Pub Date : 1995-05-09DOI: 10.1109/ICASSP.1995.479748
David V. Anderson, Richard W. Harris, D. M. Chabries
A new hearing compensation algorithm based on a homomorphic multiplicative AGC (automatic gain control) is evaluated and compared against commercially available digitally programmable analog hearing aids. Both quantitative (speech recognition threshold and speech discrimination) and qualitative tests (estimation of perceived quality) were used in the evaluation. The new algorithm is shown to have made significant progress in restoring normal or near normal hearing for hearing impaired individuals.
{"title":"Evaluation of a hearing compensation algorithm","authors":"David V. Anderson, Richard W. Harris, D. M. Chabries","doi":"10.1109/ICASSP.1995.479748","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.479748","url":null,"abstract":"A new hearing compensation algorithm based on a homomorphic multiplicative AGC (automatic gain control) is evaluated and compared against commercially available digitally programmable analog hearing aids. Both quantitative (speech recognition threshold and speech discrimination) and qualitative tests (estimation of perceived quality) were used in the evaluation. The new algorithm is shown to have made significant progress in restoring normal or near normal hearing for hearing impaired individuals.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"19 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":"127768504","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.1995.480062
S. Barbarossa, L. Parodi
The authors propose a method for optimizing the decomposition law of a tree-structured wavelet transform in order to maximize the capability of discriminating different textures. The optimization criterion is the maximization of the Fisher's distance. The analysis is carried out theoretically and by simulation on Gaussian Markov random fields and is then applied to the classification of real synthetic aperture radar images.
{"title":"Tree-structured wavelet decomposition based on the maximization of Fisher's distance","authors":"S. Barbarossa, L. Parodi","doi":"10.1109/ICASSP.1995.480062","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480062","url":null,"abstract":"The authors propose a method for optimizing the decomposition law of a tree-structured wavelet transform in order to maximize the capability of discriminating different textures. The optimization criterion is the maximization of the Fisher's distance. The analysis is carried out theoretically and by simulation on Gaussian Markov random fields and is then applied to the classification of real synthetic aperture radar images.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"58 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":"126210368","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.1995.480503
O. Arikan, Murat Belge, A. Çetin, E. Erzin
A large class of physical phenomenon observed in practice exhibit non-Gaussian behavior. In this paper, /spl alpha/-stable distributions, which have heavier tails than Gaussian distribution, are considered to model non-Gaussian signals. Adaptive signal processing in the presence of such kind of noise is a requirement of many practical problems. Since, direct application of commonly used adaptation techniques fail in these applications, new approaches for adaptive filtering for /spl alpha/-stable random processes are introduced.
{"title":"Adaptive filtering approaches for non-Gaussian stable processes","authors":"O. Arikan, Murat Belge, A. Çetin, E. Erzin","doi":"10.1109/ICASSP.1995.480503","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480503","url":null,"abstract":"A large class of physical phenomenon observed in practice exhibit non-Gaussian behavior. In this paper, /spl alpha/-stable distributions, which have heavier tails than Gaussian distribution, are considered to model non-Gaussian signals. Adaptive signal processing in the presence of such kind of noise is a requirement of many practical problems. Since, direct application of commonly used adaptation techniques fail in these applications, new approaches for adaptive filtering for /spl alpha/-stable random processes are introduced.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"62 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":"126221533","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.1995.479407
M. Saerens
When using hidden Markov models for speech recognition, it is usually assumed that the probability that a particular acoustic vector is emitted at a given time only depends on the current state and the current acoustic vector observed. We introduce another idea, i.e., we assume that, in a given state, the acoustic vectors are generated by a linear stochastic differential equation. This work is motivated by the fact that the time evolution of the acoustic vector is inherently dynamic and continuous. So that the modelling could be performed in the continuous-time domain instead of the discrete-time domain. By the way, the links between the discrete-time model obtained after sampling, and the original continuous-time signal are not so trivial. In particular, the relationship between the coefficients of a continuous-time linear process and the coefficients of the discrete-time linear process obtained after sampling is nonlinear. We assign a probability density to the continuous-time trajectory of the acoustic vector inside the state, reflecting the probability that this particular path has been generated by the stochastic differential equation associated with this state. This allows us to compute the likelihood of the uttered word. Reestimation formulae for the parameters of the process, based on the maximization of the likelihood, can be derived for the Viterbi algorithm. As usual, the segmentation can be obtained by sampling the continuous process, and by applying dynamic programming to find the best path over all the possible sequences of states.
{"title":"Viterbi algorithm for acoustic vectors generated by a linear stochastic differential equation on each state","authors":"M. Saerens","doi":"10.1109/ICASSP.1995.479407","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.479407","url":null,"abstract":"When using hidden Markov models for speech recognition, it is usually assumed that the probability that a particular acoustic vector is emitted at a given time only depends on the current state and the current acoustic vector observed. We introduce another idea, i.e., we assume that, in a given state, the acoustic vectors are generated by a linear stochastic differential equation. This work is motivated by the fact that the time evolution of the acoustic vector is inherently dynamic and continuous. So that the modelling could be performed in the continuous-time domain instead of the discrete-time domain. By the way, the links between the discrete-time model obtained after sampling, and the original continuous-time signal are not so trivial. In particular, the relationship between the coefficients of a continuous-time linear process and the coefficients of the discrete-time linear process obtained after sampling is nonlinear. We assign a probability density to the continuous-time trajectory of the acoustic vector inside the state, reflecting the probability that this particular path has been generated by the stochastic differential equation associated with this state. This allows us to compute the likelihood of the uttered word. Reestimation formulae for the parameters of the process, based on the maximization of the likelihood, can be derived for the Viterbi algorithm. As usual, the segmentation can be obtained by sampling the continuous process, and by applying dynamic programming to find the best path over all the possible sequences of states.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"9 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":"128065866","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.1995.480184
Jian Zhan, Jin S. Zhang, Fu Li, Z. Fan
Adaptive equalization can be used to improve digital data transmission on wireless links with time-varying multipath distortion. It was proved by Zhou, Proakis and Ling (see IEEE Transactions on Communications, vol.38, p.8-24, no.1, 1990) that feedforward schemes are universally capable of approximating any measurable function to any desired degree of accuracy. We propose a new realization of such feedforward scheme to the channel equalization problem, parallel feedforward equalization (PFE). An important feature of the new approach is the decomposition of any equalization into linear and nonlinear components. The new approach chooses F/sub j/(/spl middot/) (j=1, ...) from a family of nonlinear functions to approximate the nonlinear component decomposed from the desired mapping f(/spl middot/). The other new idea proposed in this paper is a measure called nonlinearity distribution which characterizes the nonlinearity in multipath fading channels. The architecture of the new equalization consists of parallel feedforward nonlinear filters, each of them has a specifically tailored nonlinear function F/sub j/(/spl middot/).
{"title":"Parallel feedforward equalization-a new nonlinear adaptive algorithm","authors":"Jian Zhan, Jin S. Zhang, Fu Li, Z. Fan","doi":"10.1109/ICASSP.1995.480184","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480184","url":null,"abstract":"Adaptive equalization can be used to improve digital data transmission on wireless links with time-varying multipath distortion. It was proved by Zhou, Proakis and Ling (see IEEE Transactions on Communications, vol.38, p.8-24, no.1, 1990) that feedforward schemes are universally capable of approximating any measurable function to any desired degree of accuracy. We propose a new realization of such feedforward scheme to the channel equalization problem, parallel feedforward equalization (PFE). An important feature of the new approach is the decomposition of any equalization into linear and nonlinear components. The new approach chooses F/sub j/(/spl middot/) (j=1, ...) from a family of nonlinear functions to approximate the nonlinear component decomposed from the desired mapping f(/spl middot/). The other new idea proposed in this paper is a measure called nonlinearity distribution which characterizes the nonlinearity in multipath fading channels. The architecture of the new equalization consists of parallel feedforward nonlinear filters, each of them has a specifically tailored nonlinear function F/sub j/(/spl middot/).","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"15 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":"125495442","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.1995.480066
O. Alata, P. Baylou, M. Najim
In the framework of model based image processing, we propose a new parametric approach for classifying textured images. The image, considered as a two-dimensional stochastic process, is characterized by a set of reflection coefficients computed using a two-dimensional adaptive lattice filter based on the recursive least squares (RLS) criterion. The corresponding algorithm is named the two-dimensional fast lattice RLS. In order to evaluate this method, classification rates are calculated on a set of 8 different textures from the Brodatz album. We carry out performance comparisons with methods of characterization based on two-dimensional AR coefficients computed with two-dimensional transversal filters or based on statistical features calculated from co-occurrence matrices and neighbouring matrices.
{"title":"Texture characterization based on 2-D reflection coefficients","authors":"O. Alata, P. Baylou, M. Najim","doi":"10.1109/ICASSP.1995.480066","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480066","url":null,"abstract":"In the framework of model based image processing, we propose a new parametric approach for classifying textured images. The image, considered as a two-dimensional stochastic process, is characterized by a set of reflection coefficients computed using a two-dimensional adaptive lattice filter based on the recursive least squares (RLS) criterion. The corresponding algorithm is named the two-dimensional fast lattice RLS. In order to evaluate this method, classification rates are calculated on a set of 8 different textures from the Brodatz album. We carry out performance comparisons with methods of characterization based on two-dimensional AR coefficients computed with two-dimensional transversal filters or based on statistical features calculated from co-occurrence matrices and neighbouring matrices.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"40 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":"125650786","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.1995.479792
Qiang Huo, Chorkin Chan
On-line adaptation of semi-continuous (or tied mixture) hidden Markov model (SCHMM) is studied. A theoretical formulation of the segmental quasi-Bayes learning of the mixture coefficients in SCHMM for speech recognition is presented. The practical issues related to the use of this algorithm for on-line speaker adaptation are addressed. A pragmatic on-line adaptation approach to combine the long-term adaptation of the mixture coefficients and the short-term adaptation of the mean vectors of the Gaussian mixture components are also proposed. The viability of these techniques are confirmed in a series of comparative experiments using a 26-word English alphabet vocabulary.
{"title":"On-line Bayes adaptation of SCHMM parameters for speech recognition","authors":"Qiang Huo, Chorkin Chan","doi":"10.1109/ICASSP.1995.479792","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.479792","url":null,"abstract":"On-line adaptation of semi-continuous (or tied mixture) hidden Markov model (SCHMM) is studied. A theoretical formulation of the segmental quasi-Bayes learning of the mixture coefficients in SCHMM for speech recognition is presented. The practical issues related to the use of this algorithm for on-line speaker adaptation are addressed. A pragmatic on-line adaptation approach to combine the long-term adaptation of the mixture coefficients and the short-term adaptation of the mean vectors of the Gaussian mixture components are also proposed. The viability of these techniques are confirmed in a series of comparative experiments using a 26-word English alphabet vocabulary.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"453 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":"125787215","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.1995.480411
S. Zazo, J. M. Páez-Borrallo, I. Pérez-Álvarez
Existing blind adaptive equalizers that use nonconvex cost functions (as Bussgang type algorithms) and stochastic gradient descent suffer from lack of global convergence to an equalizer tap set that removes sufficient ISI when an FIR equalizer is used. In this paper we propose a new algorithm including tap anchoring and gain recovery into the classical schemes. The combined effect of these strategies is to establish the preservation of the transmitted symbol preventing ill convergence, and therefore providing the ability of implementation of the inverse filter regardless of the initial ISI. Under certain hypotheses, we suggest that a globally convex scheme can be proposed overcoming the existing structures. Several computer simulations support our theoretical results.
{"title":"A linearly constrained blind equalization scheme based on Bussgang type algorithms","authors":"S. Zazo, J. M. Páez-Borrallo, I. Pérez-Álvarez","doi":"10.1109/ICASSP.1995.480411","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480411","url":null,"abstract":"Existing blind adaptive equalizers that use nonconvex cost functions (as Bussgang type algorithms) and stochastic gradient descent suffer from lack of global convergence to an equalizer tap set that removes sufficient ISI when an FIR equalizer is used. In this paper we propose a new algorithm including tap anchoring and gain recovery into the classical schemes. The combined effect of these strategies is to establish the preservation of the transmitted symbol preventing ill convergence, and therefore providing the ability of implementation of the inverse filter regardless of the initial ISI. Under certain hypotheses, we suggest that a globally convex scheme can be proposed overcoming the existing structures. Several computer simulations support our theoretical results.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"2004 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":"125794057","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.1995.480039
Hsin-Piao Lin, S. Jeng, I. Parra, Guanghan Xu, W. Vogel, G. W. Torrence
This paper presents some preliminary results of experimental studies of space-division-multiple-access (SDMA) systems for wireless communications to expand capacity, increase coverage, and improve quality. Although SDMA schemes have been studied by a number of researchers, most of these studies are based on theoretical analyses and computer simulations. Very few real RF or microwave experiments have been conducted to validate the feasibility of various signal processing algorithms, such as direction finding and signal copy techniques. Also, no extensive experiments have been conducted to study the channel propagation associated with multiple antennas. The purpose of this paper is to present our preliminary experimental results using our recently developed antenna array testbed. We will also discuss the implications of these results on various array signal processing algorithms.
{"title":"Experimental studies of SDMA schemes for wireless communications","authors":"Hsin-Piao Lin, S. Jeng, I. Parra, Guanghan Xu, W. Vogel, G. W. Torrence","doi":"10.1109/ICASSP.1995.480039","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.480039","url":null,"abstract":"This paper presents some preliminary results of experimental studies of space-division-multiple-access (SDMA) systems for wireless communications to expand capacity, increase coverage, and improve quality. Although SDMA schemes have been studied by a number of researchers, most of these studies are based on theoretical analyses and computer simulations. Very few real RF or microwave experiments have been conducted to validate the feasibility of various signal processing algorithms, such as direction finding and signal copy techniques. Also, no extensive experiments have been conducted to study the channel propagation associated with multiple antennas. The purpose of this paper is to present our preliminary experimental results using our recently developed antenna array testbed. We will also discuss the implications of these results on various array signal processing algorithms.","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":"122012044","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.1995.479551
T. Manickam, R. Vaccaro
We address the problem of multipath time-delay estimation. When the received data is very long compared to the transmitted signal, the data is expected to consist of a large number of paths. Modeling the entire data becomes computationally expensive. We propose a technique to break the data into short segments and model each segment individually without misfitting or truncating any paths at the ends of any segment. By effectively using overlapping segments, the estimates of time-delays are combined to model the entire data record. The method is extended to the case where only basebanded data are available. The proposed technique is demonstrated on an experimental sea-test data.
{"title":"Multipath time-delay estimation for long data records","authors":"T. Manickam, R. Vaccaro","doi":"10.1109/ICASSP.1995.479551","DOIUrl":"https://doi.org/10.1109/ICASSP.1995.479551","url":null,"abstract":"We address the problem of multipath time-delay estimation. When the received data is very long compared to the transmitted signal, the data is expected to consist of a large number of paths. Modeling the entire data becomes computationally expensive. We propose a technique to break the data into short segments and model each segment individually without misfitting or truncating any paths at the ends of any segment. By effectively using overlapping segments, the estimates of time-delays are combined to model the entire data record. The method is extended to the case where only basebanded data are available. The proposed technique is demonstrated on an experimental sea-test data.","PeriodicalId":300119,"journal":{"name":"1995 International Conference on Acoustics, Speech, and Signal Processing","volume":"104 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":"127969998","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：