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

共 168 条记录，以下是121-130 订阅

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An efficient array processor for accurate downdating of ULV decomposition

An efficient array processor for accurate downdating of ULV ...

引用

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

作者： Yoon, PA Sibul, LH Trinity Coll Dept Comp Sci Hartford CT 06106 USA

ISBN: (纸本)0819441880

We present an efficient array processor for downdating the ULVD. The array processor exploits the block structure of the decomposition to determine the effective rank at every step of modification and maintains the rank-revealing nature of the decomposition throughout the steps by tracking exact quantities of Frobenius norms of all three blocks of the lower triangular factor in the decomposition. Thus. a deflation step that is necessary to compute its numerical rank can be often avoided. This feature of the algorithm is particularly attractive to its array processor implementation because the deflation steps usually require some condition estimation. and most of good condition estimators are not well pipelined.

关键词： signal processing null space rank detection array processors

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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

引用

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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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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Design of a color image processing algorithm using on-line arithmetic modules

Design of a color image processing algorithm using on-line a...

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

作者： Sinky, MH Tenca, AF Shantilal, AC Lucchese, L Oregon State Univ Corvallis OR 97331 USA

ISBN: (纸本)0819454974

A new class of nonlinear filters for color image processing was proposed by Lucchese and Mitra. This type of color filter processes the chromatic component of images encoded in the International Commission. on Illumination (CIE) u'v' color space. Images processed by this filter do not show color shifts near edges between regions with different intensities. The filter uses linear convolution operations internally and is effective and efficient for denoising and regularizing color images. Image processing systems are computationally intensive and usually require a large amount of area in order to reach desirable levels of performance. The use off on-line arithmetic can decrease the area of the hardware implementation and still maintain a reasonable throughput. This work presents the design of the color filter as a network of on-line arithmetic modules. The network topology and some detail of each arithmetic module are provided. The final implementation targets FPGAs and it is compared in terms of area against an estimate of a conventional design. The throughput of this solution is capable of supporting real-time processing of common image formats.

关键词： achromatic chromatic convolution DSP DIP FPGA on-line arithmetic RGB SRT division

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Color characterization for image indexing and machine vision

Color characterization for image indexing and machine vision

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

作者： Lin, SM Nair, D Natl Instruments Austin TX 78759 USA

ISBN: (纸本)0819437611

Color representation and comparison based on the histogram has proved to be very efficient for image indexing in content-based image retrieval and machine vision applications. However, the issues of color constancy and accurate color similarity measures remain unsolved. This paper presents a new algorithm for intensity-insensitive color characterization for image retrieval and machine vision applications. The color characterization algorithm divides the HSI (hue, saturation and intensity) color space into a given number of bins in such a way that the color characterization represents all the colors in the hue/saturation plane as well as black, white and gray colors. The color distribution in these bins of the HSI space is represented in the form of a one-dimensional vector called Color Spectrum Vector (CSV). The color information that is stored in the CSV is insensitive to changes in the luminance. A weighted version of CSV called WCSV is introduced to take the similarity of the neighboring bins into account A Fuzzy Color Spectrum Vector (FCSV) color representation vector that takes into account the human uncertainty in color classification process is also introduced here. The accuracy and speed of the algorithm is demonstrated in this paper through a series of experiments on image indexing and machine vision applications.

关键词： color comparison pattern matching clustering algorithm fuzzy set theory image indexing machine vision image processing

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