Pub Date : 2000-06-05DOI: 10.1109/ICASSP.2000.861962
L. Ying, D. Munson
We consider approximation of the optimal Yen algorithm (1956) for interpolation from a nonuniformly-spaced grid. Although the Yen interpolator is optimal in many senses, it suffers from severe numerical ill conditioning. We suggest a tradeoff between accuracy in computing the interpolator and accuracy in performing the interpolation. A new interpolator is proposed using Choi's expression (1998) for interpolation error. A strategy is suggested to control the error tradeoff. We also generalize the new interpolator to multiple dimensions. The newly designed sinc-kernel interpolator is compared with the Yen, Choi, and usual sinc interpolator with Jacobian weighting using simulations in both one and two dimensions. We show that the new interpolator is robust. It performs similarly to the Yen algorithm when noise is small and similarly to the Choi algorithm when noise is large.
{"title":"Approximation of the minmax interpolator","authors":"L. Ying, D. Munson","doi":"10.1109/ICASSP.2000.861962","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.861962","url":null,"abstract":"We consider approximation of the optimal Yen algorithm (1956) for interpolation from a nonuniformly-spaced grid. Although the Yen interpolator is optimal in many senses, it suffers from severe numerical ill conditioning. We suggest a tradeoff between accuracy in computing the interpolator and accuracy in performing the interpolation. A new interpolator is proposed using Choi's expression (1998) for interpolation error. A strategy is suggested to control the error tradeoff. We also generalize the new interpolator to multiple dimensions. The newly designed sinc-kernel interpolator is compared with the Yen, Choi, and usual sinc interpolator with Jacobian weighting using simulations in both one and two dimensions. We show that the new interpolator is robust. It performs similarly to the Yen algorithm when noise is small and similarly to the Choi algorithm when noise is large.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"219 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114682659","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.859066
R. R. Ghauieb, Y. Horita, T. Murai
It is known that sinusoids generate lines in their spectra but false lines may appear due to additive colored noise. Employing the fourth-order cumulant of the noisy sinusoids for retrieving the sinusoids has become an approach to handling Gaussian noise either white or colored. But the assumption that the noise is Gaussian does not exist in some applications. This paper presents a new investigation of employing cumulants for retrieving sinusoids in colored non-Gaussian noise. It is concerned with estimating the frequencies and spectrum of the sinusoids and no attention is given to the amplitudes. It is shown theoretically and experimentally that employing cumulants is an attractive approach to handling colored Rayleigh noise. Results of a presented cumulant-based forward-backward linear prediction (CBFBLP) approach are compared with that of a correlation-based counterpart.
{"title":"Retrieving sinusoids in colored Rayleigh noise by a cumulant-based FBLP approach","authors":"R. R. Ghauieb, Y. Horita, T. Murai","doi":"10.1109/ICASSP.2000.859066","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.859066","url":null,"abstract":"It is known that sinusoids generate lines in their spectra but false lines may appear due to additive colored noise. Employing the fourth-order cumulant of the noisy sinusoids for retrieving the sinusoids has become an approach to handling Gaussian noise either white or colored. But the assumption that the noise is Gaussian does not exist in some applications. This paper presents a new investigation of employing cumulants for retrieving sinusoids in colored non-Gaussian noise. It is concerned with estimating the frequencies and spectrum of the sinusoids and no attention is given to the amplitudes. It is shown theoretically and experimentally that employing cumulants is an attractive approach to handling colored Rayleigh noise. Results of a presented cumulant-based forward-backward linear prediction (CBFBLP) approach are compared with that of a correlation-based counterpart.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"12 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114747627","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.860151
S. Abid, F. Fnaiech, M. Najim
This paper presents a theoretical approach for the evaluation of a feedforward neural network covariance output error matrix. It is shown how the input signals errors and the different weights errors are linked together and spread over the neural network to form the output covariance matrix error which could may be used to determine an error bound. The formulas of the output covariance matrix error is derived arising the sensitivity of the additive weight perturbations or input perturbations. The analytical formulas is validated via simulation of a function approximation example showing that the theoretical result is in agreement with simulation result.
{"title":"Evaluation of the feedforward neural network covariance matrix error","authors":"S. Abid, F. Fnaiech, M. Najim","doi":"10.1109/ICASSP.2000.860151","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.860151","url":null,"abstract":"This paper presents a theoretical approach for the evaluation of a feedforward neural network covariance output error matrix. It is shown how the input signals errors and the different weights errors are linked together and spread over the neural network to form the output covariance matrix error which could may be used to determine an error bound. The formulas of the output covariance matrix error is derived arising the sensitivity of the additive weight perturbations or input perturbations. The analytical formulas is validated via simulation of a function approximation example showing that the theoretical result is in agreement with simulation result.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"44 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114849798","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.861168
B. Raghothaman, R. T. Derryberry, G. Mandyam
The transmit adaptive array (TxAA) is one of the promising closed-loop downlink diversity schemes being considered for the third generation wireless systems based on code division multiple access (CDMA). The TxAA technique originally proposed, requires a user-specific auxiliary pilot for coherent demodulation. This affects the capacity of the system due to additional power being used by this pilot. The mobile receiver requires additional hardware correlators for demodulating the pilot. Since different channels in spread spectrum systems are distinguished by their spreading sequences, it also uses up an additional Walsh code per user. This paper proposes a decision directed method for demodulation which does not require a user-specific pilot. Our simulations demonstrate that the performance of the technique is comparable to that of the method based on a user specific pilot.
{"title":"Transmit adaptive array without user-specific pilot for 3G CDMA","authors":"B. Raghothaman, R. T. Derryberry, G. Mandyam","doi":"10.1109/ICASSP.2000.861168","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.861168","url":null,"abstract":"The transmit adaptive array (TxAA) is one of the promising closed-loop downlink diversity schemes being considered for the third generation wireless systems based on code division multiple access (CDMA). The TxAA technique originally proposed, requires a user-specific auxiliary pilot for coherent demodulation. This affects the capacity of the system due to additional power being used by this pilot. The mobile receiver requires additional hardware correlators for demodulating the pilot. Since different channels in spread spectrum systems are distinguished by their spreading sequences, it also uses up an additional Walsh code per user. This paper proposes a decision directed method for demodulation which does not require a user-specific pilot. Our simulations demonstrate that the performance of the technique is comparable to that of the method based on a user specific pilot.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"349 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124302812","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.859290
A. Fahmy, Bassel S. Tawfik, Y. Kadah
In magnetic resonance imaging (MRI), data are collected as spectrum samples. The acquisition time is proportional to the number of the spectrum lines. Therefore, only few lines of the data space may be required in order to track rapid changes of an object. In the current techniques, the missed lines may be zeroed or replaced by the corresponding lines in a reference image, which is acquired a priori for the same anatomical cross-section. However, this always comes at the expense of the spatial-resolution. In this study, we propose an extrapolation iterative algorithm to provide an improved estimate of the missed lines. Additional spatial and spatial-frequency constraints of the reference image are incorporated to enhance the convergence and obtain a better estimate of the initial conditions of the iterations. Results from simulated data verify the theory and indicate that the algorithm may provide better reconstruction in dynamic imaging studies.
{"title":"A local-field extrapolation algorithm for improving the spatial resolution in magnetic resonance dynamic imaging","authors":"A. Fahmy, Bassel S. Tawfik, Y. Kadah","doi":"10.1109/ICASSP.2000.859290","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.859290","url":null,"abstract":"In magnetic resonance imaging (MRI), data are collected as spectrum samples. The acquisition time is proportional to the number of the spectrum lines. Therefore, only few lines of the data space may be required in order to track rapid changes of an object. In the current techniques, the missed lines may be zeroed or replaced by the corresponding lines in a reference image, which is acquired a priori for the same anatomical cross-section. However, this always comes at the expense of the spatial-resolution. In this study, we propose an extrapolation iterative algorithm to provide an improved estimate of the missed lines. Additional spatial and spatial-frequency constraints of the reference image are incorporated to enhance the convergence and obtain a better estimate of the initial conditions of the iterations. Results from simulated data verify the theory and indicate that the algorithm may provide better reconstruction in dynamic imaging studies.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"13 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124189436","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.861955
J. A. Gubner, L. Scharf
The natural models of multi-access communication and modern radar and sonar systems involve infinite-dimensional waveform spaces. A common problem in these systems is the detection of subspace signals measured in the presence of subspace interference and broadband noise. By and large, the existing theory for such problems has been developed for finite-dimensional measurement spaces rather than the infinite-dimensional waveform spaces needed here. In this paper the log of the generalized likelihood ratio detector for the waveform problem is derived and is shown to have a certain chi-squared distribution, depending on the hypothesis.
{"title":"Detection of subspace waveforms in subspace interference and noise","authors":"J. A. Gubner, L. Scharf","doi":"10.1109/ICASSP.2000.861955","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.861955","url":null,"abstract":"The natural models of multi-access communication and modern radar and sonar systems involve infinite-dimensional waveform spaces. A common problem in these systems is the detection of subspace signals measured in the presence of subspace interference and broadband noise. By and large, the existing theory for such problems has been developed for finite-dimensional measurement spaces rather than the infinite-dimensional waveform spaces needed here. In this paper the log of the generalized likelihood ratio detector for the waveform problem is derived and is shown to have a certain chi-squared distribution, depending on the hypothesis.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"25 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127755920","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.859149
P. Sivakumaran, A. Ariyaeeinia
This paper focuses on the spectral representation of the sub-band cepstrum in relation to that of the full-band cepstrum. Through theoretical analysis it is shown that the net spectral information content of the cepstral coefficients with the same index in different sub-bands is only comparable to that of a full-band cepstral parameter whose quefrency is given by the product of that specific index with the number of sub-bands. A new method is proposed to tackle this deficiency of the sub-band cepstrum when it is used in the context of text-dependent speaker verification. The experimental investigations have clearly demonstrated the effectiveness of this method in speaker verification.
{"title":"The use of sub-band cepstrum in speaker verification","authors":"P. Sivakumaran, A. Ariyaeeinia","doi":"10.1109/ICASSP.2000.859149","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.859149","url":null,"abstract":"This paper focuses on the spectral representation of the sub-band cepstrum in relation to that of the full-band cepstrum. Through theoretical analysis it is shown that the net spectral information content of the cepstral coefficients with the same index in different sub-bands is only comparable to that of a full-band cepstral parameter whose quefrency is given by the product of that specific index with the number of sub-bands. A new method is proposed to tackle this deficiency of the sub-band cepstrum when it is used in the context of text-dependent speaker verification. The experimental investigations have clearly demonstrated the effectiveness of this method in speaker verification.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127775454","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.859205
Wael Badawy, M. Bayoumi
This paper presents an efficient mesh coding technique suitable for MPEG-4 video applications. The proposed technique significantly reduces the number of bits that are used to describe the mesh topology. It uses an adaptive structured mesh from coarse to fine, which can be coded as a count of splitting instead of nodes' locations. In the case of the quadtree, less than one bit per node can be achieved. This reduction induces an improvement of either the image quality or the global bit rate.
{"title":"On minimizing hierarchical mesh coding overhead: (HASM) hierarchical adaptive structured mesh approach","authors":"Wael Badawy, M. Bayoumi","doi":"10.1109/ICASSP.2000.859205","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.859205","url":null,"abstract":"This paper presents an efficient mesh coding technique suitable for MPEG-4 video applications. The proposed technique significantly reduces the number of bits that are used to describe the mesh topology. It uses an adaptive structured mesh from coarse to fine, which can be coded as a count of splitting instead of nodes' locations. In the case of the quadtree, less than one bit per node can be achieved. This reduction induces an improvement of either the image quality or the global bit rate.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"85 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126302557","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.860125
L. González, E. Boutillon
A VLSI structure for path metric updating in the M algorithm is presented. The architecture is based on the combination of a modified Batcher's (1968) odd-even merging network and a bitonic selection procedure. A feature of the trellis structure allows to replace an existing solution based on two 2M-item sorting operations by three M-item sorting operations with an additional one-layer bitonic merge. These three sorting networks and the bitonic merging procedure permit a reduction of up to 50% in hardware complexity.
{"title":"Simplified path metric updating in the M algorithm for VLSI implementation","authors":"L. González, E. Boutillon","doi":"10.1109/ICASSP.2000.860125","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.860125","url":null,"abstract":"A VLSI structure for path metric updating in the M algorithm is presented. The architecture is based on the combination of a modified Batcher's (1968) odd-even merging network and a bitonic selection procedure. A feature of the trellis structure allows to replace an existing solution based on two 2M-item sorting operations by three M-item sorting operations with an additional one-layer bitonic merge. These three sorting networks and the bitonic merging procedure permit a reduction of up to 50% in hardware complexity.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126389971","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 : 2000-06-05DOI: 10.1109/ICASSP.2000.859323
J. J. Williams, A. Katsaggelos, M. Randolph
This paper describes a hidden Markov model (HMM) based visual synthesizer designed to assist persons with impaired hearing. This synthesizer builds on results in the area of audio-visual speech recognition. We describe how a correlation HMM can be used to integrate independent acoustic and visual HMMs for speech-to-visual synthesis. Our results show that an HMM correlating model can significantly improve synchronization errors versus techniques which compensate for rate differences through scaling.
{"title":"A hidden Markov model based visual speech synthesizer","authors":"J. J. Williams, A. Katsaggelos, M. Randolph","doi":"10.1109/ICASSP.2000.859323","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.859323","url":null,"abstract":"This paper describes a hidden Markov model (HMM) based visual synthesizer designed to assist persons with impaired hearing. This synthesizer builds on results in the area of audio-visual speech recognition. We describe how a correlation HMM can be used to integrate independent acoustic and visual HMMs for speech-to-visual synthesis. Our results show that an HMM correlating model can significantly improve synchronization errors versus techniques which compensate for rate differences through scaling.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"27 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126412336","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}
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