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[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing最新文献

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Gradient-based adaptive filters for non-Gaussian noise environments 非高斯噪声环境下基于梯度的自适应滤波器
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226392
G. Williamson, P. Clarkson
{"title":"Gradient-based adaptive filters for non-Gaussian noise environments","authors":"G. Williamson, P. Clarkson","doi":"10.1109/ICASSP.1992.226392","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226392","url":null,"abstract":"Convergence properties are studied for a class of gradient-based adaptive algorithms known as order statistic least mean square (OSLMS) algorithms. These algorithms apply an order statistic filtering operation to the gradient estimate of the standard least mean square (LMS) algorithm. The order statistic operation in OSLMS can reduce the variance of the gradient estimate (relative to LMS) when operating in non-Gaussian noise environments. A consequence is that in steady state the excess mean square error can be reduced. It is shown that the coefficient estimates for a class of OSLMS algorithms converge when the input signals are i.i.d. and symmetrically distributed.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"48 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127226899","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: 2
Bootstrap: a fast blind adaptive signal separator Bootstrap:一种快速盲自适应信号分离器
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226054
A. Dinc, Yeheskel Bar-Ness
{"title":"Bootstrap: a fast blind adaptive signal separator","authors":"A. Dinc, Yeheskel Bar-Ness","doi":"10.1109/ICASSP.1992.226054","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226054","url":null,"abstract":"A fast multidimensional adaptive algorithm, Bootstrap, is proposed for multiple signal separation. It separates multiple uncorrelated signals imposed on each other. The bootstrap adaptive algorithm, which does not require training sequences, uses an optimization criteria that is based on minimization of output signal correlations. The learning process of this algorithm is compared with that of the least mean square (LMS) algorithm for different eigenvalue spreads. It has been found from computer simulations that the Bootstrap algorithm converges much faster than the LMS algorithm. The learning process of the Bootstrap algorithm is almost independent of eigenvalue spread.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"27 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130008194","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: 30
Optimal synthetic FWI design of state-space digital filters 状态空间数字滤波器的最佳合成FWI设计
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226344
G. Li, M. Gevers
{"title":"Optimal synthetic FWI design of state-space digital filters","authors":"G. Li, M. Gevers","doi":"10.1109/ICASSP.1992.226344","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226344","url":null,"abstract":"The optimal finite word length (FWL) design problem of state-space filters is investigated. Instead of the usual L/sub 1//L/sub 2/-mixed sensitivity measure, it is argued that a sensitivity measure based on the L/sub 2/ norm only is natural and reasonable. The minimization problem of this newly defined sensitivity measure is studied. The set of optimal realizations minimizing this measure is characterized. It is shown that the FWL effects can be synthesized by what is called FWL noise gain (FNG), which is a linear combination of the classical roundoff noise gain and the L/sub 2/ sensitivity measure. By minimizing the FNG with dynamical constraint, the optimal synthetic FWL state-space design problem is formulated. The existence of optimal realizations is shown. The necessary and sufficient condition equation that should be satisfied by the optimal realizations is given.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"33 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126957851","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: 37
Exploiting recursive parameter trajectories in speech analysis 在语音分析中利用递归参数轨迹
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.225956
N. Hubing, K. Yoo
{"title":"Exploiting recursive parameter trajectories in speech analysis","authors":"N. Hubing, K. Yoo","doi":"10.1109/ICASSP.1992.225956","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.225956","url":null,"abstract":"A novel approach to extracting voicing, pitch, and pitch pulse location information by processing the trajectories produced by weighted recursive least squares algorithms is presented. Traditional approaches to extracting this information usually involve performing an LPC analysis, calculating the residual using the LPC parameters, and then processing the residual using various methods. In this work it is shown that if an RLS algorithm is used for the LPC analysis, then extracting the pitch information from the parameter trajectories produced by the algorithm is an accurate alternative to residual-based pitch estimation. This approach also produces accurate LPC parameter estimates by selecting the RLS coefficients at the point just prior to the pitch pulse indicator.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130702672","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
Conventional interference cancellation for minimum redundancy array structure 最小冗余阵列结构的常规干扰消除
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226391
E. Panayirci, Y. Bar-Ness, Wan-Ling Chen
{"title":"Conventional interference cancellation for minimum redundancy array structure","authors":"E. Panayirci, Y. Bar-Ness, Wan-Ling Chen","doi":"10.1109/ICASSP.1992.226391","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226391","url":null,"abstract":"A special thinned array structure, the minimum redundancy array (MRA), is proposed for interference cancellation. This array structure has been proven in the past to have advantages in direction finding applications. It is also shown that MRA performs as a canceler better than a comparable uniform array structure. Simulation results supplement this conclusion.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"129 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132951113","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
Kalman filtering techniques in speech coding 语音编码中的卡尔曼滤波技术
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.225968
S. Crisafulli, J. D. Mills, R. Bitmead
{"title":"Kalman filtering techniques in speech coding","authors":"S. Crisafulli, J. D. Mills, R. Bitmead","doi":"10.1109/ICASSP.1992.225968","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.225968","url":null,"abstract":"The use of Kalman filtering (KF) techniques in speech coding is investigated. The authors show that the common linear predictor (LP) is a special case of the KF based on an all-pole signal model. They also show that the KF algorithm provides fixed-lag smoothing at no additional complexity. Simulation results reveal that KF based speech coding has significant advantage over the equivalent LP based systems, particularly when used with coarsely quantized measurements.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"60 4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130884809","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: 31
An efficient method for the detection of multiple concentric circles 一种检测多个同心圆的有效方法
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226257
X. Cao, F. Deravi
{"title":"An efficient method for the detection of multiple concentric circles","authors":"X. Cao, F. Deravi","doi":"10.1109/ICASSP.1992.226257","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226257","url":null,"abstract":"The authors extend the alternative Hough transform method for the detection of multiple circles proposed by the authors (1990) to the detection of multiple concentric circles contained in an image. The parameters of a circle are determined by the groups of three edge points on the circle. The extension of the proposed rules to search for the three edge points ensures that every member of a group will be on the same circle, either on the inner circle or on the outer one, instead of some of them lying on the inner circle and some of them lying on the outer one. The application of the proposed method to a noisy gray-scale image containing washers shows that the clusters representing circle centers in the parameter space are much more compact than those obtained by the original Hough method for circle detection without involving postprocessing operations.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"44 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131022801","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
The design of low sensitivity digital filters using multi-criterion optimization strategies 基于多准则优化策略的低灵敏度数字滤波器设计
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226372
V. DeBrunner
{"title":"The design of low sensitivity digital filters using multi-criterion optimization strategies","authors":"V. DeBrunner","doi":"10.1109/ICASSP.1992.226372","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226372","url":null,"abstract":"Multicriterion optimization methods for determining low-sensitivity digital filters for a given structure are considered. Pareto optimal as well as min/max methods are considered. It is shown that the methods yield low sensitivity designs via the presented computer algorithms. Filter scaling can be directly incorporated in the designs.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"43 12 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130238519","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
Effects of convolution on chaotic signals 卷积对混沌信号的影响
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226468
S. Isabelle, A. Oppenheim, G. Wornell
{"title":"Effects of convolution on chaotic signals","authors":"S. Isabelle, A. Oppenheim, G. Wornell","doi":"10.1109/ICASSP.1992.226468","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226468","url":null,"abstract":"Because chaotic signals are potentially useful both in describing physical phenomena and in engineering applications, signal processing algorithms exploiting their unique characteristics are of interest. The authors consider issues pertaining to processing signals in convolutional distortion. Specifically, they discuss the effects of convolutional distortion on two parameters commonly used in the description of chaotic signals-the Lyapunov exponents and the fractal dimension of the attractor. In addition, a blind deconvolution technique based on minimizing a nonlinear prediction error for data generated by one-dimensional chaotic maps is presented.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130238581","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
VLSI architectures for Dirichlet arithmetic Dirichlet算法的VLSI架构
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226379
G. Ray
{"title":"VLSI architectures for Dirichlet arithmetic","authors":"G. Ray","doi":"10.1109/ICASSP.1992.226379","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226379","url":null,"abstract":"A new system of arithmetic is presented called Dirichlet arithmetic which models the arithmetic on the coefficients of a Dirichlet series. This approach has the property that output digits depend on very few of the input digits for the basic operations of addition, multiplication, and division. What is perhaps more interesting is that Dirichlet arithmetic has the same near parallelism for all the elementary transcendental functions (log, exp, sin, cos, sinh, cosh, sin/sup -1/, cos/sup -1/, etc.) as well. Furthermore, this property follows from the fact that the values of the elementary transcendental functions are represented naturally by their Dirichlet digits and can be computed by operations on the input digits as simple as those for multiplication or division.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"8 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130417363","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
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