{"title":"The Wigner distribution, multi-equilibria nonlinear systems and chaotic behaviour","authors":"P. G. Adamopoulos, J. Hammond, J. S. Lee","doi":"10.1109/ICASSP.1988.197075","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197075","url":null,"abstract":"The Wigner distribution and chaotic behavior of systems have recently received considerable attention: the Wigner-Ville distribution, due to its advantageous time-frequency plane representation of the signals; and the chaotic behavior of certain types of nonlinear system for apparently having a deterministic form. The authors bring the two together and apply the Wigner distribution to the analysis of nonlinear system response including both nonchaotic and chaotic systems. The procedure is based on simulation of time histories and the calculation of Wigner and pseudo-Wigner distributions.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"28 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":"129955584","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}
Wayne H. Ward, Alexander Hauptmann, R. Stern, T. Chanak
{"title":"Parsing spoken phrases despite missing words","authors":"Wayne H. Ward, Alexander Hauptmann, R. Stern, T. Chanak","doi":"10.1109/ICASSP.1988.196569","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196569","url":null,"abstract":"The authors compare the recognition accuracy obtained in forming sentence hypotheses using island-driven sentence parsers with parsers that hypothesize sentences in left-to-right fashion. Island-driven parsing algorithms are especially valuable in speech recognition systems because they can function more gracefully when not all of the correct words of an utterance were produced by the word hypothesizer. The inputs to both types of parsers consist of a lattice of candidate words, which are identified by their begin and end times, and the quality of the acoustic-phonetic match. Grammatical constraints are expressed by trigram models of sequences of lexical and semantic labels. The authors found that the island-driven parser produces parses with a higher percentage of correct words than the left-to-right parser is all cases considered. When the quality of the input lattices is extremely high, differences in parsing accuracy can be directly attributed to the superior ability of the island-driven parser to handle lattices with missing words. With lower-quality input, the accuracy of both types of parsers degrades, which is due to the creation of garden-path hypotheses and a lack of good words to serve as seeds for island formation.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"75 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":"128738151","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 new algorithm for the estimation of hidden Markov model parameters","authors":"L. Bahl, P. Brown, P. D. Souza, R. Mercer","doi":"10.1109/ICASSP.1988.196627","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196627","url":null,"abstract":"Discusses the problem of estimating the parameter values of hidden Markov word models for speech recognition. The authors argue that maximum-likelihood estimation of the parameters does not lead to values which maximize recognition accuracy and describe an alternative estimation procedure called corrective training which is aimed at minimizing the number of recognition errors. Corrective training is similar to a well-known error-correcting training procedure for linear classifiers and works by iteratively adjusting the parameter values so as to make correct words more probable and incorrect words less probable. There are also strong parallels between corrective training and maximum mutual information estimation. They do not prove that the corrective training algorithm converges, but experimental evidence suggests that it does, and that it leads to significantly fewer recognition errors than maximum likelihood estimation.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"36 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":"124580946","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 new adaptive echo canceller used in the ISDN U-interface","authors":"Dacheng Yang, Weiping Li","doi":"10.1109/ICASSP.1988.196916","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196916","url":null,"abstract":"A good adaptive echo canceller (EC), in terms of its performance, is an interesting problem for those who are engaged in the ISDN (integrated-services digital network) U-interface subscriber loop design. It is also important to implement an EC with low cost chips. An adaptive EC is proposed that makes its contribution in this respect. The filter combines some advantages possessed by common types of FIR (finite-impulse response) adaptive filters, transversal, and table lookup filters, together. In implementation, the filter design is mainly based on developing DSP (digital-signal-processor) chips, e.g. the Texas Instruments TMS320 family of microprocessors. The proposed architecture not only has a faster convergence speed than that of usual table lookup adaptive filters, but can also compensate the long period of echo tail caused by insufficient filter taps, especially when it is used in the full-duplex two-wire data transmission system with a net rate of 144 kb/s.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"168 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":"124645432","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":"Simulation and analysis of Suzuki fading processes","authors":"A. Krantzik, D. Wolf","doi":"10.1109/ICASSP.1988.197067","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197067","url":null,"abstract":"The authors deal with the simulation and the analysis of a Suzuki process. This stochastic process can be considered to be a model for the random fluctuations of the envelope of the received signal in land mobile radio transmission systems. Digital speech transmission over such a channel is disturbed by these random fluctuations, in particular during the fading periods. Therefore, the knowledge of the frequency of the fades and the distribution of their duration is of special interest. The simulation system is used to determine these quantities experimentally. Recent results are presented, and the average number of fades is derived theoretically.<<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":"129897197","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 criterion for identifying dominant singular values in the SVD based method of harmonic retrieval","authors":"S. Rao, D. Gnanaprakasam","doi":"10.1109/ICASSP.1988.197141","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197141","url":null,"abstract":"The problem of determining the dominant singular values in the singular value decomposition (SVD) based state-space approach to harmonic retrieval is considered. A common difficulty encountered in harmonic retrieval methods is that the covariance matrix is full rank due to noise and estimation errors, instead of the ideal low rank. Then, from the singular value decomposition of this noisy and estimated covariance matrix, a low rank approximation is normally sought by retaining the dominant singular values and zeroing out the rest. A criterion is proposed, based on the distribution of the norms of the perturbation matrix associated with the estimated covariance matrix, to identify these dominant singular values.<<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":"130340532","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":"An adaptive edge enhancing order statistic filter","authors":"Y.H. Lee, S. Tantaratana","doi":"10.1109/ICASSP.1988.196892","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196892","url":null,"abstract":"A filter which combines L-type structure with a modified median filter (MMF) is proposed for suppressing nonimpulsive as well as impulsive noise. The MMF is designed to reduce edge jitter caused by the median filter. The MMF output is obtained by modifying the median filter output, based on results of hypothesis tests. The proposed filter, named the decision-based order statistic filter (DBOSF), generates the output by using asymmetrical alpha -trimming with respect to the MMF output. In addition to noise suppression, the DBOSF can enhance the gradient of blurred edges provided that the size of the window is greater than twice the transition with width of the blurred edges. Experimental results for one- and two-dimensional signals are presented to demonstrate performances of the proposed filter.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"6 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":"130666767","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":"Geometry-free X-ray reconstruction using the theory of convex projections","authors":"P. Oskoui-Fard, H. Stark","doi":"10.1117/12.967292","DOIUrl":"https://doi.org/10.1117/12.967292","url":null,"abstract":"The authors apply the method of projections onto convex sets (POCS) to reconstruct an image in computer tomography. Although POCS has been used before to recover missing portions of the data set, this is the first time that it is directly used to invert raysum data. Since POCS is geometry-free, it can be directly applied to cases of incomplete data. The strength of POCS compared to other geometry-free methods is its systematic use of a priori information in a natural and consistent way along with the raysum data.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"21 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":"123220188","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":"Stability of adaptive FIR filters","authors":"O. Horna","doi":"10.1109/ICASSP.1988.196889","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196889","url":null,"abstract":"Svoboda's matrix inversion algorithm is used to analyze the adaptive FIR (finite-impulse response) filter. To get stable response, conditions of convergence and solvability have to be met. They determine the maximum correction step of the adaptive loop, the lower bound of the norm of output signal vector and the statistical properties of the reference signal. Methods of signal decorrelation are discussed and the upper bounds of resolution of a digital filter are derived. The long-term instability caused by parasitic DC voltages in the correction loop is analyzed and stabilization methods are suggested.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"33 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":"116459707","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":"An algorithm for order statistic determination and L-filtering","authors":"R. Hoctor, S. Kassam","doi":"10.1109/ICASSP.1988.196940","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196940","url":null,"abstract":"An algorithm is presented for order-statistic determination which operates in a number of steps fewer than or equal to the number of bits in the binary representation of the input. The algorithm uses a bit-level thresholding operation on input data in binary representation with both adaptive thresholds and adaptive binary weights, and it does not require any auxiliary data structure. A bit-level pipelined implementation for this scheme is also presented whose hardware complexity is bounded by N(log/sub 2/N) where N is the window size; this leads to a pipelined structure for L-filtering.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"11 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":"114874623","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}