Use of the Kurtosis Statistic in the Frequency Domain As an Aid In
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Moving Average Filters
CHAPTER 15 Moving Average Filters The moving average is the most common filter in DSP, mainly because it is the easiest digital filter to understand and use. In spite of its simplicity, the moving average filter is optimal for a common task: reducing random noise while retaining a sharp step response. This makes it the premier filter for time domain encoded signals. However, the moving average is the worst filter for frequency domain encoded signals, with little ability to separate one band of frequencies from another. Relatives of the moving average filter include the Gaussian, Blackman, and multiple- pass moving average. These have slightly better performance in the frequency domain, at the expense of increased computation time. Implementation by Convolution As the name implies, the moving average filter operates by averaging a number of points from the input signal to produce each point in the output signal. In equation form, this is written: EQUATION 15-1 Equation of the moving average filter. In M &1 this equation, x[ ] is the input signal, y[ ] is ' 1 % y[i] j x [i j ] the output signal, and M is the number of M j'0 points used in the moving average. This equation only uses points on one side of the output sample being calculated. Where x[ ] is the input signal, y[ ] is the output signal, and M is the number of points in the average. For example, in a 5 point moving average filter, point 80 in the output signal is given by: x [80] % x [81] % x [82] % x [83] % x [84] y [80] ' 5 277 278 The Scientist and Engineer's Guide to Digital Signal Processing As an alternative, the group of points from the input signal can be chosen symmetrically around the output point: x[78] % x[79] % x[80] % x[81] % x[82] y[80] ' 5 This corresponds to changing the summation in Eq. -
Higher-Order Asymptotics
Higher-Order Asymptotics Todd Kuffner Washington University in St. Louis WHOA-PSI 2016 1 / 113 First- and Higher-Order Asymptotics Classical Asymptotics in Statistics: available sample size n ! 1 First-Order Asymptotic Theory: asymptotic statements that are correct to order O(n−1=2) Higher-Order Asymptotics: refinements to first-order results 1st order 2nd order 3rd order kth order error O(n−1=2) O(n−1) O(n−3=2) O(n−k=2) or or or or o(1) o(n−1=2) o(n−1) o(n−(k−1)=2) Why would anyone care? deeper understanding more accurate inference compare different approaches (which agree to first order) 2 / 113 Points of Emphasis Convergence pointwise or uniform? Error absolute or relative? Deviation region moderate or large? 3 / 113 Common Goals Refinements for better small-sample performance Example Edgeworth expansion (absolute error) Example Barndorff-Nielsen’s R∗ Accurate Approximation Example saddlepoint methods (relative error) Example Laplace approximation Comparative Asymptotics Example probability matching priors Example conditional vs. unconditional frequentist inference Example comparing analytic and bootstrap procedures Deeper Understanding Example sources of inaccuracy in first-order theory Example nuisance parameter effects 4 / 113 Is this relevant for high-dimensional statistical models? The Classical asymptotic regime is when the parameter dimension p is fixed and the available sample size n ! 1. What if p < n or p is close to n? 1. Find a meaningful non-asymptotic analysis of the statistical procedure which works for any n or p (concentration inequalities) 2. Allow both n ! 1 and p ! 1. 5 / 113 Some First-Order Theory Univariate (classical) CLT: Assume X1;X2;::: are i.i.d. -
Random Signals
Chapter 8 RANDOM SIGNALS Signals can be divided into two main categories - deterministic and random. The term random signal is used primarily to denote signals, which have a random in its nature source. As an example we can mention the thermal noise, which is created by the random movement of electrons in an electric conductor. Apart from this, the term random signal is used also for signals falling into other categories, such as periodic signals, which have one or several parameters that have appropriate random behavior. An example is a periodic sinusoidal signal with a random phase or amplitude. Signals can be treated either as deterministic or random, depending on the application. Speech, for example, can be considered as a deterministic signal, if one specific speech waveform is considered. It can also be viewed as a random process if one considers the ensemble of all possible speech waveforms in order to design a system that will optimally process speech signals, in general. The behavior of stochastic signals can be described only in the mean. The description of such signals is as a rule based on terms and concepts borrowed from probability theory. Signals are, however, a function of time and such description becomes quickly difficult to manage and impractical. Only a fraction of the signals, known as ergodic, can be handled in a relatively simple way. Among those signals that are excluded are the class of the non-stationary signals, which otherwise play an essential part in practice. Working in frequency domain is a powerful technique in signal processing. While the spectrum is directly related to the deterministic signals, the spectrum of a ran- dom signal is defined through its correlation function. -
The Method of Maximum Likelihood for Simple Linear Regression
08:48 Saturday 19th September, 2015 See updates and corrections at http://www.stat.cmu.edu/~cshalizi/mreg/ Lecture 6: The Method of Maximum Likelihood for Simple Linear Regression 36-401, Fall 2015, Section B 17 September 2015 1 Recapitulation We introduced the method of maximum likelihood for simple linear regression in the notes for two lectures ago. Let's review. We start with the statistical model, which is the Gaussian-noise simple linear regression model, defined as follows: 1. The distribution of X is arbitrary (and perhaps X is even non-random). 2. If X = x, then Y = β0 + β1x + , for some constants (\coefficients", \parameters") β0 and β1, and some random noise variable . 3. ∼ N(0; σ2), and is independent of X. 4. is independent across observations. A consequence of these assumptions is that the response variable Y is indepen- dent across observations, conditional on the predictor X, i.e., Y1 and Y2 are independent given X1 and X2 (Exercise 1). As you'll recall, this is a special case of the simple linear regression model: the first two assumptions are the same, but we are now assuming much more about the noise variable : it's not just mean zero with constant variance, but it has a particular distribution (Gaussian), and everything we said was uncorrelated before we now strengthen to independence1. Because of these stronger assumptions, the model tells us the conditional pdf 2 of Y for each x, p(yjX = x; β0; β1; σ ). (This notation separates the random variables from the parameters.) Given any data set (x1; y1); (x2; y2);::: (xn; yn), we can now write down the probability density, under the model, of seeing that data: n n (y −(β +β x ))2 Y 2 Y 1 − i 0 1 i p(yijxi; β0; β1; σ ) = p e 2σ2 2 i=1 i=1 2πσ 1See the notes for lecture 1 for a reminder, with an explicit example, of how uncorrelated random variables can nonetheless be strongly statistically dependent. -
STATISTICAL FOURIER ANALYSIS: CLARIFICATIONS and INTERPRETATIONS by DSG Pollock
STATISTICAL FOURIER ANALYSIS: CLARIFICATIONS AND INTERPRETATIONS by D.S.G. Pollock (University of Leicester) Email: stephen [email protected] This paper expounds some of the results of Fourier theory that are es- sential to the statistical analysis of time series. It employs the algebra of circulant matrices to expose the structure of the discrete Fourier transform and to elucidate the filtering operations that may be applied to finite data sequences. An ideal filter with a gain of unity throughout the pass band and a gain of zero throughout the stop band is commonly regarded as incapable of being realised in finite samples. It is shown here that, to the contrary, such a filter can be realised both in the time domain and in the frequency domain. The algebra of circulant matrices is also helpful in revealing the nature of statistical processes that are band limited in the frequency domain. In order to apply the conventional techniques of autoregressive moving-average modelling, the data generated by such processes must be subjected to anti- aliasing filtering and sub sampling. These techniques are also described. It is argued that band-limited processes are more prevalent in statis- tical and econometric time series than is commonly recognised. 1 D.S.G. POLLOCK: Statistical Fourier Analysis 1. Introduction Statistical Fourier analysis is an important part of modern time-series analysis, yet it frequently poses an impediment that prevents a full understanding of temporal stochastic processes and of the manipulations to which their data are amenable. This paper provides a survey of the theory that is not overburdened by inessential complications, and it addresses some enduring misapprehensions. -
2D Fourier, Scale, and Cross-Correlation
2D Fourier, Scale, and Cross-correlation CS 510 Lecture #12 February 26th, 2014 Where are we? • We can detect objects, but they can only differ in translation and 2D rotation • Then we introduced Fourier analysis. • Why? – Because Fourier analysis can help us with scale – Because Fourier analysis can make correlation faster Review: Discrete Fourier Transform • Problem: an image is not an analogue signal that we can integrate. • Therefore for 0 ≤ x < N and 0 ≤ u <N/2: N −1 * # 2πux & # 2πux &- F(u) = ∑ f (x),cos % ( − isin% (/ x=0 + $ N ' $ N '. And the discrete inverse transform is: € 1 N −1 ) # 2πux & # 2πux &, f (x) = ∑F(u)+cos % ( + isin% (. N x=0 * $ N ' $ N '- CS 510, Image Computaon, ©Ross 3/2/14 3 Beveridge & Bruce Draper € 2D Fourier Transform • So far, we have looked only at 1D signals • For 2D signals, the continuous generalization is: ∞ ∞ F(u,v) ≡ ∫ ∫ f (x, y)[cos(2π(ux + vy)) − isin(2π(ux + vy))] −∞ −∞ • Note that frequencies are now two- dimensional € – u= freq in x, v = freq in y • Every frequency (u,v) has a real and an imaginary component. CS 510, Image Computaon, ©Ross 3/2/14 4 Beveridge & Bruce Draper 2D sine waves • This looks like you’d expect in 2D Ø Note that the frequencies don’t have to be equal in the two dimensions. hp://images.google.com/imgres?imgurl=hFp://developer.nvidia.com/dev_content/cg/cg_examples/images/ sine_wave_perturbaon_ogl.jpg&imgrefurl=hFp://developer.nvidia.com/object/ cg_effects_explained.html&usg=__0FimoxuhWMm59cbwhch0TLwGpQM=&h=350&w=350&sz=13&hl=en&start=8&sig2=dBEtH0hp5I1BExgkXAe_kg&tbnid=fc yrIaap0P3M:&tbnh=120&tbnw=120&ei=llCYSbLNL4miMoOwoP8L&prev=/images%3Fq%3D2D%2Bsine%2Bwave%26gbv%3D2%26hl%3Den%26sa%3DG CS 510, Image Computaon, ©Ross 3/2/14 5 Beveridge & Bruce Draper 2D Discrete Fourier Transform N /2 N /2 * # 2π & # 2π &- F(u,v) = ∑ ∑ f (x, y),cos % (ux + vy)( − isin% (ux + vy)(/ x=−N /2 y=−N /2 + $ N ' $ N '. -
20. the Fourier Transform in Optics, II Parseval’S Theorem
20. The Fourier Transform in optics, II Parseval’s Theorem The Shift theorem Convolutions and the Convolution Theorem Autocorrelations and the Autocorrelation Theorem The Shah Function in optics The Fourier Transform of a train of pulses The spectrum of a light wave The spectrum of a light wave is defined as: 2 SFEt {()} where F{E(t)} denotes E(), the Fourier transform of E(t). The Fourier transform of E(t) contains the same information as the original function E(t). The Fourier transform is just a different way of representing a signal (in the frequency domain rather than in the time domain). But the spectrum contains less information, because we take the magnitude of E(), therefore losing the phase information. Parseval’s Theorem Parseval’s Theorem* says that the 221 energy in a function is the same, whether f ()tdt F ( ) d 2 you integrate over time or frequency: Proof: f ()tdt2 f ()t f *()tdt 11 F( exp(j td ) F *( exp(j td ) dt 22 11 FF() *(') exp([j '])tdtd ' d 22 11 FF( ) * ( ') [2 ')] dd ' 22 112 FF() *() d F () d * also known as 22Rayleigh’s Identity. The Fourier Transform of a sum of two f(t) F() functions t g(t) G() Faft() bgt () aF ft() bFgt () t The FT of a sum is the F() + sum of the FT’s. f(t)+g(t) G() Also, constants factor out. t This property reflects the fact that the Fourier transform is a linear operation. Shift Theorem The Fourier transform of a shifted function, f ():ta Ffta ( ) exp( jaF ) ( ) Proof : This theorem is F ft a ft( a )exp( jtdt ) important in optics, because we often encounter functions Change variables : uta that are shifting (continuously) along fu( )exp( j [ u a ]) du the time axis – they are called waves! exp(ja ) fu ( )exp( judu ) exp(jaF ) ( ) QED An example of the Shift Theorem in optics Suppose that we’re measuring the spectrum of a light wave, E(t), but a small fraction of the irradiance of this light, say , takes a different path that also leads to the spectrometer. -
Introduction to Frequency Domain Processing
MASSACHUSETTS INSTITUTE OF TECHNOLOGY DEPARTMENT OF MECHANICAL ENGINEERING 2.14 Analysis and Design of Feedback Control Systems Introduction to Frequency Domain Processing 1 Introduction - Superposition In this set of notes we examine an alternative to the time-domain convolution operations describing the input-output operations of a linear processing system. The methods developed here use Fourier techniques to transform the temporal representation f(t) to a reciprocal frequency domain space F (jω) where the difficult operation of convolution is replaced by simple multiplication. In addition, an understanding of Fourier methods gives qualitative insights to signal processing techniques such as filtering. Linear systems,by definition,obey the principle of superposition for the forced component of their responses: If linear system is at rest at time t = 0,and is subjected to an input u(t) that is the sum of a set of causal inputs,that is u(t)=u1(t)+u2(t)+...,the response y(t) will be the sum of the individual responses to each component of the input,that is y(t)=y1(t)+y2(t)+... Suppose that a system input u(t) may be expressed as a sum of complex n exponentials n s t u(t)= aie i , i=1 where the complex coefficients ai and constants si are known. Assume that each component is applied to the system alone; if at time t = 0 the system is at rest,the solution component yi(t)is of the form sit yi(t)=(yh (t))i + aiH(si)e where (yh(t))i is a homogeneous solution. -
Time Domain and Frequency Domain Signal Representation
ES442-Lab 1 ES440. Lab 1 Time Domain and Frequency Domain Signal Representation I. Objective 1. Get familiar with the basic lab equipment: signal generator, oscilloscope, spectrum analyzer, power supply. 2. Learn to observe the time domain signal representation with oscilloscope. 3. Learn to observe the frequency domain signal representation with oscilloscope. 4. Learn to observe signal spectrum with spectrum analyzer. 5. Understand the time domain representation of electrical signals. 6. Understand the frequency domain representation of electrical signals. II. Pre-lab 1) Let S1(t) = sin(2pf1t) and S1(t) = sin(2pf1t). Find the Fourier transform of S1(t) + S2(t) and S1(t) * S2(t). 2) Let S3(t) = rect (t/T), being a train of square waveform. What is the Fourier transform of S3(t)? 3) Assume S4(t) is a train of square waveform from +4V to -4V with period of 1 msec and 50 percent duty cycle, zero offset. Answer the following questions: a) What is the time representation of this signal? b) Express the Fourier series representation for this signal for all its harmonics (assume the signal is odd symmetric). c) Write the Fourier series expression for the first five harmonics. Does this signal have any DC signal? d) Determine the peak magnitudes and frequencies of the first five odd harmonics. e) Draw the frequency spectrum for the first five harmonics and clearly show the frequency and amplitude (in Volts) for each line spectra. 4) Given the above signal, assume it passes through a bandlimited twisted cable with maximum bandwidth of 10KHz. a) Using Fourier series, show the time-domain representation of the signal as it passes through the cable. -
Statistical Models in R Some Examples
Statistical Models Statistical Models in R Some Examples Steven Buechler Department of Mathematics 276B Hurley Hall; 1-6233 Fall, 2007 Statistical Models Outline Statistical Models Linear Models in R Statistical Models Regression Regression analysis is the appropriate statistical method when the response variable and all explanatory variables are continuous. Here, we only discuss linear regression, the simplest and most common form. Remember that a statistical model attempts to approximate the response variable Y as a mathematical function of the explanatory variables X1;:::; Xn. This mathematical function may involve parameters. Regression analysis attempts to use sample data find the parameters that produce the best model Statistical Models Linear Models The simplest such model is a linear model with a unique explanatory variable, which takes the following form. y^ = a + bx: Here, y is the response variable vector, x the explanatory variable, y^ is the vector of fitted values and a (intercept) and b (slope) are real numbers. Plotting y versus x, this model represents a line through the points. For a given index i,y ^i = a + bxi approximates yi . Regression amounts to finding a and b that gives the best fit. Statistical Models Linear Model with 1 Explanatory Variable ● 10 ● ● ● ● ● 5 y ● ● ● ● ● ● y ● ● y−hat ● ● ● ● 0 ● ● ● ● x=2 0 1 2 3 4 5 x Statistical Models Plotting Commands for the record The plot was generated with test data xR, yR with: > plot(xR, yR, xlab = "x", ylab = "y") > abline(v = 2, lty = 2) > abline(a = -2, b = 2, col = "blue") > points(c(2), yR[9], pch = 16, col = "red") > points(c(2), c(2), pch = 16, col = "red") > text(2.5, -4, "x=2", cex = 1.5) > text(1.8, 3.9, "y", cex = 1.5) > text(2.5, 1.9, "y-hat", cex = 1.5) Statistical Models Linear Regression = Minimize RSS Least Squares Fit In linear regression the best fit is found by minimizing n n X 2 X 2 RSS = (yi − y^i ) = (yi − (a + bxi )) : i=1 i=1 This is a Calculus I problem. -
A Statistical Test Suite for Random and Pseudorandom Number Generators for Cryptographic Applications
Special Publication 800-22 Revision 1a A Statistical Test Suite for Random and Pseudorandom Number Generators for Cryptographic Applications AndrewRukhin,JuanSoto,JamesNechvatal,Miles Smid,ElaineBarker,Stefan Leigh,MarkLevenson,Mark Vangel,DavidBanks,AlanHeckert,JamesDray,SanVo Revised:April2010 LawrenceE BasshamIII A Statistical Test Suite for Random and Pseudorandom Number Generators for NIST Special Publication 800-22 Revision 1a Cryptographic Applications 1 2 Andrew Rukhin , Juan Soto , James 2 2 Nechvatal , Miles Smid , Elaine 2 1 Barker , Stefan Leigh , Mark 1 1 Levenson , Mark Vangel , David 1 1 2 Banks , Alan Heckert , James Dray , 2 San Vo Revised: April 2010 2 Lawrence E Bassham III C O M P U T E R S E C U R I T Y 1 Statistical Engineering Division 2 Computer Security Division Information Technology Laboratory National Institute of Standards and Technology Gaithersburg, MD 20899-8930 Revised: April 2010 U.S. Department of Commerce Gary Locke, Secretary National Institute of Standards and Technology Patrick Gallagher, Director A STATISTICAL TEST SUITE FOR RANDOM AND PSEUDORANDOM NUMBER GENERATORS FOR CRYPTOGRAPHIC APPLICATIONS Reports on Computer Systems Technology The Information Technology Laboratory (ITL) at the National Institute of Standards and Technology (NIST) promotes the U.S. economy and public welfare by providing technical leadership for the nation’s measurement and standards infrastructure. ITL develops tests, test methods, reference data, proof of concept implementations, and technical analysis to advance the development and productive use of information technology. ITL’s responsibilities include the development of technical, physical, administrative, and management standards and guidelines for the cost-effective security and privacy of sensitive unclassified information in Federal computer systems. -
The Wavelet Tutorial Second Edition Part I by Robi Polikar
THE WAVELET TUTORIAL SECOND EDITION PART I BY ROBI POLIKAR FUNDAMENTAL CONCEPTS & AN OVERVIEW OF THE WAVELET THEORY Welcome to this introductory tutorial on wavelet transforms. The wavelet transform is a relatively new concept (about 10 years old), but yet there are quite a few articles and books written on them. However, most of these books and articles are written by math people, for the other math people; still most of the math people don't know what the other math people are 1 talking about (a math professor of mine made this confession). In other words, majority of the literature available on wavelet transforms are of little help, if any, to those who are new to this subject (this is my personal opinion). When I first started working on wavelet transforms I have struggled for many hours and days to figure out what was going on in this mysterious world of wavelet transforms, due to the lack of introductory level text(s) in this subject. Therefore, I have decided to write this tutorial for the ones who are new to the topic. I consider myself quite new to the subject too, and I have to confess that I have not figured out all the theoretical details yet. However, as far as the engineering applications are concerned, I think all the theoretical details are not necessarily necessary (!). In this tutorial I will try to give basic principles underlying the wavelet theory. The proofs of the theorems and related equations will not be given in this tutorial due to the simple assumption that the intended readers of this tutorial do not need them at this time.