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Generalized causal moving average (GCMA) smoothing filter for real-time applications

Generalized causal moving average (GCMA) smoothing filter fo...

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conference on advanced signal processing algorithms architectures and implementations XIII

作者： Rajagopalan, R Pratt & Whitney Control & Diagnost Syst Propul Syst Anal E Hartford CT 06108 USA

ISBN: (纸本)0819450782

Moving average filters are commonly used in industries for real-time processing of noisy data. Though they perform well in filtering out the noise, they introduce significant lag in the signal. The resulting peak value of the filtered signal at the operating point is likely to be lower due to averaging of higher and lower peak signals in the averaging interval. The generalized moving average smoothing filter by Golay-Savitzky preserves the higher moments and does not suffer from the limitations imposed by the conventional moving average filter. The smoothing strategy is derived from least squares fitting of a lower order polynomial to a number of consecutive points. Due to polynomial curve fitting as opposed to a line fitting in the case of conventional moving average filter, this filter preserves the higher frequency components of the signal and their line width. This paper presents a generalized causal moving average filter deduced using the concepts in Golay-Savitzky smoothing filter for real-time applications. Golay-Savitzky filter is non-causal, relies on the future data that is not available, hence not suitable for real-time applications. Further, the designed causal filter makes use of the filtered data as opposed to the original data in the case of Golay-Savitzky. This approach allows us to conduct frequency response studies to evaluate the quality and the applicability of the filter for various signals in the aircraft engines and other engineering applications. Frequency response studies cannot carried out using the Golay-Savitzky filter. This paper also investigates the performance of various polynomial orders in-reproducing the signal from a noisy data. Some of the performance measures used are bandwidth, overshoots, and lags introduced by the filter. The mathematical technique to extract the signal and deduce the coefficients in off-line is also presented.

关键词： Filtering (signal processing) Electronic filtering signal processing Smoothing Interference (communication) Control systems Digital signal processing Data processing Electromagnetic coupling Nonlinear filtering

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Design of a hybrid prefix adder for non-uniform input arrival times

Design of a hybrid prefix adder for non-uniform input arriva...

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conference on advanced signal processing algorithms, architectures and implementations XII

作者： Choi, Y Swartzlander, EE Univ Texas Elect & Comp Engn Dept Austin TX 78712 USA

ISBN: (纸本)0819445584

This paper examines the design of a hybrid prefix adder under the condition of non-uniform input signal arrival. This is encountered in the final adder for fast parallel multipliers, which use column compression reduction. The prefix graph scheme efficiently accommodates the non-uniform arrival times. Rules are presented for designing hybrid prefix adders under such conditions. This rule produces adders which are faster and less complex than previous work.

关键词： prefix adder non-uniform input signal arrival parallel multipliers

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On the regularized solution of block banded block Toeplitz systems

On the regularized solution of block banded block Toeplitz s...

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conference on advanced signal processing algorithms, architectures, and implementations X

作者： Bini, DA Farusi, A Fiorentino, G Meini, B Dipartimento Matemat I-56127 Pisa Italy

ISBN: (纸本)0819437611

Given two n. x n Toeplitz matrices T-1 and T-2, and a vector b epsilon R-n2, consider the linear system Ax = b - eta, where eta epsilon R-n2 is an unknown vector representing the noise and A = T-1 x T-2. Recovering approximations of a, given A and b, is encountered in image restoration problems. We propose a method for the approximation of the solution a: that has good regularization properties. The algorithm is based on a modified version of Newton's iteration for matrix inversion and relies on the concept of approximate displacement rank. We provide a formal description of the regularization properties of Newton's iteration in terms of filters and determine bounds to the number of iterations that guarantee regularization. The method is extended to deal with more general systems where A = Sigma (h)(i=1) T-1((i)) x T-2((i)). The cost of computing regularized inverses is O(n log n) operations (ops), the cost of solving the system Ax = b is O(n(2) log n) ops. Numerical experiments which show the effectiveness of our algorithm are presented.

关键词： Band Toeplitz systems Newton's iteration regularization displacement rank image processing

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On second-order statistics-based blind equalization of FIR/IIR multiple-input multiple-output channels with common zeros

On second-order statistics-based blind equalization of FIR/I...

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SPIE conference on advanced signal processing algorithms, architectures, and implementations VIII

作者： Tugnait, JK Huang, B Auburn Univ Dept Elect Engn Auburn AL 36849 USA

ISBN: (纸本)0819429163

The problem of blind equalization of MIMO (multiple-input nultiple-output) communications channels is considered using the second-order statistics of the data. Such models arise when a single receiver data from multiple sources is fractionally sampled (assuming that there is excess bandwidth), or when an antenna array is used with or without fractional sampling. We focus on direct design of finite-length MMSE (minimum mean-square error) blind equalizers. We allow infinite impulse response (IIR) channels. Our approaches also work when the "subchannel" transfer functions have common zeros so long as the common zeros are minimum-phase zeros. We only require that the there exist a causal, stable left inverse (not necessarily unique) to the MIMO transfer function and that the leading coefficient matrix of the MIMO channel impulse response have its rank equal to the number of sources. The channel length or model orders need not be known. The sources are recovered up to a unitary mixing matrix and are further 'unmixed' using higher-order statistics of the data. An illustrative simulation example is provided.

关键词： spatio-temporal processing multiple-input multiple-output channels multiaccess systems blind equalization space diversity time diversity fractional sampling

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Application of symmetric redundant residues for fast and reliable arithmetic

Application of symmetric redundant residues for fast and rel...

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conference on advanced signal processing algorithms, architectures and implementations XII

作者： Parhami, B Univ Calif Santa Barbara Dept Elect & Comp Engn Santa Barbara CA 93106 USA

ISBN: (纸本)0819445584

Despite difficulties in general division, magnitude comparison, and sign detection, residue number system arithmetic has been used for many special-purpose systems in light of its parallelism and modularity for the most common arithmetic operations of addition/subtraction and multiplication. Computation in RNS requires modular reduction, both for the initial conversion from binary to RNS and after each operation to bring the result back to within a valid residue range. Use of redundant residues simplifies this critical operation, leading to even faster arithmetic. One type of redundant mod-m residue, that keeps the representational redundancy to the minimum of 1 bit per residue, has the nearly symmetric range [-m, m) and allows two values for each pseudoresidue: (m) or (m) - m. We study the extent of simplification and speed-up in the modular reduction process afforded by such redundant residues and discuss its potential implications to the design of RNS arithmetic circuits. In particular, we show that besides cost and performance benefits, introduction of error checking and fault tolerance in arithmetic computations is facilitated when such redundant residues are used.

关键词： computer arithmetic error code modular reduction parallel processing redundant representation RNS

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A DSP implementation of source location using microphone arrays 6

A DSP implementation of source location using microphone arr...

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conference on advanced signal processing algorithms, architectures, and implementations VI

作者： Rabinkin, DV Renomeron, RJ Dahl, A French, JC Flanagan, JL Bianchi, MH RUTGERS STATE UNIV CTR COMP AIDS IND PRODPISCATAWAYNJ 08855

ISBN: (纸本)0819422347

The design, implementation, and performance of a low-cost, real-time DSP system for source location is discussed. The system consists of an 8-element electret microphone array connected to a signalogic DSP daughterboard hosted by a PC. The system determines the location of a speaker in the audience in an irregularly shaped auditorium. The auditorium presents a non-ideal acoustical environment;some of the walls are acoustically treated, but there still exists significant reverberation and a large amount of low frequency noise from fans in the ceiling. The source location algorithm is implemented in a two step process: The first step determines time delay of arrival (TDOA) for select microphone pairs. A modified version of the Cross-Power Spectrum Phase Method is used to compute TDOAs and is implemented on the DSP daughterboard. The second step uses the computed TDOAs in a least mean squares gradient descent search algorithm implemented on the PC to compute a location estimate. © 1996 SPIE. All rights reserved.

关键词： microphone arrays speech source location real-time processing

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Some significant communications processing realized with adaptive Bragg gratings

Some significant communications processing realized with ada...

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SPIE conference on advanced signal processing algorithms, architectures, and implementations VIII

作者： Upton, E Wickham, M Smith, M Hart, J Flynn, R Havstad, S Nachman, P TRW Co Inc Redondo Beach CA 90278 USA

ISBN: (纸本)0819429163

Higher data capacity demands and lower interference requirements in the wireless communications arena are exploiting higher carrier frequencies and wider modulation bandwidths. Circuitry which can perform intermediate frequency processing over these more demanding ranges is needed to provide complex signal processing without commercial penalties. Photonics technologies utilizing Bragg Grating signal processing (BGSP) can bridge the gap between the very high frequency RF millimeter wave integrated circuit (MIMIC) domains at the antenna interface and the CMOS digital signal processors at the base band frequency interface. The desirable benefits of multiple tap adaptive finite impulse response (FIR) and infinite impulse response (IIR) filters and equalizers are well known;however, they are usually the province of digital signal processing and force the sample rates prior to these processors to a higher overall system power consumption level. BGSP provides these functions with discrete taps and digital controls but at the bandwidths usually reserved for RF circuitry because the actual processing occurs at optical frequencies and at wave lengths which are compatible with integrated circuit technologies. The high performance benefits of photonic processing can be realized if the stability control of the Bragg grating is derived from the same metric which induces in photonics its sensitivity to drift. We will present a orthogonally coded tap modulation technique which stabilizes the transfer function of the signal processor and enables significant adaptive IF signal processing to be obtained with very low size, weight, and power. Our demonstration of a photonic proof-of-concept architecture is a reconfigurable multiple tap FIR filter that is dynamically controlled to perform low pass, high pass, band pass, and band stop filters operating over bandwidths of 3 GHz.

关键词： signal processing Fiber Bragg gratings Filtering (signal processing) Control systems Linear filtering Digital signal processing Optical filters Servomechanisms Modulators Electronic filtering

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Separating desired image and signal invariant components from extraneous variations 6

Separating desired image and signal invariant components fro...

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conference on advanced signal processing algorithms, architectures, and implementations VI

作者： Williams, WJ Zalubas, EJ Hero, A UNIV MICHIGAN DEPT ELECT ENGN & COMP SCIANN ARBORMI 48109

ISBN: (纸本)0819422347

Images and signals can be characterized by representations invariant to time shifts, spatial shifts, frequency shifts, and scale changes as the situation dictates. Advances in time-frequency analysis and scale transform techniques have made this possible. The next step is to distinguish between invariant forms representing different classes of image or signal. Unfortunately, additional factors such as noise contamination and `style' differences complicate this. A ready example is found in text, where letters and words may vary in size and position within the image segment being examined. Examples of complicating variations include font used, corruption during fax transmission, and printer characteristics. The solution advanced in this paper is to cast the desired invariants into separate subspaces for each extraneous factor or group of factors. The first goal is to have minimal overlap between these subspaces and the second goal is to be able to identify each subspace accurately. Concepts borrowed from high-resolution spectral analysis, but adapted uniquely to this problem have been found to be useful in this context. Once the pertinent subspace is identified, the recognition of a particular invariant form within this subspace is relatively simple using well-known singular value decomposition techniques.

关键词： scale transform Mellin transform wordspotting singular value decomposition character recognition time-frequency

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Approach to the joint optimization of video communications and video signal processing

Approach to the joint optimization of video communications a...

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conference on advanced signal processing algorithms, architectures, and implementations XI

作者： Pearlman, WA Medley, MJ Suter, BW Rensselaer Polytech Inst Ctr Next Generat Video Res Troy NY 12180 USA

ISBN: (纸本)0819441880

Compressed video bitstreams require protection from channel errors in a wireless channel. The three-dimensional (3-D) SPIHT coder has proved its efficiency and its real-time capability in compression of video. A forward-error-correcting (FEC) channel (RCPC) code combined with a single ARQ (automatic-repeat-request) proved to be an effective means for protecting the bitstream. In this paper, the need for ARQ is eliminated by making the 3-D SPIHT bitstream more robust and resistant to channel errors. Packetization of the bitstream and the reorganization of these packets to achieve scalability in bit rate and/or resolution in addition to robustness is demonstrated and combined with channel coding to not only protect the integrity of the packets, but also allow detection of packet decoding failures, so that only the cleanly recovered packets are reconstructed. In extensive comparative tests, the reconstructed video is shown to be superior to that of MPEG-2, with the margin of superiority growing substantially as the channel becomes noisier.

关键词： video compression video transmission robust source coding error resilient transmission combined source-channel coding 3-D wavelet transform embedded wavelet coding

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Bandlimits and coherent modulation filtering of non-stationary signals

Bandlimits and coherent modulation filtering of non-stationa...

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conference on advanced signal processing algorithms, architectures, and Implementantions xviiI

作者： Clark, Pascal Atlas, Les Univ Washington Dept Elect Engn Seattle WA 98195 USA

ISBN: (纸本)9780819472946

Modulation filtering is a technique for filtering slowly-varying envelopes of frequency subbands of a non-stationary signal, ideally without affecting the signal's phase and fine-structure. Coherent modulation filtering is a potentially more effective subtype of such techniques where subband envelopes are determined through demodulation of the subband signal with a coherently detected subband carrier. In this paper we propose a coherent modulation filtering technique based on detecting the instantaneous frequency of a subband from its time-frequency representation. We show that coherent modulation filtering imposes a new bandlimiting constraint on the modulation product plus the ability to recover arbitrarily chosen envelopes and carriers from their modulation product. We show that a carrier estimate based on the time-varying spectral center-of-gravity satisfies the bandlimiting condition as well as Loughlin's previously derived bandlimiting constraint on the instantaneous frequency of carrier. These bandwidth constraints lead to effective and distortion-free modulation filters, offering new approaches for potential signal modification. The spectral center-of-gravity does not satisfy the condition on arbitrary recovery, however, which somewhat limits the flexibility of coherent modulation filtering. Demonstrations are provided with speech signals.

关键词： modulation time-frequency analysis time-varying filters bandwidth speech processing

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