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1993 IEEE International Conference on Acoustics, Speech, and Signal Processing最新文献

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Concatenated phoneme models for text-variable speaker recognition 文本变量说话人识别的连接音素模型
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319321
Tomoko Matsui, S. Furui
{"title":"Concatenated phoneme models for text-variable speaker recognition","authors":"Tomoko Matsui, S. Furui","doi":"10.1109/ICASSP.1993.319321","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319321","url":null,"abstract":"Methods that create models to specify both speaker and phonetic information accurately by using only a small amount of training data for each speaker are investigated. For a text-dependent speaker recognition method, in which arbitrary key texts are prompted from the recognizer, speaker-specific phoneme models are necessary to identify the key text and recognize the speaker. Two methods of making speaker-specific phoneme models are discussed: phoneme-adaptation of a phoneme-independent speaker model and speaker-adaptation of universal phoneme models. The authors also investigate supplementing these methods by adding a phoneme-independent speaker model to make up for the lack of speaker information. This combination achieves a rejection rate as high as 98.5% for speech that differs from the key text and a speaker verification rate of 100.0%.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115001836","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 134
A fast method for combining palettes of color quantized images 一种快速组合颜色量化图像调色板的方法
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319811
V. Iverson, E. Riskin
{"title":"A fast method for combining palettes of color quantized images","authors":"V. Iverson, E. Riskin","doi":"10.1109/ICASSP.1993.319811","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319811","url":null,"abstract":"The authors consider the use of a vector quantization algorithm-the pairwise nearest neighbor algorithm-as a means of quickly combining the palettes of multiple images into a single shared palette that can be used for simultaneous display. This approach yields an efficient and practical method of managing palettes that adds little distortion to the images being displayed. The implementation and results of this work are described.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115383279","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 10
Supervised and unsupervised clustering of the speaker space for connectionist speech recognition 连接主义语音识别中说话人空间的监督和无监督聚类
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319176
Y. Konig, N. Morgan
{"title":"Supervised and unsupervised clustering of the speaker space for connectionist speech recognition","authors":"Y. Konig, N. Morgan","doi":"10.1109/ICASSP.1993.319176","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319176","url":null,"abstract":"One of the challenging problems of a speaker-independent continuous speech recognition system is how to achieve good performance with a new speaker, when the only available source of information about the new speaker is the utterance to be recognized. The authors propose a first step toward a solution, based on clustering of the speaker space. The study had two steps. The first was searching for a set of features to cluster speakers. Second, using the chosen features, two kinds of clustering were investigated: supervised-using two clusters, males and females-and unsupervised-using two, three, and five clusters. The cluster information was integrated into the connectionist speech recognition system by using the speaker cluster neural network (SCNN). The SCNN attempts to share the speaker-independent parameters and to model the cluster-dependent parameters. The results show that the best performance is achieved with the supervised clusters, resulting in an overall improvement in recognition performance.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"1 2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115716406","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 6
Adaptive signal processing techniques for chaotic systems 混沌系统的自适应信号处理技术
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319560
Fawad Rauf, H. Ahmed
{"title":"Adaptive signal processing techniques for chaotic systems","authors":"Fawad Rauf, H. Ahmed","doi":"10.1109/ICASSP.1993.319560","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319560","url":null,"abstract":"The issue of modeling chaotic systems is addressed. Present methods for treating chaotic dynamics are based on state space reconstruction through delay embedding. These approaches are computationally intensive and are adversely affected by noise in the experimental time series. The authors take a different approach and apply an adaptive layered structure for estimation of chaotic dynamics. They show that presently used spatial local approximations are not necessary and that their temporal adaptive local approximations perform better, are tolerant to noise factors, and save an order of magnitude in computations, and data requirements.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124404656","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 4
Simplified Kalman estimation of fading mobile radio channels: high performance at LMS computational load 衰落移动无线电信道的简化卡尔曼估计:在LMS计算负荷下的高性能
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319507
L. Lindbom
{"title":"Simplified Kalman estimation of fading mobile radio channels: high performance at LMS computational load","authors":"L. Lindbom","doi":"10.1109/ICASSP.1993.319507","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319507","url":null,"abstract":"Low-complexity algorithms for channel estimation in Rayleigh fading environments are presented. The channel estimators are presumed to operate in conjunction with a Viterbi decoder, or an equalizer. The algorithms are based on simplified internal modeling of time-invariant channel coefficients and approximation of a Kalman estimator. A novel averaging approach is used to replace the online update of the Riccati equation with a constant matrix. The associated Kalman gain is expressed in an analytical form. Compared with RLS (recursive least squares) tracking, both a significantly lower bit error rate and a much lower computational complexity are achieved.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116796869","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 68
A practical stopping rule for iterative signal restoration 一种实用的迭代信号恢复停止规则
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319529
Kevin M. Perry, S. Reeves
{"title":"A practical stopping rule for iterative signal restoration","authors":"Kevin M. Perry, S. Reeves","doi":"10.1109/ICASSP.1993.319529","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319529","url":null,"abstract":"The randomized generalized cross-validation (RGCV) criterion is developed as an efficient stopping rule for iterative signal restoration, and it is shown that it can be used on relatively small data sets with a large degree of confidence. Various experiments demonstrate the ability of the RGCV stopping rule to perform well for a large range of blur size ratios and noise levels, and for smaller signal lengths. The RGCV stopping rule has several advantages over previous methods. It is computationally efficient and easy to implement, taking advantage of quantities already computed in the restoration algorithm. It does not require any knowledge of the noise variance. It provides a restoration closer to the ideal restoration than the residual method.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116886284","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 16
Shape calibration for a nominally linear equispaced array 标称线性阵列的形状校正
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319654
J. Fuchs
{"title":"Shape calibration for a nominally linear equispaced array","authors":"J. Fuchs","doi":"10.1109/ICASSP.1993.319654","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319654","url":null,"abstract":"The author considers a thin flexible line array of equispaced hydrophones that is towed through the sea, and develops a procedure that allows testing of the straightness of the array. The motion of the towing ship, the currents of the ocean and other forces induce deformations on the array and affect the performance of spatial processing of the data developed under the assumption that the array is straight. When the ship is maneuvering, the processing is generally turned off for long periods of time, an extremely penalizing situation that can be overcome by applying the procedure described to determine the maximum size of admissible sub-arrays on which the standard processing can be pursued.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"60 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116970142","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 10
A comparison of trajectory and mixture modeling in segment-based word recognition 基于分段词识别的轨迹建模与混合建模的比较
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319303
Ashvin Kannan, Mari Ostendorf
{"title":"A comparison of trajectory and mixture modeling in segment-based word recognition","authors":"Ashvin Kannan, Mari Ostendorf","doi":"10.1109/ICASSP.1993.319303","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319303","url":null,"abstract":"A mechanism for implementing mixtures at a phone-subsegment (microsegment) level for continuous word recognition based on the stochastic segment model (SMM) is presented. The issues that are involved in tradeoffs between the trajectory and mixture modeling in segment-based word recognition are investigated. Experimental results are reported on DAPRA's speaker-independent Resource management corpus. The results obtained suggest that there is a tradeoff in using mixture models and trajectory models, associated with the level of detail of the modeling unit. The results support the use of whole segment models in the context-dependent case, and microsegment-level (and possibly segment-level) mixtures rather than frame-level mixtures.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"49 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117149320","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 19
Estimating ocean-scene coherence-time with a multi-aperture SAR 用多孔径SAR估计海洋场景相干时间
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319847
M. F. Griffin, E. Boman, I. Metal, L. Orwig
{"title":"Estimating ocean-scene coherence-time with a multi-aperture SAR","authors":"M. F. Griffin, E. Boman, I. Metal, L. Orwig","doi":"10.1109/ICASSP.1993.319847","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319847","url":null,"abstract":"It is shown how to estimate decorrelation time using a multiple aperture SAR (synthetic aperture radar) and an AR (autoregressive) model. The experimental order-p autoregressive or AR (p) results presented here for averaged tau /sub D/ (characteristic decorrelation time) suggest that the ocean reflectivity may be modeled as an AR process. The experimental results show decorrelation times slightly larger than those inferred from the SAXON data, which is most likely due to the imposition of wide sense stationarity.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"26 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117186041","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 4
Microphone array speech enhancement in overdetermined signal scenarios 超定信号场景下的麦克风阵列语音增强
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319309
Raymond E. Slyh, R. Moses
{"title":"Microphone array speech enhancement in overdetermined signal scenarios","authors":"Raymond E. Slyh, R. Moses","doi":"10.1109/ICASSP.1993.319309","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319309","url":null,"abstract":"The problem of enhancing noisy speech by using a microphone array is considered. An algorithm called the graphic equalizer (GEQ) array is presented which trades off signal degradation for additional interference suppression. The algorithm is based on the concept of directly modifying the short-time spectral magnitude of the sum of the received signals. Simulation results are given that illustrate the advantages of using the GEQ array for diffuse-noise scenarios and for scenarios involving more interference signals than array degrees of freedom. The GEQ array was shown to outperform the Frost array for diffuse-noise and overdetermined signal scenarios, in which the Frost array was not capable of attenuating all of the interference with one broad null. While the GEQ array did not always yield a better SNR than the Frost array for these scenarios, it often outperformed the Frost array in terms of the power function spectral distance measure, a measure that is more highly correlated with human auditory perception than is the SNR.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"24 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125002917","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 4
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