ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing最新文献

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Sketch based and context sensitive coding of Laplacian pyramids 基于草图和上下文敏感的拉普拉斯金字塔编码
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196707
J. Vincent
{"title":"Sketch based and context sensitive coding of Laplacian pyramids","authors":"J. Vincent","doi":"10.1109/ICASSP.1988.196707","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196707","url":null,"abstract":"An improvement in pyramidal coding is described which goes beyond first-order statistics by exploiting structural redundancies within Laplacian pyramids. Their three-dimensional structures are described with a novel sketch pyramid, which is used in a new context-sensitive method for coding Laplacian pyramids. This approach is enhanced by using the Gaussian function as the generating kernel, and the sinc function for the expansion filter. The compression ratios obtained from pyramidal coding are improved by a factor ranging from around 2.5 to 4, depending on scene complexity. Example results are given.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"147 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125224729","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}
引用次数: 0
Regularized edge detection 正则化边缘检测
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196773
A. Katsaggelos, B. Sullivan
{"title":"Regularized edge detection","authors":"A. Katsaggelos, B. Sullivan","doi":"10.1109/ICASSP.1988.196773","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196773","url":null,"abstract":"It has been observed that the edge detection problem in image processing is ill-posed. The ill-posed aspect of this problem is the differentiation step, which is an explicit operation in many schemes for finding edges in images. Differentiation is very sensitive to noise and its discrete version is an ill-conditioned operation. The authors concentrate on regularizing the differentiation operation and incorporating it in an edge detection scheme. They consider both a direct implementation of Miller regularization and one involving projections onto convex sets.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131219531","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}
引用次数: 8
Adaptive L/sub p/ approach to speech coding 语音编码的自适应L/sub /方法
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196584
J. Lansford, R. Yarlagadda
{"title":"Adaptive L/sub p/ approach to speech coding","authors":"J. Lansford, R. Yarlagadda","doi":"10.1109/ICASSP.1988.196584","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196584","url":null,"abstract":"Some experimental results using L/sub p/ (1<or=p<or=3) normed models in speech coding are described. In particular, the residual steepest descent (RSD) algorithm has been used to generate L/sub p/ models of speech for a variety of values of p; these models have in turn been used to generate statistics of the residual sequence. The RSD algorithm has been incorporated into the government standard LPC-10 coding algorithm, where the norm p used is selected on the basis of a statistical measure of the residual.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"208 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133447145","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}
引用次数: 6
Estimation of perceptual entropy using noise masking criteria 使用噪声掩蔽准则估计感知熵
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197157
J. Johnston
{"title":"Estimation of perceptual entropy using noise masking criteria","authors":"J. Johnston","doi":"10.1109/ICASSP.1988.197157","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197157","url":null,"abstract":"The perceptual entropy of each short-term section of the audio stimuli is estimated as the number of bits required to encode the short-term spectrum of the signal to the resolution measured by this process provide an entropy estimate, for transparent coding, of 1.4 (mean) or 2.1 (peak) bits/sample for telephone speech (200-3200-Hz bandwidth sampled at 8 kHz). The entropy measures for audio signals of other bandwidths and sampling rates is also reported.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"23 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132567698","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}
引用次数: 153
Mapping linear algebra algorithms into array processors: a case study 将线性代数算法映射到数组处理器:一个案例研究
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197040
G. Bisio, F. Arduini, F. Curatelli, E. Zitti, G. Parodi, A. Zini
{"title":"Mapping linear algebra algorithms into array processors: a case study","authors":"G. Bisio, F. Arduini, F. Curatelli, E. Zitti, G. Parodi, A. Zini","doi":"10.1109/ICASSP.1988.197040","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197040","url":null,"abstract":"A general methodology, based on the concept of folding, for mapping wavefront algorithms into processor square meshes of fixed size has been implemented on a mesh of INMOS transputers, programmed in OCCAM, for the multiplication of matrices of arbitrary size. Different implementations of the code are considered and compared. The computational costs are analyzed in detail. The method and the performances of the system used are compared with other more standard solutions.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"43 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132709150","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}
引用次数: 1
A new carrier frequency estimator for modem signals 一种新的调制解调器信号载波频率估计方法
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196993
T. D. Tran, V. J. Mathews, C. Rushforth
{"title":"A new carrier frequency estimator for modem signals","authors":"T. D. Tran, V. J. Mathews, C. Rushforth","doi":"10.1109/ICASSP.1988.196993","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196993","url":null,"abstract":"A novel carrier frequency estimation scheme for a relatively broad class of voiceband data signals (modem signals) is presented. The class of signals being studied includes different types of phase-shift-keyed (PSK) and frequency-shift-keyed (FSK) signals. The frequency estimates are obtained averaging the estimates of the derivative of the instantaneous signal phase after discarding the estimates at the baud boundaries. Experimental results illustrate that the frequency estimates obtained are unbiased and have small variances. Results also show that the carrier frequency estimation scheme presented is superior to two other techniques in terms of both the mean and variance of the carrier frequency estimates.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127791153","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}
引用次数: 0
Conditions for the 2-D characteristic polynomial of a matrix to be very strict Hurwitz 一个矩阵的二维特征多项式的条件是非常严格的赫维茨
P. Agathoklis
{"title":"Conditions for the 2-D characteristic polynomial of a matrix to be very strict Hurwitz","authors":"P. Agathoklis","doi":"10.1109/29.31276","DOIUrl":"https://doi.org/10.1109/29.31276","url":null,"abstract":"Conditions for the bi-variate characteristic polynomial of a matrix to be very strict Hurwitz are presented. These conditions are based on two different formulations of the 2-D continuous Lyapunov equation. The necessary and sufficient conditions for the existence of positive definite solutions for the first formulation with constant coefficients are presented. It is shown that such an existence is only sufficient but not necessary for the characteristic polynomial to be very strict Hurwitz. Further, the testing of zeros at infinite distant points requires the use of a class of very strict positive real functions. An alternative formulation of the Lyapunov equation with frequency dependent coefficients is also presented based on a new condition for the very strict Hurwitz property. This second formulation of the Lyapunov equation leads to necessary and sufficient conditions for the very strict Hurwitz property.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"71 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127383991","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}
引用次数: 11
Code optimization technique for general-purpose-DSP compiler 通用dsp编译器的代码优化技术
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197023
I. Utsumi, Y. Mori
{"title":"Code optimization technique for general-purpose-DSP compiler","authors":"I. Utsumi, Y. Mori","doi":"10.1109/ICASSP.1988.197023","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197023","url":null,"abstract":"A description is given of code optimization techniques that have been developed for the compiler of general-power digital signal processors (DSPs). While DSPs provide the high-speed arithmetic capability, they are controlled by microinstructions that are in a low-level language, and program development for DSPs is thus not easy to achieve. In order to generate efficient codes for DSP programs, the authors have successfully developed the code optimization techniques for the DSP architecture. A language processing system that utilizes these optimization techniques has been designed for use with the floating-point DSP (MSM6992) and the effectiveness of the optimization techniques has been proven.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127390438","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}
引用次数: 0
Broadband source direction estimation using a circular array 使用圆形阵列的宽带源方向估计
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197194
Y. Rockah, H. Messer
{"title":"Broadband source direction estimation using a circular array","authors":"Y. Rockah, H. Messer","doi":"10.1109/ICASSP.1988.197194","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197194","url":null,"abstract":"The problem of estimating directions of multiple broadband sources using the eigenstructure of the tempospatial data covariance matrix (TSCM) is dealt with. K.M. Buckley and L.J. Griffiths (1986) suggested that, similarly to the narrowband case, the eigenstructure of this covariance matrix can be used to estimate the signal-only (or noise-only) subspace and then use it for direction estimation. However, even for a single source and a given arbitrary array geometry, the eigenvalues of the signal-only tempospatial covariance matrix (TSCM), and therefore its effective rank, depend on the source direction as well as on its spectral level. This leads to serious difficulties in the implementation of Buckley and Griffiths' algorithm. The authors show here that by using a special array geometry, namely a two dimensional circular array, the eigenvalues of the signal-only TSCM for one source are practically independent of the source direction. Based on this property, they suggest a more feasible algorithm for parameter estimation of multiple broadband sources using an array of circular geometry.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"76 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115772076","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}
引用次数: 5
A fast technique for automatic segmentation and classification of textured images 纹理图像的快速自动分割和分类技术
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196796
Y. Boutalis, S. Kollias, G. Carayannis, Levon Sukissian
{"title":"A fast technique for automatic segmentation and classification of textured images","authors":"Y. Boutalis, S. Kollias, G. Carayannis, Levon Sukissian","doi":"10.1109/ICASSP.1988.196796","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196796","url":null,"abstract":"A fast computationally efficient method for automatic segmentation and classification of textured images is presented. The method does not necessarily need a-priori information about the textures present in the image, thus avoiding the necessity of a training set of textures. A fast adaptive multichannel technique for autoregressive image model parameter estimation with fast tracking capabilities and a powerful statistical distance measure are appropriately interweaved to form the proposed technique. Specific properties of the estimation part of the algorithm are exploited to reduce greatly the computational complexity of the distance measure. Some interesting extensions of the method are discussed and examples are given which illustrate the performance of the algorithm.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"86 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114326434","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}
引用次数: 4
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