{"title":"Correlation analysis of ultrasonic speckle for motion estimation","authors":"H.S. Bilge, M. Karaman","doi":"10.1109/DSPWS.1996.555498","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555498","url":null,"abstract":"Synthetic aperture techniques permit low-cost ultrasonic imaging systems, but are susceptible to motion artifacts. Motion compensation is crucial in imaging of non-stationary structures. In this study, correlation analysis of ultrasonic speckle signals for correlation based motion estimation is presented. Correlation measurements with different system parameters, such as the subaperture size, inter-firing distance and kernel size, are performed using experimental data acquired from a diffuse scattering phantom. The efficiency of the correlation based motion estimation is tested qualitatively on the phantom image.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117144998","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Spectral extrapolation in sub-band coding","authors":"C. Cafforio, E. Sciascio, C. Guaragnella","doi":"10.1109/DSPWS.1996.555448","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555448","url":null,"abstract":"We examine how feasible it is to extrapolate high-pass terms from the low-pass sub-image obtained by sub-band decomposition. Only very simple techniques like morphological filtering and contours extraction have been considered at this first stage, attempting to synthesize from the low-pass sub-image the attenuated transitions. The results point out that there is room for further investigation and improvements.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"8 5","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114122120","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"On the exact recovery of higher-order moments of noisy signals","authors":"L. Cheded","doi":"10.1109/DSPWS.1996.555519","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555519","url":null,"abstract":"The importance of moments in science and engineering, as witnessed by the continuous and wide applicability of second-order moments (correlations) and the use of their higher-order brethren is clearly unquestionable. Due to the predominance of digital, rather than analogue, signal processing, it is of practical importance to investigate the impact of amplitude quantization on the exact recovery of unquantized moments from their quantized counterparts. We extend the results of Cheded (see IEEE ICASSP'95, p.1816-19, Detroit, USA) to the more general and interesting case where no a priori knowledge of the quantizer input's membership of the class L/sub p/ is available. We introduce a new moment-sense input/output function h/sub p/(x) that statistically characterizes the quantizer. Two new theorems are also stated that solve the exact moment recovery problem. Finally, two approaches to this problem are presented with some simulation results: based on a 1-bit quantizer, that substantiate very well the theory.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"26 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114807682","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Noise tolerance of output-coded neural net","authors":"K. Al-Mashouq","doi":"10.1109/DSPWS.1996.555557","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555557","url":null,"abstract":"Error correcting codes were used previously to encode the output of feed-forward neural nets. We study the effect of additive noise on the performance of a coded net and compare it to an uncoded net. Some necessary analytical tools are developed to estimate the performance of a neural net in the presence of noise. Simulation examples (isolated word utterances recognition) are also included to show the advantage of coding in reducing the probability of classification error due to noise. In addition we point the use of the estimated performance as a lower limit to the performance of a multilayer neural net.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"135 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114752351","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"A distortion analysis of image VQ-based coding using a finite mixture distribution model","authors":"C. Natale, H. Cherifi","doi":"10.1109/DSPWS.1996.555512","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555512","url":null,"abstract":"Traditional coding schemes break an image into blocks prior to coding. It is possible to classify current algorithms according to the block construction. Either the block size is kept constant, or the block size is image dependent. Since most natural images can be divided into regions of high and low detail, variable block-size coding techniques exploit more efficiently the structure of the data. The superior performance of this scheme over fixed block ones has been observed experimentally. Rate-distortion analysis is carried out for compression systems using vector quantization. Using an appropriate model of the signal we derive analytical results that assess the superiority of variable block vector quantization algorithms.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"30 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127451506","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"A two-band linear-phase QMF lattice with an improved robustness to coefficient quantization","authors":"D. Pinchon, P. Siohan","doi":"10.1109/DSPWS.1996.555481","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555481","url":null,"abstract":"In this paper we examine the design problem of a new lattice structure for two-band perfect reconstruction filter banks with linear phase analysis and synthesis filters. The aim is to provide sets of lattice coefficients which are robust to quantization. Two complementary methods are presented which are key elements to derive low dynamic range solutions while satisfying given frequency specifications. The first possible technique is based on a rearrangement of the elementary blocks involved in the cascade structure, and the second is a sequential design method which, for given weighting factors related to the frequency specifications, leads to a minimal dynamic range of the lattice coefficients. Two design examples, illustrating each of these techniques, show quantisation results with a reduced number of bits which yield frequency results close to infinite precision solutions.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124848767","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Performance of the forgetting factor RLS during the transient phase","authors":"G. Moustakides","doi":"10.1109/DSPWS.1996.555538","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555538","url":null,"abstract":"The recursive least squares (RLS) algorithm is one of the most well known algorithms used for adaptive filtering and system identification. We consider the convergence properties of the forgetting factor RLS algorithm in a stationary data environment. We study the dependence of the speed of convergence of RLS with respect to the initialization of the input sample covariance matrix and with respect to the observation noise level. By obtaining estimates of the settling time we show that RLS, in a high SNR environment, when initialized with a matrix of small norm, has a very fast convergence. The convergence speed decreases as we increase the norm of the initialization matrix. In a medium SNR environment the optimum convergence speed of the algorithm is reduced, but the RLS becomes more insensitive to initialization. Finally in a low SNR environment it is preferable to start the algorithm with a matrix of large norm.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"123 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124191166","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"A new algorithm for the automatic search of the best delay in blind equalization","authors":"M. Prandini, M. Campi, R. Leonardi","doi":"10.1109/DSPWS.1996.555561","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555561","url":null,"abstract":"This paper deals with the problem of recovering the input signal applied to a linear time-invariant system from the measures of its output and the a-priori knowledge of the input statistics (blind equalization). Under the assumption of an i.i.d. non-Gaussian input sequence, a new iterative procedure based on phase sensitive high-order cumulants for adjusting the coefficients of a transversal equalizer is introduced. The main feature of the proposed technique is that it realizes the automatic selection of the equalization delay so as to improve the equalization performance.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"578 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124973640","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"The effect of modified filter distribution on an adaptive, sub-band speech enhancement method","authors":"D. Darlington, Douglas R. Campbell","doi":"10.1109/DSPWS.1996.555483","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555483","url":null,"abstract":"An adaptive noise cancellation scheme for speech processing is described, in which adaptive filters operate in frequency-limited sub-bands. Previously the filters had been distributed in a linear fashion in the frequency domain. This work investigates the effects of spacing the filters more in sympathy with the signal power and spectral characteristics. It is found that improvements in signal-to-noise ratio of processed noisy speech signals may be obtained in certain cases when the sub-bands are spaced according to a published cochlear function.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"35 2","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131810451","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"A parallel architecture for adaptive frequency-domain Volterra filtering","authors":"M. A. Shcherbakov","doi":"10.1109/DSPWS.1996.555496","DOIUrl":"https://doi.org/10.1109/DSPWS.1996.555496","url":null,"abstract":"A parallel-serial implementation for an adaptive nonlinear Volterra filter, which exploits the symmetric properties of kernels in the frequency domain, is presented. The realization is based on the successive generation of the m-th order frequency domain of the kernel definition in terms of lower order domains. An efficient implementation structure of the frequency-domain adaptive filters is presented using a specific array of identical processor elements. A great degree of modularity and regularity ensures suitability of the proposed architecture for fast implementation using VLSI technologies.","PeriodicalId":131323,"journal":{"name":"1996 IEEE Digital Signal Processing Workshop Proceedings","volume":"11 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1996-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116460945","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}