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ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing最新文献

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Square root normalized feedback ladder algorithm for the identification of moving average systems 平方根归一化反馈阶梯算法用于移动平均系统的识别
ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1984-03-19 DOI: 10.1109/ICASSP.1984.1172390
C. Muravchik, M. Morf
{"title":"Square root normalized feedback ladder algorithm for the identification of moving average systems","authors":"C. Muravchik, M. Morf","doi":"10.1109/ICASSP.1984.1172390","DOIUrl":"https://doi.org/10.1109/ICASSP.1984.1172390","url":null,"abstract":"We have presented a square root normalized version of the feedback ladder algorithm for the identification of the parameters of a moving average model. The number of equations needed is reduced from eight in the unnormalized case to just five. The complexity of the equations increases but the procedure is justified because it seems to lead to a more convenient hardware realization. Moreover, this realization would be completely similar (for the backward and forward residuals lines) to the CORDIC processors implementation already proposed for the feedlorward ladder algorithms (FFLA). A possible disadvantage is that three of the variables used may have magnitudes greater than one. However the essential feature of the FBLA, that of being able to read out directly the estimated coefficients of the -monic-polynomial model is not modified.","PeriodicalId":112264,"journal":{"name":"ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"9 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1984-03-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129010798","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
A convergence analysis of an adaptive underwater passive tracking system 自适应水下无源跟踪系统的收敛性分析
ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1984-03-19 DOI: 10.1109/ICASSP.1984.1172778
R. Moose, Mauro J. Caputi
{"title":"A convergence analysis of an adaptive underwater passive tracking system","authors":"R. Moose, Mauro J. Caputi","doi":"10.1109/ICASSP.1984.1172778","DOIUrl":"https://doi.org/10.1109/ICASSP.1984.1172778","url":null,"abstract":"The ability of the adaptive filtering system to converge to an unbiased estimate cf those target parameters of interest such as range and depth is examined. Passive target measurements make use of difference in signal arrival time between geometrically separated sensor systems such as those described in Knapp and Carter (1976), Hassab and Boucher (1976), Hassab (1976), McCabe and Moose (1981). While generally good results of different simulated tracking scenarios have been reported upon in Moose (1983), and Moose and Dailey (1983) these results are valid only for the geometries that were specifically simulated. Thus a theoretical investigation is necessary to examine filter convergence after an initial target detection or target maneuver has occurred. Due to the complexity of the nonlinear data generation and tracking system shown for the vertical plane, and not shown, though very similar for range and bearing in the horizontal ocean plane the convergence analysis is part analytic and part computer analysis. Preliminary results show that the tracking systems converge, but converge with a small bias that is both geometry and signal to noise ratio dependent.","PeriodicalId":112264,"journal":{"name":"ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1984-03-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127902456","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
Some properties of a family of generalized time-limited window functions 一类广义限时窗函数的一些性质
ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1984-03-19 DOI: 10.1109/ICASSP.1984.1172330
J. Kaiser
{"title":"Some properties of a family of generalized time-limited window functions","authors":"J. Kaiser","doi":"10.1109/ICASSP.1984.1172330","DOIUrl":"https://doi.org/10.1109/ICASSP.1984.1172330","url":null,"abstract":"For use in the design of finite impulse response (FIR) digital filters via the window function method the first integral of the window is required in order to relate transition width, filter order, and maximum passband and stopband error values. Again this relationship for the generalized family is found to be nearly linear if maximum error is measured in logarithmic units. Approximate empirical expressions are given for these relationships. Thus one can now design FIR filters with controlled error concentrated to any prescribed degree near to the band edges. Convenient computation methods for the generalized window functions are also described as well as the location of zeros and maxima and minima of their transforms.","PeriodicalId":112264,"journal":{"name":"ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"28 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1984-03-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128135162","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
Modern, active sonar AGC design considerations 现代主动声呐AGC设计考虑
ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1984-03-19 DOI: 10.1109/ICASSP.1984.1172682
R. Seegal
{"title":"Modern, active sonar AGC design considerations","authors":"R. Seegal","doi":"10.1109/ICASSP.1984.1172682","DOIUrl":"https://doi.org/10.1109/ICASSP.1984.1172682","url":null,"abstract":"The following identifies key elements of the AGC design problem for active sonars. Because the character of the background noise and of the echo are highly dependent on a sonar environment that varies from place to place and from hour to hour, the signal statistics are unknown. Researchers have left the area of AGC design to practitioners; such designs are guided usually by heuristics. This paper follows that tradition. It postulates a microprocessor-controlled gain controller that adapts its parameters to the sonar environment. Since the sonar for which this AGC was designed is still in development, the performance of the AGC is evaluated with the aid of a simulation.","PeriodicalId":112264,"journal":{"name":"ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"28 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1984-03-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133131366","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 speech direction finder 语音测向器
ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1984-03-19 DOI: 10.1109/ICASSP.1984.1172557
D. Fischell, C. Coker
{"title":"A speech direction finder","authors":"D. Fischell, C. Coker","doi":"10.1109/ICASSP.1984.1172557","DOIUrl":"https://doi.org/10.1109/ICASSP.1984.1172557","url":null,"abstract":"The speech direction finder described here is a relatively simple device based on an off the shelf microcomputer. It can provide the direction to a talker to within 3 degrees of azimuth angle on a single spoken syllable, will only respond to speech, and when used with Wallace linear array microphones can provide this at distances of 50 feet or more. There are numerous applications for the device which may enhance the quality of audio and video teleconferences.","PeriodicalId":112264,"journal":{"name":"ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"8 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1984-03-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130824194","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: 14
A new digital voice summing technique for teleconferencing 一种新的用于电话会议的数字语音求和技术
ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1984-03-19 DOI: 10.1109/ICASSP.1984.1172465
T. Hsing
{"title":"A new digital voice summing technique for teleconferencing","authors":"T. Hsing","doi":"10.1109/ICASSP.1984.1172465","DOIUrl":"https://doi.org/10.1109/ICASSP.1984.1172465","url":null,"abstract":"The objectives of this paper are to investigate various speech coding techniques to determine their applicability to voice conferencing, to present a new technique for summing directly from the encoded signals, and to demonstrate the audio results and effectiveness of the proposed voice summing technique.","PeriodicalId":112264,"journal":{"name":"ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"30 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1984-03-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133345292","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
VLSI Architecture for signal processing with alternate low-level primitive structures (ALPS) 交替低阶原始结构(ALPS)信号处理的VLSI体系结构
ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1984-03-19 DOI: 10.1109/ICASSP.1984.1172307
T. E. Curtis, A. Constantinides, Y. Wu
{"title":"VLSI Architecture for signal processing with alternate low-level primitive structures (ALPS)","authors":"T. E. Curtis, A. Constantinides, Y. Wu","doi":"10.1109/ICASSP.1984.1172307","DOIUrl":"https://doi.org/10.1109/ICASSP.1984.1172307","url":null,"abstract":"A set of Alternate Low-Level Primitive Structures (ALPS) has been considered in this context. It is envisaged that each standalone structure consists of an input queue, an output queue, the processing primitive, and mechanisms for control and synchronization. Some of these primitives and a new system architecture, which allows orderly VLSI/VHSIC transition are described.","PeriodicalId":112264,"journal":{"name":"ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"15 6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1984-03-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125321957","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
On redefining the optimal least squares filter under floating point operations 浮点运算下最优最小二乘滤波器的重新定义
ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1984-03-19 DOI: 10.1109/ICASSP.1984.1172597
Erik I. Verriest
{"title":"On redefining the optimal least squares filter under floating point operations","authors":"Erik I. Verriest","doi":"10.1109/ICASSP.1984.1172597","DOIUrl":"https://doi.org/10.1109/ICASSP.1984.1172597","url":null,"abstract":"A novel solution on approximation to the least squares filter problem under floating point arithmetic is presented for a linear stochastic model.","PeriodicalId":112264,"journal":{"name":"ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"26 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1984-03-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125393756","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
Synthesis by rule of english intonation patterns 英语语调模式规则合成
ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1984-03-19 DOI: 10.1109/ICASSP.1984.1172427
Mark Anderson, J. Pierrehumbert, M. Liberman
{"title":"Synthesis by rule of english intonation patterns","authors":"Mark Anderson, J. Pierrehumbert, M. Liberman","doi":"10.1109/ICASSP.1984.1172427","DOIUrl":"https://doi.org/10.1109/ICASSP.1984.1172427","url":null,"abstract":"This papet reports work on synthesizing English F0 contours. One motivation for this work is to improve the naturalness and liveliness of the prosody in speech synthesis systems. However, our main goal is to develop a theory of the dimensions of variation controlling intonation, and of their interaction.","PeriodicalId":112264,"journal":{"name":"ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1984-03-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134562652","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: 87
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