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Toward blind removal of unwanted sound from orchestrated music

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

引用

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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Non-linear signal processing using index calculus DBNS arithmetic

Non-linear signal processing using index calculus DBNS arith...

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

作者： Muscedere, R Jullien, GA Dimitrov, VS Miller, WC Univ Windsor VLSI Res Grp Windsor ON N9B 3P4 Canada

ISBN: (纸本)0819437611

This paper discusses the use of a recently introduced index calculus Double-Base Number System (IDBNS) for representing and processing numbers for non-linear digital signal processing;the target application is a digital hearing aid processor. The IDBNS representation uses 2 orthogonal bases (2 and 3) to represent real numbers with arbitrary precision. By restricting the number of digits to one or two, it is possible to efficiently represent the real number using the indices of the bases rather than the distribution of the digits. In this paper we discuss the use of the two-digit form of this representation (2-IDBNS) to efficiently perform arithmetic associated with the non-linear processing required to correct the usual forms of hearing loss in a digital hearing aid. The non-linear processing takes the form of dynamic range compression as a function of frequency band. Currently developed digital hearing instrument processors require large dynamic range representations (20-24 bits) in order to accurately generate the dynamic range compression associated with typical hearing loss. We show that the natural non-linear representation afforded by the IDBNS provides both a more efficient signal representation and a more efficient technique for processing the dynamic range compression. We pay particular attention to a novel technique of converting from a linear binary input directly to the 2-IDBNS representation using an observation of partial cyclic repetition in the indices along with near unity approximants.

关键词： computer arithmetic non-linear number representation double-base number system hearing loss

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Application of higher-order cepstral techniques in problems of fetal heart signal extraction 6

Application of higher-order cepstral techniques in problems ...

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

作者： SabryRizk, M Zgallai, W Hardiman, P ORiordan, J CITY UNIV LONDON DEPT ELECT ENGNSIGNAL PROC GRPLONDON EC1V 0HBENGLAND

ISBN: (纸本)0819422347

Recently, cepstral analysis based on second order statistics and homomorphic filtering techniques have been used in the adaptive decomposition of overlapping, or otherwise, and noise contaminated ECG complexes of mothers and fetals obtained by a transabdominal surface electrodes connected to a monitoring instrument, an interface card, and a PC. Differential time delays of fetal heart beats measured from a reference point located on the mother complex after transformation to cepstra domains are first obtained and this is followed by fetal heart rate variability computations. Homomorphic filtering in the complex cepstral domain and the subuent transformation to the time domain results in fetal complex recovery. However, three problems have been identified with second-order based cepstral techniques that needed rectification in this paper. These are (1) errors resulting from the phase unwrapping algorithms and leading to fetal complex perturbation, (2) the unavoidable conversion of noise statistics from Gaussianess to non-Gaussianess due to the highly non-linear nature of homomorphic transform does warrant stringent noise cancellation routines, (3) due to the aforementioned problems in (1) and (2), it is difficult to adaptively optimize windows to include all individual fetal complexes in the time domain based on amplitude thresholding routines in the complex cepstral domain (i.e. the task of `zooming' in on weak fetal complexes requires more processing time). The use of third-order based high resolution differential cepstrum technique results in recovery of the delay of the order of 120 milliseconds.

关键词： advanced signal processing techniques non-linear adaptive filtering differential bicepstral applications non-invasive fetal ECG monitoring

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Bayesian image deblurring and boundary effects

Bayesian image deblurring and boundary effects

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advanced signal processing algorithms, architectures, and implementations XV

作者： Calvetti, Daniela Somersalo, Erkki Department of Mathematics Center for Modeling Integrated Metabolic Systems Case Western Reserve University Cleveland OH United States Department of Mathematics Helsinki University of Technology Helsinki Finland

We consider the deconvolution problem of estimating an image from a noisy blurred version of it. In particular, we are interested in the boundary effects: since the convolution operator is non-local, the blurred image depend on the scenery outside the field of view. Ignoring this dependency leads to image distortion known as boundary effect. In this article, we consider two different approaches to treat the non-locality. One is to estimate the image extended outside the field of view. The other is to treat the influence of the out of view scenery as boundary clutter. Both approaches are considered from the Bayesian point of view.

关键词： Image processing

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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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EFFECT OF RESAMPLING ON THE MUSIC ALGORITHM WHEN APPLIED TO INVERSE SYNTHETIC APERTURE RADAR

EFFECT OF RESAMPLING ON THE MUSIC ALGORITHM WHEN APPLIED TO ...

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CONF ON advanced signal processing algorithms, architectures, AND implementations 2

作者： NASH, G Defence Science and Technology Organization and Univ. of South Australia (Australia)

ISBN: (纸本)0819406945

Inverse synthetic aperture radar (ISAR) is a radar imaging method by which the rotation of a target is utilized to produce a two dimensional image. To achieve high resolution in both range and cross range, a series of stepped frequency waveforms are transmitted. These sample the target's radar response in frequency and time. Frequency transforms to time, which is proportional to range, and time transforms to doppler frequency, which is proportional to cross range. Therefore a two dimensional Fourier transform can be applied to the two dimensional data set to produce a radar image. However this is an approximation as the data is in a polar format, which only approximates a rectangular grid. Therefore resampling (interpolation) is required to change the grid from a polar to a rectangular format. The resampling in this case is straightforward. In an attempt to obtain higher resolution images, the Fourier transform has been replaced by the multiple signal classification (MUSIC) algorithm. The justification for this is that the targets of interest are manmade and so have sharp edges and corners. Therefore they consist of a number of corner reflectors with a background of continuous reflectors. The corner reflectors by their nature will generally give much stronger reflections, so one can with a certain degree of accuracy, approximate the ship as a collection of corner reflectors. Over the small change in aspect angle for which ISAR imaging is performed, corner reflectors can be approximated as point scatterers. This leads to the data being modeled as a collection of complex exponentials with added white Gaussian noise. The noise being due to thermal noise in the radar system. This type of data set is ideal for the two dimensional MUSIC algorithm. There are two major difficulties in applying the MUSIC algorithm to ISAR imaging. First, the MUSIC estimator is more sensitive to the sampling grid being polar than the Fourier transform is. Second, the resampling is less effect

关键词： Fourier transforms Radar Reflectors Computer simulations Interference (communication) Data modeling Detection and tracking algorithms Image resolution signal processing Transform theory

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New signal adaptive approach for the reduction of interference terms in quadratic time-frequency distributions

New signal adaptive approach for the reduction of interferen...

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SPIE Conference on advanced signal processing algorithms, architectures, and implementations ViiI

作者： Nickel, RM Williams, WJ Univ Michigan Dept Elect Engn & Comp Sci Ann Arbor MI 48109 USA

ISBN: (纸本)0819429163

One of the key problems in high resolution (quadratic), time-varying spectral analysis is the suppression of interference terms which can obscure the true location of auto components in the resulting time-frequency distribution (TFD). Commonly used reduced interference distributions (RIDs) tackle the problem with a properly chosen two-dimensional low pass filter (kernel). A recently published(8) novel approach uses alternative means to achieve the desired goal. The idea of the new method is to obtain an estimate of the cross terms from a given prior distribution based on the magnitude and location of its negative components. The estimate is constructed via an iterative projection method that guarantees that the resulting distribution is positive and satisfies the marginals. Even though the marginals are usually a desirable property of TFDs in general, they can impose an undesirably strong constraint on positive TFDs in particular (Examples are given in this paper). For these cases it is thus beneficial to relax the marginals-constraint. In this paper we present a new method that does not require the incorporation of this constraint and thus leads to positive TFDs with reduced interference terms but without the restrictions due to the marginals.

关键词： positive time-frequency distributions cross-terms successive projections penalty functions

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Element accessed array radar processing

Element accessed array radar processing

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SPIE Conference on advanced signal processing algorithms, architectures, and implementations ViiI

作者： Wills, GD Rees, HD Skidmore, ID Def Evaluat & Res Agcy Malvern WR14 3PS Worcs England

ISBN: (纸本)0819429163

Digitising the signals from a phased array at the element level is becoming more practicable through advances in technology. Adaptive beamforming schemes such as a sample matrix inversion involve inverting a covariance matrix of dimension equal to the number of spatial channels. Since with an element digitised system there can be several thousand elements it becomes impractical to invert such large matrices. This paper presents a weight computation scheme for a fully digitised system. The clutter returns are firstly range gated and then the adaptive process is broken down into a smaller cascade of stages, or domains. Attenuation of the clutter is then achieved using a spatial Wiener filter in each domain. From a simulation of an airborne radar at a high pulse repetition frequency, examples are given of adapted beampatterns and clutter output doppler spectra within each domain. By choosing alias free domains, a clutter-free output is obtained for the sidelobe clutter scenario investigated. A comparison is then given with the performance of a 32 sub-array system. The sub-array system does not have the necessary control of the small scale order of the array weights to spatially null the clutter.

关键词： element level digitisation radar clutter subarrays

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A total variation wavelet inpainting model with multilevel fitting parameters

A total variation wavelet inpainting model with multilevel f...

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

作者： Chan, Tony F. Shen, Jianhong Zhou, Hao-Min Univ Calif Los Angeles Dept Math Los Angeles CA 90095 USA Univ Minnesota Dept Math Minneapolis MN 55455 USA Georgia Inst Technol Sch Math Atlanta GA 30332 USA

ISBN: (纸本)0819463922

In [14], we have proposed two total variation (TV) minimization wavelet models for the problem of filling in missing or damaged wavelet coefficients due to lossy image transmission or communication. The proposed models can have effective and automatic control over geometric features of the inpainted images including sharp edges, even in the presence of substantial loss of wavelet coefficients, including in the low frequencies. In this paper, we investigate a modification of the model for noisy images to further improve the recovery properties by using multi-level parameters in the fitting term. Some new numerical examples are also shown to illustrate the effectiveness of the recovery.

关键词： Wavelets Image compression Image restoration Image processing Image quality Wavelet transforms Automatic control Cameras Image transmission Mathematics

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