Pub Date : 1987-04-06DOI: 10.1109/ICASSP.1987.1169464
G. Rigoll
This paper describes the development of the German version of the DECtalk system, which was originally designed for the American language by D.H. Klatt. The aim of this paper is not only to provide an overview on the problems and difficulties for German text-to-speech conversion, using the cascade/parallel formant synthesizer and on the use of new algorithms for parameter extraction, but also to provide a study of the modification procedure which is necessary to build a new language version for a text-to-speech system which was designed for a different, language. These experiences are important for the future design of multilingual text-to-speech systems because the modification from one language to another language gives automatically the answer to many questions which are interesting for the design of a multilingual system or at least a system which can be easily modified for another language. The paper describes the most important steps that have to be performed during the modification procedure, i.e. text normalization and letter-to-sound rules, description of the used synthesizer, description of the tools used for speech analysis and comparison of synthetic and natural speech, tuning of the stationary parts of the phonemes, transitions, prosodies and generation of different voices.
{"title":"The dectalk system for German: A study of the modification of a text-to-speech converter for a foreign language","authors":"G. Rigoll","doi":"10.1109/ICASSP.1987.1169464","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169464","url":null,"abstract":"This paper describes the development of the German version of the DECtalk system, which was originally designed for the American language by D.H. Klatt. The aim of this paper is not only to provide an overview on the problems and difficulties for German text-to-speech conversion, using the cascade/parallel formant synthesizer and on the use of new algorithms for parameter extraction, but also to provide a study of the modification procedure which is necessary to build a new language version for a text-to-speech system which was designed for a different, language. These experiences are important for the future design of multilingual text-to-speech systems because the modification from one language to another language gives automatically the answer to many questions which are interesting for the design of a multilingual system or at least a system which can be easily modified for another language. The paper describes the most important steps that have to be performed during the modification procedure, i.e. text normalization and letter-to-sound rules, description of the used synthesizer, description of the tools used for speech analysis and comparison of synthetic and natural speech, tuning of the stationary parts of the phonemes, transitions, prosodies and generation of different voices.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"67 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129739981","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}
Pub Date : 1987-04-06DOI: 10.1109/ICASSP.1987.1169427
C. Myers
This paper discusses the symbolic representation and manipulation of signals and signal processing operations in computers. Symbolic representation and manipulation of signals, in contrast to numerical representation and manipulation, is the manipulation of signal properties and signal descriptions, rather than the computation of signal values. We identify three classes of symbolic manipulations of signals - symbolic manipulation of signal properties, rearrangement of signal processing expressions, and manipulation of abstract signal classes - and we describe the Extended Signal Processing Language and Interactive Computing Environment (E-SPLICE), a system for the symbolic manipulation of signals. Examples of the symbolic manipulation of signals are given, including expression simplification, generation of equivalent forms of an expression, the manipulation of signal properties, the measurement of computational cost, and the selection of signal processing algorithm implementation.
{"title":"Symbolic representation and manipulation of signals","authors":"C. Myers","doi":"10.1109/ICASSP.1987.1169427","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169427","url":null,"abstract":"This paper discusses the symbolic representation and manipulation of signals and signal processing operations in computers. Symbolic representation and manipulation of signals, in contrast to numerical representation and manipulation, is the manipulation of signal properties and signal descriptions, rather than the computation of signal values. We identify three classes of symbolic manipulations of signals - symbolic manipulation of signal properties, rearrangement of signal processing expressions, and manipulation of abstract signal classes - and we describe the Extended Signal Processing Language and Interactive Computing Environment (E-SPLICE), a system for the symbolic manipulation of signals. Examples of the symbolic manipulation of signals are given, including expression simplification, generation of equivalent forms of an expression, the manipulation of signal properties, the measurement of computational cost, and the selection of signal processing algorithm implementation.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"14 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124827664","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}
Pub Date : 1987-04-06DOI: 10.1109/ICASSP.1987.1169624
Don H. Johnson, P. Rao
It is shown that non-Gaussian time series require new analysis methods to extract their structure. Spectral analysis does not seem to provide the precise information required to analyze important aspects of a time series. The conditional expected value can, in simple cases, be related to Components of the Barrett-Lampard expansion, which provides a mathematical tool for determining the system which can generate the time series.
{"title":"Properties and generation of non-Gaussian time series","authors":"Don H. Johnson, P. Rao","doi":"10.1109/ICASSP.1987.1169624","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169624","url":null,"abstract":"It is shown that non-Gaussian time series require new analysis methods to extract their structure. Spectral analysis does not seem to provide the precise information required to analyze important aspects of a time series. The conditional expected value can, in simple cases, be related to Components of the Barrett-Lampard expansion, which provides a mathematical tool for determining the system which can generate the time series.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"67 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122260821","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}
Pub Date : 1987-04-06DOI: 10.1109/ICASSP.1987.1169881
B. Upadhyaya, O. Glöckler, F. Wolvaardt
A fault detection approach based on the combination of the Generalized Consistency Check and the Sequential Probability Ratio Test is developed and applied for validation of signals from process sensors. The basic methodology requires at least triple redundancy of a given measurement from like sensors and analytical measurements. The separate measurement of the signal mean value and the random fluctuation improves the reliability of fault identification and signal reconstruction. The diagnostics of the source of anomaly in a sub-system is performed by multivariate autoregressive modeling of the process signals and the analysis of resulting signatures.
{"title":"Combined dynamic data analysis and process variable prediction approach for system fault detection","authors":"B. Upadhyaya, O. Glöckler, F. Wolvaardt","doi":"10.1109/ICASSP.1987.1169881","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169881","url":null,"abstract":"A fault detection approach based on the combination of the Generalized Consistency Check and the Sequential Probability Ratio Test is developed and applied for validation of signals from process sensors. The basic methodology requires at least triple redundancy of a given measurement from like sensors and analytical measurements. The separate measurement of the signal mean value and the random fluctuation improves the reliability of fault identification and signal reconstruction. The diagnostics of the source of anomaly in a sub-system is performed by multivariate autoregressive modeling of the process signals and the analysis of resulting signatures.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"14 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127805600","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}
Pub Date : 1987-04-06DOI: 10.1109/ICASSP.1987.1169735
L. Griffiths, M. Rude
In least-squares estimation problems, a desired signald(n)is estimated using a linear combination of L successive data samples, [x(n), x(n - 1), . . . , x(n-L+1)]. The weight set Woptwhich minimizes the mean-square error betweend(n)and the estimate is given by the product of the inverse data covariance matrix and the cross-correlation between the data vector and the desired signal, i.e. the P-vector. For those cases in which time samples of both the desired and data vector signals are available, a variety of adaptive methods have been proposed which will guarantee that an iterative weight vectorW_{a}(n)converges (in some sense) to the optimal solution. Two which have been extensively studied are the recursive least-squares (RLS) method and the LMS gradient approximation approach. There are several problems of interest in the communication and radar environment in which the optimal least-squares weight set is of interest and in which time samples of the desired signal are not available. Examples can be found in array processing in which only the direction of arrival of the desired signal is known and in single channel filtering where the spectrum of the desired response is known a priori. One approach to these problems which has been suggested is the P-vector algorithm which is an LMS-like approximate gradient method. Although it is easy to derive the mean and variance of the weights which result with this algorithm, there has never been an identification of the corresponding underlying error surface which the procedure searches. The purpose of this paper is to suggest an alternative approach to providing adaptive solutions to problems in which samples ofd(n)are unavailable. The method is based on the use of linearly-constrained minimum mean-square error methods. The constraint used is simply that the inner product of the filter weights with the known P-vector must be unity. The criterion employed is then minimization of total output power, subject to this constraint. Once this problem has been formulated, it can be readily implemented in either scalar or multi-channel form using the Generalized Sidelobe Canceller method. Both LMS-like and RLS algorithms may be employed to update the coefficients.
{"title":"Adaptive filtering without a desired signal","authors":"L. Griffiths, M. Rude","doi":"10.1109/ICASSP.1987.1169735","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169735","url":null,"abstract":"In least-squares estimation problems, a desired signald(n)is estimated using a linear combination of L successive data samples, [x(n), x(n - 1), . . . , x(n-L+1)]. The weight set Woptwhich minimizes the mean-square error betweend(n)and the estimate is given by the product of the inverse data covariance matrix and the cross-correlation between the data vector and the desired signal, i.e. the P-vector. For those cases in which time samples of both the desired and data vector signals are available, a variety of adaptive methods have been proposed which will guarantee that an iterative weight vectorW_{a}(n)converges (in some sense) to the optimal solution. Two which have been extensively studied are the recursive least-squares (RLS) method and the LMS gradient approximation approach. There are several problems of interest in the communication and radar environment in which the optimal least-squares weight set is of interest and in which time samples of the desired signal are not available. Examples can be found in array processing in which only the direction of arrival of the desired signal is known and in single channel filtering where the spectrum of the desired response is known a priori. One approach to these problems which has been suggested is the P-vector algorithm which is an LMS-like approximate gradient method. Although it is easy to derive the mean and variance of the weights which result with this algorithm, there has never been an identification of the corresponding underlying error surface which the procedure searches. The purpose of this paper is to suggest an alternative approach to providing adaptive solutions to problems in which samples ofd(n)are unavailable. The method is based on the use of linearly-constrained minimum mean-square error methods. The constraint used is simply that the inner product of the filter weights with the known P-vector must be unity. The criterion employed is then minimization of total output power, subject to this constraint. Once this problem has been formulated, it can be readily implemented in either scalar or multi-channel form using the Generalized Sidelobe Canceller method. Both LMS-like and RLS algorithms may be employed to update the coefficients.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"47 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115785567","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}
Pub Date : 1987-04-06DOI: 10.1109/ICASSP.1987.1169446
R. Vaccaro, Fu Li
The problem considered in this paper is the following: given a symmetric sequence{r_{j}}which is not positive, (i.e. its Fourier transform takes on negative values), find a positive sequence{bar{r}_{j}}which is a good approximation to{r_{j}}. This problem arises, for example, when{r_{j}}is an estimated covariance sequence which is not positive, and a positive covariance sequence is desired. A solution to the positivity problem is given which uses state-space models and a scaled algebraic Riccati equation.
{"title":"A state-space approach to positive sequences","authors":"R. Vaccaro, Fu Li","doi":"10.1109/ICASSP.1987.1169446","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169446","url":null,"abstract":"The problem considered in this paper is the following: given a symmetric sequence{r_{j}}which is not positive, (i.e. its Fourier transform takes on negative values), find a positive sequence{bar{r}_{j}}which is a good approximation to{r_{j}}. This problem arises, for example, when{r_{j}}is an estimated covariance sequence which is not positive, and a positive covariance sequence is desired. A solution to the positivity problem is given which uses state-space models and a scaled algebraic Riccati equation.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"317 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132607663","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}
Pub Date : 1987-04-06DOI: 10.1109/ICASSP.1987.1169517
Juin-Hwey Chen, A. Gersho
A novel least-squares formulation of the vector linear prediction (VLP) problem is presented. Based on this formulation, we develop two new design methods for obtaining the optimal vector predictor for frame-adaptive prediction: the covariance method and the autocorrelation method, which bear the names of the corresponding methods in scalar LPC analysis. Our formulation reveals several previously unrecognized properties of the resulting normal equation. Simulation results for VLP of speech waveforms confirm that the two proposed methods indeed give higher prediction gain than previously developed methods.
{"title":"Covariance and autocorrelation methods for vector linear prediction","authors":"Juin-Hwey Chen, A. Gersho","doi":"10.1109/ICASSP.1987.1169517","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169517","url":null,"abstract":"A novel least-squares formulation of the vector linear prediction (VLP) problem is presented. Based on this formulation, we develop two new design methods for obtaining the optimal vector predictor for frame-adaptive prediction: the covariance method and the autocorrelation method, which bear the names of the corresponding methods in scalar LPC analysis. Our formulation reveals several previously unrecognized properties of the resulting normal equation. Simulation results for VLP of speech waveforms confirm that the two proposed methods indeed give higher prediction gain than previously developed methods.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"67 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125721270","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}
Pub Date : 1987-04-06DOI: 10.1109/ICASSP.1987.1169563
J. Vaisey, A. Gersho
Current algorithms for the efficient coding of images generally employ one of two techniques: Vector Quantization or Transform Coding. Both of these methods are based on block coding, and the traditional algorithms require the partitioning of the original image into a number of, usually square, regions of uniform size. This paper explores techniques where the size of these regions, or blocks, is varied according to the local detail of the image. By isolating regions of differing detail, using a quad-tree based segmentation algorithm to control the size of the blocks, the low detail areas can be encoded at substantially lower rates than is otherwise possible. Satisfactory quality coding results are achieved at rates between 0.35 and 0.4 bpp; the actual rate depends on the image being encoded.
{"title":"Variable block-size image coding","authors":"J. Vaisey, A. Gersho","doi":"10.1109/ICASSP.1987.1169563","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169563","url":null,"abstract":"Current algorithms for the efficient coding of images generally employ one of two techniques: Vector Quantization or Transform Coding. Both of these methods are based on block coding, and the traditional algorithms require the partitioning of the original image into a number of, usually square, regions of uniform size. This paper explores techniques where the size of these regions, or blocks, is varied according to the local detail of the image. By isolating regions of differing detail, using a quad-tree based segmentation algorithm to control the size of the blocks, the low detail areas can be encoded at substantially lower rates than is otherwise possible. Satisfactory quality coding results are achieved at rates between 0.35 and 0.4 bpp; the actual rate depends on the image being encoded.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133263011","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}
Pub Date : 1987-04-06DOI: 10.1109/ICASSP.1987.1169507
J. Deller, D. Hsu, L. Ferrier
A method for recognition of speech of the nonverbal is presented. Results of the application of the procedure to simple utterances by cerebral palsied individuals and implications for an overall communications device are addressed.
提出了一种非语言语音识别方法。结果的应用程序，以简单的话语脑瘫患者和影响的整体通信设备进行了讨论。
{"title":"Recognition of Cerebral Palsy speech: Technical method and a study of vowel consistency","authors":"J. Deller, D. Hsu, L. Ferrier","doi":"10.1109/ICASSP.1987.1169507","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169507","url":null,"abstract":"A method for recognition of speech of the nonverbal is presented. Results of the application of the procedure to simple utterances by cerebral palsied individuals and implications for an overall communications device are addressed.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133180970","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}
Pub Date : 1987-04-06DOI: 10.1109/ICASSP.1987.1169483
S. Pei, Sy-Been Jaw
This paper deals with the design of 2D multiplierless FIR digital filters, first a special class of multiplierless 1D FIR filters are designed with coefficients which are a sum or difference of power-of-two, and then map it into 2D multiplierless FIR filters by McClellan transform; This technique has the advantage that it is very fast, because only a 1-D filter must be designed, also the hardware implementation for these 2D FIR filters is very attractive and efficient.
{"title":"Efficient design of 2D multiplierless FIR filters by transformation","authors":"S. Pei, Sy-Been Jaw","doi":"10.1109/ICASSP.1987.1169483","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169483","url":null,"abstract":"This paper deals with the design of 2D multiplierless FIR digital filters, first a special class of multiplierless 1D FIR filters are designed with coefficients which are a sum or difference of power-of-two, and then map it into 2D multiplierless FIR filters by McClellan transform; This technique has the advantage that it is very fast, because only a 1-D filter must be designed, also the hardware implementation for these 2D FIR filters is very attractive and efficient.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"103 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127646237","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}
京公网安备 11010802042870号
京ICP备2023020795号-1
扫码关注我们
求助内容：
应助结果提醒方式：