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Page Abstract .. Acknowledgements .. Table of Contents List of Illustration Glossary of Key Terms .. 1. 1.1 1.2 1.3 2. 2.1 2.2 2.3 3. 3.1 3.2 Introduction Purpose . Scope Objective .. Theorical Background Continuous Time Signals and its Fourier Transforms . Filte r .. MATLAB .. Procedure, Findings and Discussions Preparation to run MATLAB . Four ier Tra ns for m o f Co nt inuo us- Time S igna ls 3.2.1 Four ier Tra ns for m a nd spec tr um p lot o f s ine a nd cos ine .. 3.2.2 Disc uss io n o n F ind ings .. 3.3 Four ier Tra ns for m o f Recta ngular P ulses a nd t he ir p lo ts .. 3.3.1 Plo t o f Rect a ngular P ulses a nd it s Spec tr ums . 3.3.2 Disc uss io n o n F ind ings .. 3.4 Filte r ing o f Rea l- Life S igna ls . 3.4.1 Filte r ing o f Spe ec h . 3.4.2 Disc uss io n o n F ind ings .. 3.4.3 Filte r ing o f I ma ge .. 3.4.4 Disc uss io n o n F ind ings . 4. Concl us io n References Appendix i iii iv v vi 1 1 1 2 3 3 3 4 5 5 5 5 6 7 7 8 9 10 10 12 13 14

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LIST OF ILLUSTRATIONS

Page Figure 1.1 MATLAB software used in experiment .. Figure 2.1 A Continuous time signal .. Figure 2.2 Spectrum of a signal Figure 3.1 Plot of sin(t) in time domain . Figure 3.2 Plot of sin spectrum .. Figure 3.3 Plot of Cos(t) in time domain Figure 3.4 Plot of Cos spectrum . Figure 3.5 P lo t o f rec ta ngular p ulse Figure 3.6 Rectangular Pulses with Difference Pulse Width . Figure 3.7 Spect r um o f Rec ta ngular P uls es Figure 3.8 Spect r um o f speec h s igna l Figure 3.9 Spect r um o f filte red speec h s igna l wit h LPF Figure 3.10 Or igina l Speec h S igna l a nd F ilt ered S igna l . Figure 3.11 Or igina l P ict ur e Figure 3.12 P ic t ure O utp ut wit h Low Pa ss F ilte r Figure 3.13 P ic t ure O utp ut wit h High Pas s F ilter ... 1 3 4 6 6 7 7 9 9 9 11 11 11 12 13 13

GLOSSARY OF KEY TERMS Amplitudes Amplitude is a nonnegative scalar measure of a wave's magnitude of oscillation, that is, magnitude of the maximum disturbance in the medium during one wave cycle. Continus-Time Signal A continuous signal or a continuous-time signal is a varying quantity (a signal) that is expressed as a function of a real- valued domain, usually time. The function of time need not be continuous. The signal is defined over a domain, which may or may not be finite, and there is a functional mapping from the domain to the value of the signal. The continuity of the time variable, in connection with the law of density of real numbers, means that the signal value can be found at any arbitrary location, t0. Discrete-Time Signal A discrete signal or discrete-time signal is a time series, perhaps a signal that has been sampled from a continuous-time signal. Unlike a continuoustime signal, a discrete-time signal is not a function of a continuous-time argument, but is a sequence of quantities, that is, a function over a domain of discrete integers. Each value in the sequence is called a sample. Filter Filter (Electronic Filter) is electronic circuit which perform signal processing functions, specifically intended to remove unwanted signal components and/or enhance wanted ones. Electronic filters or audio filters can be: passive or active, analog or digital, discrete-time (sampled) or continuous-time, linear or non- linear, infinite impulse response (IIR type) or finite impulse response (FIR type). The most common types of electronic filters are linear filters, regardless of other aspects of their design. Fourier Transform The Fourier transform, named after Joseph Fourier, is a reversible (i.e., invertible) integral transform of one function into another. The second function, which is called a Fourier transform, gives the coefficients of sinusoidal basis functions (vs. their frequencies) whose linear combination (summation or integral) produces the original function. Frequency Frequency is the measurement of the number of times that a repeated event occurs per unit of time. It is also defined as the rate of change of phase of a sinusoidal waveform. Frequency Domain Frequency domain is a term used to describe the analysis of mathematical functions or signals with respect to frequency. Speaking non-technically, a time domain graph shows how a signal changes over time, whereas a frequency domain graph shows how much of the signal lies within each given frequency band over a range of frequencies. A frequency domain representation can also include information on the phase shift that must be applied to each sinusoid in order to be able to recombine the frequency components to recover the original time signal. The frequency domain relates to the Fourier transform or Fourier series by decomposing a function into an infinite or finite number of frequencies. This is based on the concept of Fourier series which is that any waveform can be expressed as a sum of sinusoids (sometimes infinitely many.) The impulse response of a system is its output when Impulse Response Function presented with a very brief signal, an impulse. While an impulse is a difficult concept to imagine, and an impossible thing in reality, it represents the limit case of a pulse made

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infinitely short in time while maintaining its area or integral (thus giving an infinitely high peak). While this is impossible in any real system, it is a useful concept as an idealization. Inverse Fourier Transform Refer to inverse process of the Fourier Transform and thus recovers a function from its Fourier transform back to time domain. Low-Pass Filter(LPF) A low-pass filter is a filter that passes low frequencies well, but attenuates (or reduces) frequencies higher than the cut-off frequency. The actual amount of attenuation for each frequency varies from filter to filter. It is sometimes called a high-cut filter, or treble cut filter when used in audio applications. MATLAB A numerical computing environment and programming language created by The MathWorks , MATLAB allows easy matrix manipulation, plotting of functions and data, implementation of algorithms, creation of user interfaces, and interfacing with programs in other languages. Although it specializes in numerical computing, an optional toolbox interfaces with the Maple symbolic engine, making it a full computer algebra system. Spectrums A graphic or photographic representation of the range of values of a quantity or set of related quantities distributed in various frequencies.

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EE2072: Laboratory 2B School of Electrical and Laboratory Experiment Report (L226) Electronic Engineering _____________________________________________________________________

1. INTRODUCTION

1.1 Purpos e

This report documents the laboratory experiment on Fourier representation of signals and filtering. Fourier representation of signals is important in signal analysis. MATLAB provide great computing tools for such analysis.

Signals and Systems are important in electronics engineering, especially in communication field. Signal analysis is one of the most important parts of communication system design. There are a lot of methods to perform the signal analysis like Fourier Transform Analysis, Laplace Transform Analysis or Z Transform. However, this experiment only emphasizes on Fourier Transform Analysis on signals and Fourier representations of signals are mainly focus. Filters, another major topic in electronics engineering, also play an important role. Filters can be classified in different point of view such as: Low Pass Filter, High Pass Filter, Band Pass Filter, Band Stop Filter, Passive Filter, Active Filter or even the Digital Filters which may used microprocessor to achieve the filtering job. This experiment, as an undergraduate level, will only discussed about Low Pass Filter and High Pass Filter, the two very fundamental filters in signals and system analysis and design. On the other hand, MATHLAB provide powerful tools for signals and systems analysis and design. It is not possible to cover all the MATHLAB functions in this experiment. Only the basic command will be used in this experiment and advanced MATLAB functions like Simulink are not cover in this experiment.

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EE2071: Laboratory 2A School of Electrical and Laboratory Experiment Report (L217) Electronic Engineering _____________________________________________________________________ 1.3 Obje ct ive

The main objectives of this experiment are: v v v v To learn the MATLAB environment to perform basic signals analysis. To learn the Fourier transform of signals using MATLAB environment. To learn the plotting of spectrums of different type of signals. To learn the filtering applications of signals using sound and picture as an example.

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EE2071: Laboratory 2A School of Electrical and Laboratory Experiment Report (L217) Electronic Engineering _____________________________________________________________________

2. THEORICAL BACKGROUND

2.1 Conti nuo us Ti me Sig nals a nd its Fo urie r Tra ns fo rms

A continuous signal or a continuous-time signal is a varying quantity that is expressed as a function of a real- valued domain, usually time. The function of time need not be continuous. The signal is defined over a domain, which may or may not be finite, and there is a functional mapping from the domain to the value of the signal. The continuity of the time variable, in connection with the law of density of real numbers, means that the signal value can be found at any arbitrary location, t0 .

The continuous time signals can be represented in frequency domain using the continuestime Fourier transform (CTFT). Spectrum, the Fourier transform of signals, denoted by F(j) is defined as:

F ( jw ) =

f (t )e

- jwt

dt

(1)

2.2

Fi lte r

Filter (Electronic filter) is electronic circuit which perform signal processing Functions, specifically intended to remove unwanted signal components and/or enhance wanted ones. Electronic filters or audio filters can be: passive or active, analog or digital, discrete-time (sampled) or continuous-time, linear or non- linear, infinite impulse response (IIR type) or finite impulse response (FIR type).

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EE2071: Laboratory 2A School of Electrical and Laboratory Experiment Report (L217) Electronic Engineering _____________________________________________________________________ Therefore, a filter passes some of the signals frequencies and stops others frequencies leading the output signal does not have all the frequencies of the input signal. Spectrums are plotted to analyze which frequency components are being filtered and which are present in output.

The name MATLAB stands for matrix laboratory. MATLAB is a high-performance language for technical computing. It integrates computation, visualization, and programming in an easy-to-use environment where problems and solutions are expressed in familiar mathematical notation. Created by The MathWorks , MATLAB allows easy matrix manipulation, plotting of functions and data, implementation of algorithms, creation of user interfaces, and interfacing with programs in other languages. Although it specializes in numerical computing, an optional toolbox interfaces with the Maple symbolic engine, making it a full computer algebra system. The details of the theorical background are not discussed there in order not to overload the report. The reference list at the end of the report should be referred when more detailed are required.

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EE2071: Laboratory 2A School of Electrical and Laboratory Experiment Report (L217) Electronic Engineering _____________________________________________________________________

3.1 Pre pa rat io n to run M ATLAB

The school laboratory had pre written the MATLAB function, diracplot.m, to plot the Fourier transform Spectrum of the signals. speech_dft.wav and flowers.tif files were provided for filter analysis. These three files were copy to MATLAB working directory C:\MATLAB6p5\work. 3.2 Fo urie r Tra ns fo rm of Co nt i nuous - Ti me Sig nals

A periodic signal f = 3Sin 2 x - 2Cos 3 x is defined as symbolic function and plotted the time domain signal using MATLAB ezplot command. Fourier transformed of f is found by fourier command and inverse Fourier transform of F was found using ifourier command, taking note that the result of inverse Fourier transform of F was the original f = 3Sin 2 x - 2Cos 3 x . The detailed procedure is listed in Listing 3.1. Plot of f and spectrum of F were shown in figure 2.1 and figure 2.2 respectively. Listing 3.1 1 2 3 4 5 6 7 8 9 x=sym('t'); f = sym(3*sin(2*x)-2*cos(3*x)); figure (1) ezplot(f) ezplot(f,0,20) F=fourier(f) f2=ifourier(F); figure(2) diracplot(F)

3.2.1 Fo urie r Tra ns fo rm a nd s pe ct rum plo t of s i ne and cos ine The Fourier transforms of sine and cosine functions were calculated and their spectrums were plotted for better understanding of MATLAB, signal analysis and spectrum. The negative frequencies were observed (figure 3.2) and note that this was because of mathematical equation: Sine(J ) = x and Sine ( -J ) = - x . In real world, there will not be such negative frequency and taking note that the Y-axis refer to the magnitude and thus, only have the positive value is presented.

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EE2071: Laboratory 2A School of Electrical and Laboratory Experiment Report (L217) Electronic Engineering _____________________________________________________________________

3.2.2 Dis cus s ion on Fi ndi ngs The continuous time signal, Sine(t ) = x for example, is the function of time (t). So, when the function was plotted, the X-axis (the horizontal axis) refers to input to the function, t in this case, and the Y-axis (the vertical axis) refers to the value of function at respective input value.

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When the spectrum is plotted, the X-axis refers to frequency, the domain of the function, and the Y-axis refer to magnitude of spectrum, the range of the function. The spectrum plot of Sine and Cosine Signals are the same because of both signals having same frequency and amplitude. Period, loosely speaking, is the time taken to complete one cycle of signal before it repeats. In figure 2.1, the period of the signal is about 6 second and it is related to sinusoid components by mean of the one complete cycle.

3.3

t In Mathematics, the unit rectangular function rect ( ) is defined as: T t t rect ( ) = 1 for all | t | T and rect ( ) = 0 otherwise. Therefore T refer to the width of 2 T T the rectangular pulse. rectpuls defined in MATLAB is not a symbolic function because there is no variable t or x and therefore, the MATLAB function fourier, which argument needs to be a symbolic function, could not be used to calculate the Fourier transforms of the rectangular function.

To plot the spectrum of the rectangular function, the Fourier transform integral was calculated by the basic definition and their spectrums are plotted. Listing 3.2 lists the detailed procedure.

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EE2071: Laboratory 2A School of Electrical and Laboratory Experiment Report (L217) Electronic Engineering _____________________________________________________________________ Listing 3.2 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 tarray= -10:0.1:10; T=2; figure(3) plot(tarray,rectpuls(tarray,T)) hold tarray=-10:0.01:10; plot(tarray,rectpuls(tarray,T)) x=sym('t'); w=sym('w'); P=int(exp(-j*w*x),x,-T/2,T/2); figure(4) T=2; plot(tarray,rectpuls(tarray,T), 'b') hold T=5; plot(tarray,rectpuls(tarray,T),'r') T=2; P=int(exp(-j*w*x),x,-T/2,T/2); figure(1) ezplot(P) T=5; P=int(exp(-j*w*x),x,-T/2,T/2); hold ezplot(P)

3.3.2 Dis cus s ion on Fi ndi ngs MATLAB int function was used to perform the integration to get the Fourier transform of the rectangular pulse. The result is same as equation 8c in lab manual. Before plotting the spectrum of the signal, the rectangular pulse was plotted with difference step size. From the figure 3.5, it is clearly seen that the step size plays an important role in highly precise systems. The blue pulse was plotted with step size of 0.1 and finer plot was obtained with step size of 0.01.

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In order to analysis the spectrum of rectangular pulse, two different rectangular pulses were defined and their spectrum was plotted.

Fig ure 3.7 Spe ct rums of Re ctang ula r Puls e s It is clearly that when the rectangular pulse is decreased (from red to blue in figure 3.6), the width of the frequency spectrum is increased (from red to blue in figure 3.7). 3.4 Fi lte ri ng of Re al-Li fe Signa ls

Filters, as introduced in section 2.2, can be obtained in MATLAB with fir1 function. In this experiment, a low pass filter with cut-off frequency 0.4 was used to demonstrate the affect of using filter on real- life signals.

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EE2071: Laboratory 2A School of Electrical and Laboratory Experiment Report (L217) Electronic Engineering _____________________________________________________________________ 3.4.1 Fi lte ri ng of S pe e ch A speech file, speech_dft.wav , was used as input to the filter and output speech signal was examined. Input spectrum and output spectrum were plotted on different figure as well as in the same figure. Listing 3.3 lists the detailed procedure on MATLAB. Listing 3.3 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 input = wavread('speech_dft.wav'); myfilter = fir1(20,0.4,'low'); output = filter(myfilter,1,input); sound(input,22050) sound(output,22050) X1=fft(input); X2=fftshift(X1); X3=abs(X2); figure(5) fr=-11024.8:0.200394:11025; plot(fr,X3,'r') Y1=fft(output); Y2=fftshift(Y1); Y3=abs(Y2); figure(6) plot(fr,Y3,'b') figure(7) plot(fr,X3,'r') hold plot(fr,Y3,'b')

3.4.2 Dis cus s ion on Fi ndi ngs Listing 3.3 outputs are shown in Figure 3.8, 3.9 and 3.10. In Figure 3.8, the input sound signals spectrum is plotted and observed that the frequency range is from -1 to 1 which means the full frequency range. This input signal was inputted to the low pass filter called myfilter (defined in line 2 of listing 3.3) which have cut-off frequency of 0.4 and the output signals spectrum was plotted. Since the output signal was filter through the low pass filter with 0.4 cut off frequency, only the signal within the frequency range of -0.5 to 0.5 were present at the output spectrum.

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Observed clearly with the Figure 3.10, when the two signals were plotted together, it is clearly see that, the signal was starting to attenuate at about 0.3 and totally attenuated at about 0.5. It is totally expected, since myfilter is defined as low pass filter with order 20 and 0.4 cut-off frequency. Cut-off frequency is determine by half power point and Figure 3.10 also explained that the ideal filter could not be obtained in real.

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EE2071: Laboratory 2A School of Electrical and Laboratory Experiment Report (L217) Electronic Engineering _____________________________________________________________________ 3.4.3 Fi lte ri ng of I mage Another filtering application was applied in an image file called flowers.tif . imread command is used to read the image file and pass to the filer. Figure 3.11

Listing 3.4 list the details procedure on how the filtering of image is performed in MATLAB environment. Listing 3.4 1 2 3 4 5 6 7 8 9 10 11 12 13 14 myfilter = fir1(20,0.2,low); inp2 = imread(flowers.tif,tif); myfilter2 = ftrans2(myfilter); outp2 = imfilter (inp2,myfilter2); figure(6) imshow(inp2) figure(7) imshow(outp2) myfilter = fir1(20,0.2,high); inp3 = imread(flowers.tif,tif); myfilter3 = ftrans2(myfilter); outp3 = imfilter (inp2,myfilter3); figure(8) imshow(outp3)

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Fig ure 3.12 Pict ure Out put wit h Lo w Pas s Fil te r (Cut-o ff f re que ncy 0.2) 3.4.4 Dis cus s ion on Fi ndi ngs The output of the low pass filter with cut-off frequency of 0.2 is displayed in Figure 3.12. It is obviously that the output figure is much unclear compare with original picture due to the filtering effect.

Fig ure 3.13 Pict ure Out put wit h Hig h Pas s Filte r (Cut-o ff f re que ncy 0.2) The same input picture was applied to the high pass filter and the output picture is shown on Figure 3.13. Comparing two outputs, low-pass filters output tends to white color and highpass filters output tends to black color when most of the frequencies components are filtered.

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4. CONCLUSION

This experiment introduced the powerful signal analysis tool, MATLAB, to perform the Fourier Transforms of the functions and plotting of spectrums of difference signals. Fourier transform provide a great mathematical tool when converting the functions from time domain to frequency domain. The concept of frequency domain and time domain were studies and the frequency domain spectrums were highly focused. Filter application on real life signals were also studied. Sound signals were passed to lowpass filter and their spectrums were plotted. Lastly, a picture file was filtered with both lowpass filter and high-pass filer and their results were study.

****************************

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REFERENCES

[1] Laboratory Manual, Experiment No. 217, Fourier Representation of Signals and Filtering. Nanyang Technological University, School of Electrical and Electronic Engineering. (Unpublished) [2] http://en.wikipedia.org/wiki/

[3] EE2010 Signal and System, Lectures note, Nanyang Technological University, School of Electrical and Electronic Engineering. (Unpublished) [4] MATLAB documentations

[5] M.J. Roberts, Signals and Systems Analysis Using Transform Methods and MATLAB, International Edition 2003, Mc Graw Hill.

FOURIER : Fourier integral transform F = FOURIER(f) is the Fourier transform of the sym scalar f with default independent variable x. The default return is a function of w.

SYM : Construct symbolic numbers, variables and objects S = SYM(A) constructs an object S, of class 'sym', from A. If the input argument is a string, the result is a symbolic number or variable. If the input argument is a numeric scalar or matrix, the result is a symbolic representation of the given numeric values. IFOURIER Inverse Fourier integral transform f = IFOURIER(F) is the inverse Fourier transform of the scalar sym F with default independent variable w. The default return is a function of x. The inverse Fourier transform is applied to a function of w and returns a function of x: F = F(w) => f = f(x). If F = F(x), then IFOURIER returns a function of t: f = f(t). By definition, f(x) = 1/(2*pi) * int(F(w)*exp(i*w*x),w,- inf,inf) and the integration is taken with respect to w. EZPLOT : Easy to use function plotter EZPLOT(f) plots the expression f = f(x) over the default domain -2*pi < x < 2*pi. EZPLOT(f, [a,b]) plots f = f(x) over a < x < b. For implicitly defined functions, f = f(x,y) FIGURE : Create figure window FIGURE, by itself, creates a new figure window, and returns its handle. FIGURE(H) makes H the current figure, forces it to become visible, and raises it above all other figures on the screen. If Figure H does not exist, and H is an integer, a new figure is created with handle H. SIMPLIFY : Symbolic simplification SIMPLIFY(S) simplifies each element of the symbolic matrix S.

INT : Integrate INT(S) is the indefinite integral of S with respect to its symbolic variable as defined by FINDSYM. S is a SYM (matrix or scalar). If S is a constant, the integral is with respect to 'x'. WAVREAD : Read Microsoft WAVE (".wav") sound file Y=WAVREAD(FILE) reads a WAVE file specified by the string FILE, returning the sampled data in Y. The ".wav" extension is appended if no extension is given. Amplitude values are in the range [-1,+1].

FILTER : One-dimensional digital filter Y = FILTER(B,A,X) filters the data in vector X with the filter described by vectors A and B to create the filtered data Y. The filter is a "Direct Form II Transposed" implementation of the standard difference equation: a(1)*y(n) = b(1)*x(n) + b(2)*x(n-1) + ... + b(nb+1)*x(n-nb) - a(2)*y(n-1) - ... - a(na+1)*y(n-na) SOUND : Play vector as sound SOUND(Y,FS) sends the signal in vector Y (with sample frequency FS) out to the speaker on platforms that support sound. Values in Y are assumed to be in the range 1.0 <= y <= 1.0. Values outside that range are clipped. Stereo sounds are played, on platforms that support it, when Y is an N-by-2 matrix. FFT : Discrete Fourier transform FFT(X) is the discrete Fourier transform (DFT) of vector X. For matrices, the FFT operation is applied to each column. For N-D arrays, the FFT operation operates on the first non-singleton dimension.

FFTSHIFT:Shift zero-frequency component to center of spectrum For vectors, FFTSHIFT(X) swaps the left and right halves of X. For matrices, FFTSHIFT(X) swaps the first and third quadrants and the second and fourth quadrants. For N-D arrays, FFTSHIFT(X) swaps "half-spaces" of X along each dimension. IMREAD : Read image from graphics file A = IMREAD(FILENAME,FMT) reads the image in FILENAME into A. If the file contains a grayscale intensity image, A is a two-dimensional array. If the file contains a truecolor (RGB) image, A is a three-dimensional (M-by-N-by-3) array.

FTRANS2:Design 2-D FIR filter using frequency transformation H = FTRANS2(B,T) produces the 2-D FIR filter H that corresponds to the 1-D FIR filter B using the transform T. (FTRANS2 returns H as a computational molecule, which is the appropriate form to use with FILTER2.) B must be a 1-D odd- length (type I) filter such as can be returned by FIR1, FIR2, or REMEZ in the Signal Processing Toolbox. IMFILTER : Multidimensional image filtering B = IMFILTER(A,H) filters the multidimensional array A with the multidimensional filter H. A can be logical or it can be a nonsparse numeric array of any class and dimension. The result, B, has the same size and class as A. IMSHOW : Display image IMSHOW(I,N) displays the intensity image I with N discrete levels of gray. If you omit N, IMSHOW uses 256 gray levels on 24-bit displays, or 64 gray levels on other systems.

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