ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing最新文献_第9页

Set-membership theory applied to linear prediction analysis of speech 集合隶属度理论在语音线性预测分析中的应用

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169574

J. Deller, T. Luk

引用次数: 10

Mixed-phase deconvolution of speech based on a sine-wave model 基于正弦波模型的语音混合相位反卷积

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169573

T. Quatieri, R. McAulay

{"title":"Mixed-phase deconvolution of speech based on a sine-wave model","authors":"T. Quatieri, R. McAulay","doi":"10.1109/ICASSP.1987.1169573","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169573","url":null,"abstract":"This paper describes a new method of deconvolving the vocal cord excitation and vocal tract system response. The technique relies on a sine-wave representation of the speech waveform and forms the basis of an analysis-synthesis method which yields synthetic speech essentially indistinguishable from the original. Unlike an earlier sinusoidal analysis-synthesis technique that used a minimum-phase system estimate, the approach in this paper generates a \"mixed-phase\" system estimate and thus an improved decomposition of excitation and system components. Since a mixed-phase system estimate is removed from the speech waveform, the resulting excitation residual is less dispersed than the previous sinusoidal-based excitation estimate or the more commonly used linear prediction residual. A method of time-varying linear filtering is given as an alternative to sinusoidal reconstruction, similar to conventional time-domain synthesis used in certain vocoders, but without the requirement of pitch and voicing decisions. Finally, speech modification with a mixed-phase system estimate is shown to be capable of more closely preserving waveform shape in time-scale and pitch transformations than the earlier approach.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"19 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":"123652862","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: 12

An efficient speaker-independent automatic speech recognition by simulation of some properties of human auditory perception 通过模拟人类听觉感知的某些特性，实现了一种高效的独立于说话人的自动语音识别

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169803

H. Hermansky

引用次数: 34

Localization of coherent sources using a modified spatial smoothing technique 利用改进的空间平滑技术定位相干源

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169420

Ronald T. Williams, S. Prasad, A. Mahalanabis, L. Sibul

引用次数: 4

A new approach to partially adaptive arrays 部分自适应阵列的一种新方法

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169367

L. Griffiths

{"title":"A new approach to partially adaptive arrays","authors":"L. Griffiths","doi":"10.1109/ICASSP.1987.1169367","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169367","url":null,"abstract":"When an adaptive array operates in the presence of white noise only, the resulting beam pattern is referred to as the quiescent response. Typically, these patterns have mainlobe and sidelobe shapes differing from those designed for use in deterministic, non-adaptive arrays. This paper describes a simple method which allows nearly arbitrary specification of the quiescent response in a linearly-constrained power minimization adaptive array. The only restriction on the quiescent is that it must meet the constraints defined for the adaptive array. Since many well-known deterministic designs such as Chebychev are not likely to meet the linear constraint conditions used in adaptive arrays for mainlobe and other pattern control functions, a procedure is presented which modifies the deterministic design to force it to meet the linear constraints in a least-squares manner. Once this has been accomplished, the methods outlined in this paper can be used to cause the modified deterministic design to become the quiescent response of the adaptive array. As a result, the adaptive array can be configured to closely resemble a deterministic array when the noise is white. Under conditions of correlated interference, or jamming, however, the response changes so as to effectively steer nulls in the appropriate directions. The method is based on the use of a generalized sidelobe canceller and requires one additional linear constraint for both narrow-band and broad-band arrays. This added flexibility in a partially adaptive array allows the system to be configured so as to meet an arbitrary number M of linear constraints either at all times (using M degrees of freedom) or only under quiescent conditions (using a single constraint). Any intermediate mixture of these extreme positions is also possible.","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":"125861956","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: 11

A segment vocoder algorithm for real-time implementation 一个分段声码器算法的实时实现

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169363

Salim Roukos, A. Wilgus, W. Russell

引用次数: 10

A minimum discrimination information approach for hidden Markov modeling 隐马尔可夫建模的最小判别信息方法

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169727

Y. Ephraim, A. Dembo, L. Rabiner

引用次数: 110

Comparative performance of ESPRIT and MUSIC for direction-of-arrival estimation ESPRIT和MUSIC在到达方向估计中的比较性能

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169322

R. Roy, A. Paulraj, T. Kailath

引用次数: 40

Time delay estimation by autoregressive modelization 基于自回归模型的时延估计

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169733

M. Pallas, N. Martin, J. Martin

引用次数: 8

Dynamic programming speech recognition using a context-free grammar 使用上下文无关语法的动态编程语音识别

ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169746

H. Ney

引用次数: 40