transfer characteristic has heavy attenuation below 300 Hz. Synchronization; where bit clock recovery can be simplified. Error detection capabilities; It should be possible to detect some patterns of errors.
System Modules
To complete the experiment the following modules are needed: SEQUENCE GENERATOR, LINE-CODE ENCODER, and LINE-CODE DECODER. SEQUENCE GENERATOR Using a common external clock signal, the SEQUENCE GENERATOR produces two independent pseudorandom sequences X and Y. In this experiment we need only one output. The SEQUENCE GENERATOR will be clocked by B.CLK from the LINER ENCODER module. A SYNC output is provided which is coincident with the start of the sequences. The synch out from the SEQUENCE GENERATOR can be used to trigger the oscilloscope. The sequences may be stopped and restarted at any time via front panel controls. Sequences X and Y are available as either standard TTL or analog level output. The SEQUENCE GENERATOR is a basic module and you can read more about it in TIMS 301 User Manual. The module and its block diagram are shown in the figure below.
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LINE ENCODER and LINE DEODER Modules In a digital transmission system line encoding is the final digital processing performed on the signal before it is connected to the analog channel, although there may be simultaneous bandlimiting and wave shaping. Thus in TIMS the LINE-CODE ENCODER accepts a TTL input, and produce an output that is suitable for transmission via an analog channel. The TIMS LINE-CODE DECODER decodes it back to the binary TTL format. The LINE-CODE ENCODER serves as a source of the system bit clock. It is driven by a master clock (M.CLK) at 8.3 kHz (from the TIMS MASTER SIGNALS module). The LINE-CODE ENCODER module divides M.CLK by a factor of four, in order to derive some necessary internal timing signals at a rate of 2.083 kHz (B.CLK). The latter becomes a convenient source for use as the system bit clock. The reason we are using a slower clock (clock/4) is that the encoder requires some cycles to provide the proper output and hence the data should arrive at a slower rate. Because the LINE-CODE DECODER has some processing to do, it introduces a time delay. To allow for this, it provides a re-timed clock (STROBE) if required by any further digital processing circuits (eg, for decoding, or error counting modules). For a TTL input signal the following output formats are available from the LINE-CODE ENCODER: NRZ-L, NRZ-M, UNI-RZ, BIP-RZ, RZ-AMI, Bi-L (Manchester), DICODE-NRZ. Rather than defining each of the previous codes, you will find what they mean experimentally. Resetting the LINE-CODE ENCODER and the LINE-CODE DECODER is recommended every time after the clock is connected, or interrupted You can read more about the LINE ENCODER and LINE DECODER Modules in the Advanced Modules Manuals.
Lab Work
This experiment has three parts. The first part takes you through defining the available line codes and observing their spectra. The second part verifies the overall operation of the line encoding-decoding system. Differential encoding and signal inversion test are examined in part III.
Kousa & Muqaibel | Exp 11: Line Coding
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Post-Lab Work
1. Based on your experimental understanding of the law of transformation of the line codes, complete the figure below. 2. Comment on the DC component, synchronization capability and error detection capability of each code.
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Objective:
modulate binary digital signals using FSK. understand the design issue related to choosing the frequencies in the FSK signal. differentiate continuous phase FSK (CPFSK) from general FSK. realize ASK and PSK as special cases of FSK.
Introduction
Digital modulation techniques include: amplitude shift keying (ASK), frequency shift keying (FSK), and phase shift keying (PSK). In this experiment we will concentrate on FSK. The other techniques can be related to it. The block diagram of Figure 1 illustrates the operation of an FSK generator.
Figure 1: an FSK transmitter In principle the three frequencies f1, f2, and fs are independent. In practice this is often not so - there are certain advantages in having them related in some way (eg, as submultiples). Oscillator #1 and #2 can be taken from the same source (say a VCO), whose frequency is changed by the message, leading to a continuous phase output (CPFSK). This is illustrated in Figure 2, which shows a VCO as the source of f1 and f2, and the corresponding CPFSK output waveform.
To complete this experiment the following modules are needed: ADDER, AUDIO OSCILLATOR, DUAL ANALOG SWITCH, SEQUENCE GENERATOR, and VCO.
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System Modules
VCO as FSK generator
The VCO module was used before for FM generation. It can also be sued for FSK generation. In this case it must be switched to the FSK mode using the on-board switch SW2. A +5V DC to the DATA input of the VCO module allows the setting of f1 with RV8, while a 0V allows the setting of f2 using RV7. These frequencies will be in the audio range with the front panel switch set to LO, or near 100 kHz when set to HI. The two other front panel controls have no influence in this FSK mode. See the TIMS User Manual for details of FSK mode for the VCO.
Lab Work
This experiment has two parts. The former examines the generation of a continuous phase frequency shift keyed (FSK) signal, while the latter demonstrates a general FSK modulator with higher degree of freedom.
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1. Connect the circuit in Figure 5. o f1 is 2 kHz obtained from the MASTER SIGNALS modules. o f2 is tunable and obtained from VCO. Make sure that the on board switch SW2 is set to VCO mode. Also set the toggle switch to LO. o fs, the bit rate of the message, is determined by the AUDIO OSCILLATOR. o The SEQUENCE GENERATOR produces a TTL (0 , 5V) sequence of rate fs. o DC shift of around -2.5V is needed to adjust the TTL level to V in order to generate FSK. Set the DC source to full counter clockwise (about -2.5V). The system generates two carriers at f1 and f2. The DUAL ANALGO SWITCH will alternate between the two according to the message sequences. 2. Vary f2. How does that affect the waveform and the spectrum of the modulated signal? 3. Set fs to minimum then to maximum. Which setting generates a better modulated signal? Why?
Kousa & Muqaibel | Exp 12: Digital Modulation: FSK
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4. Zoom in to a transition between any two bits. Is the phase continuous? Why? How does that affect the bandwidth? 5. Set the VARIABLE DC to zero. What type of modulated signal is generated? Why? 6. How can you use the above system to generate PSK?
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The following Regulations and Safety Rules must be observed in all concerned laboratory locations. It is the duty of all concerned parties who use any electrical laboratory to take all reasonable steps to safeguard the HEALTH and SAFETY of themselves and all other users and visitors. Be sure that all equipment is properly working before using them for laboratory exercises. Any defective equipment must be reported immediately to the Lab. Instructors or Lab. Technical Staff. Students are allowed to use only the equipment provided in the experiment manual or equipment used for senior project laboratory. Power supply terminals connected to any circuit are only energized with the presence of the Instructor or Lab. Staff. Students should keep a safe distance from the circuit breakers, electric circuits or any moving parts during the experiment. Avoid any part of your body to be connected to the energized circuit and ground. Switch off the equipment and disconnect the power supplies from the circuit before leaving the laboratory. Observe cleanliness and proper laboratory housekeeping of the equipment and other related accessories. Wear proper clothes and safety gloves or goggles required in working areas that involves fabrications of printed circuit boards, chemicals process control system, antenna communication equipment and laser facility laboratories. Double check your circuit connections specifically in handling electrical power machines, AC motors and generators before switching ON the power supply. Make sure that the last connection to be made in your circuit is the power supply and first thing to be disconnected is also the power supply. Equipment should not be removed, transferred to any location without permission from the laboratory staff. Software installation in any computer laboratory is not allowed without the permission from the Laboratory Staff. Computer games are strictly prohibited in the computer laboratory. Students are not allowed to use any equipment without proper orientation and actual hands on equipment operation. Smoking and drinking in the laboratory are not permitted.
All these rules and regulations are necessary precaution in Electrical Laboratory to safeguard the students, laboratory staff, the equipment and other laboratory users.
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Objectives:
1- Perform basic operations on sinusoids in the lab. 2- Demonstrate time and frequency domain representations of signals. 3- Realize practical limitations.
(a)
(b)
Figure 2: (a) Waveform and (b) Spectrum of the sum of 2 kHz and 3 KHz sinusoid
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Discussion and Comments (Answers to questions posed in the lab manual and/or Lab Sheets): (1) The sum is period with period 1 msec. (2) The noise level in the spectrum plot is -40 dB (3) The sinusoids appeared in the spectrum as spikes at 2 and 3 kHz. (4) (5) Part II: The Product of Two Sinusoids The following block diagram was implemented in the lab
Signal 1
Signal 2
M U L T I P L I E R
Figure 3. Block Diagram of Sinusoids Multiplier Plots: Place Waveform Here Place Spectrum Here
Figure 4: (a) Waveform and (b) Spectrum of the product of 2 kHz and 3 KHz sinusoid Discussion and Comments (Answers to questions posed in the lab manual and/or Lab Sheets): (1) The product is period with period 1/3 msec. (2) The noise level in the spectrum plot is -40 dB. No change to Part I (3) The product appeared in the spectrum as spikes at 1 and 5 kHz. (4) . (5) ..
Kousa & Muqaibel | Appendix B: Sample Report
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Conclusion:
In this experiment we implemented the basic operations on sinusoids. The experiment was smooth, and all results were as expected. This is my first time to see and quantify the noise of the electrical and electronic circuit. I had a little difficulty in working on the picoscope, but hopefully with time I will be better.
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