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 (15)  |  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 (13)
    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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  • Effects of ultrasonic components on perceived tone quality

    Publication Year: 1989, Page(s):1997 - 2000 vol.3
    Cited by:  Papers (2)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (200 KB)

    A psychophysical experiment was carried out to study the effects of ultrasonic components on perceived tone qualities. The physical attribute of the experiment was the decay rate of the envelope of the ultrasonic components, and its psychological attributes were the distinctness, brightness, and loudness of tones. It is shown that slower decay tends to make brightness higher and loudness lower. It... View full abstract»

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  • Speaker adaptation for large vocabulary speech recognition systems using speaker Markov models

    Publication Year: 1989, Page(s):5 - 8 vol.1
    Cited by:  Papers (10)  |  Patents (60)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (320 KB)

    An alternative approach to speaker adaptation for a large-vocabulary hidden-Markov-model-based speech recognition system is described. The goal of this investigation was to train the IBM speech recognition system with only five minutes of speech data from a new speaker instead of the usual 20 minutes without the recognition rate dropping by more than 1-2%. The approach is based on the use of a sto... View full abstract»

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  • Perceptual transform coding of wideband stereo signals

    Publication Year: 1989, Page(s):1993 - 1996 vol.3
    Cited by:  Papers (19)  |  Patents (59)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (200 KB)

    The author reports on the extension of the entropy-coded perceptual transform coder (ECPTC) to stereo signals. The stereo ECPTC exploits both the redundancy in the stereo signal and the effects of acoustic mixing in the listening environment in order to encode the stereo signal at a bit rate much less than twice that for a monophonic signal. This coder uses the perceptual threshold developed by th... View full abstract»

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  • Large vocabulary word recognition based on demi-syllable hidden Markov model using small amount of training data

    Publication Year: 1989, Page(s):1 - 4 vol.1
    Cited by:  Papers (6)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (280 KB)

    The authors present a large-vocabulary speech recognition method based on hidden Markov models (HMMs) and aimed at high recognition performance with a small amount of training data. The recognition model is designed to treat contextual and allophonic variations utilizing acoustic-phonetic knowledge. The demisyllable is used as a recognition unit to treat contextual variations caused by the coartic... 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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  • A general-purpose hearing aid prescription, simulation and testing system

    Publication Year: 1989, Page(s):1989 - 1992 vol.3
    Cited by:  Papers (2)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (364 KB)

    The authors describe a comprehensive, integrated, microcomputer-based facility for research on hearing aid fitting and for clinical use. The system first applies formal prescription rules to audiometric measures that have been obtained from an individual patient to predict which hearing aid amplification function is likely to provide the optimal benefit for a hearing-impaired listener. It then des... 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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  • An interpolative spatial domain technique for coding image sequences

    Publication Year: 1989, Page(s):1917 - 1920 vol.3
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (296 KB)

    The authors identify the main mechanism through which bits are consumed in the RBN (recursive binary nesting) coding scheme and determine ways in which this can be reduced. To this end, various forms of local motion-compensation are implemented and described, but the results are found to be disappointing. Different quantizers are then considered, but again the improvements were minimal. A scaled q... View full abstract»

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  • Speech intelligibility and MTF in non-exponential decay fields

    Publication Year: 1989, Page(s):1985 - 1988 vol.3
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (184 KB)

    SI (speech intelligibility) tests were performed in nonexponentially-decaying reverberant fields using both Japanese monosyllables and Dutch PB words. It is shown that Si depends mainly on the initial portion of the reverberant energy decay curve (REDC). The authors also compared SI scores with the MTF (modulation transfer function), and it is demonstrated that the initial portion of REDC governs ... View full abstract»

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  • Phase coherence in speech reconstruction for enhancement and coding applications

    Publication Year: 1989, Page(s):207 - 210 vol.1
    Cited by:  Papers (13)  |  Patents (11)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (312 KB)

    It has been shown that an analysis-synthesis system based on a sinusoidal representation leads to synthetic speech that is essentially perceptually indistinguishable from the original. A change in speech quality has been observed, however, when the phase relation of the sine waves is altered. This occurs in practice when sine waves are processed for speech enhancement and for speech coding. A desc... 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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  • Image sequence coding at 64 kbps using vector quantization and block matching

    Publication Year: 1989, Page(s):1913 - 1916 vol.3
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (388 KB)

    The authors propose algorithms for compressing monochrome and color image sequences depicting videotelephone/videoconference scenery down to 64 kb/s. This rate is of interest as it is one of the primary rates within the ISDN (integrated services digital network) network architecture. Emphasis is given to synchronous transmission strategies in which successive frames are updated at a fixed bit rate... View full abstract»

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  • A study of English word category prediction based on neutral networks

    Publication Year: 1989, Page(s):731 - 734 vol.2
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (244 KB)

    Using traditional statistical approaches, it is difficult to develop an N-gram word prediction model for constructing an accurate word recognition system because of the increased demand for sample data and parameters to memorize probabilities. To solve this problem, NETgrams, which are neural networks for N-gram word category prediction in text are proposed. NETgrams can easily b... View full abstract»

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  • A multistage projection structure for multidimensional signal detection

    Publication Year: 1989, Page(s):1484 - 1487 vol.3
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (212 KB)

    A multistage projection detection structure that for one example is demonstrated to yield significantly higher sensitivity than single-stage projection is proposed. Instead of projecting many samples onto one sample in a single step, the projection is done in stages. Prior to each stage, filtering is done to enhance the signal-to-noise ratio as much as possible. The more signal samples there are i... View full abstract»

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  • Multiprocessor implementation of an ATC audio codec

    Publication Year: 1989, Page(s):1981 - 1984 vol.3
    Cited by:  Papers (1)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (184 KB)

    A source coding scheme for high-quality audio signals based on adaptive transform coding is presented. Using the well known ATC algorithm of R. Zelinski and P. Noll (1977) as a starting point, the authors have developed an algorithm that features psychoacoustic weighting and a joint bit allocation to stereo channels. The real-time implementation uses a four-node processor network, each node consis... View full abstract»

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  • Limited-angle reconstruction from noisy data using clustering of the solution space

    Publication Year: 1989, Page(s):1516 - 1519 vol.3
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (240 KB)

    The authors propose a stabilized method for limited angle reconstruction. The algorithm provides a framework that makes it possible to incorporate prior information and to account for the noise; this framework is the reason for the method's stability. The method entails the following steps: a space of reconstruction solutions is generated, either analytically or by Monte Carlo simulation; represen... 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 (3)  |  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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