An Introduction to Statistical Signal Processing
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Probability Based Estimation Theory for Respondent Driven Sampling
Journal of Official Statistics, Vol. 24, No. 1, 2008, pp. 79–97 Probability Based Estimation Theory for Respondent Driven Sampling Erik Volz1 and Douglas D. Heckathorn2 Many populations of interest present special challenges for traditional survey methodology when it is difficult or impossible to obtain a traditional sampling frame. In the case of such “hidden” populations at risk of HIV/AIDS, many researchers have resorted to chain-referral sampling. Recent progress on the theory of chain-referral sampling has led to Respondent Driven Sampling (RDS), a rigorous chain-referral method which allows unbiased estimation of the target population. In this article we present new probability-theoretic methods for making estimates from RDS data. The new estimators offer improved simplicity, analytical tractability, and allow the estimation of continuous variables. An analytical variance estimator is proposed in the case of estimating categorical variables. The properties of the estimator and the associated variance estimator are explored in a simulation study, and compared to alternative RDS estimators using data from a study of New York City jazz musicians. The new estimator gives results consistent with alternative RDS estimators in the study of jazz musicians, and demonstrates greater precision than alternative estimators in the simulation study. Key words: Respondent driven sampling; chain-referral sampling; Hansen–Hurwitz; MCMC. 1. Introduction Chain-referral sampling has emerged as a powerful method for sampling hard-to-reach or hidden populations. Such sampling methods are favored for such populations because they do not require the specification of a sampling frame. The lack of a sampling frame means that the survey data from a chain-referral sample is contingent on a number of factors outside the researcher’s control such as the social network on which recruitment takes place. -
Random Signals
RANDOM SIGNALS Random Variables A random variable x(ξ) is a mapping that assigns a real number x to every outcome ξ from an abstract probability space. The mapping should satisfy the following two conditions: • the interval {x(ξ) ≤ x} is an event in the abstract probabilty space for every x; • Pr[x(ξ) < ∞] = 1 and Pr[x(ξ) = −∞] = 0. Cumulative distribution function (cdf) of a random variable x(ξ): Fx(x) = Pr{x(ξ) ≤ x}. Probability density function (pdf): dF (x) f (x) = x . x dx Then Z x Fx(x) = fx(x) dx. −∞ EE 524, Fall 2004, # 7 1 Since Fx(∞) = 1, we have normalization condition: Z ∞ fx(x)dx = 1. −∞ Several important properties: 0 ≤ Fx(x) ≤ 1,Fx(−∞) = 0,Fx(∞) = 1, Z ∞ fx(x) ≥ 0, fx(x) dx = 1. −∞ Simple interpretation: Pr{x − ∆/2 ≤ x(ξ) ≤ x + ∆/2} fx(x) = lim . ∆→0 ∆ EE 524, Fall 2004, # 7 2 Expectation of an arbitrary function g(x(ξ)): Z ∞ E {g(x(ξ))} = g(x)fx(x) dx. −∞ Mean: Z ∞ µx = E {x(ξ)} = xfx(x) dx. −∞ Variance of a real random variable x(ξ): 2 2 var{x} = σx = E {(x − E {x}) } = E {x2 − 2xE {x} + E {x}2} = E {x2} − (E {x})2 2 2 = E {x } − µx. Complex random variables: A complex random variable x(ξ) = xR(ξ) + jxI(ξ). Although the definition of the mean remains unchanged, the definition of variance changes for complex x(ξ): 2 2 var{x} = σx = E {|x − E {x}| } = E {|x|2 − xE {x}∗ − x∗E {x} + |E {x}|2} = E {|x|2} − |E {x}|2 2 2 = E {|x| } − |µx| . -
Detection and Estimation Theory Introduction to ECE 531 Mojtaba Soltanalian- UIC the Course
Detection and Estimation Theory Introduction to ECE 531 Mojtaba Soltanalian- UIC The course Lectures are given Tuesdays and Thursdays, 2:00-3:15pm Office hours: Thursdays 3:45-5:00pm, SEO 1031 Instructor: Prof. Mojtaba Soltanalian office: SEO 1031 email: [email protected] web: http://msol.people.uic.edu/ The course Course webpage: http://msol.people.uic.edu/ECE531 Textbook(s): * Fundamentals of Statistical Signal Processing, Volume 1: Estimation Theory, by Steven M. Kay, Prentice Hall, 1993, and (possibly) * Fundamentals of Statistical Signal Processing, Volume 2: Detection Theory, by Steven M. Kay, Prentice Hall 1998, available in hard copy form at the UIC Bookstore. The course Style: /Graduate Course with Active Participation/ Introduction Let’s start with a radar example! Introduction> Radar Example QUIZ Introduction> Radar Example You can actually explain it in ten seconds! Introduction> Radar Example Applications in Transportation, Defense, Medical Imaging, Life Sciences, Weather Prediction, Tracking & Localization Introduction> Radar Example The strongest signals leaking off our planet are radar transmissions, not television or radio. The most powerful radars, such as the one mounted on the Arecibo telescope (used to study the ionosphere and map asteroids) could be detected with a similarly sized antenna at a distance of nearly 1,000 light-years. - Seth Shostak, SETI Introduction> Estimation Traditionally discussed in STATISTICS. Estimation in Signal Processing: Digital Computers ADC/DAC (Sampling) Signal/Information Processing Introduction> Estimation The primary focus is on obtaining optimal estimation algorithms that may be implemented on a digital computer. We will work on digital signals/datasets which are typically samples of a continuous-time waveform. -
Z-Transformation - I
Digital signal processing: Lecture 5 z-transformation - I Produced by Qiangfu Zhao (Since 1995), All rights reserved © DSP-Lec05/1 Review of last lecture • Fourier transform & inverse Fourier transform: – Time domain & Frequency domain representations • Understand the “true face” (latent factor) of some physical phenomenon. – Key points: • Definition of Fourier transformation. • Definition of inverse Fourier transformation. • Condition for a signal to have FT. Produced by Qiangfu Zhao (Since 1995), All rights reserved © DSP-Lec05/2 Review of last lecture • The sampling theorem – If the highest frequency component contained in an analog signal is f0, the sampling frequency (the Nyquist rate) should be 2xf0 at least. • Frequency response of an LTI system: – The Fourier transformation of the impulse response – Physical meaning: frequency components to keep (~1) and those to remove (~0). – Theoretic foundation: Convolution theorem. Produced by Qiangfu Zhao (Since 1995), All rights reserved © DSP-Lec05/3 Topics of this lecture Chapter 6 of the textbook • The z-transformation. • z-変換 • Convergence region of z- • z-変換の収束領域 transform. -変換とフーリエ変換 • Relation between z- • z transform and Fourier • z-変換と差分方程式 transform. • 伝達関数 or システム • Relation between z- 関数 transform and difference equation. • Transfer function (system function). Produced by Qiangfu Zhao (Since 1995), All rights reserved © DSP-Lec05/4 z-transformation(z-変換) • Laplace transformation is important for analyzing analog signal/systems. • z-transformation is important for analyzing discrete signal/systems. • For a given signal x(n), its z-transformation is defined by ∞ Z[x(n)] = X (z) = ∑ x(n)z−n (6.3) n=0 where z is a complex variable. Produced by Qiangfu Zhao (Since 1995), All rights reserved © DSP-Lec05/5 One-sided z-transform and two-sided z-transform • The z-transform defined above is called one- sided z-transform (片側z-変換). -
Basics on Digital Signal Processing
Basics on Digital Signal Processing z - transform - Digital Filters Vassilis Anastassopoulos Electronics Laboratory, Physics Department, University of Patras Outline of the Lecture 1. The z-transform 2. Properties 3. Examples 4. Digital filters 5. Design of IIR and FIR filters 6. Linear phase 2/31 z-Transform 2 N 1 N 1 j nk n X (k) x(n) e N X (z) x(n) z n0 n0 Time to frequency Time to z-domain (complex) Transformation tool is the Transformation tool is complex wave z e j e j j j2k / N e e With amplitude ρ changing(?) with time With amplitude |ejω|=1 The z-transform is more general than the DFT 3/31 z-Transform For z=ejω i.e. ρ=1 we work on the N 1 unit circle X (z) x(n) z n And the z-transform degenerates n0 into the Fourier transform. I -z z=R+jI |z|=1 R The DFT is an expression of the z-transform on the unit circle. The quantity X(z) must exist with finite value on the unit circle i.e. must posses spectrum with which we can describe a signal or a system. 4/31 z-Transform convergence We are interested in those values of z for which X(z) converges. This region should contain the unit circle. I Why is it so? |a| N 1 N 1 X (z) x(n) zn x(n) (1/zn ) n0 n0 R At z=0, X(z) diverges ROC The values of z for which X(z) diverges are called poles of X(z). -
3F3 – Digital Signal Processing (DSP)
3F3 – Digital Signal Processing (DSP) Simon Godsill www-sigproc.eng.cam.ac.uk/~sjg/teaching/3F3 Course Overview • 11 Lectures • Topics: – Digital Signal Processing – DFT, FFT – Digital Filters – Filter Design – Filter Implementation – Random signals – Optimal Filtering – Signal Modelling • Books: – J.G. Proakis and D.G. Manolakis, Digital Signal Processing 3rd edition, Prentice-Hall. – Statistical digital signal processing and modeling -Monson H. Hayes –Wiley • Some material adapted from courses by Dr. Malcolm Macleod, Prof. Peter Rayner and Dr. Arnaud Doucet Digital Signal Processing - Introduction • Digital signal processing (DSP) is the generic term for techniques such as filtering or spectrum analysis applied to digitally sampled signals. • Recall from 1B Signal and Data Analysis that the procedure is as shown below: • is the sampling period • is the sampling frequency • Recall also that low-pass anti-aliasing filters must be applied before A/D and D/A conversion in order to remove distortion from frequency components higher than Hz (see later for revision of this). • Digital signals are signals which are sampled in time (“discrete time”) and quantised. • Mathematical analysis of inherently digital signals (e.g. sunspot data, tide data) was developed by Gauss (1800), Schuster (1896) and many others since. • Electronic digital signal processing (DSP) was first extensively applied in geophysics (for oil-exploration) then military applications, and is now fundamental to communications, mobile devices, broadcasting, and most applications of signal and image processing. There are many advantages in carrying out digital rather than analogue processing; among these are flexibility and repeatability. The flexibility stems from the fact that system parameters are simply numbers stored in the processor. -
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. -
Analog Transmit Signal Optimization for Undersampled Delay-Doppler
Analog Transmit Signal Optimization for Undersampled Delay-Doppler Estimation Andreas Lenz∗, Manuel S. Stein†, A. Lee Swindlehurst‡ ∗Institute for Communications Engineering, Technische Universit¨at M¨unchen, Germany †Mathematics Department, Vrije Universiteit Brussel, Belgium ‡Henry Samueli School of Engineering, University of California, Irvine, USA E-Mail: [email protected], [email protected], [email protected] Abstract—In this work, the optimization of the analog transmit achievable sampling rate fs at the receiver restricts the band- waveform for joint delay-Doppler estimation under sub-Nyquist width B of the transmitter and therefore the overall system conditions is considered. Based on the Bayesian Cramer-Rao´ performance. Since the sampling rate forms a bottleneck with lower bound (BCRLB), we derive an estimation theoretic design rule for the Fourier coefficients of the analog transmit signal respect to power resources and hardware limitations [2], it when violating the sampling theorem at the receiver through a is necessary to find a trade-off between high performance wide analog pre-filtering bandwidth. For a wireless delay-Doppler and low complexity. Therefore we discuss how to design channel, we obtain a system optimization problem which can be the transmit signal for delay-Doppler estimation without the solved in compact form by using an Eigenvalue decomposition. commonly used restriction from the sampling theorem. The presented approach enables one to explore the Pareto region spanned by the optimized analog waveforms. Furthermore, we Delay-Doppler estimation has been discussed for decades demonstrate how the framework can be used to reduce the in the signal processing community [3]–[5]. In [3] a subspace sampling rate at the receiver while maintaining high estimation based algorithm for the estimation of multi-path delay-Doppler accuracy. -
10. Linear Models and Maximum Likelihood Estimation ECE 830, Spring 2017
10. Linear Models and Maximum Likelihood Estimation ECE 830, Spring 2017 Rebecca Willett 1 / 34 Primary Goal General problem statement: We observe iid yi ∼ pθ; θ 2 Θ n and the goal is to determine the θ that produced fyigi=1. Given a collection of observations y1; :::; yn and a probability model p(y1; :::; ynjθ) parameterized by the parameter θ, determine the value of θ that best matches the observations. 2 / 34 Estimation Using the Likelihood Definition: Likelihood function p(yjθ) as a function of θ with y fixed is called the \likelihood function". If the likelihood function carries the information about θ brought by the observations y = fyigi, how do we use it to obtain an estimator? Definition: Maximum Likelihood Estimation θbMLE = arg max p(yjθ) θ2Θ is the value of θ that maximizes the density at y. Intuitively, we are choosing θ to maximize the probability of occurrence for y. 3 / 34 Maximum Likelihood Estimation MLEs are a very important type of estimator for the following reasons: I MLE occurs naturally in composite hypothesis testing and signal detection (i.e., GLRT) I The MLE is often simple and easy to compute I MLEs are invariant under reparameterization I MLEs often have asymptotic optimal properties (e.g. consistency (MSE ! 0 as N ! 1) 4 / 34 Computing the MLE If the likelihood function is differentiable, then θb is found from @ log p(yjθ) = 0 @θ If multiple solutions exist, then the MLE is the solution that maximizes log p(yjθ). That is, take the global maximizer. Note: It is possible to have multiple global maximizers that are all MLEs! 5 / 34 Example: Estimating the mean and variance of a Gaussian iid 2 yi = A + νi; νi ∼ N (0; σ ); i = 1; ··· ; n θ = [A; σ2]> n @ log p(yjθ) 1 X = (y − A) @A σ2 i i=1 n @ log p(yjθ) n 1 X = − + (y − A)2 @σ2 2σ2 2σ4 i i=1 n 1 X ) Ab = yi n i=1 n 2 1 X 2 ) σc = (yi − Ab) n i=1 Note: σc2 is biased! 6 / 34 Example: Stock Market (Dow-Jones Industrial Avg.) Based on this plot we might conjecture that the data is \on average" increasing. -
CONVOLUTION: Digital Signal Processing .R .Hamming, W
CONVOLUTION: Digital Signal Processing AN-237 National Semiconductor CONVOLUTION: Digital Application Note 237 Signal Processing January 1980 Introduction As digital signal processing continues to emerge as a major Decreasing the pulse width while increasing the pulse discipline in the field of electrical engineering, an even height to allow the area under the pulse to remain constant, greater demand has evolved to understand the basic theo- Figure 1c, shows from eq(1) and eq(2) the bandwidth or retical concepts involved in the development of varied and spectral-frequency content of the pulse to have increased, diverse signal processing systems. The most fundamental Figure 1d. concepts employed are (not necessarily listed in the order Further altering the pulse to that of Figure 1e provides for an [ ] of importance) the sampling theorem 1 , Fourier transforms even broader bandwidth, Figure 1f. If the pulse is finally al- [ ][] 2 3, convolution, covariance, etc. tered to the limit, i.e., the pulsewidth being infinitely narrow The intent of this article will be to address the concept of and its amplitude adjusted to still maintain an area of unity convolution and to present it in an introductory manner under the pulse, it is found in 1g and 1h the unit impulse hopefully easily understood by those entering the field of produces a constant, or ``flat'' spectrum equal to 1 at all digital signal processing. frequencies. Note that if ATe1 (unit area), we get, by defini- It may be appropriate to note that this article is Part II (Part I tion, the unit impulse function in time. -
Thompson Sampling on Symmetric Α-Stable Bandits
Thompson Sampling on Symmetric α-Stable Bandits Abhimanyu Dubey and Alex Pentland Massachusetts Institute of Technology fdubeya, [email protected] Abstract Thompson Sampling provides an efficient technique to introduce prior knowledge in the multi-armed bandit problem, along with providing remarkable empirical performance. In this paper, we revisit the Thompson Sampling algorithm under rewards drawn from α-stable dis- tributions, which are a class of heavy-tailed probability distributions utilized in finance and economics, in problems such as modeling stock prices and human behavior. We present an efficient framework for α-stable posterior inference, which leads to two algorithms for Thomp- son Sampling in this setting. We prove finite-time regret bounds for both algorithms, and demonstrate through a series of experiments the stronger performance of Thompson Sampling in this setting. With our results, we provide an exposition of α-stable distributions in sequential decision-making, and enable sequential Bayesian inference in applications from diverse fields in finance and complex systems that operate on heavy-tailed features. 1 Introduction The multi-armed bandit (MAB) problem is a fundamental model in understanding the exploration- exploitation dilemma in sequential decision-making. The problem and several of its variants have been studied extensively over the years, and a number of algorithms have been proposed that op- timally solve the bandit problem when the reward distributions are well-behaved, i.e. have a finite support, or are sub-exponential. The most prominently studied class of algorithms are the Upper Confidence Bound (UCB) al- gorithms, that employ an \optimism in the face of uncertainty" heuristic [ACBF02], which have been shown to be optimal (in terms of regret) in several cases [CGM+13, BCBL13]. -
On Products of Gaussian Random Variables
On products of Gaussian random variables Zeljkaˇ Stojanac∗1, Daniel Suessy1, and Martin Klieschz2 1Institute for Theoretical Physics, University of Cologne, Germany 2 Institute of Theoretical Physics and Astrophysics, University of Gda´nsk,Poland May 29, 2018 Sums of independent random variables form the basis of many fundamental theorems in probability theory and statistics, and therefore, are well understood. The related problem of characterizing products of independent random variables seems to be much more challenging. In this work, we investigate such products of normal random vari- ables, products of their absolute values, and products of their squares. We compute power-log series expansions of their cumulative distribution function (CDF) based on the theory of Fox H-functions. Numerically we show that for small arguments the CDFs are well approximated by the lowest orders of this expansion. For the two non-negative random variables, we also compute the moment generating functions in terms of Mei- jer G-functions, and consequently, obtain a Chernoff bound for sums of such random variables. Keywords: Gaussian random variable, product distribution, Meijer G-function, Cher- noff bound, moment generating function AMS subject classifications: 60E99, 33C60, 62E15, 62E17 1. Introduction and motivation Compared to sums of independent random variables, our understanding of products is much less comprehensive. Nevertheless, products of independent random variables arise naturally in many applications including channel modeling [1,2], wireless relaying systems [3], quantum physics (product measurements of product states), as well as signal processing. Here, we are particularly motivated by a tensor sensing problem (see Ref. [4] for the basic idea). In this problem we consider n1×n2×···×n tensors T 2 R d and wish to recover them from measurements of the form yi := hAi;T i 1 2 d j nj with the sensing tensors also being of rank one, Ai = ai ⊗ai ⊗· · ·⊗ai with ai 2 R .