Perfect Reconstructing Nonlinear Filter Banks
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Linearisation of Analogue to Digital and Digital to Analogue Converters Can Be Achieved by Threshold Tracking
LINEARISATION OF ANALOGUE TO DIGITAL AND DIGITAL TO ANALOGUE CONVERTERS ALAN CHRISTOPHER DENT, B.Sc, AMIEE, MIEEE A thesis submitted for the degree of Doctor of Philosophy, to the Faculty of Science, at the University of Edinburgh, May 1990. 7. DECLARATION OF ORIGINALITY I hereby declare that this thesis and the work reported herein was composed and originated entirely by myself in the Department of Electrical Engineering, at the University of Edinburgh, between October 1986 and May 1990. Iffl ABSTRACT Monolithic high resolution Analogue to Digital and Digital to Analogue Converters (ADC's and DAC's), cannot currently be manufactured with as much accuracy as is desirable, due to the limitations of the various fabrication technologies. The tolerance errors introduced in this way cause the converter transfer functions to be nonlinear. This nonlinearity can be quantified in static tests measuring integral nonlinearity (INL) and differential nonlinearity (DNL). In the dynamic testing of the converters, the transfer function nonlinearity is manifested as harmonic distortion, and intermodulation products. In general, regardless of the conversion technique being used, the effects of the transfer function nonlinearities get worse as converter resolution and speed increase. The result is that these nonlinearities can cause severe problems for the system designer who needs to accurately convert an input signal. A review is made of the performance of modern converters, and the existing methods of eliminating the nonlinearity, and of some of the schemes which have been proposed more recently. A new method is presented whereby code density testing techniques are exploited so that a sufficiently detailed characterisation of the converter can be made. -
Linear Filtering of Random Processes
Linear Filtering of Random Processes Lecture 13 Spring 2002 Wide-Sense Stationary A stochastic process X(t) is wss if its mean is constant E[X(t)] = µ and its autocorrelation depends only on τ = t1 − t2 ∗ Rxx(t1,t2)=E[X(t1)X (t2)] ∗ E[X(t + τ)X (t)] = Rxx(τ) ∗ Note that Rxx(−τ)=Rxx(τ)and Rxx(0) = E[|X(t)|2] Lecture 13 1 Example We found that the random telegraph signal has the autocorrelation function −c|τ| Rxx(τ)=e We can use the autocorrelation function to find the second moment of linear combinations such as Y (t)=aX(t)+bX(t − t0). 2 2 Ryy(0) = E[Y (t)] = E[(aX(t)+bX(t − t0)) ] 2 2 2 2 = a E[X (t)] + 2abE[X(t)X(t − t0)] + b E[X (t − t0)] 2 2 = a Rxx(0) + 2abRxx(t0)+b Rxx(0) 2 2 =(a + b )Rxx(0) + 2abRxx(t0) −ct =(a2 + b2)Rxx(0) + 2abe 0 Lecture 13 2 Example (continued) We can also compute the autocorrelation Ryy(τ)forτ =0. ∗ Ryy(τ)=E[Y (t + τ)Y (t)] = E[(aX(t + τ)+bX(t + τ − t0))(aX(t)+bX(t − t0))] 2 = a E[X(t + τ)X(t)] + abE[X(t + τ)X(t − t0)] 2 + abE[X(t + τ − t0)X(t)] + b E[X(t + τ − t0)X(t − t0)] 2 2 = a Rxx(τ)+abRxx(τ + t0)+abRxx(τ − t0)+b Rxx(τ) 2 2 =(a + b )Rxx(τ)+abRxx(τ + t0)+abRxx(τ − t0) Lecture 13 3 Linear Filtering of Random Processes The above example combines weighted values of X(t)andX(t − t0) to form Y (t). -
Active Non-Linear Filter Design Utilizing Parameter Plane Analysis Techniques
1 ACTIVE NONLINEAR FILTER DESIGN UTILIZING PARAMETER PLANE ANALYSIS TECHNIQUES by Wi 1 ?am August Tate It a? .1 L4 ULUfl United States Naval Postgraduate School ACTIVE NONLINEAR FILTER DESIGN UTILIZING PARAMETER PLANE ANALYSIS TECHNIQUES by William August Tate December 1970 ThU document ka6 bzzn appnjovtd ^oh. pubtic. K.Z.- tza&c and 6 ate; Ltb dUtAtbuutlon -U untunitzd. 13781 ILU I I . Active Nonlinear Filter Design Utilizing Parameter Plane Analysis Techniques by William August Tate Lieutenant, United States Navy B.S., Leland Stanford Junior University, 1963 Submitted in partial fulfillment of the requirement for the degrees of ELECTRICAL ENGINEER and MASTER OF SCIENCE IN ELECTRICAL ENGINEERING from the NAVAL POSTGRADUATE SCHOOL December 1970 . SCHOSf NAVAL*POSTGSi&UATE MOMTEREY, CALIF. 93940 ABSTRACT A technique for designing nonlinear active networks having a specified frequency response is presented. The technique, which can be applied to monolithic/hybrid integrated-circuit devices, utilizes the parameter plane method to obtain a graphical solution for the frequency response specification in terms of a frequency-dependent resistance. A limitation to this design technique is that the nonlinear parameter must be a slowly-varying quantity - the quasi-frozen assumption. This limitation manifests itself in the inability of the nonlinear networks to filter a complex signal according to the frequency response specification. Within the constraints of this limitation, excellent correlation is attained between the specification and measurements of the frequency response of physical networks Nonlinear devices, with emphasis on voltage-controlled nonlinear FET resistances and nonlinear device frequency controllers, are realized physically and investigated. Three active linear filters, modified with the nonlinear parameter frequency are constructed, and comparisons made to the required specification. -
Chapter 3 FILTERS
Chapter 3 FILTERS Most images are a®ected to some extent by noise, that is unexplained variation in data: disturbances in image intensity which are either uninterpretable or not of interest. Image analysis is often simpli¯ed if this noise can be ¯ltered out. In an analogous way ¯lters are used in chemistry to free liquids from suspended impurities by passing them through a layer of sand or charcoal. Engineers working in signal processing have extended the meaning of the term ¯lter to include operations which accentuate features of interest in data. Employing this broader de¯nition, image ¯lters may be used to emphasise edges | that is, boundaries between objects or parts of objects in images. Filters provide an aid to visual interpretation of images, and can also be used as a precursor to further digital processing, such as segmentation (chapter 4). Most of the methods considered in chapter 2 operated on each pixel separately. Filters change a pixel's value taking into account the values of neighbouring pixels too. They may either be applied directly to recorded images, such as those in chapter 1, or after transformation of pixel values as discussed in chapter 2. To take a simple example, Figs 3.1(b){(d) show the results of applying three ¯lters to the cashmere ¯bres image, which has been redisplayed in Fig 3.1(a). ² Fig 3.1(b) is a display of the output from a 5 £ 5 moving average ¯lter. Each pixel has been replaced by the average of pixel values in a 5 £ 5 square, or window centred on that pixel. -
A Multidimensional Filtering Framework with Applications to Local Structure Analysis and Image Enhancement
Linkoping¨ Studies in Science and Technology Dissertation No. 1171 A Multidimensional Filtering Framework with Applications to Local Structure Analysis and Image Enhancement Bjorn¨ Svensson Department of Biomedical Engineering Linkopings¨ universitet SE-581 85 Linkoping,¨ Sweden http://www.imt.liu.se Linkoping,¨ April 2008 A Multidimensional Filtering Framework with Applications to Local Structure Analysis and Image Enhancement c 2008 Bjorn¨ Svensson Department of Biomedical Engineering Linkopings¨ universitet SE-581 85 Linkoping,¨ Sweden ISBN 978-91-7393-943-0 ISSN 0345-7524 Printed in Linkoping,¨ Sweden by LiU-Tryck 2008 Abstract Filtering is a fundamental operation in image science in general and in medical image science in particular. The most central applications are image enhancement, registration, segmentation and feature extraction. Even though these applications involve non-linear processing a majority of the methodologies available rely on initial estimates using linear filters. Linear filtering is a well established corner- stone of signal processing, which is reflected by the overwhelming amount of literature on finite impulse response filters and their design. Standard techniques for multidimensional filtering are computationally intense. This leads to either a long computation time or a performance loss caused by approximations made in order to increase the computational efficiency. This dis- sertation presents a framework for realization of efficient multidimensional filters. A weighted least squares design criterion ensures preservation of the performance and the two techniques called filter networks and sub-filter sequences significantly reduce the computational demand. A filter network is a realization of a set of filters, which are decomposed into a structure of sparse sub-filters each with a low number of coefficients. -
The Unscented Kalman Filter for Nonlinear Estimation
The Unscented Kalman Filter for Nonlinear Estimation Eric A. Wan and Rudolph van der Merwe Oregon Graduate Institute of Science & Technology 20000 NW Walker Rd, Beaverton, Oregon 97006 [email protected], [email protected] Abstract 1. Introduction The Extended Kalman Filter (EKF) has become a standard The EKF has been applied extensively to the field of non- technique used in a number of nonlinear estimation and ma- linear estimation. General application areas may be divided chine learning applications. These include estimating the into state-estimation and machine learning. We further di- state of a nonlinear dynamic system, estimating parame- vide machine learning into parameter estimation and dual ters for nonlinear system identification (e.g., learning the estimation. The framework for these areas are briefly re- weights of a neural network), and dual estimation (e.g., the viewed next. Expectation Maximization (EM) algorithm) where both states and parameters are estimated simultaneously. State-estimation This paper points out the flaws in using the EKF, and The basic framework for the EKF involves estimation of the introduces an improvement, the Unscented Kalman Filter state of a discrete-time nonlinear dynamic system, (UKF), proposed by Julier and Uhlman [5]. A central and (1) vital operation performed in the Kalman Filter is the prop- (2) agation of a Gaussian random variable (GRV) through the system dynamics. In the EKF, the state distribution is ap- where represent the unobserved state of the system and proximated by a GRV, which is then propagated analyti- is the only observed signal. The process noise drives cally through the first-order linearization of the nonlinear the dynamic system, and the observation noise is given by system. -
Improved Audio Coding Using a Psychoacoustic Model Based on a Cochlear Filter Bank Frank Baumgarte
IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, VOL. 10, NO. 7, OCTOBER 2002 495 Improved Audio Coding Using a Psychoacoustic Model Based on a Cochlear Filter Bank Frank Baumgarte Abstract—Perceptual audio coders use an estimated masked a band and the range of spectral components that can interact threshold for the determination of the maximum permissible within a band, e.g., two sinusoids creating a beating effect. This just-inaudible noise level introduced by quantization. This es- interaction plays a crucial role in the perception of whether a timate is derived from a psychoacoustic model mimicking the properties of masking. Most psychoacoustic models for coding sound is noise-like which in turn corresponds to a significantly applications use a uniform (equal bandwidth) spectral decomposi- more efficient masking compared with a tone-like signal [2]. tion as a first step to approximate the frequency selectivity of the The noise or tone-like character is basically determined by human auditory system. However, the equal filter properties of the the amount of envelope fluctuations at the cochlear filter uniform subbands do not match the nonuniform characteristics of outputs which widely depend on the interaction of the spectral cochlear filters and reduce the precision of psychoacoustic mod- eling. Even so, uniform filter banks are applied because they are components in the pass-band of the filter. computationally efficient. This paper presents a psychoacoustic Many existing psychoacoustic models, e.g., [1], [3], and [4], model based on an efficient nonuniform cochlear filter bank employ an FFT-based transform to derive a spectral decom- and a simple masked threshold estimation. -
Median Filter Performance Based on Different Window Sizes for Salt and Pepper Noise Removal in Gray and RGB Images
International Journal of Signal Processing, Image Processing and Pattern Recognition Vol.8, No.10 (2015), pp.343-352 http://dx.doi.org/10.14257/ijsip.2015.8.10.34 Median Filter Performance Based on Different Window Sizes for Salt and Pepper Noise Removal in Gray and RGB Images Elmustafa S.Ali Ahmed1, Rasha E. A.Elatif2 and Zahra T.Alser3 Department of Electrical and Electronics Engineering, Red Sea University, Port Sudan, Sudan [email protected], [email protected], [email protected] Abstract Noised image is a major problem of digital image systems when images transferred between most electronic communications devices. Noise occurs in digital images due to transmit the image through the internet or maybe due to error generated by noisy sensors. Other types of errors are related to the communication system itself, since image needed to be transferred from analogue to digital and vice versa, also to be transmitted in most of the communication systems. Impulse noise added to image signal due to process of converting image signal or error from communication channel. The noise added to the original image by changes the intensity of some pixels while other remain unchanged. Salt-and-pepper noise is one of the impulse noises, to remove it a simplest way used by windowing the noisy image with a conventional median filter. Median filters are the most popular nonlinear filters extensively applied to eliminate salt- and-pepper noise. This paper evaluates the performance of median filter based on the effective median per window by using different window sizes and cascaded median filters. The performance of the proposed idea has been evaluated in MATLAB simulations on a gray and RGB images. -
Using Nonlinear Kalman Filtering to Estimate Signals
Using Nonlinear Kalman Filtering to Estimate Signals Dan Simon Revised September 10, 2013 It appears that no particular approximate [nonlinear] filter is consistently better than any other, though ... any nonlinear filter is better than a strictly linear one.1 The Kalman filter is a tool that can estimate the variables of a wide range of processes. In mathematical terms we would say that a Kalman filter estimates the states of a linear system. There are two reasons that you might want to know the states of a system: • First, you might need to estimate states in order to control the system. For example, the electrical engineer needs to estimate the winding currents of a motor in order to control its position. The aerospace engineer needs to estimate the velocity of a satellite in order to control its orbit. The biomedical engineer needs to estimate blood sugar levels in order to regulate insulin injection rates. • Second, you might need to estimate system states because they are interesting in their own right. For example, the electrical engineer needs to estimate power system parameters in order to predict failure probabilities. The aerospace engineer needs to estimate satellite position in order to intelligently schedule future satellite activities. The biomedical engineer needs to estimate blood protein levels in order to evaluate the health of a patient. The standard Kalman filter is an effective tool for estimation, but it is limited to linear systems. Most real-world systems are nonlinear, in which case Kalman filters do not directly apply. In the real world, nonlinear filters are used more often than linear filters, because in the real world, systems are nonlinear. -
Nonlinear Matched Filter
A MUTUAL INFORMATION EXTENSION TO THE MATCHED FILTER Deniz Erdogmus1, Rati Agrawal2, Jose C. Principe2 1 CSEE Department, Oregon Graduate Institute, OHSU, Portland, OR 97006, USA 2 CNEL, ECE Department, University of Florida, Gainesville, FL 32611, USA Abstract. Matched filters are the optimal linear filters demonstrated the superior performance of information for signal detection under linear channel and white noise theoretic measures over second order statistics in signal conditions. Their optimality is guaranteed in the additive processing [4]. In this general framework, we assume an white Gaussian noise channel due to the sufficiency of arbitrary instantaneous channel and additive noise with an second order statistics. In this paper, we introduce a nonlinear arbitrary distribution. Specifically in the case of linear filter for signal detection based on the Cauchy-Schwartz channels, we will consider Cauchy noise, which is a member quadratic mutual information (CS-QMI) criterion. This filter of the family of symmetric α-stable distributions, and is an is still implementing correlation but now in a high impulsive noise source. These types of noise distributions are dimensional transformed space defined by the kernel utilized known to plague signal detection [5, 6]. in estimating the CS-QMI. Simulations show that the Specifically, the proposed nonlinear signal detection nonlinear filter significantly outperforms the traditional filter is based on the Cauchy-Schwartz Quadratic Mutual matched filter in nonlinear channels, as expected. In the Information (CS-QMI) measure that has been proposed by linear channel case, the proposed filter outperforms the Principe et al. [7]. This definition of mutual information is matched filter when the received signal is corrupted by preferred because of the existence of a simple nonparametric impulsive noise such as Cauchy-distributed noise, but estimator for the CS-QMI, which in turn forms the basis of performs at the same level in Gaussian noise. -
Signal Processing with Scipy: Linear Filters
PyData CookBook 1 Signal Processing with SciPy: Linear Filters Warren Weckesser∗ F Abstract—The SciPy library is one of the core packages of the PyData stack. focus on a specific subset of the capabilities of of this sub- It includes modules for statistics, optimization, interpolation, integration, linear package: the design and analysis of linear filters for discrete- algebra, Fourier transforms, signal and image processing, ODE solvers, special time signals. These filters are commonly used in digital signal functions, sparse matrices, and more. In this chapter, we demonstrate many processing (DSP) for audio signals and other time series data of the tools provided by the signal subpackage of the SciPy library for the design and analysis of linear filters for discrete-time signals, including filter with a fixed sample rate. representation, frequency response computation and optimal FIR filter design. The chapter is split into two main parts, covering the two broad categories of linear filters: infinite impulse response Index Terms—algorithms, signal processing, IIR filter, FIR filter (IIR) filters and finite impulse response (FIR) filters. Note. We will display some code samples as transcripts CONTENTS from the standard Python interactive shell. In each interactive Python session, we will have executed the following without Introduction....................1 showing it: IIR filters in scipy.signal ..........1 IIR filter representation.........2 >>> import numpy as np >>> import matplotlib.pyplot as plt Lowpass filter..............3 >>> np.set_printoptions(precision=3, linewidth=50) Initializing a lowpass filter.......5 Bandpass filter..............5 Also, when we show the creation of plots in the transcripts, we Filtering a long signal in batches....6 won’t show all the other plot commands (setting labels, grid Solving linear recurrence relations...6 lines, etc.) that were actually used to create the corresponding FIR filters in scipy.signal ..........7 figure in the paper. -
Filters CW100322-01
PHY3128 Electronics for Measurement Systems Linear Filters Linear Filters Introduction A linear filter is a circuit with an output Vout that is a linear function of its input Vin and that is intended to remove unwanted parts of a signal. Such filters are often described by specifying how they modify the amplitude and phase of each frequency component of the signal with a transfer function € € 2 Vout (ω) a0 + a1ω + a2ω + ... G(ω) = = 2 . (1) Vin (ω) b0 + b1ω + b2ω + ... This quotient of two polynomials is known as a Padé function and is usually expressed as the product of its complex poles and zeros, e.g. € (ωz0 + jω) × (ωz1 + jω) × ... G(ω) = (2) (ωp0 + jω) × (ωp1 + jω) × ... Much cryptic engineering-jargon arises from this convention. For example a ‘single-pole op-amp’ is one with an open-loop gain accurately described by € a A(ω ) = 0 (3) ω 0 + jω which has a single complex pole at ω = jω0 . In principle, designing a filter is a straightforward matter – one uses a computer and/or creates a chain of stages until the desired transfer function is obtained. However, non-trivial designs often do not work well; they tend to require extremely precise component values that are not available in practice. There are many of types of filter; the most commonly encountered include: Butterworth (maximum flatness in pass-band), Chebyshev (sharpest transition from pass-band to stop-band), Bessel (best linear phase vs frequency response). General Remarks In the context of test and measurement systems, filters irreversibly remove information from the signal and should be used sparingly, particularly if it would be possible to post-process signals digitally.