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Realization of linguistic information in the voice fundamental frequency contour of the spoken Japanese 日语口语语音基频轮廓中语言信息的实现
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196673
H. Fujisaki, H. Kawai
{"title":"Realization of linguistic information in the voice fundamental frequency contour of the spoken Japanese","authors":"H. Fujisaki, H. Kawai","doi":"10.1109/ICASSP.1988.196673","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196673","url":null,"abstract":"Although it has been well known that prosody plays an important role both in the intelligibility and in the naturalness of speech, the process of generating natural prosody from linguistic information has not been fully understood. The authors first define units of prosody of spoken Japanese on the basis of analysis of fundamental frequency contours. Prosodic words are defined by the presence of an accent component, while prosodic phrases and clauses are defined by the presence/absence of a pause and resetting/addition of a phrase component. It is shown that the accent components, representing the information concerning lexical word accent, are modified systematically by the syntactic and the discourse information. Classifications of prosodic boundaries are also presented, and the relationship between these two kinds of boundary is described.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"53 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":"124617370","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: 54
Image edge detection and segmentation based on the Hilbert transform 基于希尔伯特变换的图像边缘检测与分割
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196801
Gerassimos M. Livadas, A. Constantinides
{"title":"Image edge detection and segmentation based on the Hilbert transform","authors":"Gerassimos M. Livadas, A. Constantinides","doi":"10.1109/ICASSP.1988.196801","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196801","url":null,"abstract":"An efficient edge detection model based on a two-stage procedure is presented. The first stage consists of a linear transformation of an artificially created signal that has the property of detecting step edges irrespective of the distribution of the intervals between two consecutive edges. It is demonstrated that the Hilbert transform possesses this property and also outperforms the derivative operation in the detection of step edges in the presence of noise. The second stage consists of an appropriate mapping of the filtered results into edge detection primitives. Practical confirmation of the results is given by means of examples.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"104 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":"124807391","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: 17
Neural net based pattern recognition on the graph search machine 基于神经网络的图搜索机模式识别
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197062
H. Na, S. Glinski
{"title":"Neural net based pattern recognition on the graph search machine","authors":"H. Na, S. Glinski","doi":"10.1109/ICASSP.1988.197062","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197062","url":null,"abstract":"The Hopfield net algorithm for pattern recognition has been implemented on a single GSM (graph search machine). The authors report on the methods used to implement the algorithm as well as on the algorithm performance as compared with competing architectures. It is shown that the GSM provides a viable alternative architecture for neural net research and development.<<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":"129385142","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
Unbiased parameter estimation of non-stationary signals on the block processing 对非平稳信号进行无偏参数估计的块处理
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197073
T. Kiryu, T. Iijima
{"title":"Unbiased parameter estimation of non-stationary signals on the block processing","authors":"T. Kiryu, T. Iijima","doi":"10.1109/ICASSP.1988.197073","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197073","url":null,"abstract":"The authors present a nonlinear nonstationary (NN) model which represents time-varying characteristics of interest as the evolution over successive blocks in block processing. The NN model assumes that a nonstationary signal consists of a time-invariant component and a time-varying component over blocks. A set of parameters estimated up to the last block is used to model the time-varying parameters in the current block. Subtracting the time-varying component just modeled from the observed signal provides a transformed signal in the current block. The least-squares (LS) estimation with respect to the transformed signal again gives a new set of parameters. As a result less variance and unbiased estimation of time-varying parameters are achieved.<<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":"129415835","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
High resolution model based 2-D spectrum estimation 基于二维光谱估计的高分辨率模型
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196688
R. R. Hansen, R. Chellappa
{"title":"High resolution model based 2-D spectrum estimation","authors":"R. R. Hansen, R. Chellappa","doi":"10.1109/ICASSP.1988.196688","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196688","url":null,"abstract":"A noncausal autoregressive (NCAR) plus additive noise model is presented for model-based spectrum estimation of two-dimensional sinusoidal signals in noise. The maximum-likelihood (ML) procedure provides consistent and efficient parameter estimates for NCAR models with bilateral neighbor sets, and these properties carry over to the maximum-likelihood estimates of parameters for Gaussian-NCAR-plus-noise models. By assuming a toroidal lattice the complexity of the ML equation is significantly reduced with little impact on the observed accuracy of the estimated spectra. Initial conditions for starting the ML computation are proposed. Experimental results are presented for various signal-to-noise ratios.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"34 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":"129842473","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
DSP system architecture using signed-digit number representation DSP系统架构采用符号数字表示
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196944
P. A. Ramamoorthy, B. Potu, G. Govind
{"title":"DSP system architecture using signed-digit number representation","authors":"P. A. Ramamoorthy, B. Potu, G. Govind","doi":"10.1109/ICASSP.1988.196944","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196944","url":null,"abstract":"Signed-digit (SD) arithmetic techniques are evaluated for applicability to DSP (digital signal-processing) architectures used for high-speed applications. Binary number representations limit the speed of the system due to carry propagation in addition, Residue arithmetic has been tried to alleviate this problem but its use introduces other problems in algebraic comparison, conversion, division, and floating-point representation. It is shown that signed-digit arithmetic offers the advantage of parallelism in computation without the problems associated with the residue number system. An overview of the basic features of SD arithmetic is given, followed by structures for primitive operations required for a general-purpose signal processor.<<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":"128357044","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
Decision- and classification-directed methods in nonstationary signal analysis 非平稳信号分析中的决策导向和分类导向方法
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197072
R. Muir, W. Stirling
{"title":"Decision- and classification-directed methods in nonstationary signal analysis","authors":"R. Muir, W. Stirling","doi":"10.1109/ICASSP.1988.197072","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197072","url":null,"abstract":"An examination is made of alternatives to traditional frequency analysis of nonstationary signals using a decision-directed estimation methodology. This methodology is used to estimate the probability structure of signal energy at discrete frequencies. The method presented utilizes possible harmonic structure in signals of interest by using adaptive coupling of detectors at harmonically related frequencies. This coupling is directed according to signal classifications made on the marginal detector decision outputs. The method given is less computationally intensive than estimation of the full joint probability distribution. Results show improvement over marginal detection alone given true Bayesian statistics.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"3 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":"128186833","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 structured matrix approximation problems 若干结构化矩阵逼近问题
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197104
A. Shaw, R. Kumaresan
{"title":"Some structured matrix approximation problems","authors":"A. Shaw, R. Kumaresan","doi":"10.1109/ICASSP.1988.197104","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197104","url":null,"abstract":"An improved structured matrix approximation approach for simultaneous estimation of frequencies and wavenumbers from 2-D array data is proposed. A quasi-linear relationship of the error with the polynomial coefficients of both the spatial and temporal domains is derived. This leads to an iterative optimization of the error criterion simultaneously in both the domains. By performing simulations it is shown that the method is capable of resolving signals closely spaced in frequency and wavenumber at low SNR. Next, the extendibility of the method for least-squares fitting of Toeplitz/Hankel/data matrix to a given non-Toeplitz/Hankel/data matrix is also discussed.<<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":"128221590","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: 19
Variants of cepstrum based speaker identity verification 基于倒频谱的说话人身份验证变体
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196652
G. Velius
{"title":"Variants of cepstrum based speaker identity verification","authors":"G. Velius","doi":"10.1109/ICASSP.1988.196652","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196652","url":null,"abstract":"Analysis parameters and various distance measures are investigated for a template matching scheme for speaker identity verification (SIV). Two parameters are systematically varied-the length of the signal analysis window, and the order of the linear predictive coding/-cepstrum analysis. Computational costs associated with the choice of parameters are also considered. The distance measures tested are the Euclidean, inverse variance weighting, differential mean weighting, Kahn's simplified weighting, the Mahalanobis distance, and the Fisher linear discriminant. Using the equal error rate (EER) of pairwise utterance dissimilarity distributions, performance is estimated for prespecified and (a simulation of) user-determined input vocabulary. Performance varies significantly across vocabulary, and average performance is approximately 5% EER for the better algorithms on telephone speech.<<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":"129575204","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: 23
A graphics tool for model based signal processing 基于模型的信号处理的图形工具
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196936
A. Raper, J. Mardell
{"title":"A graphics tool for model based signal processing","authors":"A. Raper, J. Mardell","doi":"10.1109/ICASSP.1988.196936","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196936","url":null,"abstract":"Model-based signal processing is concerned with breaking up a system into its components. While this can make the system easier to understand, it imposes a programming overhead on the user. A program is presented that aims to provide a simple way of working with models of signal-generating systems. The program is based around the use of a data structure to represent each component. Three sets of procedures act on these structures, to display the model as a graphical interface, to manipulate the components, and to keep the model consistent after changes have been made.<<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":"129810656","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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