Experimental Fourier Transforms

Similar documents
Solutions. Chapter 5. Problem 5.1. Solution. Consider the driven, two-well Duffing s oscillator. which can be written in state variable form as

Fourier Analysis and Power Spectral Density

Signals, Instruments, and Systems W5. Introduction to Signal Processing Sampling, Reconstruction, and Filters

SEISMIC WAVE PROPAGATION. Lecture 2: Fourier Analysis

Continuous Fourier transform of a Gaussian Function

SNR Calculation and Spectral Estimation [S&T Appendix A]

Sensors. Chapter Signal Conditioning

MEDE2500 Tutorial Nov-7

ANALOG AND DIGITAL SIGNAL PROCESSING ADSP - Chapter 8

L6: Short-time Fourier analysis and synthesis

Review: Continuous Fourier Transform

OSE801 Engineering System Identification. Lecture 05: Fourier Analysis

Distortion Analysis T

Signal Processing Signal and System Classifications. Chapter 13

EE 435. Lecture 32. Spectral Performance Windowing

Vibration Testing. an excitation source a device to measure the response a digital signal processor to analyze the system response

Vibration Testing. Typically either instrumented hammers or shakers are used.

Chapter 6: Applications of Fourier Representation Houshou Chen

Radar Systems Engineering Lecture 3 Review of Signals, Systems and Digital Signal Processing

Principles of Communications

The Discrete Fourier Transform (DFT) Properties of the DFT DFT-Specic Properties Power spectrum estimate. Alex Sheremet.

Time Series Analysis: 4. Digital Linear Filters. P. F. Góra

The Discrete Fourier Transform

Sound & Vibration Magazine March, Fundamentals of the Discrete Fourier Transform

Each of these functions represents a signal in terms of its spectral components in the frequency domain.

In this Lecture. Frequency domain analysis

EEO 401 Digital Signal Processing Prof. Mark Fowler

8/19/16. Fourier Analysis. Fourier analysis: the dial tone phone. Fourier analysis: the dial tone phone

Multimedia Signals and Systems - Audio and Video. Signal, Image, Video Processing Review-Introduction, MP3 and MPEG2

Signal processing Frequency analysis

A6523 Signal Modeling, Statistical Inference and Data Mining in Astrophysics Spring

X(t)e 2πi nt t dt + 1 T

Data Processing and Analysis

Signals and Systems. Lecture 14 DR TANIA STATHAKI READER (ASSOCIATE PROFESSOR) IN SIGNAL PROCESSING IMPERIAL COLLEGE LONDON

Discrete Fourier Transform

Correlator I. Basics. Chapter Introduction. 8.2 Digitization Sampling. D. Anish Roshi

Fourier Analysis of Signals Using the DFT

Information and Communications Security: Encryption and Information Hiding

A523 Signal Modeling, Statistical Inference and Data Mining in Astrophysics Spring 2011

Sistemas de Aquisição de Dados. Mestrado Integrado em Eng. Física Tecnológica 2016/17 Aula 3, 3rd September

EE 435. Lecture 30. Data Converters. Spectral Performance

Linear models. Chapter Overview. Linear process: A process {X n } is a linear process if it has the representation.

EEM 409. Random Signals. Problem Set-2: (Power Spectral Density, LTI Systems with Random Inputs) Problem 1: Problem 2:

ELEN E4810: Digital Signal Processing Topic 11: Continuous Signals. 1. Sampling and Reconstruction 2. Quantization

Data Processing and Analysis

Fourier transform. Stefano Ferrari. Università degli Studi di Milano Methods for Image Processing. academic year

ENSC327 Communications Systems 2: Fourier Representations. Jie Liang School of Engineering Science Simon Fraser University

Practical Spectral Estimation

Various signal sampling and reconstruction methods

Time Series Analysis: 4. Linear filters. P. F. Góra

ESS Finite Impulse Response Filters and the Z-transform

VALLIAMMAI ENGINEERING COLLEGE. SRM Nagar, Kattankulathur DEPARTMENT OF INFORMATION TECHNOLOGY. Academic Year

CE 513: STATISTICAL METHODS

Fourier Methods in Digital Signal Processing Final Exam ME 579, Spring 2015 NAME

Slide Set Data Converters. Background Elements

Aspects of Continuous- and Discrete-Time Signals and Systems

Lab 4: Quantization, Oversampling, and Noise Shaping

Filter Analysis and Design

DFT & Fast Fourier Transform PART-A. 7. Calculate the number of multiplications needed in the calculation of DFT and FFT with 64 point sequence.

Centre for Mathematical Sciences HT 2017 Mathematical Statistics

Department of Electrical and Telecommunications Engineering Technology TEL (718) FAX: (718) Courses Description:

Chapter 4 Discrete Fourier Transform (DFT) And Signal Spectrum

Sampling. Alejandro Ribeiro. February 8, 2018

Solutions. Chapter 1. Problem 1.1. Solution. Simulate free response of damped harmonic oscillator

DEPARTMENT OF EI DIGITAL SIGNAL PROCESSING ASSIGNMENT 1

V(t) = Total Power = Calculating the Power Spectral Density (PSD) in IDL. Thomas Ferree, Ph.D. August 23, 1999

Chapter 8 The Discrete Fourier Transform

Discrete-time Signals and Systems in

Question Bank. UNIT 1 Part-A

EE 435. Lecture 28. Data Converters Linearity INL/DNL Spectral Performance

Discrete Fourier Transform

Summary of lecture 1. E x = E x =T. X T (e i!t ) which motivates us to define the energy spectrum Φ xx (!) = jx (i!)j 2 Z 1 Z =T. 2 d!

R13 SET - 1

Unstable Oscillations!

Chirp Transform for FFT

Quality Improves with More Rays

INTRODUCTION TO DELTA-SIGMA ADCS

Discrete-time Fourier Series (DTFS)

Contents. Digital Signal Processing, Part II: Power Spectrum Estimation

Chapter 5 Frequency Domain Analysis of Systems

Ch.11 The Discrete-Time Fourier Transform (DTFT)

[ ], [ ] [ ] [ ] = [ ] [ ] [ ]{ [ 1] [ 2]

DFT and Matlab with some examples.

Massachusetts Institute of Technology Department of Electrical Engineering and Computer Science : Discrete-Time Signal Processing

!Sketch f(t) over one period. Show that the Fourier Series for f(t) is as given below. What is θ 1?

3. Lecture. Fourier Transformation Sampling

OSE801 Engineering System Identification. Lecture 09: Computing Impulse and Frequency Response Functions

Course Name: Digital Signal Processing Course Code: EE 605A Credit: 3

EDISP (NWL3) (English) Digital Signal Processing DFT Windowing, FFT. October 19, 2016

0 for all other times. Suppose further that we sample the signal with a sampling frequency f ;. We can write x # n =

Chapter 5 Frequency Domain Analysis of Systems

The (Fast) Fourier Transform

An Introduction to Discrete-Time Signal Processing

DHANALAKSHMI COLLEGE OF ENGINEERING DEPARTMENT OF ELECTRICAL AND ELECTRONICS ENGINEERING EC2314- DIGITAL SIGNAL PROCESSING UNIT I INTRODUCTION PART A

ECG782: Multidimensional Digital Signal Processing

Variable and Periodic Signals in Astronomy

1 Understanding Sampling

Laboratory handout 5 Mode shapes and resonance

Time and Spatial Series and Transforms

Chapter 2: Problem Solutions

Transcription:

Chapter 5 Experimental Fourier Transforms 5.1 Sampling and Aliasing Given x(t), we observe only sampled data x s (t) = x(t)s(t; T s ) (Fig. 5.1), where s is called sampling or comb function and can be described as s(t; T s ) = δ(t nt s ), (5.1) where T s is sampling period and f s = 1/T s is sampling frequency (Fig. 5.2). Figure 5.1: We observe only sampled data of a continuous signal every sampling period of T s...... Figure 5.2: Sampling or comb function s(t) 41

42 CHAPTER 5. EXPERIMENTAL FOURIER TRANSFORMS Therefore, using the sifting property of the Dirac delta function, we get x s (t) = x(t) δ(t nt s ) = x(nt s )δ(t nt s ). (5.2) To take the Fourier Transform of x s, it is useful to note that s(t; T s ) is periodic with period T s. Thus, the corresponding Fourier Series will be: where So s(t; T s ) = X n = 1 [ Ts/2 T s T s/2 s(t; T s ) = 1 T s δ(t nt s ) 2π in X n e Ts t, (5.3) ] 2π in e Ts t dt = 1 e = 1. (5.4) T s T s e i2π n Ts t. (5.5) Now, from previously calculated δ function Fourier Transform pairs, we get S(f) = 1 ) δ (f nts. (5.6) T s that is or Therefore, since x s (t) = x(t)s(t; T s ), by convolution we get X s (f) = 1 T s F [x s ] = X s (f) = X(f) S(f), (5.7) X s (f) = 1 T s ( X(s)δ f n ) s ds, (5.8) T s ) X (f nts. (5.9) That is, sampling results in an number of periodically spaced copies of the true X(f). Now, we see that if f s < 2f BW, we will get spectral overlap or aliasing: high frequencies are folded onto lower ones. This leads to the Nyquist sampling criterion: f s 2f BW for accurate frequency domain estimates with sampled data (see Fig. 5.3). Shannon [14] was the one who credited Nyquist when he stated his theorem: Figure 5.3: (a) true Fourier transform, and (b) Fourier transform of sampled data

5.2. WINDOWING 43 Shannon s Sampling Theorem: If a function x(t) contains no frequencies higher than f BW hertz, it is completely determined by giving its ordinates at a series of points spaced T s = 1 2f BW seconds apart. This means that the time series can be completely reconstructed from its Fourier transform if the Nyquist Criteria is satisfied. Remark 1: In practice, we may need f s = 3F BW or even 4f BW to avoid aliasing. Remark 2: Also, we usually use an anti-aliasing (low-pass) analog filter, such as shown in Fig. 5.4, which is meant as an example only. If f c f BW, sample at 4f c = 4f BW and discard bottom one-half of the Fourier transform data. -5 Figure 5.4: Example of analog low-pass filter frequency response function (FRF) magnitude versus frequency 5.2 Windowing Another artifact of data collection is that we take only a finite time record of our data (see Fig. 5.5). Mathematically we can express this as we are sampling the following signal x w (t) = x(t)w(t), where w(t) is a data collection window function. If our data collection window w(t) is a rectangular window function, we get what is shown in Fig. 5.6. Figure 5.5: We only take a finite time record of data during the data collection window

44 CHAPTER 5. EXPERIMENTAL FOURIER TRANSFORMS Figure 5.6: Rectangular window function w(t) and the resultant observed signal Now, we know how to find the Fourier transform of a rectangular window: W (f) = = 1 2πif w(t)e 2πift dt = a a ( e 2πifa e 2πifa) = e 2πift dt sin 2πfa πf = 2a sinc(2πfa), (5.1) where the sinc function has the form shown in Fig. 5.7. 2 a -5/(2a) -2/a -3/(2a) -1/a -1/(2a) 1/(2a) 1/a 3/(2a) 2/a 5/(2a) Figure 5.7: Fourier transform of w(t) in Fig. 5.6 As an example of the effect of the window, let x(t) = cos 2πf t, then using convolution property we have the following: X w (f) = X(f) W (f) 1 = 2 [δ (f + f sin 2πsa s) + δ (f f s)] ds πs = 1 [ sin (2π(f + f )a) + sin (2π(f f ] )a) 2 π(f + f ) π(f f ) (5.11) The results of this transform are shown in Fig. 5.8, which shows that windowing causes spectral leackage into neighboring frequencies. The solution to spectral leakage is to use different window functions. Some of the widely used window functions are shown in Fig. 5.9.

5.2. WINDOWING 45 Figure 5.8: True Fourier transforms X(f) = F (cos 2πf t) (a) and windowed Fourier transform X w (f) = F (w(t) cos 2πf t) (b) showing spectral leakage Amplitude 1.8.6.4.2 -.2 Time domain hamming hanning gaussian blackman chebyshev flattop kaiser tukey 1 2 3 4 5 6 Samples Magnitude (db) 4 2-2 -4-6 -8-1 -12 Frequency domain hamming hanning gaussian blackman chebyshev flattop kaiser tukey -14.1.2.3.4.5.6.7.8.9 Normalized Frequency ( rad/sample) Figure 5.9: Some of the windows from the window design tool of MATLAB.

46 CHAPTER 5. EXPERIMENTAL FOURIER TRANSFORMS 5.3 Discrete Fourier Transfrom Continuous Fourier transfrom was defined in Eq. (4.9): X(f) = x(t) e i2πft dt. (5.12) However, we only measure over a finite time T, with equally-spaced samples with t = T N, where N is the number of samples (see Fig. 5.1). Therefore, Figure 5.1: We observe only sampled data of a continuous signal every sampling period of T s X(f) T x(t) e i2πft dt n= x n e i2πf(n ), (5.13) where we ignore the fact that X(f) is no longer true F(x(t)), and x n x(t n ) = x(n ). Now, let f = m T f m and X(f m ) = X m ( scaling is needed to have correct units), then we get X m = 1 n= x n e i2π m T (n ) = n= x n e i2π T (mn). (5.14) However, = T N, so T = 1 N. Then, X m = which is discrete Fourier transform or DFT. n= mn i2π x n e N, (5.15) We cannot possibly get x(t) or X(f) for t or f between the t n = n (i.e., n < t < (n + 1) ) or m f < f < (m + 1) f ( f = m+1 T m T = 1 T ), respectively. However, given {x n} n= we get {X m } m=, and xn = 1 X m e i2πnm N, (5.16) N m= which is called inverse discrete Fourier transfor (IDFT) and discrete values are transformed exactly. We can prove the above as follows. Given X m = p= mp i2π x p e N. (5.17)

5.3. DISCRETE FOURIER TRANSFROM 47 we can write: x n = 1 N = 1 N m= p= ( p= ( x p = 1 N x nn = x n. mp i2π x p e N m= ) m(n p) i2π e N e i2π nm N, ), m= m(n p) i2π e N = for n p, N for n = p. (5.18) In conclusion, the discrete and inverse discrete Fourier transforms are exact transforms. 5.3.1 Notes on Fast Fourier Transforms Fast Fourier Transform (FFT) is a fast algorithm for DFT (see, e.q., Numerical Recipes). It has the following requirements and parameters: Time records of N points, t between points must satisfy 1 t 2f BW of signal. Simplest and fastest implementation is when N = 2 m, so data records (buffer) are usually 2 m (the DP 42 has 4K or 496 point buffer). We get N 2 lines in the frequency domail. Heuristically, the N real data points give N 2 complex (amplitude and phase) data points in the frequency domain. f in the FFT id 1 T f max = 1 2 t = fs 2. of data record. Thus, f max = N 2 f = N 2T. However, T = N t and, thus,

48 CHAPTER 5. EXPERIMENTAL FOURIER TRANSFORMS Problems Problem 5.1 Consider the driven, two-well Duffing s oscillator ẍ + ɛγẋ x + x 3 = ɛf cos(ωt) which can be written in state variable form as ẋ = v v = x x 3 + ɛ( γv + F cos(ωt)). In the above expressions, γ is the linear (viscous) damping coefficient, F is the forcing amplitude, and ω is the forcing frequency. 1 Develop a MATLAB code to simulate this system. You need to be able to generate uniformly sampled time series of any length, with any sampling time t s (recall that the sampling frequency is f s = 1/t s, and be very careful when choosing your sampling time). Experiment with the simulations, and by varying the system parameters and examples of: 1. a periodic, approximately simple harmonic response; 2. a periodic response that is not simple harmonic; and 3. a chaotic (i.e. aperiodic) response. Plot the steady state results in the time domain and in the x v phase plane. To help you get started, reasonable values for the parameters to use in the simulations are ɛ =.1, γ [.7; 2.], ω [.9; 1, 1], F [1.5; 2.8]. Problem 5.2 Using the time series {x i } generated in Exercise 1, make an observable via {y i } = {x i + n i }, where the n i are independent identically-distributed (IID) mean zero Gaussian random variable with variance σ 2 (be careful when choosing your parameters we want to approximate real-life situations). Find the digital Power Spectral Density (PSD) for the three types of response found in Exercise 1, using the MATLAB functions pwelch or periodogram (e.g., at MATLAB prompt type help pwelch ). By experimentation, examine the effect of σ, t s, the number of points 11 The parameter ɛ is included here only for convenience when comparing our results to certain theoretical treatments. In particular, the damping and forcing are scaled by ɛ to indicate that the oscillator can be thought of as a perturbation of of a conservative system, with the conservative system corresponding to the case ɛ =. See, for example, Guckenheimer and Holmes (1983).

5.3. DISCRETE FOURIER TRANSFROM 49 used in the FFT, and windowing (including no window, which is called rectwin in MATLAB). A good window to use is the Hanning, however you can experiment to see the differences. Type help window at the command line for more information. Plot your results using a semilog or Decibel (db) scale. Discuss your results: what do you see? References John Guckenheimer and Philip Holmes (1983) Nonlinear Oscillations, Dynamical Systems, and Bifurcations of Vector Fields, Springer-Verlag, New York. See in particular Ch. 4.5, Eq. (4.5.18).