SPTM-11.1

Rational Signal Subspace Approximation with Applications to DOA Estimation
Mohammed A. Hasan (University of Minnesota Duluth), Jawad A. Hasan (University of Baghdad)

In this paper, we have developed an approach for approximating the signal and noise subspaces which avoid the costly eigendecomposition or SVD. These subspaces were approximated using rational and power-like methods applied to the sample covariance matrix. It is shown that MUSIC and Minimum Norm frequency estimators can be derived using these approximated subspaces. These approximate estimators are shown to be robust against noise and overestimation of number of sources. A substantial computational saving would be gained compared with those associated with the eigendecomposition-based methods. Simulations results show that these approximated estimators have comparable performance at low signal-to-noise ration (SNR) to their standard counterparts and are robust against overestimating the number of impinging signals.

SPTM-11.2

3-D Emitter Localization Using Inhomogeneous Bistatic Scattering
Scott D Coutts (MIT Lincoln Laboratory)

The purpose of this work is to establish how a moving emitter can be localized by a passive receiver using inhomogeneous bistatic scattering. This is a novel localization technique that assumes no a priori knowledge of the location of the reflecting sources. The emitter parameters of range, heading, velocity, and altitude are estimated and the variances of the estimates are determined. The proposed estimator is successfully demon-strated using field data collected at White Sands Missile Range during the DARPA/Navy Mountaintop program

SPTM-11.3

A New Technique to Filter Reduction for Speech Signal Processing Systems
Luowen Li (Computer Control Lab. School of EEE. Nanyang Technological University. Singapore 639798), Lihua Xie, Gang Li, Yeng Chai Soh (School of EEE. Nanyang Technological University. Singapore 639798)

In many applications, one needs to approximate a filter of very high order with that of lower order. To reduce the order of the filter, some techniques such as balanced model reduction approach are often applied. In this paper, we will introduce a new technique which is based on minimizing the $H_2$-norm between the filter of very high order and the reduced one. This technique shows much better performance than other existing model reduction methods and is applied to estimating the vocal tract filter for speech processing systems. A speech processing example is presented to demonstrate the design procedure and the performance of the proposed algorithm.

SPTM-11.4

PIRANHA FILTER FOR COMMUNICATION SYSTEM ROBUSTNESS
LANDRY M Ren� Jr. (SupAero - ONERA/CERT), CALMETTES M Vincent, BOUSQUET Michel (SupAero)

The designed rejection filter is of recursive prediction error (RPE) form and uses a special constrained model of infinite impulse response (IIR) with a minimal number of parameters. The so-called PIRANHA Filter is made up independent cascaded adaptive cells realising high rejection at certain frequencies. The convergent filter is characterised by highly narrow-bandwidth and uniform notches of desired shape. Results from simulations illustrate the performance of the algorithm used in the PIRANHA Filter under a wide range of conditions and situations. This paper intends to give a description of the PIRANHA Structure, the mechanism of its interference detection monitoring and the filter stability control. The PIRANHA Filter has shown to be an efficient solution for detection, tracking and elimination of multiple high power CWIs and Narrow Band Jammers.

SPTM-11.5

Warped Linear Time Invariant Systems And Their Application In Audio Signal Processing
John Garas, Piet C.W. Sommen (Eindhoven University of Technology)

The main goal behind coordinate transformation (warping) of a Linear Time Invariant (LTI) system is to represent its signals in terms of new basis functions that better suit the application in hand. Unitary operators simplify the analysis considerably; therefore, they are used to derive the relations between variables in the original and warped domains. These relations show that an LTI system can be warped by processing its input signals with a unitary warping transformation. An efficient implementation of this warping transformation, that is based on a nonuniform sampling theorem, is given; which allows applying the warping principle in real-time applications. As an example of exploiting this technique, it is shown that sampling an audio signal at exponentially spaced moments changes the underlying coordinates of its signal processing system to suit those of the human auditory system.

SPTM-11.6

Efficiency of Radix-K Transforms on Computers with Cache
Ryszard M. Stasinski (Hoegskolen i Narvik, Norway)

In the paper impact of the most critical part of the up-to-date computer memory hierarchy, the cache, on the efficiency of fast transform algorithms (e.g. FFT, DCT/DST, DWHT, including multidimensional generalizations) is analyzed. Cache misses can severely deteriorate efficiency of a computer program, and indeed, it is shown that for large data vectors a modification of a fast transform algorithm realization may change their number dramatically. Several memory managing problems are pointed out, and suggestions for their amelioration are given. Formulae on the minimum of data-related cache misses for radix-2^s transform realizations are given, s is an integer.

SPTM-11.7

A Novel Method for Power Line Interference Suppression
George Keratiotis, Larry Lind (ESE Department, University of Essex, Colchester CO4 3SQ, United Kingdom), John W Cook, Minesh Patel, Dave Croft, Pete Whelan, Peter Hughes (British Telecommunications Laboratories, Martlesham Heath, Ipswich, Suffolk IP5 3RE, United Kingdom)

In this paper, a novel method to suppress the power line interference induced into telephone lines that are in proximity to power conductors is proposed. A phase-locked loop is used to synchronize the interference signal with an anti-phase waveform, which is stored in a buffer. The anti-phase signal is injected on the line and the samples of the residual signal are used to update the anti-phase waveform. Computer simulations are used to compare the novel Adaptive Phase-Locked Buffer (APLB) approach with the traditional Least Mean Squares (LMS) algorithm in an adaptive noise cancellation configuration. The new technique achieves 15 dB further suppression compared to LMS and since it can be implemented on a single Digital Signal Processor (DSP) chip it proves to be a very efficient solution to the power line interference problem.

SPTM-11.8

ADAPTIVE POWER-LINE DISTURBANCE DETECTION SCHEME USING A PREDICTION ERROR FILTER AND A STOP-AND-GO CA CFAR DETECTOR
Jaehak Chung, Edward J Powers, W Mack Grady (Department of Electrical and Computer Engineering, The University of Texas at Austin), Sid C Bhatt (Electric Power research Institute)

This paper presents a new power-line disturbance detection algorithm. The utilized recursive least square (RLS) prediction error filter extracts the power-line disturbance signal from recorded data, and the modified stop-and-go cell average constant false alarm rate (CA CFAR) detector makes a decision based on the squared output of the previous stage. The detection performance of the proposed algorithm is determined by simulations, and actual high voltage transmission line data is utilized to demonstrate the performance of the proposed algorithm.

SPTM-11.9

Time Series Analysis for ECG Data Compression
Hazem M Abbas (Ain Shams University, Faculty of Engineering, Dept. Computer and Systems Engineering)

This work presents a new electrocardiogram (ECG) data compression method. By differentiating the signal and using proper thresholding, the ECG is first segmented into a sequence of straight lines. The vertices of these lines are used to encode the signal. The decoding part works by applying Korenberg's Fast Orthogonal Search (FOS) method to reconstruct the original signal. Simulation results have demonstrated the efficiency of the algorithm.

SPTM-11.10

A PARALLEL RESIDUE-TO-BINARY CONVERTER
Wei Wang, M.N.S Swamy, M. Omair Ahmad, Yuke Wang (Concordia University)

In this paper, a high-speed parallel residue-to-binary converter is proposed for the recently introduced moduli set S^k={2^m-1,2^((2^0)m)+1,2^((2^1)m)+1,..., 2^((2^k)m) +1} for a general value of k. The proposed converter replaces the multiplications of the residue-to-binary conversion by simple cyclic shift and concatenation operations. For the purpose of comparison, the individual converters for the cases of k=0 and 1 are derived from the general architecture. The converter for S^0 is twice as fast as the previous converter using only one-half of the hardware, while that of S^1 is three times as fast, but requiring only 60% of the hardware.

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Last Update: February 4, 1999 Ingo Höntsch