ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing最新文献

The analog voice privacy system 模拟语音保密系统

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-01-02 DOI: 10.1109/ICASSP.1986.1169066

R. Cox, D. Bock, K. Bauer, J. Johnston, J. Snyder

{"title":"The analog voice privacy system","authors":"R. Cox, D. Bock, K. Bauer, J. Johnston, J. Snyder","doi":"10.1109/ICASSP.1986.1169066","DOIUrl":"https://doi.org/10.1109/ICASSP.1986.1169066","url":null,"abstract":"The Analog Voice Privacy System is based on individual sample permutation of the output samples of a sub-band coder analysis filterbank. The system has a large number of digital keys, giving it the strength of a digital encryption system, but also retains the good quality characteristics of analog scramblers. It has been implemented in a real-time hardware prototype designed for evaluation in the field. The units work with any modular telephone and standard 120 volts AC electricity. The device contains two circuitry boards, one for analog and one for digital processing which contain four digital signal processors. There are 125! possible permutation keys. These prototypes were designed to be tested in real telephone environments. To date, the device has been successfully tested over long distance telephone connections, several different analog and digital PBXs and telephone switches, and a channel simulator. The quality of the decrypted speech is considered very natural, and in particular, speaker recognition is retained. This is a significant advantage over digital vocoders. This paper describes the underlying principles of the algorithm, the details of its implementation and laboratory test results.","PeriodicalId":242072,"journal":{"name":"ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"11 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-01-02","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123927787","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: 41

A proposal of a knowledge based isolated word recognition 一种基于知识的孤立词识别方法

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-12-01 DOI: 10.1109/ICASSP.1986.1169202

S. Morishima, H. Harashima, H. Miyakawa

引用次数: 11

A network model dealing with focus of conversation for speech understanding system 语音理解系统中处理会话焦点的网络模型

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-12-01 DOI: 10.1109/ICASSP.1986.1168665

Tetsunori Kobayashi, K. Shirai

引用次数: 0

Continuously variable duration hidden Markov models for speech analysis 语音分析的连续变时隐马尔可夫模型

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-12-01 DOI: 10.1109/ICASSP.1986.1168801

S. Levinson

{"title":"Continuously variable duration hidden Markov models for speech analysis","authors":"S. Levinson","doi":"10.1109/ICASSP.1986.1168801","DOIUrl":"https://doi.org/10.1109/ICASSP.1986.1168801","url":null,"abstract":"During the past decade, the applicability of hidden Markov models (HMM) to various facets of speech analysis had been demonstrated in several different experiments. These investigations all rest on the assumption that speech is a quasi-stationary process whose stationary intervals can be identified with the occupancy of a single state of an appropriate HMM. In the traditional form of the HMM, the probability of duration of a state decreases exponentially with time. This behavior does not provide an adequate representation of the temporal structure of speech. The solution proposed here is to replace the probability distributions of duration with continuous probability density functions to form a continuously variable duration hidden Markov model (CVDHMM). The gamma distribution is ideally suited to specification of the durational density since it is one-sided and has only two parameters which, together, define both mean and variance. The main result is a derivation and proof of convergence of reestimation formulae for all the parameters of the CVDHMM. It is interesting to note that if the state durations are gamma distributed, one of the formulae is nonalgebraic but, fortuitously, has properties such that it is easily and rapidly solved numerically to any desired degree of accuracy. Other results are presented including the performance of the formulae on simulated data.","PeriodicalId":242072,"journal":{"name":"ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"9 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1986-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114950935","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: 84

Performance of narrowband signal-subspace processing 窄带信号子空间处理性能

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-12-01 DOI: 10.1109/ICASSP.1986.1169009

Hong Wang, M. Kaveh

引用次数: 2

Spectrum enhancement using linear programming 使用线性规划的频谱增强

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-12-01 DOI: 10.1109/ICASSP.1986.1168700

R. Rothacker, R. Mammone, S. Davidovici

引用次数: 3

Speech coder using phase equalization and vector quantization 语音编码器使用相位均衡和矢量量化

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-12-01 DOI: 10.1109/ICASSP.1986.1169253

T. Moriya, M. Honda

引用次数: 23

Range extended speech decoder for Modulo-PCM 范围扩展语音解码器的模pcm

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-12-01 DOI: 10.1109/ICASSP.1986.1168987

M. Hagiwara, M. Nakagawa

引用次数: 0

Hardware implementation of FM multipath distortion canceller 调频多径失真消除器的硬件实现

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-12-01 DOI: 10.1109/ICASSP.1986.1168729

M. Itami, Takashi Mochizuki, M. Hatori

引用次数: 1

Spectral analysis of order statistic filters 阶统计滤波器的频谱分析

ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-12-01 DOI: 10.1109/ICASSP.1986.1168707

A. Restrepo, A. Bovik

引用次数: 8