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Optimal parameter tracking and control of nonlinear systems with time-varying parameters 时变非线性系统的最优参数跟踪与控制
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197077
Carlos March-Leuba, R. B. Perez
{"title":"Optimal parameter tracking and control of nonlinear systems with time-varying parameters","authors":"Carlos March-Leuba, R. B. Perez","doi":"10.1109/ICASSP.1988.197077","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197077","url":null,"abstract":"A new formulation of a class of nonlinear optimal control problems in which the system's parameters are changing arbitrarily with time is presented. A variational technique based on the Pontryagin maximum principle is used to track the time-varying parameters of the system and to calculate the optimal control. The formulation allows online solution for parameters and optimal control calculations by recasting it into the form of an initial-value problem. An application to a nonlinear nuclear reactor in which feedback coefficients and detector time constants are varying is presented.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"10 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":"124453798","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
Low bit-rate image coding using 2-D linear prediction and 2-D stochastic excitation 基于二维线性预测和二维随机激励的低比特率图像编码
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196706
K. Paliwal
{"title":"Low bit-rate image coding using 2-D linear prediction and 2-D stochastic excitation","authors":"K. Paliwal","doi":"10.1109/ICASSP.1988.196706","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196706","url":null,"abstract":"The use of stochastic excited linear predictive coding method is investigated for image coding and its parameters are studied. It is shown that it is not necessary to transmit local bias values of the image frames. It is also shown that the stochastic excitation is not adequate to represent the prediction residual signal. In order to get good performance from this coder, it is necessary to generate the codebook from the actual prediction residual signal.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"74 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":"124092434","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
Event-based multiple-resolution analysis of speech signals 基于事件的语音信号多分辨率分析
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196582
T. Altosaar, M. Karjalainen
{"title":"Event-based multiple-resolution analysis of speech signals","authors":"T. Altosaar, M. Karjalainen","doi":"10.1109/ICASSP.1988.196582","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196582","url":null,"abstract":"A methodology for multiple-resolution analysis and event-based representation of speech signals is presented. The computation of multiple-resolution filtering, event detection and parsing of event structures is described with examples and discussions of auditory modeling aspects. The approach and its implementation are entirely based on object-oriented programming, which provides a systematic framework for the hierarchical nature of the method. Implementation of the analysis system is based on an object-oriented signal processing system called QuickSig running on the Symbolics Lisp machine. The QuickSig system has object classes such as signals, windows and filter banks, each with its own set of method functions for signal processing operations.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"16 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":"127563373","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
Estimating steady-state response of a resonant transducer in a reverberant underwater environment 水下混响环境下谐振换能器稳态响应的估计
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197216
J. D. George, V. Jain, P. Ainsleigh
{"title":"Estimating steady-state response of a resonant transducer in a reverberant underwater environment","authors":"J. D. George, V. Jain, P. Ainsleigh","doi":"10.1109/ICASSP.1988.197216","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197216","url":null,"abstract":"The authors examine the estimation of steady-state amplitude and phase using short, noisy records of the transient response of systems excited by a stepped sinusoid close to a resonance. Linear prediction estimation strategies are tested, and near maximum-likelihood (ML) performance is obtained by combining FIR (finite-impulse response) cancellation of the excitation poles with the truncated singular-value decomposition approach of R. Kumaresan and D.W. Tufts (1982). For the model tested, this excitation constrained estimation strategy departs from ML performance at a threshold signal-to-noise ratio that depends on the separation between the excitation and resonant frequencies.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"44 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":"126496329","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
Connected sentence recognition using diphone-like templates 使用类似电话的模板连接句子识别
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196621
A. Rosenberg
{"title":"Connected sentence recognition using diphone-like templates","authors":"A. Rosenberg","doi":"10.1109/ICASSP.1988.196621","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196621","url":null,"abstract":"A template-based connected speech recognition system which represents words as sequences of diphone-like segments has been implemented and tested on a database of 50 phonetically balanced sentences uttered 5 times by a single male talker. The sentences contain 250 words, of which, 80% are monosyllabic. The inventory of segments is divided into two principal classes, single phone segments, such as vowels, nasals, fricatives, and stop bursts, and diphone segments including consonant-vowel, vowel-consonant, and consonant-consonant combinations. Words are represented by network models whose nodes are these segments. Word models incorporate juncture branches to and from other words. 400 segments are required to represent the 250 vocabulary words. Templates representing these segments are extracted from a database of 450 training sentences uttered by the same talker. Recognition is carried out by a series of matching and search processes, successively for segments, words, word strings, and sentences. The performance obtained to data has yielded 63% correct recognition of content words and approximately 30% recognition of function words.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"84 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":"125631592","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
Non-quadratic recursive algorithms (RLK) for transversal plant identification 横向植物识别的非二次递归算法
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196858
A. Figueiras-Vidal, J. M. Páez-Borrallo, Francisco Lorenz Speranzini
{"title":"Non-quadratic recursive algorithms (RLK) for transversal plant identification","authors":"A. Figueiras-Vidal, J. M. Páez-Borrallo, Francisco Lorenz Speranzini","doi":"10.1109/ICASSP.1988.196858","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196858","url":null,"abstract":"A generalization of the RLS algorithm is presented. The objective measure to be minimized is composed of the sum of arbitrarily weighted kth powers of the observed error (RLK algorithm). The authors formulate general recursive algorithm in the context of noisy transversal plant identification. An approximate analysis of its performance based on the convergence of the mean and covariance matrix of the adaptive filter coefficients is carried out. This analysis evidences the importance of the choice of the order k under the knowledge of the plant noise statistics. The coherence of some computer simulation results for two different algorithms (k=2, 4) and plant noise statistics (binary and Laplacian) with the theoretical analysis is shown.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"154 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":"122257383","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
A comparison of wavelet deconvolution techniques for ultrasonic NDT 超声无损检测中小波反卷积技术的比较
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196725
C. Chen, W. Hsu, S. Sin
{"title":"A comparison of wavelet deconvolution techniques for ultrasonic NDT","authors":"C. Chen, W. Hsu, S. Sin","doi":"10.1109/ICASSP.1988.196725","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196725","url":null,"abstract":"Wavelet deconvolution is a problem of fundamental importance in many signal processing applications. The authors are concerned with deconvolution to extract impulse responses of materials tested in ultrasonic nondestructive testing (NDT). The impulse responses can provide vital information about the flaws and material properties. Several deconvolution techniques including Wiener filtering in three versions, the spiking filter, time domain deconvolution, L1 norm deconvolution, and others are examined and critical comparisons are made for the ultrasonic NDT data.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"45 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":"131441968","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: 13
Hardware and software considerations for active noise control 主动噪声控制的硬件和软件考虑
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197178
M. Allie, Cary D. Bremigan, L. Eriksson, R. Greiner
{"title":"Hardware and software considerations for active noise control","authors":"M. Allie, Cary D. Bremigan, L. Eriksson, R. Greiner","doi":"10.1109/ICASSP.1988.197178","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197178","url":null,"abstract":"Active noise control is a real-time application of adaptive digital filtering which requires the full capabilities of a modern digital signal processing module. Even though frequencies are low, so that sample rates of a few kilohertz can be used, the extensive computations involved require efficient, high-speed processing. System hardware was designed to allow software-controlled versatility as well as fully automatic operation of a complete active noise control system. The ability to self-calibrate and self-model are important system features. Adaptive filter lengths determine the algorithm speed and necessary memory requirements. System limitations were primarily caused by on-chip memory size. The use of the Texas Instruments TMS32010 for this application and the improvements possible with the TMS32020 are described.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"13 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":"131940140","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: 9
Bearing estimation using neural networks 基于神经网络的方位估计
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197059
S. Jha, R. Chapman, T. Durrani
{"title":"Bearing estimation using neural networks","authors":"S. Jha, R. Chapman, T. Durrani","doi":"10.1109/ICASSP.1988.197059","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197059","url":null,"abstract":"Two modifications to the neural-network algorithm originally proposed by J.J. Hopfield (1982), gain annealing and iterated descent, are proposed that yield better convergence to the global minimum. Simulation results are presented to illustrate the performance of the proposed algorithm for bearing estimation.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"24 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":"131947093","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: 15
Stochastic model of diphone-like segments based on trajectory concepts 基于轨迹概念的圆盘状线段随机模型
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196660
P. Marteau, G. Bailly, M. T. Janot-Giorgetti
{"title":"Stochastic model of diphone-like segments based on trajectory concepts","authors":"P. Marteau, G. Bailly, M. T. Janot-Giorgetti","doi":"10.1109/ICASSP.1988.196660","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196660","url":null,"abstract":"A new global approach to coarse classification of speech segments is presented. Markov modeling is applied on an analytic approach to coarticulation. Speech signal evolution of diphone-like segments is modelized by a point moving frame by frame in a factorial space. Kinematic segmentation applied to the trajectory covered by this point enables the authors to build stochastic models of these segments. Input parameters of a Markov model are extracted from a skeleton of this trajectory considered as a functional model of overlapping segments. The evaluation of such representations in a recognition task gives some elements of discussion about the relative information contained in steady states versus transient segments and acoustical trajectories in general.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"10 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":"132083529","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: 7
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