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ADAPTIVE CASCADE FILTER FOR speech analysis

IEE PROCEEDINGS-E COMPUTERS AND DIGITAL TECHNIQUES 1983年第1期130卷 11-18页

作者： CHING, PC GOODYEAR, CC UNIV BATH SCH ELECT ENGNBATH BA2 7AYAVONENGLAND

Two techniques are described for sequentially adapting a finite impulse response digital filter which is constructed as a cascade of 2nd-order sections. It is shown that the radius and angle co-ordinates of each zero pair may be separately adapted. The first method uses a modified LMS algorithm, while the second, by employing prequantised coefficients, avoids the need for a control parameter. The performance, when used for linear prediction of speech, is discussed.

关键词： prequantised coefficients finite impulse response digital filter speech analysis and processing linear prediction speech analysis cascade networks LMS algorithm cascade filter digital filters Digital filters speech and audio signal processing filtering and prediction theory

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IMPROVED PITCH DETECTION ALGORITHM EMPLOYING TEMPORAL STRUCTURE INVESTIGATION OF THE speech WAVEFORM

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IEE PROCEEDINGS-F RADAR AND SIGNAL processing 1988年第2期135卷 169-174页

作者： SUTHERLAND, AM JACK, MA LAVER, J Centre for Speech Technology Research University of Edinburgh Edinburgh UK

The paper examines the performance of several versions of the parallel processing method of pitch period estimation of speech, highlighting the limitations of each. An improved algorithm, based on the temporal investigation of the speech waveform, is described and evaluated. It is shown to offer improved performance in both long term pitch period estimation accuracy, and cycle to cycle accuracy. A new method of evaluating the performance of pitch detection algorithms, based on measuring their stability with respect to variation of the time origin of the input speech, is also described.

关键词： stability pitch detection algorithm speech recognition speech analysis and processing pitch period estimation acoustic variables measurement signal detection speech waveform temporal investigation speech and audio signal processing speech recognition and synthesis equipment computerised signal processing speech recognition parallel processing

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IMPROVING speech QUALITY OF CELP CODER

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ELECTRONICS LETTERS 1989年第19期25卷 1275-1277页

作者： LEE, JI UN, CK Communications Research Laboratory Department of Electrical Engineering Korea Advanced Institute of Science & Technology Seoul Korea

Although the speech quality of the code excited linear prediction (CELP) coder at 4800bit/s is relatively good, it is still perceived as rough or noisy. In the letter we propose a residual shaping method that produces, without signal distortion, quality comparable to that obtained by adaptive postfiltering. The proposed method is particularly effective in the multiple tandeming environment.

关键词： multiple tandeming environment speech recognition speech analysis and processing residual shaping method encoding speech intelligibility Codes CELP coder speech coding speech and audio signal processing 4800 bit/s code excited linear prediction speech quality improvement filtering and prediction theory

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BLANK AND BURST TRANSMISSION OF DATA OVER ACTIVE speech CHANNELS

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ELECTRONICS LETTERS 1988年第11期24卷 670-672页

作者： WONG, WC GOODMAN, DJ AT&T BELL LABS HOLMDELNJ 07733

To increase the capacity of blank and burst transmission of control information in cellular communication systems, we explore the application of a speech segment reconstruction technique. The transmitter selects a blanking interval on the basis of a segment classification measure, while the receiver replaces the missing speech with an estimate based on previously received speech. Using this approach, approximately 3.2 data bursts per second, each lasting 16 ms, may be sent without degrading speech quality.

关键词： Mobile radio systems speech analysis and processing burst transmission cellular communication systems speech quality data transmission data communication systems cellular radio speech and audio signal processing blank transmission speech segment reconstruction technique Telecommunication applications active speech channels

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Precise glottal closure instant detector for voiced speech

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ELECTRONICS LETTERS 1996年第23期32卷 2117-2118页

作者： Hahn, M Kang, DG Acoustic Communication Section Electronics and Telecommunications Research Institute Taejon Korea

An algorithm which can extract precise GCI information directly from speech is described. By utilising the average pitch and the area informations, accurate GCI positions are obtained. The result compared with the man... 详细信息

关键词： speech analysis and processing signal processing

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HOMOMORPHIC VECTOR QUANTIZATION

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ELECTRONICS LETTERS 1987年第11期23卷 561-562页

作者： VICH, R Institute of Radio Engineering & Electronics Czechoslovak Academy of Sciences Prague Czechoslovakia

A new approach is proposed for vector quantisation in linear predictive speech coding. The problem is formulated as speech model recognition by minimising the Euclidean distance measure of real cepstra of models with unit power transmission. The procedure is robust with respect to quantisation of both the cepstral coefficients and operational results.

关键词： speech recognition speech analysis and processing speech model recognition linear predictive speech coding homomorphic vector quantisation speech and audio signal processing speech recognition Euclidean distance measure of real cepstra

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LINEAR PREDICTION USING L(1) NORM IN ORTHOGONAL VECTOR-SPACE

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ELECTRONICS LETTERS 1995年第6期31卷 430-431页

作者： HU, HT Department of Electronic Engineering National I-Lan Institute of Agriculture and Technology I-Lan Republic of China

Linear prediction is formulated in a vector space by means of the orthogonal transformation, with which the L(1) criterion can be easily incorporated to yield an efficient iterative algorithm. An improvement over the ... 详细信息

关键词： LINEAR PREDICTIVE CODING speech analysis and processing

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MODULATED QMF FILTER BANKS WITH PERFECT RECONSTRUCTION

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ELECTRONICS LETTERS 1990年第13期26卷 906-907页

作者： MALVAR, HS Dept. de Engenharia Elétrica Universidade de Brasília Brasilia Brazil

Necessary and sufficient conditions for perfect reconstruction (PR) in a modulated filter bank are derived. It is shown that, for a bank of M filters of length L, PR can be obtained when L = 2KM, for any positive inte... 详细信息

关键词： image processing modulated filter bank speech analysis and processing perfect reconstruction speech and audio signal processing Signal processing and detection quadrature mirror filter signal processing Filters and other networks QMF filter picture processing filtering and prediction theory

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SYSTOLIC ARRAY FOR NONLINEAR MULTIDIMENSIONAL INTERPOLATION USING RADIAL BASIS FUNCTIONS

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ELECTRONICS LETTERS 1990年第1期26卷 7-9页

作者： BROOMHEAD, DS JONES, R MCWHIRTER, JG SHEPHERD, TJ Royal Signals & Radar Establishment Malvern UK

A fully systolic network is proposed for rapid and efficient multidimensional interpolation using radial basis functions (RBFs). The resulting processor, which constitutes a form of nonlinear adaptive filter, resemble... 详细信息

关键词： least squares approximations nonlinear perceptron multilayer perceptron model Interpolation and function approximation (numerical analysis) neural network computerised signal processing Filters and other networks Multiprocessing systems radial basis functions nonlinear adaptive filter speech analysis and processing nonlinear multidimensional interpolation cellular arrays parallel architectures Signal processing and detection speech and audio signal processing interpolation neural nets feedforward perceptron Analogue and hybrid computing techniques adaptive filters systolic network Information theory Numerical analysis Communications computing

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BUFFER analysis FOR ASYNCHRONOUS DATA INTERPOLATION IN ANALOG speech

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IEE PROCEEDINGS-E COMPUTERS AND DIGITAL TECHNIQUES 1981年第6期128卷 229-238页

作者： KEKRE, HB KHALID, M Computer Centre Indian Institute of Technology Bombay India

The store-and-forward buffer required for asynchronous data interpolation (ADI) in analogue speech is analysed using a queueing model with single server having 1st-order Markov interruptions. The buffer-content probability generating function and probability mass function are derived. It is shown that the empty-buffer probability in the finite buffer equals that of the infinite buffer with its offered load equal to the carried load of the finite buffer. This concept proves helpful in obtaining a solution for the finite buffer through the solution for infinite buffer which is obtained in a closed form. This results in considerable saving in computer time. Although the results presented are for analogue speech, they can serve as a worst-case design for the STDM case.

关键词： queueing model speech analysis and processing finite buffer infinite buffer markov interruptions speech analysis store-and-forward buffer asynchronous data interpolation speech and audio signal processing analogue speech channel capacity speech processing Queueing theory single server queueing theory

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