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

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Chandrasekhar adaptive regularizer for adaptive filtering 用于自适应滤波的Chandrasekhar自适应正则化器
ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-04-07 DOI: 10.1109/ICASSP.1986.1168766
A. Houacine, G. Demoment
{"title":"Chandrasekhar adaptive regularizer for adaptive filtering","authors":"A. Houacine, G. Demoment","doi":"10.1109/ICASSP.1986.1168766","DOIUrl":"https://doi.org/10.1109/ICASSP.1986.1168766","url":null,"abstract":"Adaptivity, stability, fast initial convergence, and low complexity are contradictory exigences in adaptive filtering. The least-mean-squares (LMS) algorithms suffer from a slow initial convergence, and the fast recursive least-squares (RLS) ones present numerical stability problems. In this paper we address this last-mentioned problem and perform a regularization of the initial LS problem by using a priori information about the solution and a finite memory. A new, fast, adaptive, recursive algorithm is presented, based on a state-space representation and Chandrasekhar factorizations.","PeriodicalId":242072,"journal":{"name":"ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1986-04-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132093503","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: 13
Digital signal processing in a 16kbps APC-AB codec by fixed point digital signal processor (FDSP-3) 定点数字信号处理器(FDSP-3)在16kbps APC-AB编解码器中的数字信号处理
ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-04-07 DOI: 10.1109/ICASSP.1986.1169050
Y. Tomita, S. Unagami, T. Taniguchi, Y. Tada, M. Taka
{"title":"Digital signal processing in a 16kbps APC-AB codec by fixed point digital signal processor (FDSP-3)","authors":"Y. Tomita, S. Unagami, T. Taniguchi, Y. Tada, M. Taka","doi":"10.1109/ICASSP.1986.1169050","DOIUrl":"https://doi.org/10.1109/ICASSP.1986.1169050","url":null,"abstract":"Recently much intensive research of 16kbps Speech coding algorithm has been conducted aiming to reduce the transmission bit rate and yet provides high speech quality. Adaptive predictive coding with adaptive bit allocation (APC-AB)[1] is considered to be one promising approach. However, the processing of this coding algorithm is so complicated that the implementation of the algorithm on a general-purpose signal processor, especially if fixed-point arithmetic DSPs are used, requires careful study of arithmetic operation precision and same way to reduce the number of processing cycles. Taking account of these points, real-time signal processing using a fixed-point signal processing chip (FDSP-3) has been studied, and a prototype codec has been realized. The prototype codec satisfied the CCITT mask of signal-to-total distortion ratio for PCM codecs and showed quality good enough for \"toll\" speech.","PeriodicalId":242072,"journal":{"name":"ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"24 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1986-04-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133865048","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: 0
Digital data equalizing in multitrack digital audio recording 多轨数字音频记录中的数字数据均衡
ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-04-07 DOI: 10.1109/ICASSP.1986.1169166
R. Lagadec
{"title":"Digital data equalizing in multitrack digital audio recording","authors":"R. Lagadec","doi":"10.1109/ICASSP.1986.1169166","DOIUrl":"https://doi.org/10.1109/ICASSP.1986.1169166","url":null,"abstract":"Digital Audio stationary-head recording is performed with recording densities requiring playback equalization to reduce intersymbol interference. In today's digital audio recorders, data retrieval is usually implemented with analog circuits (equalizing filters, DC component restoration, data detectors) with manual adjustment. Several approaches are reviewed for digitizing data retrieval and making it adaptive.","PeriodicalId":242072,"journal":{"name":"ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"39 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1986-04-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122198921","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: 0
The C-MU phonetic classification system C-MU语音分类系统
ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-04-07 DOI: 10.1109/ICASSP.1986.1168527
R. Cole, M. Phillips, B. Brennan, B. Chigier
{"title":"The C-MU phonetic classification system","authors":"R. Cole, M. Phillips, B. Brennan, B. Chigier","doi":"10.1109/ICASSP.1986.1168527","DOIUrl":"https://doi.org/10.1109/ICASSP.1986.1168527","url":null,"abstract":"The Carnegie-Mellon Speech Group is working with a number of institutions in the DARPA community to develop \"A New Generation English Language System\" (ANGELS) to perform large vocabulary speaker-independent recognition of natural continuous speech. A major focus of this effort is the development of an acoustic-phonetic module that provides an accurate phonetic transcription of an unknown utterance. This paper describes the phonetic classification system now under development, the research approach and some preliminary results.","PeriodicalId":242072,"journal":{"name":"ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1986-04-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123780218","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: 9
Efficient multi-processor implementation of recursive digital filters 递归数字滤波器的高效多处理器实现
ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-04-07 DOI: 10.1109/ICASSP.1986.1169082
Wonyong Sung, S. Mitra
{"title":"Efficient multi-processor implementation of recursive digital filters","authors":"Wonyong Sung, S. Mitra","doi":"10.1109/ICASSP.1986.1169082","DOIUrl":"https://doi.org/10.1109/ICASSP.1986.1169082","url":null,"abstract":"Efficient computation method for the recursive digital filtering is studied in the multi-processor environment. The method solves the dependency problem by separate computations of the particular and transient solutions. The throughput of the algorithm increases linearly with the number of processors, making it possible to increase the throughput effectively by using multiple number of processors. The implementations of the algorithm using a vector-processor and a multiprocessor in a ring network are also studied.","PeriodicalId":242072,"journal":{"name":"ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"55 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1986-04-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122620291","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: 25
Noise cancellation for hearing aids 助听器降噪
ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-04-07 DOI: 10.1109/ICASSP.1986.1168857
D. Chazan, Y. Medan, U. Shvadron
{"title":"Noise cancellation for hearing aids","authors":"D. Chazan, Y. Medan, U. Shvadron","doi":"10.1109/ICASSP.1986.1168857","DOIUrl":"https://doi.org/10.1109/ICASSP.1986.1168857","url":null,"abstract":"The feasibility of using multi microphone noise cancellation techniques to enhance the performance of hearing aids was investigated in a project carried out by the IBM Israel scientific center. A short survey of some of the work in this field and a summary of the results and the motivation for the project will be given below, together with a description of an listening test which was carried out on the system. The results of the tests indicate that a marked improvement in intelligibility may be obtained using these techniques.","PeriodicalId":242072,"journal":{"name":"ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"24 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1986-04-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125302543","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
A fast codebook search algorithm for nearest-neighbor pattern matching 一种最近邻模式匹配的快速码本搜索算法
ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-04-07 DOI: 10.1109/ICASSP.1986.1169084
D. Cheng, A. Gersho
{"title":"A fast codebook search algorithm for nearest-neighbor pattern matching","authors":"D. Cheng, A. Gersho","doi":"10.1109/ICASSP.1986.1169084","DOIUrl":"https://doi.org/10.1109/ICASSP.1986.1169084","url":null,"abstract":"A critical issue in the use of Vector Quantization is to circumvent the complexity bottleneck that prevents the full power of this data compression technique from being exploited. The demanding exhaustive codebook search can be avoided by more efficient algorithms for locating the nearest codevector in a codebook to a given input vector. This paper introduces a fast nearest neighbor search algorithm, the Binary Hyperplane Testing (BHT) algorithm, for fast codebook searching when least Euclidean distance is the appropriate distortion or dissimilarity measure. LetNdenote the size of a codebook, andkdenote the dimension of a vector. The search complexity of the BHT algorithm increases linearly with the resolutionr = (log_{2}N)/kfor fixed dimensionk. The search complexity of the BHT algorithm is sufficiently low that it has been used to implement a real-time speech waveform vector quantizer with sampling rate 8 KHz, dimensionk = 4, and codebook sizeN = 128using a single programmable signal processor, the TMS32010.","PeriodicalId":242072,"journal":{"name":"ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1986-04-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125454822","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: 76
Deciphering speech waveforms 解码语音波形
ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-04-07 DOI: 10.1109/ICASSP.1986.1168540
M. O'Kane, Judy Gillis, Philip Rose, Michael Wagner
{"title":"Deciphering speech waveforms","authors":"M. O'Kane, Judy Gillis, Philip Rose, Michael Wagner","doi":"10.1109/ICASSP.1986.1168540","DOIUrl":"https://doi.org/10.1109/ICASSP.1986.1168540","url":null,"abstract":"Many phoneticians are remarkably expert at 'reading' speech waveforms. This paper describes an attempt to capture this knowledge for use as a segmentation and early labelling knowledge source for a continuous speech recognition system. As well as deriving information from the waveform directly, the decisions made by the waveform deciphering knowledge source are based on a related series of functions derived from the waveform. These functions, which relate to both valley-to-peak and zero crossing measures, are computationally very efficient and it would seem that the frequency analogues of these functions could provide an alternative means of deriving a certain amount of the spectral information more usually obtained through spectrograms.","PeriodicalId":242072,"journal":{"name":"ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1986-04-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121767751","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: 3
Spectral transformations through canonical correlation analysis for speaker adptation in ASR 基于典型相关分析的ASR中说话人自适应频谱变换
ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-04-07 DOI: 10.1109/ICASSP.1986.1168759
K. Choukri, G. Chollet, Y. Grenier
{"title":"Spectral transformations through canonical correlation analysis for speaker adptation in ASR","authors":"K. Choukri, G. Chollet, Y. Grenier","doi":"10.1109/ICASSP.1986.1168759","DOIUrl":"https://doi.org/10.1109/ICASSP.1986.1168759","url":null,"abstract":"This paper describes a technique of spectral transformation for improved adaptation of a knowledge data base or reference templates to new speakers in automatic speech recognition (ASR). Based on a statistical analysis tool (Canonical correlation analysis) the proposed method permits to improve speaker independance in Large vocabulary ASR. Application to an isolated word recognizer improved a 70% correct score to 87%.","PeriodicalId":242072,"journal":{"name":"ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1986-04-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131122612","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: 28
Data flow chip ImPP and its system for image processing 数据流芯片ImPP及其图像处理系统
ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1986-04-07 DOI: 10.1109/ICASSP.1986.1169141
M. Iwashita, T. Temma
{"title":"Data flow chip ImPP and its system for image processing","authors":"M. Iwashita, T. Temma","doi":"10.1109/ICASSP.1986.1169141","DOIUrl":"https://doi.org/10.1109/ICASSP.1986.1169141","url":null,"abstract":"A data flow image pipelined processor VLSI chip (ImPP:µPD7281) has been developed. Its hardware architecture and the experimental Template Controlled Image Processing System -3 (TIP-3), which includes 8 ImPP chips, are described. The ImPP is characterized by its data flow architecture and flexible pipeline processing. The ImPP has a uni-directional pipeline bus. The connection between ImPPs is easy and does not require extra circuits. The processing performance can be increased by connecting many chips to each other. In this multiple ImPP configuration, individual ImPPs share different portions of the pipeline programs and different portions of spatial data. The ImPP can be used in various system configurations. A single ImPP chip is sufficient for simple processing and multiple ImPPs are useful for a faster image processing system. Performance estimation using a software simulator, and also using an actual hardware system has been carried out. TIP-3's top performance is 40 MIPS. The main factors which influence execution efficiency are discussed and analyzed.","PeriodicalId":242072,"journal":{"name":"ICASSP '86. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"32 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1986-04-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127877166","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: 5
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