International Conference on Acoustics, Speech, and Signal Processing,

23-26 May 1989

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Displaying Results 1 - 25 of 711
  • A locus model of coarticulation in an HMM speech recognizer

    Publication Year: 1989, Page(s):97 - 100 vol.1
    Cited by:  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (324 KB)

    A novel type of hidden Markov model (HMM) has been developed to account explicitly for the context-dependent vowel acoustic transitions in consonant-vowel and vowel consonant phonetic environments. The major difference between this type of HMM and the standard Gaussian HMM is that the Gaussian mean vectors associated with the vowel HMM states, which are intended to model the vowel acoustic transit... View full abstract»

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  • ICASSP-89: 1989 International Conference on Acoustics, Speech and Signal Processing (IEEE Cat. No.89CH2673-2)

    Publication Year: 1989
    Request permission for commercial reuse | PDF file iconPDF (54 KB)
    Freely Available from IEEE
  • An order recursive algorithm for synthesizing linear recursive filters

    Publication Year: 1989, Page(s):1131 - 1133 vol.2
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (149 KB)

    The author proposes an order recursive algorithm to compute efficiently the solution to the set of linear equations for finding the parameters of a recursive filter whose unit impulse response best approximates that of the prescribed ideal response. The algorithm requires only O(p/sup 2/) operations. The inherent structure of the coefficient matrix in the system of linear equations and the inversi... View full abstract»

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  • Knowledge based parallel recognition of handwritten alphanumerics

    Publication Year: 1989, Page(s):1807 - 1810 vol.3
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (238 KB)

    A novel knowledge-based parallel processing system has been designed for recognition of handwritten characters. With five quadtree-linked microprocessors, this system can extract features from the character image in four directions simultaneously. Through repetitive order-giving and information-gathering between the master and the slaves, the system can process the information at two levels: globa... View full abstract»

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  • Digital audio gain control for hearing aids

    Publication Year: 1989, Page(s):2049 - 2052 vol.3
    Cited by:  Papers (6)  |  Patents (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (198 KB)

    It is suggested that the digital realization of an audio gain controller can bypass some of the problems commonly encountered with analog hearing aid automatic gain controllers, such as accurate setting of its input/output state characteristic (SC). The author describes the embedding of a digital audio gain controller in a TMS32010 DSP (digital signal processor). Its SC is graphically programmable... View full abstract»

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  • A tutorial overview of modern spectral estimation

    Publication Year: 1989, Page(s):2152 - 2157 vol.4
    Cited by:  Papers (17)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (575 KB)

    A summary of several modern spectral estimation methods is presented. Most of the methods can be explained in the context of parametric time-series modeling. A few methods involve nonparametric treatment. The techniques discussed include classical spectral estimation, autoregressive (maximum entropy), ARMA (autoregressive moving average), Prony, maximum-likelihood, Pisarenko, and MUSIC methods. Ma... View full abstract»

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  • Performance of normalized matched filters

    Publication Year: 1989, Page(s):2704 - 2707 vol.4
    Cited by:  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (183 KB)

    In sonar or radar, the operation called normalization consists of getting a constant false alarm rate receiver by using a background noise power estimation to set the threshold. This study compares the performance of the two test functions obtained, in the white Gaussian noise case, by using two different maximum-likelihood noise power estimates, one under the hypotheses H/sub 0/ and the other und... View full abstract»

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  • New simple implementation of the coherent signal subspace method for wide band direction of arrival estimation

    Publication Year: 1989, Page(s):2764 - 2767 vol.4
    Cited by:  Papers (14)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (297 KB)

    The authors address the problem of source localization estimation (SLE), given the output of a sensor array, in the case of aerial acoustics in an indoor environment. The presence of echos similar to highly correlated sources necessitates the study of high-resolution methods in the case of wideband emitters. As the field of possible applications includes robotics, the authors do not assume any a p... View full abstract»

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  • Fast algorithms for Nevanlinna-Pick interpolation and H∞ optimization

    Publication Year: 1989, Page(s):1306 - 1309 vol.2
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (248 KB)

    Fast algorithms are derived for the scalar and tangential directional Nevanlinna-Pick interpolation problems. These fast algorithms require fewer multiplications than standard algorithms for these problems. The scalar interpolation problem, which arises in SISO (single input single output) H∞ optimization, is solved using a three-term recurrence similar to that of the so-called spli... View full abstract»

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  • Feature-based recognition of nonsonorant consonants in Chinese speech

    Publication Year: 1989, Page(s):334 - 337 vol.1
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (228 KB)

    The recognition of nonsonorant consonants is a key problem in unlimited-vocabulary Chinese speech recognition. A feature-based nonsonorant consonant classification scheme is assumed. The unknown single-syllabic consonant-vowel utterance is first divided into consonant and vowel segments; then the features of the consonant segment are extracted. Next, the unknown consonant is categorized into subcl... View full abstract»

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  • A fast convergence algorithm for adaptive FIR filters

    Publication Year: 1989, Page(s):892 - 895 vol.2
    Cited by:  Papers (2)  |  Patents (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (264 KB)

    The authors present a fast convergence algorithm for adaptive FIR (finite impulse response) filters. The algorithm controls the step size on the basis of the moving average of the misadjustment level for a faster convergence. Several operations in addition to the LMS operations are used to estimate the misadjustment and average it to provide a stable step size. The algorithm is applied to adaptive... View full abstract»

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  • Very high efficiency VLSI chip-pair for full search block matching with fractional precision

    Publication Year: 1989, Page(s):2437 - 2440 vol.4
    Cited by:  Papers (7)  |  Patents (35)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (276 KB)

    VLSI architecture design and implementation of a chip pair for the motion compensation full search block matching algorithm are described. This pair of ASICs (application-specific integrated circuits) is motivated by the intensive computational demands for performing motion compensation in real time. They have been developed to calculate fractional motion vectors with quarter-pel precision. The VL... View full abstract»

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  • Efficient FFT implementation on an IEEE floating-point digital signal processor

    Publication Year: 1989, Page(s):1302 - 1305 vol.2
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (312 KB)

    The authors describe the implementation of real and complex FFT (fast Fourier transform) algorithms on the Motorola DSP96002. The DSP96002 is a general-purpose, dual-bus IEEE standard floating-point digital signal processor (DSP). At a 74-ns instruction cycle, the DSP96002 implements a 1024-point real FFT in 0.905 ms and a 1024-point complex FFT in 1.55 ms. This performance is achieved by calculat... View full abstract»

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  • Event detection in multisource imaging using contextual estimation

    Publication Year: 1989, Page(s):1647 - 1650 vol.3
    Cited by:  Papers (2)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (284 KB)

    The authors propose a novel approach to the problem of detecting events, i.e. significant differences between pictures of a given scene taken at different wavelengths or with different sensors. It is shown that event detection can be expressed, within a Bayesian decision framework, as a contextual estimation problem. The unknown process to be estimated corresponds to the significant interimage cha... View full abstract»

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  • Phonetic recognition using peak weighted binary spectrum

    Publication Year: 1989, Page(s):330 - 333 vol.1
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (276 KB)

    Invariant acoustic-phonetic features conveyed simultaneously by the stable and the transient portion of the speech signal are investigated quantitatively using a relational cue. To represent the relational cue effectively, a peak-weighted binary spectrum is proposed. Prevocalic nasal recognition results show speaker independence and improved performance in comparison with dynamic time-warping meth... View full abstract»

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  • An LSP based speech quality measure

    Publication Year: 1989, Page(s):596 - 599 vol.1
    Cited by:  Papers (8)  |  Patents (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (240 KB)

    An objective measure is introduced for predicting the subjective quality of a distorted speech signal. The unique feature of the measure is that it explicitly uses a parametric model of speech perception (rather than a signal fidelity measure) in the design of an objective measure for speech quality. Specifically, the known sensitivity of human auditory perception to perturbations in the speech fo... View full abstract»

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  • Error analysis and comparison of FFT implementation structures

    Publication Year: 1989, Page(s):888 - 891 vol.2
    Cited by:  Papers (17)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (296 KB)

    A statistical model for roundoff errors is used to predict the output noise of most common forms of the fast Fourier transform (FFT) algorithm, the radix-2 and radix-4, with both decimation-in-time (DIT) and decimation-in-frequency (DIF). Two's complement fixed-point data representation with either truncation, rounding, or convergent rounding is treated. The mean values and the variances of the er... View full abstract»

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  • A single chip VLSI chrominance/luminance separator based on a silicon compiler

    Publication Year: 1989, Page(s):2433 - 2436 vol.4
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (232 KB)

    The authors present a single-chip VLSI chrominance/luminance (Y/C) separator that is economically fabricated for NTSC TV signals at 13.5-MHz CCIR standard sampling rate. In order to realize compactness and low power dissipation, two FIR filter architectures and a multiplier structure are proposed. A silicon compiler, which uses these structures, also contributes to fast and error-free VLSI develop... View full abstract»

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  • An orthogonal method for solving systems of linear equations without square roots and with few divisions

    Publication Year: 1989, Page(s):1298 - 1301 vol.2
    Cited by:  Papers (9)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (248 KB)

    An algorithm is presented that requires only multiplications, additions, and a single division for the orthogonal solution of a system of linear equations. For that purpose the QR-decomposition of an extended system matrix, called the orthogonal Faddeeva algorithm, is computed by a square-root- and division-free Givens rotation, called scaled standard Givens rotation (SSGR). A special kin... View full abstract»

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  • Channel interactions in visible pattern analysis

    Publication Year: 1989, Page(s):1643 - 1646 vol.3
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (256 KB)

    The efficacy of the 2-D Gabor channel filters for segmenting images based on the division of the image into textures was established previously. In the present work, the authors extend the model for the direct estimation of the emergent spatially localized orientation/frequencies of visible patterns using a variational scheme. The most significant result of this extension is that patterns with spa... View full abstract»

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  • A high quality speech coder at 400 bps

    Publication Year: 1989, Page(s):204 - 206 vol.1
    Cited by:  Papers (4)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (200 KB)

    A high-quality speech coder at 400 b/s has been developed. This coder is designed to provide reliable communication over an HF channel in a hostile environment. The major techniques utilized for coding include vector quantization, interpolation, and trellis coding. The test results in the presence of speaker variations, background noise, and acoustic impairments are discussed. The three speaker av... View full abstract»

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  • Speech recognition using dynamic transformation of phoneme templates depending on acoustic/phonetic environments

    Publication Year: 1989, Page(s):326 - 329 vol.1
    Cited by:  Papers (5)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (212 KB)

    A description is given of a phoneme-based speech recognition method using a linear model to represent the contextual variations in acoustic features caused by the phonemic context. A feature vector in a phonetic segment is decomposed into a context-independent vector, a context-dependent vector, and a residual vector. The context-independent vector is calculated by weighting a coefficient matrix t... View full abstract»

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  • The ST18940/41: an advanced single-chip digital signal processor

    Publication Year: 1989, Page(s):2493 - 2496 vol.4
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (300 KB)

    A description is given of the architecture and the instruction set of the ST18940/41 processor. The use of an advanced CMOS technology (1.2 μm) combined with a highly parallel architecture provides a powerful and efficient DSP as demonstrated on the usual signal processing algorithms. To cover a wide range of applications, two versions are provided: a closed version with on-chip masked program ... View full abstract»

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  • Stereoscopic images disparity for predictive coding

    Publication Year: 1989, Page(s):1976 - 1979 vol.3
    Cited by:  Papers (18)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (280 KB)

    In order to create a 3-D system with two images (right and left) that must be transmitted simultaneously, an efficient coding appropriate for stereoscopic images must be developed to reduce the large amount of information contained in the images. As an initial step in realizing such a coding scheme, the statistical characteristics of stereoscopic images and the possibility of stereoscopic image da... View full abstract»

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  • A new speech modification method by signal reconstruction

    Publication Year: 1989, Page(s):592 - 595 vol.1
    Cited by:  Papers (1)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (196 KB)

    The authors propose a new speech modification algorithm using short-time Fourier transform (STFT) synthesis. This algorithm is developed using the criterion that the mean-square-error signals between the STFT spectra of the estimated and the modified should be minimized. The most important and unique ideas of the algorithm are liftering that passes all cepstra except cepstra in the pitch frequency... View full abstract»

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