{"title":"Normalized lattice pole-zero adaptive filters","authors":"K. Miao, H. Fan","doi":"10.1109/ICASSP.1992.226452","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226452","url":null,"abstract":"Adaptive infinite impulse response (IIR) filtering algorithms in the cascade IR normalized lattice forms are developed. These algorithms correspond to the SHARF algorithm, the prefiltering algorithm, and the Stearns algorithm in the direct form case. Their stability properties are discussed and computer simulations provided to verify their convergence.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"104 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":"116015729","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":"UF/sup 3/-a 4-D DSP hypercube with a robust programming environment","authors":"Ahmad R. Ansari, F. Taylor","doi":"10.1109/ICASSP.1992.226541","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226541","url":null,"abstract":"The UF/sup 3/ is a 4-D hypercube multiprocessor system which is integrated with an innovative programming environment. The architecture of UF/sup 3/ contains fully programmable nodes which allow use of the system to address problems with arbitrary entities. The self-scheduling programming environment of UF/sup 3/ uses its bottom-up approach to balance the workload among the processors. The architecture provides the digital signal processor (DSP) community with a high-bandwidth general-purpose accelerator array having a high degree of computational accuracy and a robust programming environment. The principal application of the UF/sup 3/ is serving the numerically intensive computational needs of the DSP community.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"45 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":"116030901","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 general framework for the incorporation of uncertainty in set theoretic estimation","authors":"P. L. Combettes, M. Benidir, B. Picinbono","doi":"10.1109/ICASSP.1992.226229","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226229","url":null,"abstract":"In digital signal processing, the two main sources of uncertainty encountered in estimation problems are model uncertainty and noise. In many instances, probabilistic information is available to partially describe these sources of uncertainty. It is shown how such information can be exploited in a broad class of set theoretic estimation problems relevant to digital signal processing. A general framework is developed to construct sets in the solution space by constraining the estimation residual based on the known component of the model to be consistent with those known properties of a so-called uncertainty process consisting of the contribution of the unknown component of the model and the noise. Specific digital signal processing applications are discussed.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"30 2 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":"116346345","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":"Programmable audiogram matching using a frequency sampling filter implemented on the Texas TMS 320C30","authors":"N. Black, M. Lydon, N. Waterman, M. Powderly","doi":"10.1109/ICASSP.1992.226079","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226079","url":null,"abstract":"A frequency sampling filter which has been implemented on a master digital hearing aid comprising a Texas TMS 320C30 DSP and associated circuitry is presented. The filter runs in real time and is used to provide a response, determined from the human audiogram, which compensates for loss of gain. The algorithm matches given requirements using a filter structure consisting of a comb filter having 400 zeros in cascade with 167 digital resonators.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"15 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":"121937477","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 multi-frame pel-recursive algorithm for varying frame-to-frame displacement estimation","authors":"J. Huang, S. Liu, M. Hayes, R. Mersereau","doi":"10.1109/ICASSP.1992.226206","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226206","url":null,"abstract":"A pel-recursive two-dimensional motion estimation algorithm using three or more image frames is presented. It uses more information from the image sequences and gives a closed-form solution to update motion vectors among frame pairs simultaneously. Compared to other multiframe-based algorithms, the proposed algorithm is in a more general form. It is suitable for object motion with acceleration since it places no constraints on the displacement vectors among consecutive frames.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"75 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":"122140226","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}
C. Montacié, Paul Deléglise, F. Bimbot, Marie-José Caraty
{"title":"Cinematic techniques for speech processing: temporal decomposition and multivariate linear prediction","authors":"C. Montacié, Paul Deléglise, F. Bimbot, Marie-José Caraty","doi":"10.1109/ICASSP.1992.225949","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.225949","url":null,"abstract":"Two models, the temporal decomposition and the multivariate linear prediction, of the spectral evolution of speech signals capable of processing some aspects of the speech variability are presented. A series of acoustic-phonetic decoding experiments, characterized by the use of spectral targets of the temporal decomposition techniques and a speaker-dependent mode, gives good results compared to a reference system (i.e., 70% vs. 60% for the first choice). Using the original method developed by Laforia, a series of text-independent speaker recognition experiments, characterized by a long-term multivariate auto-regressive modelization, gives first-rate results (i.e., 98.4% recognition rate for 420 speakers) without using more than one sentence. Taking into account the interpretation of the models, these results show how interesting the cinematic models are for obtaining a reduced variability of the speech signal representation.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"64 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":"122143772","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 error function selection for the analysis of nonlinear time series","authors":"D. F. Drake, Douglas B. Williams","doi":"10.1109/ICASSP.1992.226616","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226616","url":null,"abstract":"The extreme sensitivity of a chaotic system's steady state response to small changes in its initial conditions makes long term prediction of the evolution of such a system difficult, if not impossible. In the framework of parameter estimation, it is shown how this sensitivity can hinder attempts to determine model parameters that will reproduce a target chaotic time sequence. Specifically, a waveform error minimization technique based on gradient descent optimization is not well suited for estimating the parameters of a strongly chaotic system. A modification of this minimization procedure that avoids some of the obstacles present when estimating the parameters of a chaotic system is proposed.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"31 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":"117116995","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":"Computer algebra and fast algorithms","authors":"G. Sobelman","doi":"10.1109/ICASSP.1992.226404","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226404","url":null,"abstract":"The utility of modern symbolic computation packages in a course of convolution and discrete Fourier transform (DFT) algorithms is presented. It is shown how Mathematica has been used to help illustrate the number-theoretic, polynomial, and finite-field computations involved. The result is that students are relieved of much of the algebraic drudgery and can concentrate on learning the fundamental aspects of the subject. It has been found that large problems can be successfully solved by students in a way that they find to be both interesting and satisfying. In addition, symbolic verification techniques can be used to prove the correctness of the results that are obtained.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"15 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":"117191022","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":"Predictor codebooks for speaker-independent speech recognition","authors":"T. Kawabata","doi":"10.1109/ICASSP.1992.225899","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.225899","url":null,"abstract":"The authors examine the speech recognition capabilities of predictor codebooks under multi-speaker and speaker-independent conditions. Three structures of spectrum predictors, a forward predictor, a backward predictor, and an interpolator, are examined. Predictor codebooks are generated by the LBG algorithm with a small modification for predictor quantization. The predictor codebooks are then tested on a phone recognition task with three different measurements. The degradation in predictor-codebook performance was reduced by one-third under speaker-independent conditions. Finally, continuous-speech recognition experiments are carried out using the predictor codebook for multi-speaker and speaker-independent conditions. The results show that the backward-predictor codebook is very effective.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"269 ","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"120869433","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 whole word recurrent neural network for keyword spotting","authors":"K. Li, J. Naylor, M. L. Rossen","doi":"10.1109/ICASSP.1992.226115","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226115","url":null,"abstract":"The authors present a neural network which is trained on word examples to perform the wordspotting task. This network has multiple recurrent connections with time delay to account for temporal dynamics. A single network may be trained to recognize one word or many words. A hybrid wordspotter is evaluated in which a conventional wordspotter (based on dynamic time warping word matching) is used to screen incoming speech for potential keywords which are then passed to the network for the final accept/reject decision. Initial tests on a standard wordspotting test corpora resulted in improved keyword recognition at false alarm rates above zero.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"19 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":"124466220","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}