Showing papers on "Infinite impulse response published in 1997"

Blind spatio-temporal equalization and impulse response estimation for MIMO channels using a Godard cost function

Abstract: Equalization and estimation of the matrix impulse response function of multiple-input multiple-output (MIMO) digital communications channels in the absence of any training sequences is considered. An iterative, Godard (1980) cost-based approach is considered for spatio-temporal equalization and MIMO impulse response estimation. Stationary points of the cost function are investigated with particular attention to the case when finite-length equalizers exist. Sufficient conditions are derived under which all stable local minima correspond to desirable minima. The inputs are extracted and cancelled one by one. The matrix impulse response is then obtained by cross-correlating the extracted inputs with the observed outputs. Identifiability conditions are analyzed.

Discrete-time linear and nonlinear aerodynamic impulse responses for efficient cfd analyses

Abstract: This dissertation discusses the mathematical existence and the numerical identification of linear and nonlinear aerodynamic impulse response functions. Differences between continuous-time and discrete-time system theories, which permit the identification and efficient use of these functions, will be detailed. Important input/output definitions and the concept of linear and nonlinear systems with memory will also be discussed. It will be shown that indicial (step or steady) responses (such as Wagner's function), forced harmonic responses (such as Theodorsen's function or those from doublet lattice theory), and responses to random inputs (such as gusts) can all be obtained from an aerodynamic impulse response function. This will establish the aerodynamic discrete-time impulse response function as the most fundamental and computationally efficient aerodynamic function that can be extracted from any given discrete-time, aerodynamic system. The results presented in this dissertation help to unify the understanding of classical two-dimensional continuous-time theories with modern three-dimensional, discrete-time theories. Nonlinear aerodynamic impulse responses are identified using the Volterra theory of nonlinear systems. The theory is described and a discrete-time kernel identification technique is presented. The kernel identification technique is applied to a simple nonlinear circuit for illustrative purposes. The method is then applied to the nonlinear viscous Burger's equation as an example of an application to a simple CFD model. Finally, the method is applied to a three-dimensional aeroelastic model using the CAP-TSD (Computational Aeroelasticity Program - Transonic Small Disturbance) code and then to a two-dimensional model using the CFL3D Navier-Stokes code. Comparisons of accuracy and computational cost savings are presented. Because of its mathematical generality, an important attribute of this methodology is that it is applicable to a wide range of nonlinear, discrete-time systems.

System and method for spatial temporal-filtering for improving compressed digital video

Abstract: A filter that filters in the spatial and temporal domain in a single step with filtering coefficients that can be varied depending upon the complexity of the video and the motion between the adjacent frames comprises: a IIR filter, a threshold unit, and a coefficient register. The IIR filter and threshold unit are coupled to receive video data. The IIR filter is also coupled to the coefficient register and the threshold unit. The IIR filter receives coefficients, a, from the coefficient register and uses them to filter the video data received. The IIR filter filters the data in the vertical, horizontal and temporal dimensions in a single step. The filtered data output by the IIR filter is sent to the threshold unit. The threshold unit compares the absolute value of the difference between the filtered data and the raw video data to a threshold value from the coefficient register, and then outputs either the raw video data or the filtered data. The present invention is advantageous because it preserves significant edges in video sequence; it preserves motion changes in video sequences; it reduces noise; and it uses minimal memory storage and introduces minimal processing delay. The present invention also includes methods for filtering in parallel the pixel data in one step for the horizontal, vertical and temporal dimensions.

Finite memory-length linear multiuser detection for asynchronous CDMA communications

Abstract: Decorrelating, linear, minimum mean-squared error (LMMSE), and noise-whitening multiuser detectors for code-division multiple-access systems (CDMA) are ideally infinite memory-length (referred to as IIR) detectors. To obtain practical detectors, which have low implementation complexity and are suitable for CDMA systems with time-variant system parameters (e.g., the number of users, the delays of users, and the signature waveforms), linear finite-memory-length (referred to as FIR) multiuser detectors are studied in this paper. They are obtained by truncating the IIR detectors or by finding optimal FIR detectors. The signature waveforms are not restricted to be time-invariant (periodic over symbol interval). Thus, linear multiuser detection is generalized to systems with spreading sequences longer than the symbol interval. Conditions for the stability of the truncated detectors are discussed. Stable truncated detectors are shown to be near-far resistant if the received powers are upper bounded, and if the memory length is large enough (but finite). Numerical examples demonstrate that moderate memory lengths are sufficient to obtain the performance of the IIR detectors even with a severe near-far problem.

Low-order modeling of head-related transfer functions using balanced model truncation

Abstract: We propose a novel technique for the design of low-order infinite impulse response (IIR) filter models of head-related transfer functions (HRTFs) that uses balanced model truncation. We design tenth-order IIR filters, suitable for efficient real-time implementation, from preprocessed HRTF impulse responses that are of significantly superior quality to current IIR models derived with the Prony and Yule-Walker methods.

Multiuser demodulation for DS-CDMA systems in fading channels

Abstract: Multiuser demodulation algorithms for centralized receivers of asynchronous direct-sequence (DS) spread-spectrum code-division multiple-access (CDMA) systems in frequency-selective fading channels are studied. Both DS-CDMA systems with short (one symbol interval) and long (several symbol intervals) spreading sequences are considered. Linear multiuser receivers process ideally the complete received data block. The approximation of ideal infinite memory-length (IIR) linear multiuser detectors by finite memory-length (FIR) detectors is studied. It is shown that the FIR detectors can be made near-far resistant under a given ratio between maximum and minimum received power of users by selecting an appropriate memory-length. Numerical examples demonstrate the fact that moderate memory-lengths of the FIR detectors are sufficient to achieve the performance of the ideal IIR detectors even under severe near-far conditions. Multiuser demodulation in relatively fast fading channels is analyzed. The optimal maximum likelihood sequence detection receiver and suboptimal receivers are considered. The parallel interference cancellation (PIC) receiver is demonstrated to achieve better performance in known channels than the decorrelating receiver, but it is observed to be more sensitive to channel coefficient estimation errors than the decorrelator. At high channel loads the PIC receiver suffers from bit error rate (BER) saturation, whereas the decorrelating receiver does not. Choice of channel estimation filters is shown to be crucial if low BER is required. Data-aided channel estimation is shown to be more robust than decision-directed channel estimation, which may suffer from BER saturation caused by hang-ups at high signal-to-noise ratios. Multiuser receivers for dynamic CDMA systems are studied. Algorithms for ideal linear detector computation are derived and their complexity is analyzed. The complexity of the linear detector computation is a cubic function of KL, where K and L are the number of users and multipath components, respectively. Iterative steepest descent, conjugate gradient, and preconditioned conjugate gradient algorithms are proposed to reduce the complexity. The computational requirements for one iteration are a quadratic function of KL. The iterative detectors are also shown to be applicable for parallel implementation. Simulation results demonstrate that a moderate number of iterations yields the performance of the corresponding ideal linear detectors. A quantitative analysis shows that the PIC receivers are significantly simpler to implement than the linear receivers and only moderately more complex than the conventional matched filter bank receiver.

Method and apparatus for three-dimensional deconvolution of optical microscope images

Abstract: An adaptive structure of a Wiener filter is used to deconvolve three-dimensional wide-field microscope images for the purposes of improving spatial resolution and removing out-of-focus light. The filter is a three-dimensional kernel representing a finite-impulse-response (FIR) structure requiring on the order of one thousand (1000) taps or more to achieve an acceptable mean-square-error. Converging to a solution is done in the spatial-domain and therefore does not experience many of the problems of frequency-domain solutions. Alternatively, a three-dimensional kernel representing an infinite-impulse-response (IIR) structure may be employed. An IIR structure typically requires fewer taps to achieve the same or better performance, resulting in higher resolution images with less noise and faster computations.

A time-domain implementation of surface acoustic impedance condition with and without flow

Abstract: The impedance condition in computational aeroacoustic applications is required in order to model acoustically treated walls. The application of this condition in time-domain methods, however, is extremely difficult because of the convolutions involved. In this paper, a time-domain method is developed which overcomes the computational difficulties associated with these convolutions. This method builds on the z-transform from control and signal processing theory and the z-domain model of the impedance. The idea of using the z-domain operations originates from the computational electromagnetics community. When the impedance is expressed in the z-domain with a rational function, the inverse z-transform of the impedance condition results in only infinite impulse response type, digital, recursive filter operations. These operations, unlike convolutions, require only limited past-time knowledge of the acoustic pressures and velocities on the surface. Examples of one- and two-dimensional problems with and without flow indicate that the method promises success in aeroacoustic applications.

Suppression of multiple narrow-band interference using real-time adaptive notch filters

Abstract: The process of detection of narrow-band interference to a broad-band signal in a noisy environment using adaptive filtering techniques attributable to Kwan and Martin (1989) is further refined by modifying the gradient-search algorithm to reduce hardware complexity. The Kwan and Martin algorithm implies a geometric increase in hardware as the number of interferers increases. The improved method admits to only an arithmetic increase to achieve the similar results. Thus Kwan and Martin filters designed to handle 3 to 10 notches require 12.5% to 124% more hardware than the new filters. Simulations of the new method show it to be robust, functioning well in a variety of practical signal processing applications.

Adaptive Laguerre-lattice filters

Abstract: Adaptive Laguerre-based filters provide an attractive alternative to adaptive FIR filters in the sense that they require fewer parameters to model a linear time-invariant system with a long impulse response. We present an adaptive Laguerre-lattice structure that combines the desirable features of the Laguerre structure (i.e., guaranteed stability, unique global minimum, and small number of parameters M for a prescribed level of modeling error) with the numerical robustness and low computational complexity of adaptive FIR lattice structures. The proposed configuration is based on an extension to the IIR case of the FIR lattice filter; it is a cascade of identical sections but with a single-pole all-pass filter replacing the delay element used in the conventional (FIR) lattice filter. We utilize this structure to obtain computationally efficient adaptive algorithms (O(M) computations per time instant). Our adaptive Laguerre-lattice filter is an extension of the gradient adaptive lattice (GAL) technique, and it demonstrates the same desirable properties, namely, (1) excellent steady-state behavior, (2) relatively fast initial convergence (comparable with that of an RLS algorithm for Laguerre structure), and good numerical stability. Simulation results indicate that for systems with poles close to the unit circle, where an (adaptive) FIR model of very high order would be required to meet a prescribed modeling error, an adaptive Laguerre-lattice model of relatively low order achieves the prescribed bound after just a few updates of the recursions in the adaptive algorithm.

A decision feedback recurrent neural equalizer as an infinite impulse response filter

Abstract: An adaptive decision feedback recurrent neural equalizer (DFRNE), which models a kind of an IIR structure, is proposed. Its performance is compared with the traditional linear and nonlinear equalizers with FIR structures for various communication channels. The small size and high performance of the DFRNE makes it suitable for high-speed channel equalization.

Modulated lapped transform method

Abstract: A fast modulated lapped transform (MLT) method and architecture for image compression and decompression systems minimizes blocking artifacts associated with JPEG based discrete cosine transform (DCT) image compression systems. The MLT method combines fast block processing capabilities of wavelet transforms and fast block processing of DCT image compression systems. The modular and pipeline MLT architecture is fast by block processing but avoids the visual blocking artifacts that can be seen in most DCT-based compression systems. Improved MLT processors are implemented by an infinite impulse response filter operating on a product of the MLT window function and the input data stream. Forward and reverse MLT processors include a new fused multiply-add logic for fast computations and localized interconnections. The MLT processors can be combined into a bank of parallel processors in a one dimensional MLT architecture, which can be used for two-dimensional image transformation. The improved MLT implementation enables a modular architecture having a reduce number of multipliers and interconnects well suited for practical VLSI implementation.

Multistage adaptive filters for in-phase processing of line-frequency signals

Abstract: The authors describe two adaptive multistage digital filters for 50/60-Hz line-frequency signal processing in zero-crossing detectors and synchronous power systems. These filters combine a median filter with adaptive predictors, either finite-impulse response (FIR)- or infinite-impulse response (IIR)-based, thus making it possible to extract the sinusoidal signals from noise and strong disturbances without phase shifting the primary frequency signal. The median filter is used as a prefilter because it can remove deep commutation notches from the waveform. Adaptation allows the filters to track the exact instantaneous line frequency and avoids the selectivity problem encountered with a fixed filter.

Adaptive IIR filtered-v algorithms for active noise control

Abstract: This paper reports on the development of “full-gradient” and “simplified-gradient” versions of the filtered-u algorithm for active noise control. After discussing the general principles of active noise control, the paper presents full mathematical derivations of the full-gradient, simplified-gradient, and Feintuch-based versions of the filtered-u algorithm, clearly showing the various levels of simplifying assumptions that are made along the way. Finally, some illustrative simulation results are presented.

Method and apparatus for data detection for PRML data channels

Abstract: A method and apparatus are provided for data detection for a partial-response maximum-likelihood (PRML) data detection channel in a direct access storage device. A class-IV partial response (PR4) signal is applied to a PR4 Viterbi detector to provide a PR4 Viterbi output and is applied to a first matching delay circuit to provide a delayed PR4 signal. The PR4 Viterbi output is subtracted from the delayed PR4 signal and a resulting signal is applied to a first filter having a frequency response of 1/(1-αD2). The filtered output signal is applied to a second filter providing a second filtered output signal. The PR4 Viterbi output is applied to a second matching delay circuit to provide a delayed PR4 Viterbi output signal. The delayed PR4 Viterbi output is corrected responsive to the second filtered output signal. The first filter is an infinite impulse response (IIR) filter and the filtered output signal represents whitened noise and modified PRML error events. The second filter is a matched filter used to identify dominant error events.

Efficient ASIC and FPGA implementations of IIR filters for real time edge detection

Abstract: To reduce the computation cost, Deriche (1987, 1990) extended the work from Canny (1986) on optimal edge detectors to the use of recursive filters. Nevertheless, this cost is still too high for real time implementation on FGGA circuits. Here, we optimized both the algorithmic and architectural aspects of the original Deriche filter. A new organization of the filter is proposed at the 2D and 1D levels which reduces the memory size and the computation cost by a factor of two for both software and hardware implementations. We prove that the use of only 3 bits to code the scale parameter does not reduce the quality. The result from this choice is that the first order recursive filter which is the basic block of the entire architecture can be built with only 4 adders. The architecture of a 10 Mpixels/second filter on an unique FPGA is described.

Design of computationally efficient elliptic IIR filters with a reduced number of shift-and-add operations in multipliers

Abstract: In this paper, a new design method for elliptic IIR filters that provides the implementation of half of the multiplication constants with few shifters and adders is proposed. An IIR filter, when derived by the bilinear transformation from an elliptic minimal Q-factor analog prototype, has its z-plane poles on the circle that is orthogonal to the unit circle and has the center on the real axis of the plane. Due to this property, the center of the circle can be used as a parameter for the representation of a pole, whereas the second parameter is the radius of the pole. It is shown in this paper that the center of the circle is uniquely determined by the frequency for which the filter attenuation is 3 dB. This result is used for the realization based on the parallel connection of two all-pass networks. It is shown that all second-order all-pass sections can be implemented with one common multiplication constant determined by the center of the circle. The design method is presented that, by an appropriate distribution of a margin in the filter performance, predetermines the value of the common constant according to the desired number of shift-and-add operations. This way, half of the multipliers are replaced with a limited number of shifters and adders. Conventional computer programs for IIR elliptic digital filters can be used. The direct approach for the distribution of the z-plane poles among two all-pass functions is developed. The application and efficiency of the proposed design method are demonstrated by examples.

Advanced filter design

Abstract: Classical filter design techniques return only one design from an infinite collection of alternative designs, or fail to design filters when solutions exist. These classical techniques hide a wealth of alternative filter designs that are more robust when implemented in analog circuits, digital hardware, and embedded software. We present (1) case studies of optimal analog and digital IIR filters that cannot be designed with classical techniques, and (2) the formal, mathematical framework that underlies their solutions. We have automated the advanced filter design techniques in software.

Realizable warped IIR filters and their properties

Abstract: Digital filters where unit delays are replaced with frequency dependent delays, such as first order allpass sections, are often called warped filters since they implement filter specifications on a warped non-uniform frequency scale. Warped IIR (WIIR) filters cannot be realized directly due to delay free loops. Specific solutions have been known that make WIIR filters realizable but no general approach has been available so far. In this paper we will explore the generation of such filters, including new filter structures. The robustness and computational efficiency of WIIR filters are studied and most potential applications are discussed.

Recurrent Neural Network Architectures: An Overview

Abstract: In this paper, we have first considered a number of popular recurrent neural network architectures. Then, two subclasses of general recurrent neural network architectures are introduced. It is shown that all these popular recurrent neural network architectures can be grouped under either of these two subclasses of general recurrent neural network architectures. It is also inferred that these two subclasses of recurrent neural network architectures are distinct, in that it is not possible to transform from one form to the other. Two recently introduced recurrent neural network architectures specifically designed for special purposes, viz., for overcoming long term temporal dependency, and for data structure classifications are also considered.

A perceptually based embedded subband speech coder

Abstract: A new scheme for robust, high-quality, embedded speech coding based on subband decomposition and perceptually optimized bit allocation and prioritization is presented. An infinite impulse response (IIR) quadrature mirror filterbank (QMF) performs subband decomposition. A perceptual model, computed using subband spectral analysis, optimizes the coder's perceptual quality. Dynamic bit allocation and prioritization is combined with embedded quantization resulting in little performance degradation relative to a nonembedded implementation. The coder output is scalable from high quality at higher bit rates to lower quality at lower bit rates, supporting a wide range of service and resource utilization. The lower bit-rate representation is obtained simply through truncation of the higher bit-rate representation. Since source-rate adaptation is performed through truncation of the encoded stream, interaction with the coder is not required, making the embedded coder ideally suited for rate-adaptive communication systems. Performance for both speech and music was verified through subjective listening tests.

Nonminimum phase channel equalization using noncausal filters

Abstract: The Viterbi algorithm is the optimum method for detection of a data sequence in the presence of intersymbol interference and additive white Gaussian noise. Since its computational complexity is very large, several simplifications and alternative methods have been proposed, most of which are more effective when dealing with minimum phase channels. We present a novel technique for the equalization of nonminimum phase channels that employs noncausal all-pass filters operating in reversed time. The impulse response of the equalized channel approximates a minimum phase sequence with higher energy concentration at its left-hand end than at the right-hand end. The method can be modified to obtain a desired impulse response with few nonzero samples with only minor variations in noise level, providing significant complexity reduction in the Viterbi algorithm for detection. In addition, a twopass decoding strategy is developed, leading to significant improvement in performance with little increase in computational cost. Simulation results are included to verify the advantages of the proposed techniques.

Design of fractional delay filters using convex optimization

Abstract: Fractional sample delay (FD) filters are useful and necessary in many applications, such as the accurate steering of acoustic arrays, delay lines for physical models of musical instruments, and time delay estimation. This paper addresses the design of finite impulse response (FIR) FD filters. The problem is posed as a convex optimization problem in which the maximum modulus of the complex error is minimized. Several design examples are presented, along with an empirical formula for the filter order required to meet a given worst case group delay error specification.

Efficient implementations of 2-D noncausal IIR filters

Abstract: In this paper, we propose a framework for the efficient implementation of two-dimensional (2-D) noncausal infinite impulse response (IIR) filters, i.e., filter systems described implicitly by difference equations and boundary conditions. A number of common 2-D LSI filter operations, (including low-pass, high-pass, and zero-phase filters), are efficiently realized and implemented in this paper as noncausal IIR filters. The basic concepts involved in our approach include the adaptation of so-called direct methods for solving partial differential equations (PDEs), and the introduction of an approximation methodology that is particularly well suited to signal processing applications and leads to very efficient implementations. In particular, for an input and output with N/spl times/N samples, the algorithm requires only O(N/sup 2/) storage and computations (yielding a per pixel computational load that is independent of image size), and has a parallel implementation (yielding a per pixel computational load that decreases with increasing image size). Also, because our approach allows for the implementation of filters with space-varying coefficients on irregularly shaped domains, it should have applications in related areas like linear estimation, geophysical signal processing, or any field requiring approximate solutions to elliptic PDEs.

Design of predictive IIR filters via feedback extension of FIR forward predictors

Abstract: Finite impulse response (FIR) predictors for polynomial signals and sinusoids are easy to design because of the available closed-form design formulae. On the other hand, those FIR predictors have two major drawbacks: the passband gain peak is usually greater than +3 dB, and a long FIR structure is needed to attain high attenuation in the stopband. Both of these characteristics cause severe problems, particularly in control instrumentation when the predictor operates inside a closed control loop. In this paper, we present a novel feedback extension scheme for FIR forward predictors. This extension makes it possible to easily design infinite impulse response (IIR) predictors with low passband ripple and high stopband attenuation. The new approach is illustrated with design examples.

Architectural Synthesis of Digital Signal Processing Algorithms Using IRIS

Abstract: In this paper, we present the IRIS architectural synthesis system for high-performance digital signal processing. This tool allows non-specialists to automatically derive VLSI circuit architectures from high-level, algorithmic representations, and provides a quick route to silicon implementation. By incorporating a novel synthesis methodology, called the Modular Design Procedure, within the IRIS system, parameterised models of complex and innovative DSP hardware can be derived and automatically assembled to create new DSP systems. The nature of this synthesis methodology is such that designers can explore a large range of architectural alternatives, whilst considering all the architectural implications of using specific hardware to realise the circuit. The applicability of IRIS is demonstrated using the design examples of a second order Infinite Impulse Response filter and a one-dimensional Discrete Cosine Transform circuit.

Statistically optimum pre- and postfiltering in quantization

Abstract: We consider the optimization of pre- and postfilters surrounding a quantization system. The goal is to optimize the filters such that the mean square error is minimized under the key constraint that the quantization noise variance is directly proportional to the variance of the quantization system input. Unlike some previous work, the postfilter is not restricted to be the inverse of the prefilter. With no order constraint on the filters, we present closed-form solutions for the optimum pre- and postfilters when the quantization system is a uniform quantizer. Using these optimum solutions, we obtain a coding gain expression for the system under study. The coding gain expression clearly indicates that, at high bit rates, there is no loss in generality in restricting the postfilter to be the inverse of the prefilter. We then repeat the same analysis with first-order pre- and postfilters in the form 1+/spl alpha/z/sup -1/ and 1/(1+/spl gamma/z/sup -1/). In specific, we study two cases: 1) FIR prefilter, IIR postfilter and 2) IIR prefilter, FIR postfilter. For each case, we obtain a mean square error expression, optimize the coefficients /spl alpha/ and /spl gamma/ and provide some examples where we compare the coding gain performance with the case of /spl alpha/=/spl gamma/. In the last section, we assume that the quantization system is an orthonormal perfect reconstruction filter bank. To apply the optimum preand postfilters derived earlier, the output of the filter bank must be wide-sense stationary WSS which, in general, is not true. We provide two theorems, each under a different set of assumptions, that guarantee the wide sense stationarity of the filter bank output. We then propose a suboptimum procedure to increase the coding gain of the orthonormal filter bank.

Image content search by color and texture properties

Abstract: A new scheme for color and texture feature extraction for image content search is presented. We introduce a scheme using the two chrominance components for color information and a computationally efficient infinite impulse response (IIR) quadrature mirror filter bank (QMF) energy measure of the luminance component for texture information. The color and texture information is combined into one feature vector, and the components are balanced with respect to dimensionality. We illustrate the utility of our features with experiments in searching for a specific color texture in a large database of images. Several different sub-band decompositions are evaluated. Features extracted using a previously published Gabor filter bank are also evaluated against the proposed scheme. We conclude that the proposed scheme outperforms the Gabor features in both quality and complexity.

CDMA power control channel estimation using dynamic coefficient scaling

Abstract: By time-sharing demodulator hardware between a primary data path (165), a power control data path (161), and a received signal strength indicator (RSSI) path (163), an entire power control data path (161) can be implemented in a demodulator (140) of a spread spectrum subscriber unit receiver with a low increase in gate count. The primary data path (165) and the power control data path (161) time-share a complex conjugate generator (270), a complex multiplier (280), and a real component extractor (290). Due to timing requirements, though, the channel estimation filter (240) of the primary data path cannot be time-shared with the power control data path. Instead, dynamic coefficient scaling is added to an infinite-duration impulse response (IIR) filter in the RSSI path (163) so that the IIR filter (250) with dynamic coefficient scaling can be time-shared between the RSSI path (163) and the power control data path (161).