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检索条件"任意字段=Conference on Advanced Signal Processing Algorithms, Architectures, and Implementations XI"

共 172 条记录，以下是121-130 订阅

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Design tradeoffs using truncated multipliers in FIR filter implementations

Design tradeoffs using truncated multipliers in FIR filter i...

引用

conference on advanced signal processing algorithms, architectures and implementations xiI

作者： Walters, EG Schulte, MJ Lehigh Univ Comp Sci & Engn Dept Comp Architecture & Arithmet Lab Bethlehem PA 18015 USA

ISBN: (纸本)0819445584

This paper presents a general FIR filter architecture utilizing truncated tree multipliers for computation. The average error, maximum error, and variance of error due to truncation are derived for the proposed architecture. A novel technique that reduces the average error of the filter is presented, along with equations for computing the signal-to-noise ratio of the truncation error. A software tool written in Java is described that automatically generates structural VHDL models for specific filters based on this architecture, given parameters such as the number of taps, operand lengths, number of multipliers, and number of truncated columns. We show that a 22.5 % reduction in area can be achieved for a 24-tap filter with 16-bit operands, 4 parallel multipliers, and 12 truncated columns. For this implementation, the average reduction error is only 9.18 x 10(-5) ulps, and the reduction error SNR is only 2.4 dB less than the roundoff SNR of an equivalent filter without truncation.

关键词： FIR filters truncated multipliers automatic VHDL generation computer arithmetic application-specific designs

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An efficient high-resolution time-frequency representation

An efficient high-resolution time-frequency representation

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

作者： Nelson, D. J. Smith, D. C. US Dept Def R61 Ft George G Meade MD 20755 USA

ISBN: (纸本)0819463922

We address the problem of efficient resolution, detection and estimation of weak tones in a potentially massive amount of data. Our goal is to produce a relatively small reduced data set characterizing the signals in the environment in time and frequency. The requirements for this problem are that the process must be computationally efficient, high gain and able to resolve signals and efficiently compress the signal information into a form that may be easily displayed and further processed. We base our process on the cross spectral representation we have previously applied to other problems. In selecting this method, we have considered other representations and estimation methods such as the Wigner distribution and Welch's method. We compare our method to these methods. The spectral estimation method we propose is a variation of Welch's method and the cross-power spectral (CPS) estimator which was first applied to signal estimation and detection in the mid 1980's. The CPS algorithm and the method we present here are based on the principles first described by Kodera et at. now frequently called the reassignment principle.

关键词： signal detection signal estimation transient signals time-frequency representations

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Blind source separation using joint signal representations for arbitrary variables.

Blind source separation using joint signal representations f...

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conference on advanced signal processing - algorithms, architectures, and implementations VII

作者： Belouchrani, A Amin, MG Villanova Univ. (United States)

ISBN: (纸本)0819425842

Blind source separation is an emerging field of fundamental research with a broad range of applications. It is motivated by practical problems that involve several source signals and several sensors. Each sensor receives an instantaneous linear mixture of the source signals. The problem of the blind source separation consists then of recovering the original waveforms of the sources without any knowledge of the mixture structure. So far, the problem of the blind source separation has been solved using statistical information available on the source signals. A blind source separation approach for non-stationary signals based on time-frequency representations (TFR) have been recently introduced by the authors (SPIE 1996). Herein, we generalize the TFR based blind source separation approach to arbitrary variables, including time and frequency. 'Spatial joint arbitrary variable distributions' are introduced and used for blind source separation via joint diagonalization techniques.

关键词： blind source separation time-frequency representation joint signal representation spatial joint arbitrary variable distributions joint diagonalization

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Optimum microphone placement for array sound capture

Optimum microphone placement for array sound capture

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conference on advanced signal processing - algorithms, architectures, and implementations VII

作者： Rabinkin, DV Renomeron, RJ French, JC Flanagan, JL Rutgers Univ. (United States)

ISBN: (纸本)0819425842

Microphone arrays can be used for high-quality sound pick up in reverberant and noisy environments. The beamforming capabilities of microphone array systems allow highly directional sound capture, providing superior signal-to-noise ratio (SNR) when compared to single microphone performance. There are two aspects in microphone array system performance: The ability of the system to locate and track sound sources, and its ability to selectively capture sound from those sources. Both aspects of system performance are strongly affected by the spatial placement of microphone sensors. A method is needed to optimize sensor placement based on geometry of the environment and assumed sound source behavior. The objective of the optimization is to obtain the greatest average system SNR using a specified number of sensors. A method is derived to evaluate array performance for a given array configuration defined by the above mentioned metrics. An overall performance function is described based on these metrics. A framework for optimum placement of sensors under the practical considerations of possible sensor placement and potential location of sound sources is also characterized.

关键词： signal to noise ratio Sensors Error analysis Interference (communication) Acoustics Monte Carlo methods signal attenuation Statistical analysis Environmental sensing signal processing

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Time-frequency domain reflectometry for smart wiring systems

Time-frequency domain reflectometry for smart wiring systems

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conference on advanced signal processing algorithms, architectures and implementations xiI

作者： Shin, YJ Song, ES Kim, JW Park, JB Yok, JG Powers, EJ Univ Texas Dept Elect & Comp Engn Austin TX 78712 USA

ISBN: (纸本)0819445584

In this paper, a new high resolution reflectometry scheme, time-frequency domain reflectometry, is proposed to detect and locate a fault in wiring. Traditional reflectometry methods have been achieved in either the time domain or frequency domain only. However, time-frequency domain reflectometry utilizes time and frequency information of a transient signal to detect and locate the fault. The time-frequency domain reflectometry approach described in this paper is characterized by time-frequency reference signal design and post-processing of the reference and reflected signals to detect and locate the fault. Using a computational electromagnetic model of a coaxial cable with a fault, time-frequency domain reflectometry has been demonstrated. Knowledge of time and frequency localized information for the reference and reflected signal gained via time-frequency analysis, allows one to detect the fault and estimate the location accurately.

关键词： time-frequency domain reflectometry chirp signal fault detection fault location smart wiring

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Time-varying spectral analysis in exercise and sport science

Time-varying spectral analysis in exercise and sport science

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

作者： Frishberg, Barry A. Galleani, Lorenzo Cohen, Leon S Carolina State Univ 300 Coll St Orangeburg SC 29117 USA Politecn Torino I-10129 Turin Italy CUNY New York NY 10021 USA

ISBN: (纸本)9780819468451

We discuss the application of time-frequency analysis to biomechanical-type signals, and in particular to signals that would be encountered in the study of rotation rates of bicycle pedaling. We simulate a number of such signals and study how well they are represented by various time-frequency methods. We show that time-frequency representations track very well the instantaneous frequency even when there are very fast changes. In addition. we do a correlation analysis between time-series whose instantaneous frequency is changing and show that the traditional correlation coefficient is insufficient to characterize the correlations. We instead show that the correlation coefficient should be evaluated directly from the instantaneous frequencies of the time series, which can be easily estimated from their time-frequency distributions.

关键词： Time-frequency analysis Heart signal generators Biological research Computer simulations signal analyzers Fourier transforms Information operations signal detection Physical phenomena

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Efficient implementations of pipelined CORDIC based IIR digital filters using fast orthonormal μ-rotations

Efficient implementations of pipelined CORDIC based IIR digi...

引用

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

作者： Ma, J Parhi, KK Hekstra, GJ Deprettere, EF Univ Minnesota Dept ECE Minneapolis MN 55455 USA

ISBN: (纸本)0819429163

CORDIC based IIR digital filters are orthogonal filters whose internal computations consist of orthogonal transformations. These filters possess desirable properties for VLSI implementations such as regularity, local connection, low sensitivity to finite word-length implementation, and elimination of limit cycles. Recently, fine-grain pipelined CORDIC based IIR digital filter architectures which can perform the filtering operations at arbitrarily high sample rates at the cost of linear increase in hardware complexity have been developed. These pipelined architectures consist of only Givens rotations and a few additions which can be mapped onto CORDIC arithmetic based processors. However, in practical applications, implementations of Givens rotations using traditional CORDIC arithmetic are quite expensive. For example, for 16 bit accuracy, using floating point data format with 16 bit mantissa and 5 bit exponent, it will require approximately 20 pairs of shift-add operations for one Givens rotation. In this paper, we propose an efficient implementation of pipelined CORDIC based IIR digital filters based on fast orthonormal mu-rotations. Using this method, the Givens rotations are approximated by angles corresponding to orthonormal mu-rotations, which are based on the idea of CORDIC and can perform rotation with minimal number of shift-add operations. We present various methods of construction for such orthonormal mu-rotations. A significant reduction (over 76%) of the number of required shift-add operations is achieved. All types of fast rotations can be implemented as a cascade of only four basic types of shift-add stages. These stages can be executed on a modified floating-point CORDIC architecture, making the pipelined filter highly suitable for VLSI implementations.

关键词： orthogonal IIR digital filters pipelining approximate rotations orthonormal mu-rotations CORDIC architecture

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Convergence results on adaptive approximate filtering 6

Convergence results on adaptive approximate filtering

引用

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

作者： Ludwig, JT Nawab, SH Chandrakasan, AP MIT ELECT RES LABCAMBRIDGEMA 02139

ISBN: (纸本)0819422347

We present an IIR filtering technique based upon a recently proposed approach for reducing power consumption in implementation of frequency-selective filters. The basic idea in such techniques is to utilize the most recent input and output signal samples to estimate the current SNR (defined as the ratio of the in-band signal power to the out-of-band signal power) at the filter's input. This estimated input SNR is then used to update the filter order to the minimum value which would guarantee a minimum tolerable SNR at the filter's output. A key issue addressed in this paper is how well the estimated filter order converges to the theoretical minimum for situations satisfying the assumptions behind the derivation of the technique. Experimental results are used to verify that convergence to the correct filter order depends (1) upon the number of input and output samples used for estimating the input SNR, (2) upon the filter order applied in generating the output samples that are used in estimating the input SNR and (3) upon the proximity of the actual input SNR to boundaries in the input-SNR space corresponding to changes in the optimal choice for filter order.

关键词： low power approximate processing incremental refinement structure frequency-selective filtering Butterworth filters adaptive filtering

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Discrete time processing of linear scale invariant signals and systems

Discrete time processing of linear scale invariant signals a...

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

作者： Izzetoglu, M Onaral, B Chitrapu, P Bilgutay, N Drexel Univ Dept Elect & Comp Engn Philadelphia PA 19104 USA

ISBN: (纸本)0819437611

In this paper, we formulate a framework for discrete-time processing of Linear Scale Invariant (LSI) systems which are invariant to scale changes in time. Continuous-time LSI systems can be processed by Mellin and Modified Scale Transforms analogous to the use of the Laplace and Fourier Transforms in the continuous-time processing of LTI systems. In this work, we present the geometric sampling theorem to prevent aliasing in the scale domain. We also derive the perfect reconstruction filter in time domain, the discrete-time convolution sum, the Discrete Time Modified Scale Transform (DTMST) and the Discrete Modified Scale Transform (DMST) for geometrically sampled signals.

关键词： Transform theory Fourier transforms Convolution Electronic filtering signal processing Statistical analysis Toxic industrial chemicals Biomedical optics Computing systems Telecommunications

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Toward blind removal of unwanted sound from orchestrated music

Toward blind removal of unwanted sound from orchestrated mus...

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

作者： Chang, SY Chun, J Univ Suwon Dept Elect Engn Kyungki do South Korea

ISBN: (纸本)0819437611

The problem addressed in this paper is to removing unwanted sounds from music sound. The sound to be removed could be disturbance such as cough. We shall present some preliminary results on this problem using statistical properties of signals. Our approach consists of three steps. We first estimate the fundamental frequencies and partials given noise-corrupted music sound. This gives us the autoregressive (AR) model of the music sound. Then se filter the noise-corrupted sound using the AR parameters. The filtered signal is then subtracted from the original noise-corrupted signal to get the disturbance. Finally the obtained disturbance is used as reference signal to eliminate the disturbance from the noise-corrupted music signal. Above three steps are carried out in a recursive manner using a sliding window or an infinitely growing window with an appropriate forgetting factor.

关键词： Wiener filter music signal processing interference cancellation

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