Showing papers on "Filter design published in 2004"

Acoustic Echo and Noise Control: A Practical Approach

Abstract: List of Figures.List of Tables.Preface.Acknowledgments.Abbreviations and Acronyms.Part I: Basics.1 Introduction.1.1 Some History.1.2 Overview of the Book.2 Acoustic Echo and Noise Control Systems.2.1 Notation.2.2 Applications.3 Fundamentals.3.1 Signals.3.2 Acoustic Echoes.3.3 Standards.Part II: Algorithms.4 Error Criteria and Cost Functions.4.1 Error Criteria for Adaptive Filters.4.2 Error Criteria for Filter Design.4.3 Error Criteria for Speech Processing and Control Purposes.5 Wiener Filter.5.1 Time-Domain Solution.5.2 Frequency-Domain Solution.6 Linear Prediction.6.1 Normal Equations.6.2 Levinson{Durbin Recursion.7 Algorithms for Adaptive Filters.7.1 The Normalized Least Mean Square Algorithm.7.2 The Affine Projection Algorithm.7.3 The Recursive Least Squares Algorithm.7.4 The Kalman Algorithm.Part III: Acoustic Echo and Noise Control.8 Traditional Methods for Stabilization of Electroacoustic Loops.8.1 Adaptive Line Enhancement.8.2 Frequency Shift.8.3 Controlled Attenuation.9 Echo Cancellation.9.1 Processing Structures.9.2 Stereophonic and Multichannel Echo Cancellation.10 Residual Echo and Noise Suppression.10.1 Basics.10.2 Suppression of Residual Echoes.10.3 Suppression of Background Noise.10.4 Combining Background Noise and Residual Echo Suppression.11 Beamforming.11.1 Basics.11.2 Characteristics of Microphone Arrays.11.3 Fixed Beamforming.11.4 Adaptive Beamforming.Part IV: Control and Implementation Issues.12 System Control-Basic Aspects.12.1 Convergence versus Divergence Speed.12.2 System Levels for Control Design.13 Control of Echo Cancellation Systems.13.1 Pseudooptimal Control Parameters for the NLMS Algorithm.13.2 Pseudooptimal Control Parameters for the Affine Projection Algorithm.13.3 Summary of Pseudooptimal Control Parameters.13.4 Detection and Estimation Methods.13.5 Detector Overview and Combined Control Methods.14 Control of Noise and Echo Suppression Systems.14.1 Estimation of Spectral Power Density of Background Noise.14.2 Musical Noise.14.3 Control of Filter Characteristics.15 Control for Beamforming.15.1 Practical Problems.15.2 Stepsize Control.16 Implementation Issues.16.1 Quantization Errors.16.2 Number Representation Errors.16.3 Arithmetical Errors.16.4 Fixed Point versus Floating Point.16.5 Quantization of Filter Taps.Part V: Outlook and Appendixes.17 Outlook.Appendix A: Subband Impulse Responses.A.1 Consequences for Subband Echo Cancellation.A.2 Transformation.A.3 Concluding Remarks.Appendix B: Filterbank Design.B.1 Conditions for Approximately Perfect Reconstruction.B.2 Filter Design Using a Product Approach.B.3 Design of Prototype Lowpass Filters.B.4 Analysis of Prototype Filters and the Filterbank System.References.Index.

Nonlinear Signal Processing: A Statistical Approach

Abstract: Preface. Acknowledgments. Acronyms. 1. Introduction. 1.1 Non--Gaussian Random Processes. 1.1.1 Generalized Gaussian Distributions and Weighted Medians. 1.1.2 Stable Distributions and Weighted Myriads. 1.2 Statistical Foundations. 1.3 The Filtering Problem. 1.3.1 Moment Theory. PART I: STATISTICAL FOUNDATIONS. 2. Non--Gaussian Models. 2.1 Generalized Gaussian Distributions. 2.2 Stable Distributions. 2.2.1 Definitions. 2.2.2 Symmetric Stable Distributions. 2.2.3 Generalized Central Limit Theorem. 2.2.4 Simulation of Stable Sequences. 2.3 Lower Order Moments. 2.3.1 Fractional Lower Order Moments. 2.3.2 Zero Order Statistics. 2.3.3 Parameter Estimation of Stable Distributions. Problems. 3. Order Statistics. 3.1 Distributions of Order Statistics. 3.2 Moments of Order Statistics. 3.2.1 Order Statistics From Uniform Distributions. 3.2.2 Recurrence Relations. 3.3 Order Statistics Containing Outliers. 3.4 Joint Statistics of Ordered and Non--Ordered Samples. Problems. 4. Statistical Foundations of Filtering. 4.1 Properties of Estimators. 4.2 Maximum Likelihood Estimation. 4.3 Robust Estimation. Problems. PART II: SIGNAL PROCESSING WITH ORDER STATISTICS. 5. Median and Weighted Median Smoothers. 5.1 Running Median Smoothers. 5.1.1 Statistical Properties. 5.1.2 Root Signals (Fixed Points). 5.2 Weighted Median Smoothers. 5.2.1 The Center Weighted Median Smoother. 5.2.2 Permutation Weighted Median Smoothers. 5.3 Threshold Decomposition Representation. 5.3.1 Stack Smoothers. 5.4 Weighted Medians in Least Absolute Deviation (LAD) Regression. 5.4.1 Foundation and Cost Functions. 5.4.2 LAD Regression with Weighted Medians. 5.4.3 Simulation. Problems. 6. Weighted Median Filters. 6.1 Weighted Median Filters With Real--Valued Weights. 6.1.1 Permutation Weighted Median Filters. 6.2 Spectral Design of Weighted Median Filters. 6.2.1 Median Smoothers and Sample Selection Probabilities. 6.2.2 SSPs for Weighted Median Smoothers. 6.2.3 Synthesis of WM Smoothers. 6.2.4 General Iterative Solution. 6.2.5 Spectral Design of Weighted Median Filters Admitting Real--Valued Weights. 6.3 The Optimal Weighted Median Filtering Problem. 6.3.1 Threshold Decomposition for Real--Valued Signals. 6.3.2 The Least Mean Absolute (LMA) Algorithm. 6.4 Recursive Weighted Median Filters. 6.4.1 Threshold Decomposition Representation of Recursive WM Filters. 6.4.2 Optimal Recursive Weighted Median Filtering. 6.5 Mirrored Threshold Decomposition and Stack Filters. 6.5.1 Stack Filters. 6.5.2 Stack Filter Representation of Recursive WM Filters. 6.6 Complex Valued Weighted Median Filter. 6.6.1 Phase Coupled Complex WM Filters. 6.6.2 Marginal Phase Coupled Complex WM Filter. 6.6.3 Complex Threshold Decomposition. 6.6.4 Optimal Marginal Phase Coupled Complex WM. 6.6.5 Spectral Design of Complex Valued Weighted Medians. 6.7 Weighted Median Filters for Multichannel Signals. 6.7.1 Marginal WM Filter. 6.7.2 Vector WM Filter. 6.7.3 Weighted Multichannel Median Filtering Structures. 6.7.4 Filter Optimization. Problems. 7. Linear Combination or Order Statistics. 7.1 L--Estimates of Location. 7.2 L--Smoothers. 7.3 L --Filters. 7.3.1 Design and Optimization of L Filters. 7.4 Lj Permutation Filters. 7.5 Hybrid Median/Linear FIR Filters. 7.5.1 Median and FIR Affinity Trimming. 7.6 Linear Combination of Weighted Medians. 7.6.1 LCWM Filters. 7.6.2 Design of LCWM Filters. 7.6.3 Symmetric LCWM Filters. Problems. PART III: SIGNAL PROCESSING WITH THE STABLE MODEL. 8. Myriad Smoothers. 8.1 FLOM Smoothers. 8.2 Running Myriad Smoothers. 8.3 Optimality of the Sample Myriad. 8.4 Weighted Myriad Smoothers. 8.5 Fast Weighted Myriad Computation. 8.6 Weighted Myriad Smoother Design. 8.6.1 Center Weighted Myriads for Image Denoising. 8.6.2 Myriadization. Problems. 9. Weighted Myriad Filters. 9.1 Weighted Myriad Filters with Real--Valued Weights. 9.2 Fast Real--Valued Weighted Myriad Computation. 9.3 Weighted Myriad Filter Design. 9.3.1 Myriadization. 9.3.2 Optimization. Problems. References. Appendix A: Software Guide. Index.

Dual-band bandpass filters using equal-length coupled-serial-shunted lines and Z-transform technique

Abstract: A synthesizing method is presented to design and implement digital dual-band filters in the microwave frequency range. A dual-band filter consists of a bandstop filter and a wide-band bandpass filter in a cascade connection, wherein the transfer functions of both the bandpass filter and bandstop filter are expressed in the Z domain. The bandstop filter is implemented by using a coupled-serial-shunted line structure, while the wide-band bandpass filter is constructed by using a serial-shunted line configuration. In particular, the bandwidth of each passband of the dual-band filter is controllable by adjusting the characteristics of both the bandpass filter and bandstop filter. By neglecting the dispersion effect between microstrip lines of different widths over a wide bandwidth, a dual-band filter is realized in the form of microstrip lines and its frequency responses are measured to validate this method.

A delay-dependent approach to robust H/sub /spl infin// filtering for uncertain discrete-time state-delayed systems

Abstract: A delay-dependent approach to robust H/sub /spl infin// filtering is proposed for linear discrete-time uncertain systems with multiple delays in the state. The uncertain parameters are supposed to reside in a polytope and the attention is focused on the design of robust filters guaranteeing a prescribed H/sub /spl infin// noise attenuation level. The proposed filter design methodology incorporates some recently appeared results, such as Moon's new version of the upper bound for the inner product of two vectors and de Oliveira's idea of parameter-dependent stability, which greatly reduce the overdesign introduced in the derivation process. In addition to the full-order filtering problem, the challenging reduced-order case is also addressed by using different linearization procedures. Both full- and reduced-order filters can be obtained from the solution of convex optimization problems in terms of linear matrix inequalities, which can be solved via efficient interior-point algorithms. Numerical examples have been presented to illustrate the feasibility and advantages of the proposed methodologies.

Robust H/sub /spl infin// filter design of uncertain descriptor systems with discrete and distributed delays

Abstract: The robust H/sub /spl infin// filtering problem for a class of continuous-time uncertain linear descriptor systems with time-varying discrete and distributed delays is investigated. The time delays are assumed to be constant and known. The uncertainties under consideration are norm-bounded, and possible time-varying, uncertainties. Sufficient condition for the existence of an H/sub /spl infin// filter is expressed in terms of strict linear matrix inequalities (LMIs). Instead of using decomposition technique, a unified form of LMIs is proposed to show the exponential stability of the augmented systems. The condition for assuring the stability of the "fast" subsystem is implied from the unified form of LMIs, which is shown to be less conservative than the characteristic equation based conditions or matrix norm-based conditions. The suitable filter is derived through a convex optimization problem. A numerical example is given to show the effectiveness of the method.

Static Analysis of Digital Filters

Abstract: We present an Abstract Interpretation-based framework for automatically analyzing programs containing digital filters. Our framework allows refining existing analyses so that they can handle given classes of digital filters. We only have to design a class of symbolic properties that describe the invariants throughout filter iterations, and to describe how these properties are transformed by filter iterations. Then, the analysis allows both inference and proofs of the properties about the program variables that are tied to any such filter.

A cascaded continuous-time /spl Sigma//spl Delta/ Modulator with 67-dB dynamic range in 10-MHz bandwidth

Abstract: This paper presents the design of a 2-2 cascaded continuous-time sigma-delta modulator. The cascaded modulator comprises two stages with second-order continuous-time resonator loopfilters, 4-bit quantizers, and feedback digital-to-analog converters. The digital noise cancellation filter design is determined using continuous-time to discrete-time transformation of the sigma-delta loopfilter transfer functions. The required matching between the analog and digital filter coefficients is achieved by means of simple digital calibration of the noise cancellation filter. Measurement results of a 0.18-/spl mu/m CMOS prototype chip demonstrate 67-dB dynamic range in a 10-MHz bandwidth at 8 times oversampling for a single continuous-time cascaded modulator. Two cascaded modulators in quadrature configuration provide 20-MHz aggregate bandwidth. Measured anti-alias suppression is over 50 dB for input signals in the band from 150 to 170 MHz around the sampling frequency of 160 MHz.

Design and analysis of UWB band pass filter with ring filter

Abstract: To realize an ultra wideband (UWB) bandpass filter (BPF) in wireless communications, a device is developed using a new ring filter. The ring filter is compact, with low insertion loss, sharp rejection and a constant group delay within the UWB pass band. The ring filter is made to control the attenuation pole frequency by adjusting both the ring and the stub impedance. A center frequency and a bandwidth of the 5-stage UWB BPF are designed based on the transmission line model. The circuit conditions of two attenuation poles at either side of the passband are given together with controlling them.

Practical analog and digital filter design

Abstract: Introduction to Filters and Filter Design Software. Analog Filter Approximation Functions. Analog Lowpass, Highpass, Bandpass, and Bandstop Filters. Analog Filter Implementation Using Active Filters. Introduction to Discrete-Time Systems. Infinite Impulse Response Digital Filter Design. Finite Impulse Response Digital Filter Design. Digital Filter Implementation Using C. Digital Filtering Using the FFT. Appendices.

Distributed Active Filter Systems (DAFSs): A New Approach to Power System Harmonics

Abstract: This paper proposes a distributed active filter system (DAFS) for alleviating the harmonic distortion of power systems. The proposed DAFS consists of multiple active filter units installed on the same location or different locations within the power system. The active filter units of the proposed DAFS can cooperate, without any communication among them, to reduce the voltage harmonic distortion of the power lines. Each individual active filter unit functions like a harmonic conductance to reduce voltage harmonics. A droop relationship between the harmonic conductance and the volt-ampere of the active filter unit is programmed into the controller of each unit so multiple active filter units can share the workload of harmonic filtering. The slope of the droop is determined by the volt-ampere rating of the active filter unit in order to distribute the harmonic filtering workload in proportion to the rated capacity of each unit. The principle of operation is explained in this paper and test results based on computer simulation and laboratory test bench are provided to validate the functionalities of the proposed DAFS.

Design of parallel-coupled microstrip filters with suppression of spurious resonances using substrate suspension

Abstract: Substrate suspension is used to suppress the spurious response of microstrip bandpass filters at twice the passband frequency (2f/sub o/). It is known that a proper height of substrate suspension can be used to equalize the even- and odd-mode phase velocities for coupled microstrip lines. In this paper, this property is applied to design the coupled stages of a parallel-coupled line filter so that the spurious response at 2f/sub o/ can be completely suppressed. The individual image impedance for each coupled stage is changed accordingly. Required filter design formulas are derived for a series of coupled stages having different image impedances. Several filters made on a substrate of relative high dielectric constant are designed and fabricated. The measured results show that a rejection level of better than -40 dB to the spurious resonance at 2f/sub o/ can be obtained.

An 8Gb/s source-synchronous I/O link with adaptive receiver equalization, offset cancellation and clock deskew

Abstract: A source-synchronous I/O link with adaptive receiver-side equalization has been implemented in 0.13-/spl mu/m bulk CMOS technology. The transceiver is optimized for small area (360 /spl mu/m /spl times/ 360 /spl mu/m) and low power (280 mW). The analog equalizer is implemented as an 8-way interleaved, 4-tap discrete-time linear filter. The equalization improved the data rate of a 102 cm backplane interconnect by 110%. On-die adaptive logic determines optimal receiver settings through comparator offset cancellation, data alignment of the transmitter and receiver, clock de-skew and setting filter coefficients for equalization. The noise-margin degradation due to statistical variation in converged coefficient values was less than 3%.

An LMI approach to the H/sub /spl infin// filter design for uncertain systems with distributed delays

Abstract: This paper investigates the problem of robust H/sub /spl infin// filter design for linear distributed delay systems with norm-bounded time-varying parameter uncertainties. The distributed delays are assumed to appear in both the state and measurement equations. The problem we address is the design of a filter, such that for all admissible uncertainties, the resulting error system is asymptotically stable and satisfies a prescribed H/sub /spl infin// performance level. A sufficient condition is obtained to guarantee the existence of desired H/sub /spl infin// filters, which can be constructed by solving certain linear matrix inequalities. The effectiveness of the proposed design method is demonstrated by a numerical example.

SVD-based design and new structures for variable fractional-delay digital filters

Abstract: This paper shows that the problem of designing one-dimensional (1-D) variable fractional-delay (VFD) digital filter can be elegantly reduced to the easier subproblems that involve one-dimensional (1-D) constant filter (subfilter) designs and 1-D polynomial approximations. By utilizing the singular value decomposition (SVD) of the variable design specification, we prove that both 1-D constant filters and 1-D polynomials possess either symmetry or anti-symmetry simultaneously. Therefore, a VFD filter can be efficiently obtained by designing 1-D constant filters with symmetrical or antisymmetrical coefficients and performing 1-D symmetrical or antisymmetrical approximations. To perform the weighted-least-squares (WLS) VFD filter design, a new WLS-SVD method is also developed. Moreover, an objective criterion is proposed for selecting appropriate subfilter orders and polynomial degrees. Our computer simulations have shown that the SVD-based design and WLS-SVD design can achieve much higher design accuracy with significantly reduced filter, complexity than the existing WLS design method. Another important part of the paper proposes two new structures for efficiently implementing the resulting VFD filter, which require less computational complexity than the so-called Farrow structure.

Power system having a phase locked loop with a notch filter

Abstract: A power system having a phase locked loop (PLL) with a notch filter for synchronous reference frame sequence separation. The notch filter includes a generalized integrator tuned to the notch frequency. The output of the generalized integrator is summed with the filter input signal and the output of the summer is fed back to the integrator input. In one embodiment, the notch filter is programmable by a control signal generated in response to the filter output signal, thereby causing the filter to self-regulate to a frequency related to the PLL input signal. Illustrative power systems include active VAR generators and active rectifiers. One or more additional notch filters may be used to remove harmonic components from derived synchronous reference frame sequence components.

A novel common-subexpression-elimination method for synthesizing fixed-point FIR filters

Abstract: We propose a common-subexpression-elimination (CSE) method for the synthesis of fixed-point finite-impulse response (FIR) filters. The proposed CSE algorithm considers both the redundancy among the canonic-signed-digit (CSD) filter coefficients and the length of the critical path in the multiplier block of a transposed-form FIR filter. Therefore, the proposed CSE method can perform tradeoff designs between complexity and the throughput rate. The number of adders synthesized by our method is commensurate with that by the graph-dependence algorithms. On the other hand, our method can synthesize a high-order complicated FIR filter in a few seconds.

A dual-passband filter of canonical structure for satellite applications

Abstract: Due to the complex arrangement of frequency plans and spatial coverages in modern satellite communication systems, channels that are noncontiguous in frequency may be amplified by a single power amplifier and transmitted to the ground through one beam. In this letter, a dual-mode canonical filter with dual-passband is presented. The filter adopts dual-mode technique for mass and volume reduction. Canonical structure is adopted for maximum zero realization. To validate the design technique, a six-pole dual-mode dual-passband filter of canonical structure for Ka-band (30/20GHz) satellite transponder is realized. The measured frequency response of the filter shows good agreement with the computed one.

Systems and methods for implementing a sample rate converter using hardware and software to maximize speed and flexibility

Abstract: Systems and methods for converting a digital input data stream from a first sample rate to a second, fixed sample rate using a combination of hardware and software components. In one embodiment, a system includes a rate estimator configured to estimate the sample rate of an input data stream, a phase selection unit configured to select a phase for interpolation of a set of polyphase filter coefficients based on the estimated sample rate, a coefficient interpolator configured to interpolate the filter coefficients based on the selected phase, and a convolution unit configured to convolve the interpolated filter coefficients with samples of the input data stream to produce samples of a re-sampled output data stream. One or more hardware or software components are shared between multiple channels that can process data streams having independently variable sample rates.

Delay-constrained scheduling: power efficiency, filter design, and bounds

Abstract: In this paper, packet scheduling with maximum delay constraints is considered with the objective to minimize average transmit power over Gaussian channels. The main emphasis is on deriving robust schedulers which do not rely on the knowledge of the source arrival process. Towards that end, we first show that all schedulers (robust or otherwise) which guarantee a maximum queuing delay for each packet are equivalent to a time-varying linear filter. Using the connection between filtering and scheduling, we study the design of optimal power minimizing robust schedulers. Two cases, motivated by filtering connection, are studied in detail. First, a time-invariant robust scheduler is presented and its performance is completely characterized. Second, we present the optimal time-varying robust scheduler, and show that it has a very intuitive time water-filling structure. We also present upper and lower bounds on the performance of power-minimizing schedulers as a function of delay constraints. The new results form an important step towards understanding of the packet time-scale interactions between physical layer metric of power and network layer metric of delay

A system and method for excluding narrow band noise from a communication channel

Abstract: A signal filtering system and method that may be used in conjunction with a repeater or an input stage of a base-station. The system may include an analog to digital converter adapted to sample a received signal and to produce a data stream corresponding to the received signal in the time domain, a filtering block having one or more digital filter elements, wherein each of said one or more filter elements is adapted to filter one or more sets of frequency bands associated with one or more communication channel, and a controller adapted to configure said one or more digital filter elements based on parameters stored on a database and/or based on parameters received via a modem.

Electronically tunable current-mode second-order universal filter using minimum elements

Abstract: A new electronically tunable universal filter with single-input, triple-output employing only four elements (two capacitors and two negative-type second generation current controlled conveyors (CCCII)) is proposed. The proposed filter realises three basic filter functions simultaneously: lowpass, highpass and bandpass. The validity of the proposed filter is verified through PSPICE simulations.

Method and apparatus for improved high-speed adaptive equalization

Abstract: Improved high-speed adaptive equalization is presented that may involve converting an optical signal into an electrical signal and performing equalization by (i) filtering the electrical signal with an analog filter according to at least one filter coefficient to produce a filtered output, (ii) generating an error signal from the filtered output according to an error function, (iii) providing at least one control signal to the analog filter for adjusting the at least one filter coefficient, (iv) detecting a relationship between a change in the at least one filter coefficient and a change in the error signal, and (v) adjusting the at least one filter coefficient according to the relationship to minimize the error signal. The least one coefficient may comprise a plurality of coefficients, and the relationship may be a gradient estimate having multiple components, each determined by varying only one of the coefficients and detecting a resulting change in the error signal.

Adaptive two-pass rank order filter to remove impulse noise in highly corrupted images

Abstract: In this paper, we present an adaptive two-pass rank order filter to remove impulse noise in highly corrupted images. When the noise ratio is high, rank order filters, such as the median filter for example, can produce unsatisfactory results. Better results can be obtained by applying the filter twice, which we call two-pass filtering. To further improve the performance, we develop an adaptive two-pass rank order filter. Between the passes of filtering, an adaptive process is used to detect irregularities in the spatial distribution of the estimated impulse noise. The adaptive process then selectively replaces some pixels changed by the first pass of filtering with their original observed pixel values. These pixels are then kept unchanged during the second filtering. In combination, the adaptive process and the second filter eliminate more impulse noise and restore some pixels that are mistakenly altered by the first filtering. As a final result, the reconstructed image maintains a higher degree of fidelity and has a smaller amount of noise. The idea of adaptive two-pass processing can be applied to many rank order filters, such as a center-weighted median filter (CWMF), adaptive CWMF, lower-upper-middle filter, and soft-decision rank-order-mean filter. Results from computer simulations are used to demonstrate the performance of this type of adaptation using a number of basic rank order filters.

Harmonic-suppression LTCC filter with the step-impedance quarter-wavelength open stub

Abstract: An effective method to design a low-temperature co-fired ceramic RF bandpass filter with suppression of the harmonic frequency is demonstrated in this paper. The second harmonic, which appears in the frequency band of 4.8-5.0 GHz, is very significant and should be reduced in the 2.4-GHz wireless local area network and Bluetooth application. This feature of harmonic frequency suppression is very important in a communication system to improve linearity, output power, intermodulation performance, etc. The harmonic-suppression filter can be easily obtained by adopting the characteristic of equivalent quarter-wavelength resonators. The detailed three-dimensional layout of each layer is disclosed. By analyzing the influences of the dielectric constant and layer thickness of a ceramic sheet by electromagnetic simulation, the optimal condition for the filter design can be obtained. The measured results agree well with the simulation.

Design of signal-adapted multidimensional lifting scheme for lossy coding

Abstract: This paper proposes a new method for the design of lifting filters to compute a multidimensional nonseparable wavelet transform Our approach is stated in the general case, and is illustrated for the 2-D separable and for the quincunx images Results are shown for the JPEG2000 database and for satellite images acquired on a quincunx sampling grid The design of efficient quincunx filters is a difficult challenge which has already been addressed for specific cases Our approach enables the design of less expensive filters adapted to the signal statistics to enhance the compression efficiency in a more general case It is based on a two-step lifting scheme and joins the lifting theory with Wiener's optimization The prediction step is designed in order to minimize the variance of the signal, and the update step is designed in order to minimize a reconstruction error Application for lossy compression shows the performances of the method