1993 IEEE International Conference on Acoustics, Speech, and Signal Processing最新文献_第10页

A dynamic cepstrum incorporating time-frequency masking and its application to continuous speech recognition 含时频掩蔽的动态倒频谱及其在连续语音识别中的应用

1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319399

K. Aikawa, H. Singer, Hideki Kawahara, Y. Tohkura

引用次数: 36

Bispectral techniques for spherical functions 球函数的双谱技术

1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319633

R. Kakarala, B. M. Bennett, G. Iverson, M. D'Zmura

引用次数: 9

Image filtering by gradient inverse inhomogeneous diffusion 梯度逆非均匀扩散图像滤波

1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319750

A. El-Fallah, G. Ford

引用次数: 4

Polyspectra-based, blind, MMSE, fractionally-spaced equalization of a cyclostationary signal 基于多光谱，盲，MMSE，分数间隔均衡的循环平稳信号

1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319648

M. Webster

引用次数: 2

Toward efficient morphological shape representation 走向有效的形态表示

1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319763

J. Reinhardt, W. Higgins

引用次数: 7

Cancellation of ISI in non-linear voice-band data channels 非线性话音波段数据信道中ISI的消除

1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319516

Z. Fejzo, H. Lev-Ari

{"title":"Cancellation of ISI in non-linear voice-band data channels","authors":"Z. Fejzo, H. Lev-Ari","doi":"10.1109/ICASSP.1993.319516","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319516","url":null,"abstract":"A technique for cancellation of intersymbol interference (ISI) in nonlinear voice band data channels that contain second and third order nonlinear elements is presented. A data-aided nonlinear canceller structure based on a configuration proposed by E. Biglieri et al. (IEEE J. Sel. Areas Commun. vol.SAC-2, no.5, p.765-77, Sept. 1984) is used to remove nonlinear ISI without excessive noise enhancement. While the scheme proposed by Biglieri et al. uses a least mean square (LMS) adaptation algorithm, the authors choose to use a recursive least squares (RLS) adaptation algorithm in order to provide a faster convergence of the algorithm and improved robustness when applied to a variety of channels. A fast version of the RLS algorithm with computational complexity of order O(9N) (where N is the order of the nonlinear filter) is derived. The performance of the nonlinear canceller with both fast RLS and LMS adaptation is evaluated by simulation for 16-QAM (quadrature amplitude modulation) data transmission over nonlinear channels with a rate of 9600 bit/s. The simulation results show a performance improvement when the fast RLS version is used as compared with the LMS.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"3 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":"129467043","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: 2

The estimation of powerful language models from small and large corpora 从大小语料库中估计强大的语言模型

1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319222

P. Placeway, R. Schwartz, Pascale Fung, L. Nguyen

{"title":"The estimation of powerful language models from small and large corpora","authors":"P. Placeway, R. Schwartz, Pascale Fung, L. Nguyen","doi":"10.1109/ICASSP.1993.319222","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319222","url":null,"abstract":"The authors consider the estimation of powerful statistical language models using a technique that scales from very small to very large amounts of domain-dependent data. They begin with improved modeling of the grammar statistics, based on a combination of the backing-off technique and zero-frequency techniques. These are extended to be more amenable to the particular system considered here. The resulting technique is greatly simplified, more robust, and gives improved recognition performance over either of the previous techniques. The authors also consider the problem of robustness of a model based on a small training corpus by grouping words into obvious semantic classes. This significantly improves the robustness of the resulting statistical grammar. A technique that allows the estimation of a high-order model on modest computation resources is also presented. This makes it possible to run a 4-gram statistical model of a 50-million word corpus on a workstation of only modest capability and cost. Finally, the authors discuss results from applying a 2-gram statistical language model integrated in the HMM (hidden Markov model) search, obtaining a list of the N-best recognition results, and rescoring this list with a higher-order statistical model.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"21 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":"128282818","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: 105

Dynamic bit allocation in CELP excitation coding CELP激励编码中的动态位分配

1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319261

T. Eriksson, Johan Sjöberg

引用次数: 4

Subspace tracking based on the projection approach and the recursive least squares method 基于投影法和递推最小二乘法的子空间跟踪

1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319615

Bin Yang

引用次数: 43

Time-recursive architectures and wavelet transform 时间递归结构和小波变换

1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319151

Emmanuel N. Frantzeskakis, J. Baras, K. J. Liu

引用次数: 3