Pub Date : 1995-05-09DOI: 10.1109/ICASSP.1995.480433
M. Karjalainen, M. Rahkila
A computer-based education (CBE) system is described that is built on top of the QuickSig DSP programming environment. As two CBE applications the authors discuss the implementation and use of "Introduction to Signal Processing" and "Fundamentals of Psychoacoustics" that exploit modern multi and hypermedia features on new computers. The signal processing that is applied, the sound I/O, the graphical user interface, and the CBE navigation system are presented.
{"title":"Learning signal processing concepts and psychoacoustics in the QuickSig DSP environment","authors":"M. Karjalainen, M. Rahkila","doi":"10.1109/ICASSP.1995.480433","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480433","url":null,"abstract":"A computer-based education (CBE) system is described that is built on top of the QuickSig DSP programming environment. As two CBE applications the authors discuss the implementation and use of \"Introduction to Signal Processing\" and \"Fundamentals of Psychoacoustics\" that exploit modern multi and hypermedia features on new computers. The signal processing that is applied, the sound I/O, the graphical user interface, and the CBE navigation system are presented.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"12 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121144701","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.480578
A. Swindlehurst
This paper is concerned with the problem of optimal (maximum likelihood) direction of arrival (DOA) estimation in situations where the sensor array is calibrated over only a portion of the DOA space. Situations such as this often arise in airborne direction finding when skywave multipath is present. A parameterization is proposed for partially calibrated arrays (PCAs), and the identifiability of the model is discussed for both uncorrelated and correlated signals. It is shown how the signal and noise subspace fitting algorithms are generalized to handle PCAs, and a detection scheme is proposed for individually determining the number of signals arriving from calibrated and uncalibrated directions. The results of several simulation examples are included to validate the analysis.
{"title":"Optimal direction finding with partially calibrated arrays","authors":"A. Swindlehurst","doi":"10.1109/ICASSP.1995.480578","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480578","url":null,"abstract":"This paper is concerned with the problem of optimal (maximum likelihood) direction of arrival (DOA) estimation in situations where the sensor array is calibrated over only a portion of the DOA space. Situations such as this often arise in airborne direction finding when skywave multipath is present. A parameterization is proposed for partially calibrated arrays (PCAs), and the identifiability of the model is discussed for both uncorrelated and correlated signals. It is shown how the signal and noise subspace fitting algorithms are generalized to handle PCAs, and a detection scheme is proposed for individually determining the number of signals arriving from calibrated and uncalibrated directions. The results of several simulation examples are included to validate the analysis.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"103 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127133890","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.480046
H. Winkler, H. Fahrner, M. Lang
An efficient system for structural analysis of handwritten mathematical expressions is proposed. To handle the problems caused by handwriting, this system is based on a soft-decision approach. This means that alternatives for the solution are generated during the analysis process if the relation between two symbols within the expression is ambiguous. Finally a string containing the mathematical information is generated and syntactical verified for each alternative. Strings failing this verification are considered as invalid.
{"title":"A soft-decision approach for structural analysis of handwritten mathematical expressions","authors":"H. Winkler, H. Fahrner, M. Lang","doi":"10.1109/ICASSP.1995.480046","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480046","url":null,"abstract":"An efficient system for structural analysis of handwritten mathematical expressions is proposed. To handle the problems caused by handwriting, this system is based on a soft-decision approach. This means that alternatives for the solution are generated during the analysis process if the relation between two symbols within the expression is ambiguous. Finally a string containing the mathematical information is generated and syntactical verified for each alternative. Strings failing this verification are considered as invalid.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"33 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124813961","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.479423
T. M. Chin, A. Mariano
Large-scale extended Kalman filters for atmospheric and oceanic circulation models can readily be approximated using a wavelet transform or a Markov random field model. For a filtering problem where the unknown field of the state variables is highly correlated and the observations are relatively sparse, the wavelet-approximated filter seems more appropriate. For a problem in which the covariance matrix is non-singular and where a relatively large quantity of independent observations are processed, the MRF-approximated filter seems more appropriate.
{"title":"Kalman filtering of large-scale geophysical flows by approximations based on Markov random field and wavelet","authors":"T. M. Chin, A. Mariano","doi":"10.1109/ICASSP.1995.479423","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.479423","url":null,"abstract":"Large-scale extended Kalman filters for atmospheric and oceanic circulation models can readily be approximated using a wavelet transform or a Markov random field model. For a filtering problem where the unknown field of the state variables is highly correlated and the observations are relatively sparse, the wavelet-approximated filter seems more appropriate. For a problem in which the covariance matrix is non-singular and where a relatively large quantity of independent observations are processed, the MRF-approximated filter seems more appropriate.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"47 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124828935","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.479463
X. Kong, V. Goel, N. Thakor
Accurate detection and characterization of changes in the EEG signal is crucial for clinical assessment of the neurological system condition. Several distance measures are tested and evaluated for their effectiveness of detecting injury-related changes in EEG. Itakura distance is found to be a very efficient means to characterize changes in EEG for both signaling injury and predicting recovery. The efficiency of the Itakura distance measure is further established through a comparison study of spectral distance measure and Kullback-Leibler information.
{"title":"Quantification of injury-related EEG signal changes using Itakura distance measure","authors":"X. Kong, V. Goel, N. Thakor","doi":"10.1109/ICASSP.1995.479463","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.479463","url":null,"abstract":"Accurate detection and characterization of changes in the EEG signal is crucial for clinical assessment of the neurological system condition. Several distance measures are tested and evaluated for their effectiveness of detecting injury-related changes in EEG. Itakura distance is found to be a very efficient means to characterize changes in EEG for both signaling injury and predicting recovery. The efficiency of the Itakura distance measure is further established through a comparison study of spectral distance measure and Kullback-Leibler information.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"78 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124947826","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.480128
D. Korompis, Arthur Wang, K. Yao
We perform simulation of various digital signal processing microphone array architectures to compare their suitability for a hearing aid pre-processor. The architectures include fixed narrowband array, general sidelobe canceler array and maximum energy array. The arrays are all equi-spaced linear array. In particular, arrays with 6 microphones and various number of taps have been simulated under typical hearing aid environment of reverberance, short speaker distance and competing speakers. The sampling frequency is chosen to be 10 kHz, with the physical length of the array being 17 cm. The improvements of various array designs are demonstrated and discussed.
{"title":"Comparison of microphone array designs for hearing aid","authors":"D. Korompis, Arthur Wang, K. Yao","doi":"10.1109/ICASSP.1995.480128","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480128","url":null,"abstract":"We perform simulation of various digital signal processing microphone array architectures to compare their suitability for a hearing aid pre-processor. The architectures include fixed narrowband array, general sidelobe canceler array and maximum energy array. The arrays are all equi-spaced linear array. In particular, arrays with 6 microphones and various number of taps have been simulated under typical hearing aid environment of reverberance, short speaker distance and competing speakers. The sampling frequency is chosen to be 10 kHz, with the physical length of the array being 17 cm. The improvements of various array designs are demonstrated and discussed.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"27 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125137993","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.478481
L. Frenkel, M. Feder
We investigate the application of EM algorithm to the classical problem of multiple target tracking (MTT) for a known number of targets. Conventional algorithms, have a computational complexity that depends exponentially on the targets' number, and usually divide the problem into a localization stage and a tracking stage. The new algorithms achieve a linear dependency, and integrate those hire stages. Three major optimization criteria are proposed, using deterministic and stochastic dynamic models for the targets.
{"title":"Recursive estimate-maximize (EM) algorithms for time varying parameters with applications to multiple target tracking","authors":"L. Frenkel, M. Feder","doi":"10.1109/ICASSP.1995.478481","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.478481","url":null,"abstract":"We investigate the application of EM algorithm to the classical problem of multiple target tracking (MTT) for a known number of targets. Conventional algorithms, have a computational complexity that depends exponentially on the targets' number, and usually divide the problem into a localization stage and a tracking stage. The new algorithms achieve a linear dependency, and integrate those hire stages. Three major optimization criteria are proposed, using deterministic and stochastic dynamic models for the targets.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"38 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125820588","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.480416
M. Honig
Minimum mean squared error (MMSE) detection has been proposed for direct sequence-code division multiple access (DS-CDMA) systems. The MMSE detectors are near-far resistant, and can be adapted with standard adaptive algorithms without knowledge of user parameters (i.e., spreading codes). These algorithms rely on a known training sequence for initial adaptation, and subsequently switch to a decision-directed mode. After the switch, the performance of the adaptive algorithm may degrade substantially if a strong interferer suddenly appears (i.e., if power control is relaxed). We present a "rescue" algorithm that monitors for sudden changes in the signal space, which may be caused by the appearance of a strong interferer. If a new interferer is detected, decision-directed adaptation is suspended, and an estimate of the optimal filter coefficients is obtained without a training sequence. It is shown that in the presence of low-level background noise, a good estimate can be obtained within a few symbol intervals. A numerical example is given which illustrates the performance of the rescue algorithm in a synchronous DS-CDMA system.
{"title":"Rapid detection and suppression of multi-user interference in DS-CDMA","authors":"M. Honig","doi":"10.1109/ICASSP.1995.480416","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480416","url":null,"abstract":"Minimum mean squared error (MMSE) detection has been proposed for direct sequence-code division multiple access (DS-CDMA) systems. The MMSE detectors are near-far resistant, and can be adapted with standard adaptive algorithms without knowledge of user parameters (i.e., spreading codes). These algorithms rely on a known training sequence for initial adaptation, and subsequently switch to a decision-directed mode. After the switch, the performance of the adaptive algorithm may degrade substantially if a strong interferer suddenly appears (i.e., if power control is relaxed). We present a \"rescue\" algorithm that monitors for sudden changes in the signal space, which may be caused by the appearance of a strong interferer. If a new interferer is detected, decision-directed adaptation is suspended, and an estimate of the optimal filter coefficients is obtained without a training sequence. It is shown that in the presence of low-level background noise, a good estimate can be obtained within a few symbol intervals. A numerical example is given which illustrates the performance of the rescue algorithm in a synchronous DS-CDMA system.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"24 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125921354","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.480414
Tracie A. Schirtzinger, Xiaohui Li, W. Jenkins
Three constant modulus algorithms (CMA), the fast quasi-Newton CMA, the transform domain CMA, and the genetic search based CMA are proposed in this paper. The performances of these three algorithms are compared with each other via computer simulation. It is shown that the fast quasi-Newton CMA and the transform domain CMA achieve much faster convergence rate than the constant modulus algorithm based on the LMS algorithm. This fact shows that the whitening technique is not only useful but also necessary for the CMA.
{"title":"A comparison of three algorithms for blind equalization based on the constant modulus error criterion","authors":"Tracie A. Schirtzinger, Xiaohui Li, W. Jenkins","doi":"10.1109/ICASSP.1995.480414","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480414","url":null,"abstract":"Three constant modulus algorithms (CMA), the fast quasi-Newton CMA, the transform domain CMA, and the genetic search based CMA are proposed in this paper. The performances of these three algorithms are compared with each other via computer simulation. It is shown that the fast quasi-Newton CMA and the transform domain CMA achieve much faster convergence rate than the constant modulus algorithm based on the LMS algorithm. This fact shows that the whitening technique is not only useful but also necessary for the CMA.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"45 2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123253598","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 : 1995-05-09DOI: 10.1109/ICASSP.1995.480473
T. Abatzoglou, A. Jaffer
A design of 2-D complex FIR filters is proposed by minimizing the p/sup th/ power norm used to measure the deviation of the FIR filter response from a desired filter response. The solution of this problem cannot be obtained in closed form except for p=2; for arbitrary p>2 the authors present an approach which treats the problem from a complex variable point of view. An iterative scheme is described based on the complex Newton method to find the solution. It has the feature that, starting with p=2, the value of p is increased after each iteration. Because the objective function is convex any local extremum is the global minimum. Convergence can be attained after a moderate number of iterations. A characterization theorem for factorization of 2-D FIR filters in terms of 1D filters is derived. This has strong implications for large order 2-D filter design. Two filter design examples are included.
{"title":"Least p/sup th/ power design of complex FIR 2-D filters using the complex Newton method","authors":"T. Abatzoglou, A. Jaffer","doi":"10.1109/ICASSP.1995.480473","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480473","url":null,"abstract":"A design of 2-D complex FIR filters is proposed by minimizing the p/sup th/ power norm used to measure the deviation of the FIR filter response from a desired filter response. The solution of this problem cannot be obtained in closed form except for p=2; for arbitrary p>2 the authors present an approach which treats the problem from a complex variable point of view. An iterative scheme is described based on the complex Newton method to find the solution. It has the feature that, starting with p=2, the value of p is increased after each iteration. Because the objective function is convex any local extremum is the global minimum. Convergence can be attained after a moderate number of iterations. A characterization theorem for factorization of 2-D FIR filters in terms of 1D filters is derived. This has strong implications for large order 2-D filter design. Two filter design examples are included.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"52 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1995-05-09","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123324904","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：