07 Sampling and Reconstruction
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Convolution! (CDT-14) Luciano Da Fontoura Costa
Convolution! (CDT-14) Luciano da Fontoura Costa To cite this version: Luciano da Fontoura Costa. Convolution! (CDT-14). 2019. hal-02334910 HAL Id: hal-02334910 https://hal.archives-ouvertes.fr/hal-02334910 Preprint submitted on 27 Oct 2019 HAL is a multi-disciplinary open access L’archive ouverte pluridisciplinaire HAL, est archive for the deposit and dissemination of sci- destinée au dépôt et à la diffusion de documents entific research documents, whether they are pub- scientifiques de niveau recherche, publiés ou non, lished or not. The documents may come from émanant des établissements d’enseignement et de teaching and research institutions in France or recherche français ou étrangers, des laboratoires abroad, or from public or private research centers. publics ou privés. Convolution! (CDT-14) Luciano da Fontoura Costa [email protected] S~aoCarlos Institute of Physics { DFCM/USP October 22, 2019 Abstract The convolution between two functions, yielding a third function, is a particularly important concept in several areas including physics, engineering, statistics, and mathematics, to name but a few. Yet, it is not often so easy to be conceptually understood, as a consequence of its seemingly intricate definition. In this text, we develop a conceptual framework aimed at hopefully providing a more complete and integrated conceptual understanding of this important operation. In particular, we adopt an alternative graphical interpretation in the time domain, allowing the shift implied in the convolution to proceed over free variable instead of the additive inverse of this variable. In addition, we discuss two possible conceptual interpretations of the convolution of two functions as: (i) the `blending' of these functions, and (ii) as a quantification of `matching' between those functions. -
Eliminating Aliasing Caused by Discontinuities Using Integrals of the Sinc Function
Sound production – Sound synthesis: Paper ISMR2016-48 Eliminating aliasing caused by discontinuities using integrals of the sinc function Fabián Esqueda(a), Stefan Bilbao(b), Vesa Välimäki(c) (a)Aalto University, Dept. of Signal Processing and Acoustics, Espoo, Finland, fabian.esqueda@aalto.fi (b)Acoustics and Audio Group, University of Edinburgh, United Kingdom, [email protected] (c)Aalto University, Dept. of Signal Processing and Acoustics, Espoo, Finland, vesa.valimaki@aalto.fi Abstract A study on the limits of bandlimited correction functions used to eliminate aliasing in audio signals with discontinuities is presented. Trivial sampling of signals with discontinuities in their waveform or their derivatives causes high levels of aliasing distortion due to the infinite bandwidth of these discontinuities. Geometrical oscillator waveforms used in subtractive synthesis are a common example of audio signals with these characteristics. However, discontinuities may also be in- troduced in arbitrary signals during operations such as signal clipping and rectification. Several existing techniques aim to increase the perceived quality of oscillators by attenuating aliasing suf- ficiently to be inaudible. One family of these techniques consists on using the bandlimited step (BLEP) and ramp (BLAMP) functions to quasi-bandlimit discontinuities. Recent work on antialias- ing clipped audio signals has demonstrated the suitability of the BLAMP method in this context. This work evaluates the performance of the BLEP, BLAMP, and integrated BLAMP functions by testing whether they can be used to fully bandlimit aliased signals. Of particular interest are cases where discontinuities appear past the first derivative of a signal, like in hard clipping. These cases require more than one correction function to be applied at every discontinuity. -
Generalizing Sampling Theory for Time-Varying Nyquist Rates Using Self-Adjoint Extensions of Symmetric Operators with Deficiency Indices (1,1) in Hilbert Spaces
Generalizing Sampling Theory for Time-Varying Nyquist Rates using Self-Adjoint Extensions of Symmetric Operators with Deficiency Indices (1,1) in Hilbert Spaces by Yufang Hao A thesis presented to the University of Waterloo in fulfillment of the thesis requirement for the degree of Doctor of Philosophy in Applied Mathematics Waterloo, Ontario, Canada, 2011 c Yufang Hao 2011 I hereby declare that I am the sole author of this thesis. This is a true copy of the thesis, including any required final revisions, as accepted by my examiners. I understand that my thesis may be made electronically available to the public. ii Abstract Sampling theory studies the equivalence between continuous and discrete representa- tions of information. This equivalence is ubiquitously used in communication engineering and signal processing. For example, it allows engineers to store continuous signals as discrete data on digital media. The classical sampling theorem, also known as the theorem of Whittaker-Shannon- Kotel'nikov, enables one to perfectly and stably reconstruct continuous signals with a con- stant bandwidth from their discrete samples at a constant Nyquist rate. The Nyquist rate depends on the bandwidth of the signals, namely, the frequency upper bound. Intuitively, a signal's `information density' and ‘effective bandwidth' should vary in time. Adjusting the sampling rate accordingly should improve the sampling efficiency and information storage. While this old idea has been pursued in numerous publications, fundamental problems have remained: How can a reliable concept of time-varying bandwidth been defined? How can samples taken at a time-varying Nyquist rate lead to perfect and stable reconstruction of the continuous signals? This thesis develops a new non-Fourier generalized sampling theory which takes samples only as often as necessary at a time-varying Nyquist rate and maintains the ability to perfectly reconstruct the signals. -
The Best Approximation of the Sinc Function by a Polynomial of Degree N with the Square Norm
Hindawi Publishing Corporation Journal of Inequalities and Applications Volume 2010, Article ID 307892, 12 pages doi:10.1155/2010/307892 Research Article The Best Approximation of the Sinc Function by a Polynomial of Degree n with the Square Norm Yuyang Qiu and Ling Zhu College of Statistics and Mathematics, Zhejiang Gongshang University, Hangzhou 310018, China Correspondence should be addressed to Yuyang Qiu, [email protected] Received 9 April 2010; Accepted 31 August 2010 Academic Editor: Wing-Sum Cheung Copyright q 2010 Y. Qiu and L. Zhu. This is an open access article distributed under the Creative Commons Attribution License, which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited. The polynomial of degree n which is the best approximation of the sinc function on the interval 0, π/2 with the square norm is considered. By using Lagrange’s method of multipliers, we construct the polynomial explicitly. This method is also generalized to the continuous function on the closed interval a, b. Numerical examples are given to show the effectiveness. 1. Introduction Let sin cxsin x/x be the sinc function; the following result is known as Jordan inequality 1: 2 π ≤ sin cx < 1, 0 <x≤ , 1.1 π 2 where the left-handed equality holds if and only if x π/2. This inequality has been further refined by many scholars in the past few years 2–30. Ozban¨ 12 presented a new lower bound for the sinc function and obtained the following inequality: 2 1 4π − 3 π 2 π2 − 4x2 x − ≤ sin cx. -
Oversampling of Stochastic Processes
OVERSAMPLING OF STOCHASTIC PROCESSES by D.S.G. POLLOCK University of Leicester In the theory of stochastic differential equations, it is commonly assumed that the forcing function is a Wiener process. Such a process has an infinite bandwidth in the frequency domain. In practice, however, all stochastic processes have a limited bandwidth. A theory of band-limited linear stochastic processes is described that reflects this reality, and it is shown how the corresponding ARMA models can be estimated. By ignoring the limitation on the frequencies of the forcing function, in the process of fitting a conventional ARMA model, one is liable to derive estimates that are severely biased. If the data are sampled too rapidly, then maximum frequency in the sampled data will be less than the Nyquist value. However, the underlying continuous function can be reconstituted by sinc function or Fourier interpolation; and it can be resampled at a lesser rate corresponding to the maximum frequency of the forcing function. Then, there will be a direct correspondence between the parameters of the band-limited ARMA model and those of an equivalent continuous-time process; and the estimation biases can be avoided. 1 POLLOCK: Band-Limited Processes 1. Time-Limited versus Band-Limited Processes Stochastic processes in continuous time are usually modelled by filtered versions of Wiener processes which have infinite bandwidth. This seems inappropriate for modelling the slowly evolving trajectories of macroeconomic data. Therefore, we shall model these as processes that are limited in frequency. A function cannot be simultaneously limited in frequency and limited in time. One must choose either a band-limited function, which extends infinitely in time, or a time-limited function, which extents over an infinite range of frequencies. -
Shannon Wavelets Theory
Hindawi Publishing Corporation Mathematical Problems in Engineering Volume 2008, Article ID 164808, 24 pages doi:10.1155/2008/164808 Research Article Shannon Wavelets Theory Carlo Cattani Department of Pharmaceutical Sciences (DiFarma), University of Salerno, Via Ponte don Melillo, Fisciano, 84084 Salerno, Italy Correspondence should be addressed to Carlo Cattani, [email protected] Received 30 May 2008; Accepted 13 June 2008 Recommended by Cristian Toma Shannon wavelets are studied together with their differential properties known as connection coefficients. It is shown that the Shannon sampling theorem can be considered in a more general approach suitable for analyzing functions ranging in multifrequency bands. This generalization R ff coincides with the Shannon wavelet reconstruction of L2 functions. The di erential properties of Shannon wavelets are also studied through the connection coefficients. It is shown that Shannon ∞ wavelets are C -functions and their any order derivatives can be analytically defined by some kind of a finite hypergeometric series. These coefficients make it possible to define the wavelet reconstruction of the derivatives of the C -functions. Copyright q 2008 Carlo Cattani. This is an open access article distributed under the Creative Commons Attribution License, which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited. 1. Introduction Wavelets 1 are localized functions which are a very useful tool in many different applications: signal analysis, data compression, operator analysis, and PDE solving see, e.g., 2 and references therein. The main feature of wavelets is their natural splitting of objects into different scale components 1, 3 according to the multiscale resolution analysis. -
Chapter 11: Fourier Transform Pairs
CHAPTER 11 Fourier Transform Pairs For every time domain waveform there is a corresponding frequency domain waveform, and vice versa. For example, a rectangular pulse in the time domain coincides with a sinc function [i.e., sin(x)/x] in the frequency domain. Duality provides that the reverse is also true; a rectangular pulse in the frequency domain matches a sinc function in the time domain. Waveforms that correspond to each other in this manner are called Fourier transform pairs. Several common pairs are presented in this chapter. Delta Function Pairs For discrete signals, the delta function is a simple waveform, and has an equally simple Fourier transform pair. Figure 11-1a shows a delta function in the time domain, with its frequency spectrum in (b) and (c). The magnitude is a constant value, while the phase is entirely zero. As discussed in the last chapter, this can be understood by using the expansion/compression property. When the time domain is compressed until it becomes an impulse, the frequency domain is expanded until it becomes a constant value. In (d) and (g), the time domain waveform is shifted four and eight samples to the right, respectively. As expected from the properties in the last chapter, shifting the time domain waveform does not affect the magnitude, but adds a linear component to the phase. The phase signals in this figure have not been unwrapped, and thus extend only from -B to B. Also notice that the horizontal axes in the frequency domain run from -0.5 to 0.5. That is, they show the negative frequencies in the spectrum, as well as the positive ones. -
CHAPTER 3 ADC and DAC
CHAPTER 3 ADC and DAC Most of the signals directly encountered in science and engineering are continuous: light intensity that changes with distance; voltage that varies over time; a chemical reaction rate that depends on temperature, etc. Analog-to-Digital Conversion (ADC) and Digital-to-Analog Conversion (DAC) are the processes that allow digital computers to interact with these everyday signals. Digital information is different from its continuous counterpart in two important respects: it is sampled, and it is quantized. Both of these restrict how much information a digital signal can contain. This chapter is about information management: understanding what information you need to retain, and what information you can afford to lose. In turn, this dictates the selection of the sampling frequency, number of bits, and type of analog filtering needed for converting between the analog and digital realms. Quantization First, a bit of trivia. As you know, it is a digital computer, not a digit computer. The information processed is called digital data, not digit data. Why then, is analog-to-digital conversion generally called: digitize and digitization, rather than digitalize and digitalization? The answer is nothing you would expect. When electronics got around to inventing digital techniques, the preferred names had already been snatched up by the medical community nearly a century before. Digitalize and digitalization mean to administer the heart stimulant digitalis. Figure 3-1 shows the electronic waveforms of a typical analog-to-digital conversion. Figure (a) is the analog signal to be digitized. As shown by the labels on the graph, this signal is a voltage that varies over time. -
Fourier Analysis and Sampling Theory
Reading Required: Shirley, Ch. 9 Recommended: Ron Bracewell, The Fourier Transform and Its Applications, McGraw-Hill. Fourier analysis and Don P. Mitchell and Arun N. Netravali, “Reconstruction Filters in Computer Computer sampling theory Graphics ,” Computer Graphics, (Proceedings of SIGGRAPH 88). 22 (4), pp. 221-228, 1988. Brian Curless CSE 557 Fall 2009 1 2 What is an image? Images as functions We can think of an image as a function, f, from R2 to R: f(x,y) gives the intensity of a channel at position (x,y) Realistically, we expect the image only to be defined over a rectangle, with a finite range: • f: [a,b]x[c,d] Æ [0,1] A color image is just three functions pasted together. We can write this as a “vector-valued” function: ⎡⎤rxy(, ) f (,xy )= ⎢⎥ gxy (, ) ⎢⎥ ⎣⎦⎢⎥bxy(, ) We’ll focus in grayscale (scalar-valued) images for now. 3 4 Digital images Motivation: filtering and resizing In computer graphics, we usually create or operate What if we now want to: on digital (discrete)images: smooth an image? Sample the space on a regular grid sharpen an image? Quantize each sample (round to nearest enlarge an image? integer) shrink an image? If our samples are Δ apart, we can write this as: In this lecture, we will explore the mathematical underpinnings of these operations. f [n ,m] = Quantize{ f (n Δ, m Δ) } 5 6 Convolution Convolution in 2D One of the most common methods for filtering a In two dimensions, convolution becomes: function, e.g., for smoothing or sharpening, is called convolution. -
Filter Effects and Filter Artifacts in the Analysis of Electrophysiological Data
GENERAL COMMENTARY published: 09 July 2012 doi: 10.3389/fpsyg.2012.00233 Filter effects and filter artifacts in the analysis of electrophysiological data Andreas Widmann* and Erich Schröger Institute of Psychology, University of Leipzig, Leipzig, Germany *Correspondence: [email protected] Edited by: Guillaume A. Rousselet, University of Glasgow, UK Reviewed by: Rufin VanRullen, Centre de Recherche Cerveau et Cognition, France Lucas C. Parra, City College of New York, USA A commentary on unity gain at DC (the step response never FILTER EFFECTS VS. FILTER ARTIFACTS returns to one). These artifacts are due to a We also recommend to distinguish between Four conceptual fallacies in mapping the known misconception in FIR filter design in filter effects, that is, the obligatory effects time course of recognition EEGLAB1. The artifacts are further ampli- any filter with equivalent properties – cutoff by VanRullen, R. (2011). Front. Psychol. fied by filtering twice, forward and back- frequency, roll-off, ripple, and attenuation 2:365. doi: 10.3389/fpsyg.2011.00365 ward, to achieve zero-phase. – would have on the data (e.g., smoothing With more appropriate filters the under- of transients as demonstrated by the filter’s Does filtering preclude us from studying estimation of signal onset latency due to step response), and filter artifacts, that is, ERP time-courses? the smoothing effect of low-pass filtering effects which can be minimized by selection by Rousselet, G. A. (2012). Front. Psychol. could be narrowed down to about 4–12 ms of filter type and parameters (e.g., ringing). 3:131. doi: 10.3389/fpsyg.2012.00131 in the simulated dataset (see Figure 1 and Appendix for a simulation), that is, about an CAUSAL FILTERING In a recent review, VanRullen (2011) con- order of magnitude smaller than assumed. -
Efficient Implementations of Discrete Wavelet Transforms Using Fpgas Deepika Sripathi
Florida State University Libraries Electronic Theses, Treatises and Dissertations The Graduate School 2003 Efficient Implementations of Discrete Wavelet Transforms Using Fpgas Deepika Sripathi Follow this and additional works at the FSU Digital Library. For more information, please contact [email protected] THE FLORIDA STATE UNIVERSITY COLLEGE OF ENGINEERING EFFICIENT IMPLEMENTATIONS OF DISCRETE WAVELET TRANSFORMS USING FPGAs By DEEPIKA SRIPATHI A Thesis submitted to the Department of Electrical and Computer Engineering in partial fulfillment of the requirements for the degree of Master of Science Degree Awarded: Fall Semester, 2003 The members of the committee approve the thesis of Deepika Sripathi defended on November 18th, 2003. Simon Y. Foo Professor Directing Thesis Uwe Meyer-Baese Committee Member Anke Meyer-Baese Committee Member Approved: Reginald J. Perry, Chair, Department of Electrical and Computer Engineering The office of Graduate Studies has verified and approved the above named committee members ii ACKNOWLEDGEMENTS I would like to express my gratitude to my major professor, Dr. Simon Foo for his guidance, advice and constant support throughout my thesis work. I would like to thank him for being my advisor here at Florida State University. I would like to thank Dr. Uwe Meyer-Baese for his guidance and valuable suggestions. I also wish to thank Dr. Anke Meyer-Baese for her advice and support. I would like to thank my parents for their constant encouragement. I would like to thank my husband for his cooperation and support. I wish to thank the administrative staff of the Electrical and Computer Engineering Department for their kind support. Finally, I would like to thank Dr. -
Digital Signal Processing Tom Davinson
First SPES School on Experimental Techniques with Radioactive Beams INFN LNS Catania – November 2011 Experimental Challenges Lecture 4: Digital Signal Processing Tom Davinson School of Physics & Astronomy N I VE R U S I E T H Y T O H F G E D R I N B U Objectives & Outline Practical introduction to DSP concepts and techniques Emphasis on nuclear physics applications I intend to keep it simple … … even if it’s not … … I don’t intend to teach you VHDL! • Sampling Theorem • Aliasing • Filtering? Shaping? What’s the difference? … and why do we do it? • Digital signal processing • Digital filters semi-gaussian, moving window deconvolution • Hardware • To DSP or not to DSP? • Summary • Further reading Sampling Sampling Periodic measurement of analogue input signal by ADC Sampling Theorem An analogue input signal limited to a bandwidth fBW can be reproduced from its samples with no loss of information if it is regularly sampled at a frequency fs 2fBW The sampling frequency fs= 2fBW is called the Nyquist frequency (rate) Note: in practice the sampling frequency is usually >5x the signal bandwidth Aliasing: the problem Continuous, sinusoidal signal frequency f sampled at frequency fs (fs < f) Aliasing misrepresents the frequency as a lower frequency f < 0.5fs Aliasing: the solution Use low-pass filter to restrict bandwidth of input signal to satisfy Nyquist criterion fs 2fBW Digital Signal Processing D…igi wtahl aSt ignneaxtl? Processing Digital signal processing is the software controlled processing of sequential data derived from a digitised analogue signal. Some of the advantages of digital signal processing are: • functionality possible to implement functions which are difficult, impractical or impossible to achieve using hardware, e.g.