(fix) proyek akhir sinsis dan pembagian kelompok (prof dadang)

7/26/2019 (FIX) Proyek Akhir SINSIS Dan Pembagian Kelompok (Prof Dadang)

1/18

KELOMPOK PROYEK AKHIR SINYAL DAN SISTEM

1. Muh Ilham Akbar Iffagano PROJECT 1

2. Reyfista Pangestu - Ahmad Salaam Mirfananda PROJECT 3

3. Ubay M Noor - Amirsyah Rayhan M PROJECT 6

4. Dita Tessa Parastika - Ismi Rosyiana Fitri PROJECT 30

5. M. Audy F. - Muhammad Raditya Gumelar PROJECT 25

6. Widi Destrianda - Benni Mustafa PROJECT 27

7. Ginas Alvianingsih - Dwinanri Egyna PROJECT 7

8. Muhammad Erfinza - Angga Hilman Hizrian PROJECT 24

9. Andreas P. Aji - Ferdy Kurniawan PROJECT 2010.M.Hilmy Iskandar - Martino Adisuwono PROJECT 19

11.Aiman Setiawan- Suharsono Halim PROJECT 22

12.Muhammad Haekal - Daniel Moses PROJECT 17

13.Ilham Dwi P - Harianto Adriprasetyo PROJECT 29

14.Claudia Khansa - Faya Safirra PROJECT 11

15.Dawud Shibghotulloh - Diamod Ravi PROJECT 36

16.Jendra Riyan Dwiputra - Andik Suprayogi PROJECT 9

17.Qashtalani Haramaini - Ilyasa Rafif PROJECT 23

18.Fajar Tri Wardana - Rahmat Sigalingging PROJECT 8

19.Agnes Grace S N - Susan Wiguna PROJECT 5

20.Luthfan Fauzan - Restu Nugroho PROJECT 33

21.Ahmad Dzul Faiq - M. Al Fatih PROJECT 34

22.Samuel Zakaria - Ester Nugraheny N.P PROJECT 18

23.Rachmat Romario Akbara - Yohan Binsar H.G PROJECT 26

24.Rialdo Stefan Josua - Hirzi Hasan PROJECT 12

25.Adhelia Irawan - Maulidya Falah PROJECT 14

26.Chaizar Ali Fachrudien - Achyar Maulana Pratama PROJECT 2

27.Erdi Nindito RumonoSamsudiat PROJECT 28

28.Zaky Ramadhan - Antonius Listyo PROJECT 10

29.Findal Darmaja - Josan Putra PROJECT 16

30.Jodi Malikan - Bagus Setiawan PROJECT 21


2/18

31.Ganang Rizky - Pradana Damara PROJECT 37

32.Kenny PrasetyoRizky Muhammad Reza PROJECT 13

33.Andhika Kumara D - Ahmad Fauzi Arief PROJECT 4

34.Budianto - Andre Jatmiko PROJECT 32

35.Dara AzkaAbi Iqbal Prasetyo PROJECT 35

36.Erasmus NKDhani Teja K. PROJECT 15

37.Muhammad IqbalBarry Muhammad PROJECT 31

Pembagian Proyek dilaksanakan secara undi, bila perlu penjelasan hubungi

@ilhamiff 08976692668

SIGNALS AND SYSTEMS PROJECT 2014

General instructions:

1. Make a user-friendly GUI application program by using MATLAB

2. Generalize the input parameters in each application you make, if you are

given a specific value, you must be able to input some other values as well.

PROJECT 1

Generate analog and discrete signals and do some basic operations on those signals,

such as adding, subtracting, multiplying, flipping, delaying, etc.

PROJECT 2

Make an application that can be used to determine if a given system is LTI or not. In

particular, if the system is linear or not, if the system is time invariant or not.

PROJECT 3

Make an application that is bale to calculate the average power of continuous and

discrete time periodic signals. The input signals can be generated by using the well-

known functions or the arbitrary functions.


3/18

PROJECT 4

Write a MATLAB program to decompose a continuous or a discrete time signal into its

odd and even parts.

PROJECT 5

A Sample and Hold circuit is used to help the conversion of an analog signal to be

digital one. The circuit conducts the sample operation and followed by uniform

quantizer. You are required to simulate the digitalization process using MATLAB. Also,

demonstrate the effect of sampling frequency.

PROJECT 6

You are supposed to write a MATLAB program to plot the magnitude spectrum of

0.23 cos10tx t e t for 10t . Because this is an infinite length signal, you should

truncate the signal by choosing the proper length. Try to simulate and analyze the

effect of signal length and also the number of frequency sampling points. Also,

demonstrate the effect of windowing. In particular, you are asked to compare for

different window functions.


4/18

PROJECT 7

You are asked to simulate the impact of DFT length on resolution. For example,

consider a test signal sin 2 1000 sin 2 1100 sin 2 3000x t t t t . You

generate an 0.0024 second sample of that signal using a sample rate if 10000 kHz.You use rectangular or Hamming window to compute the windowed DFTs with three

lengths: 25, 128, and 1024. Your program is expected to answer these questions:

1. How do the rectangular window results compare to the hamming window results?

2. Are you able to see from the frequency response magnitude that the signal contains

three separate sinusoids? If so, for which DFT length can you distinguish the

sinusoids?

3. What does increasing the DFT length accomplish?

PROJECT 8

The same as PROJECT 7, but in this project you must evaluate the impact of window

length of resolution. In specific, your program is expected to answer these questions:

1. How do the rectangular window results compare to the Hamming window results?

2. Are you able to see form the frequency response magnitude that the signal contains

three separate sinusoids? If so, for which DFT length can you distinguish the

sinusoids?

3. What does increasing the length of the data window accomplish?

PROJECT 9

In this project, you asked to analyse the sidelobe levels and leakage of a window.

Consider two sinusoidal signals: 1 sin 2 50x t t and 2 0.031sin 2 77.5x t t .You sample the signals at a rate of 1000 Hz over an interval of 0.127 seconds. Let

1 2x n x n x n . Use a 1024-DFT to compute the windowed transforms of the total

signal x n and 1x n . Compare the results for 128-point rectangular and 128-point

Hamming windows. Your program is expected to answer these questions:

1. Describe what you observe in these plots.

2. How do the sidelobe levels of the windows affect the results?

3. Which window(s) allow you to detect the presence of both sinusoids? Why?


5/18

PROJECT 10

An application of signals and systems course is an echo cancellation. In this project

youa re asked to demonstrate an echo cancellation system. Echos can result from

reflections of walls in a room or mismatched switching equipment in telephone lines.

The effect of a single echo can be modelled as y n x n x n N , where x n is

the voice signal, N is the delay, and is a constant. An echo removal system must

be described by difference equation of the echo equation above with input x n

replaced by y n , and output y n replaced by x n , which is equivalent to

y n x n y n N . In this simulation, try 1000N and 0.5 .

PROJECT 11

In this project you will filter an audio signal. First, you should have an audio signal.

You design a 40th-order lowpass filter with cutoff frequency 0.125 sf (you can use fir1

command). Then, you pass both the left and right channel signals of your audio snippet

through the filter.

PROJECT 12

Using an audio signal, demonstrate the Nyquist sampling theorem.

PROJECT 13

In this project, you are asked to do a digital music synthesis: Beethovens Fifth

Symphony using MATLAB. Please consult

http://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab3.pdf

Note that you can also compose other music!!!

PROJECT 14

Write a MATLAB program to show the correlation between two signals. In particular,

your program should be used to determine the periodicity of a signal by using the

autocorrelation property. Autocorrelation also has an application to quantify the effect

of noise on a periodic signal. Show this in your program.
http://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab3.pdfhttp://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab3.pdfhttp://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab3.pdf


6/18

PROJECT 15

You are asked to develop an understanding of how AM radio signals are modulated

and demodulated. Please consult

http://www.ece.tamu.edu/~hpfister/courses/ecen314/project2.pdf for furtherinformation.

PROJECT 16

Provide a program to analyse the frequency response of a causal discrete-time LTI

system implemented using the difference equation. For example, we have

0.1 0.1176 1 0.1 2 1.7119 1 0.81 2y n x n x n x n y n y n

You are asked to plot H f . Also, provide an output signal if given an input signal,

for example cos 0.1x n n u n .

PROJECT 17

Demonstrate the time shifting and frequency shifting properties of the DTFT.

PROJECT 18

You are asked to design a noise reduction system. Suppose you record a noisy sound

(signal) and then you sample it to be x n . You are asked to identify the spectrum

corresponds to the noise and speech signals by using DTFT. You can remove the

noise afterwards.
http://www.ece.tamu.edu/~hpfister/courses/ecen314/project2.pdfhttp://www.ece.tamu.edu/~hpfister/courses/ecen314/project2.pdfhttp://www.ece.tamu.edu/~hpfister/courses/ecen314/project2.pdf


7/18

PROJECT 19

This project is similar to PROJECT 10, however, in this project you must estimate the

echo parameters. In a real problem you do not know the parameters: delay and

constant. Suppose you are given the data y n (which is corrupted by an echo) but

suppose you do not know the value of the delay N . Let yyr n be the autocorrelation

of y n , *yyr n y n y n and let xxr n be the autocorrelation of x n ,

*xx

r n x n x n . You must first find a formula for the signal yyr n in terms of xxr n

.

PROJECT 20

Make a program to find the Fourier series coefficients for a periodic signal. For

example, cosx t t . Also, you should show the convergence of the Fourier series

of the periodic pulse signal. Sometimes the integration formula for the Fourier series

coefficient is difficult to carry out. In these cases, the coefficients can be obtained

numerically. Remember a formula m

f t dt T f m T . For example,

01 jk t

Tc k x t e dt

T

can be approximated as

01

0

Mjk m T

m

Tc k x m T e

T

. You

compare the results.

PROJECT 21

In this project, you will examine the effects of a low pass filter on speech. Write a

MATLAB script to read anaudio file into Matlab. (wavread), filter the sound file with a

40th order low pass FIR filter with a cutoff of 1000 Hz. (fir1, filter), a note on using fir1

to generate the coefficients for the low-pass filter: you specify a cutoff frequency thats

been normalized to the Nyquist frequency = fs/2. Generate and plot the amplitude of

the transfer function of the filter in dB vs. frequency in Hz. (freqz) Play the audio file

after it has been sent through the filter. In addition to plotting the frequency response

of the filter, do the following plots: use fft to plot the frequency response of the original

signal, use fft to plot the frequency response of the filtered signal (should be very

enlightening), do amplitude vs. index plots of the signal, before and after filtering
http://www.ece.uah.edu/~wilderj/ee384/nipitbud.wavhttp://www.ece.uah.edu/~wilderj/ee384/nipitbud.wav


8/18

PROJECT 22

You use high pass filter in PROJECT 21. Note that filter the sound file with a 40th

order high pass FIR filter with a cutoff of 3000 Hz. (fir1, filter)

PROJECT 23

Use a band pass filter to reduce noise in a set of simulated audio tones. Write a

MATLAB script to generate a sequence in MATLAB which samples the signal

1 2 31

sin 2 sin 2 sin 210

f t f t f t f t , where1 440f ,

2 554.365f , and

3 659.255f where it simulates the A-major chord. Generate a noisy musical chord

consisting of the sampled signal plus a noisy time-domain signal, you can use randn

in MATLAB. Filter the sound file with a 60 thorder band pass FIR filter with a cutoff of

10.9LOf f and 31.1HIf f . Use fir1. Listen to the music before and after being filtered.

PROJECT 24

In this project you are asked to learn signal sampling, manipulation, and playback.

Please see


PROJECT 25

In this project you should do Fourier sound synthesis. Please see

http://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab2.pdfhttp://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab2.pdfhttp://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab4.pdfhttp://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab4.pdfhttp://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab4.pdfhttp://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab2.pdf


9/18

PROJECT 26

You will create a signal with added high frequency noise

1. Typeload mtlb2. You can hear a voice say "MATLAB." This is the signal to which you will add

noisesoundsc(mtlb,Fs)3. Create a noise signal noise = cos(2*pi*3*Fs/8*(0:length(mtlb)-1)/Fs)';(You can

hear the noise signal by typing soundsc(noise,Fs))(You can also use randomfunction to introduce noise)

4. Add the noise to the original signalu = mtlb + noise;5. Scale the signal with noiseu = u/max(abs(u));(You scale the signal to try to

avoid overflows later on. You can hear the scaled signal with noise by typingsoundsc(u,Fs))

6. Display the frequency spectrum using FFT7. View the scaled signal with noisespecgram(u,256,Fs);colorbar(In the

spectrogram, you can see the noise signal as a horizontal line at about 2800Hz,which is equal to 3*Fs/8).

8. Use low-pass filters to eliminate high frequencyb = ones(1,10)/10; % 10 pointaveraging filterfy = filtfilt(b,1,x); % Noncausal filteringfyy = filter(b,1,x); %Normal filtering

9. Use FFT to plot the power spectrum of the filter signal and compare it to both,the originaland the corrupted signals.

You will be provided an ECG ( Electrocardiogram) signal with noise.Your task is to

create a matlab program to determine the frequency of the noise and eliminate

thenoise signal.


10/18

PROJECT 27

1. The periodic signal x(t) is shown below. Determine the fundamental period T0.

Write a MATLAB code to plot x(t), using enough points to get "smooth" curve.This is just reproducing the curve provided below.

2. Compute the Fourier series coefficients for x(t) (if you can find it in the book,that is ok). Plot the single-sided and double-sided spectra up to the 10thharmonic.

3. Plot partial sums of the Fourier series for x(t) (terms 1 through N in the infiniteseries). In your opinion, what is the value of N that results in a "good"reproduction of x(t).

Document your work, include plots to illustrate your work and your conclusions.Include your MATLAB code.


11/18

PROJECT 28

Consider the following two signals:

,0

,1)(

1 tx

elsewhere

t 3||

,0

,1)(

2 tx

elsewhere

t 1||

a) Plot these two signals on the same figure. Use a time axis of [-5,5]. Add labels, etc

b) Derive (mathematically) the Fourier Transforms X1(f) and X2(f) of these two

signals

c) Write a program to compute the Fourier Transforms numerically, and compare

with b)

Note: you can use the Matlab command trapzfor numerical integration

c) Plot the magnitude spectrum for the signals on the same figure.

d) Discuss your results and explain the relationship between the spectra X1(f) and

X2(f).

PROJECT 29

The impulse response h(t)for a particular LTI system is shown below. All parts

of this lab make use of this h (t).

h(t) = [3e-3t+ 5 e-2t+ e-t(4 cos(3t)+ 6 sin(3t)) + e-4t] . u(t)

1. Plot the impulse response for h(t)directly from the above equation by creatinga time vector.

2. Use the residuefunction to determine the transfer function H(s). Determine thelocations of the poles and zeros of H(s)with the rootsfunction, and plot themin the s-plane (x for poles, o for zeros).

3. Use the freqsfunction to plot the magnitude and phase of the transfer function.

Document your work, include plots to illustrate your work and your

conclusions. Include your MATLAB code.


12/18

PROJECT 30

Consider the system block diagram shown below:

(a) Use the series function to find the numerator and denominator polynomialcoefficient vectors for the cascade connection of G1(s) and G2(s). Use the printsysfunction to find the overall transfer function

=.Hint: in the Matlab help, terms like SYS, SYS1, etc., refer to the specificationof a system by a pair of vectors that give the coefficients for the numerator anddenominator polynomials of the transfer function. For example,>> numG1= [1 1];>> denG1=[1 2];>> numG2=[1];>> denG2=[500 0 0];

>> [num,den]=series(numG2,denG2,numG1,denG1);>> printsys(num,den);

(b) Use impulse to plot the system impulse response g(t). Note: impulse assumesthat the system is causal (ROC of the transfer function is a right half-plane).(c) Use step to plot the system step response.(d) Plot the signal

= cos5

for t [0, 5] using a time resolution of 0.002 sec. Use lsim with a left-hand

argument to plot the system response for t [0, 5] when r(t) is the system input.Use a time resolution of 0.002 sec for your plot.(e) Use pzmapwith left-hand arguments to observe the poles and zeros of G(s). Isthe system stable? (Justify your answer).(f) Usepzmapwithout left-hand arguments to generate a pole-zero plot for G(s).


13/18

PROJECT 31

Consider the feedback system depicted in the block diagram below:

Note that the gain of the feedback path is unity (1).(a) Use feedback to find the numerator and denominator polynomial coefficient

vectors of the closed-loop transfer function

=

For feedback, you will need the numerator and denominator polynomialcoefficient vectors for the series connection of G1(s) and G2(s) that youobtained in the above (problem (1)). The transfer function of the feedbackpath in the system block diagram above is equal to one, so you can describeit with the numerator polynomial coefficient vector [1] and the denominatorpolynomial coefficient vector [1].

(b) Use printsysto find G(s).(c) Use impulseto plot the system impulse response g(t). Give a brief

qualitativedescription of the main effect of adding feedback to this system ascompared tothe system in problem (1).

(d) Use pzmapwith left-hand arguments to observe the poles and zeros of G(s).Isthe system stable? (Justify your answer).

(e) Use pzmapwithout left-hand arguments to generate a pole-zero plot for G(s).(f) Plot the signal

= cos 500

for t [0, 20000] using a time resolution of 10 sec. Use lsim to plot the systemresponse when r(t) is the system input for t [0, 20000] using a timeresolution of 10 sec.


14/18

PROJECT 32

Consider the feedback system depicted in the block diagram below:

(a) Use feedback to find the numerator and denominator polynomialcoefficient vectors of the closed-loop transfer function

=

(b) Use printsys to find G(s).(c) Use impulse to plot the system impulse response g(t).(d) Use pzmap with left-hand arguments to observe the poles and zeros of

G(s). Is itcausal? Is the system stable? (Justify your answers).(e) Use pzmap without left-hand arguments to generate a pole-zero plot for

G(s).


15/18

PROJECT 33

H is a causal discrete-time LTI system with input x[n] and output y[n] related by thelinear constant coefficients difference equation

y[n] = 0.6y[n 1] + x[n]

(a) Find the transfer function H(z) analytically (using paper and pencil).Use zplane to generate a pole-zero plot for H(z). Is the system stable?(Justify your answer).

Note:You should get the vector [1] for the numerator polynomialcoefficientvector of H(z). On some versions of Matlab, this will createan incorrect polezeroplot; the plot may show an incorrect zero at z = 1.If this happens, you canfix it by using [1 0] for the numerator coefficient

vector. The vectors [1] and [1 0]actually specify the same numerator forH(z), since 1 = 1 + 0z1.

(b) Use freqzto plot the magnitude |H(ej!)| and phase ]H(ej!) of the system

frequency response ()using N = 1024 points.

(c) Use impzto plot the system impulse response h[n].

PROJECT 34

Second Order ResponseConsider a transfer function of the following form corresponding to a right-sidedresponse.

= 2[] ||

The polep is complex. Let = For this question, r=0.9 and = /2.

(a) Use the Matlab function zplane to plot the poles and zeros of H(z) .(b) Use freqz to calculate the complex frequency response. Plot the magnitudeand phase of thefrequency response corresponding to H(z) . Categorize thisresponse in terms of is it lowpass,highpass, bandpass, or bandstop.

(c) The group delay of a response is given by

=

A good group delay is one which is constant with frequency. What does thisconstancy imply for thephase? Why would filter designers prefer to deal withgroup delay rather than phase response? Use the Matlab function grpdelay toget the group delay associated with the filter given above. Plot the group

delay. What are the units of group delay? Label the axis appropriately.(d) Plot the impulse response of the filter (first 20 or so samples).


16/18

PROJECT 35

DTFT for FIR signalsWe have been using the routine freqz to compute frequency responses. In thisproblem you will write your own DTFT routine for finite length sequences. The DTFTwill be of the form

function H = DTFTFIR (h, n, w)nw = length(w);nh = length(h);for (k = 1:nw)H(k) = endreturn

The input to this routine is a set of response values (h) with corresponding values oftime indices (n), and a vector of frequencies for which the response is to beevaluated. Test your routine against freqz using the following two sets of responses(use frequencies from 0 to in, say, 100 steps). Plot the magnitude response andthe phase response for each case.(a) h = [4 3 2 1 1 2 3 4];

n = -2:5;Explain the shape of the phase response curvespecifically, why are therejumps in the function? Hint: where do the jumps occur with respect to themagnitude curve?

(b) n = 0:20;

h = n .* (0.9).^n;


17/18

PROJECT 36

DTFT for IIR signalsWe have been using the routine freqz to compute frequency responses. In thisproblem you will write your own DTFT routine for infinite length sequences specifiedby a numerator polynomial and a denominator polynomial (polynomials in 1 z ). TheDTFT will be of the formfunction H = DTFTIIR (b, a, w)nw = length(w);nb = length(b);na = length(a);for (k = 1:nw)H(k) = endreturn

To do this first consider the rational polynomial

== 11 11

It is then clear that the frequency response corresponding to H(z) for a series offrequencies is the point-bypoint division of the frequency response of B(z) divided bythe frequency response ofA(z) .

Write your DTFT routine and test it on the following responses. Plot the magnitude

and phase for each, comparing the results with those from freqz.(a) []= 0.5(b) = 1 +

+ +31.18+ .81


18/18

PROJECT 37

The following program can be used to verify the time shifting property ofdiscrete-timeFourier transform(DTFT):

clf;w = -pi:2*pi/255:pi; wo = 0.4*pi; D = 10;num = [1 2 3 4 5 6 7 8 9];h1 = freqz(num, 1, w);h2 = freqz([zeros(1,D) num], 1, w);subplot(2,2,1)plot(w/pi,abs(h1));gridtitle('Magnitude Spectrum of Original Sequence')subplot(2,2,2)

plot(w/pi,abs(h2));gridtitle('Magnitude Spectrum of Time-Shifted Sequence')subplot(2,2,3)plot(w/pi,angle(h1));gridtitle('Phase Spectrum of Original Sequence')subplot(2,2,4)plot(w/pi,angle(h2));gridtitle('Phase Spectrum of Time-Shifted Sequence')

Q.1 which parameter controls the amount of time-shift?Q.2 Repeat the above program for signal in 5.21 and for -5

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