Scheduled Maintenance on December 18, 2017:
IEEE Xplore will undergo system maintenance from 1:00 - 5:00 PM EST. During this time there may be intermittent impact on performance. We apologize for any inconvenience.

International Conference on Acoustics, Speech, and Signal Processing,

23-26 May 1989

Filter Results

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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Speech enhancement based upon hidden Markov modeling

    Publication Year: 1989, Page(s):353 - 356 vol.1
    Cited by:  Papers (8)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (316 KB)

    A maximum a posteriori approach for enhancing speech signals which have been degraded by statistically independent additive noise is proposed. The approach is based upon statistical modeling of the clean speech signal and the noise process using long training sequences from the two processes. Hidden Markov models (HMMs) with mixtures of Gaussian autoregressive (AR) output probability distributions... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Adaptive algorithms for tracking roots of spectral polynomials

    Publication Year: 1989, Page(s):1162 - 1165 vol.2
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (216 KB)

    The authors propose two fast adaptive algorithms, namely Newton's gradient algorithm and the modified Rayleigh-quotient adaptive algorithm. These methods work in association with adaptive eigensubspace algorithms for tracking the zeros of a nonstationary spectrum polynomial. Newton's gradient algorithm is developed under a linearly constrained minimization procedure, whereas the modified Rayleigh-... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Filtering of colored noise for speech enhancement and coding

    Publication Year: 1989, Page(s):349 - 352 vol.1
    Cited by:  Papers (7)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (256 KB)

    A report is presented on experiments using a colored-noise assumption Kalman filter to enhance speech additively contaminated by colored noise, such as helicopter noise and jeep noise, with a particular application to linear predictive coding (LPC) of noisy speech. The results indicate that the colored-noise Kalman filter provides a significant gain in SNR, a clear improvement in the sound spectro... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Polynomial factorization algorithms for adaptive root estimation

    Publication Year: 1989, Page(s):1158 - 1161 vol.2
    Cited by:  Papers (5)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (252 KB)

    The authors present two new polynomial factorization algorithms suitable for use in adaptive signal processing applications where it is required to track the movements of roots. Their distinguishing feature is that they provide methods for updating the roots optimally (and efficiently) in response to coefficient perturbations. This is useful, for example, for online estimation and tracking of time... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Spectral quantization and interpolation for CELP coders

    Publication Year: 1989, Page(s):69 - 72 vol.1
    Cited by:  Papers (43)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (340 KB)

    The authors present results on the comparative performance of nonuniform scalar quantizers using three different LPC (linear predictive coding) representations: the arcsine of reflection coefficients, the log area ratios, and the line spectral frequencies. On comparing the spectral distortion introduced by quantizers based on these representations, it was found that the average distortion was very... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Matrix fast match: a fast method for identifying a short list of candidate words for decoding

    Publication Year: 1989, Page(s):345 - 348 vol.1
    Cited by:  Papers (5)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (220 KB)

    A rapid method is presented for identifying a short list of candidate words that match well with some acoustic input to serve as a fast matching stage in a large-vocabulary speech recognition system that uses hidden Markov models and maximum a posteriori decoding. Given hidden Markov models for all the words in the vocabulary the authors derive a class of algorithms that are faster than a detailed... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Time-varying filter modelling of the sound field due to a moving source and time-delay estimation

    Publication Year: 1989, Page(s):2105 - 2108 vol.4
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (224 KB)

    In many measurement situations sources are in motion, and the source motion can significantly degrade the correlator output since the time delay varies during the correlator integration time. The authors propose a novel way of modeling the sound field of moving sources in multipath and/or multisensor measurement environments using a concept that has been termed the covariance-equivalent model. The... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Temporal decomposition: a framework for enhanced speech recognition

    Publication Year: 1989, Page(s):655 - 658 vol.1
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (228 KB)

    Short term spectral analysis of source-filter modeling gives a parameterized description of the acoustic signal in terms of a sequence of vectors. These parameter vectors change slowly with time corresponding to a slowly moving vocal tract. The authors consider a model (temporal decomposition) that approximates the time variation by a set of target vectors and interpolation functions that overlap ... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Time-frequency domain adaptive filters

    Publication Year: 1989, Page(s):1154 - 1157 vol.2
    Cited by:  Papers (4)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (212 KB)

    A linear system representation called the double convolution model, which combines the impulse response and transfer function, is presented. It is based on Fourier transformation by block of the impulse response. Its main advantage is that the model order and the window size are not imposed by the response length. This property makes the adaptation of systems with long impulse response easier beca... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Formant extraction from Fourier transform phase

    Publication Year: 1989, Page(s):484 - 487 vol.1
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (200 KB)

    A method of extracting formant information from the short-time Fourier transform phase spectrum of speech is proposed. Fourier transform phase has not been used for formant extraction because it appears to be noisy and difficult to interpret. The effects of wrapping of phase (due to zeros close to the unit circle and the linear phase component) make it difficult to derive useful information. The a... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Implementing algorithms for convolution on arrays of adders

    Publication Year: 1989, Page(s):1127 - 1130 vol.2
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (216 KB)

    The authors consider the problem of developing VLSI signal processors for computing convolutions. Convolutions can be efficiently computed by VLSI processors that consist of arrays of adders when they are stated in terms of matrices with elements consisting of only 1, 0, or -1. Unfortunately, when stated in matrix form the published algorithms have matrices with elements other than 1, 0, or -1. Th... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Speech enhancement for hearing aids using adaptive beamformers

    Publication Year: 1989, Page(s):1322 - 1325 vol.2
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (172 KB)

    The author presents two techniques based on adaptive noise canceling to suppress the ambient noise in hearing aids. The first technique uses two microphones and gives very good results in the case of one noise source. If two noise sources are situated at the same side of the head, there is also noise suppression, although less effective than with only one noise source. In the second technique the ... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Analysis/synthesis systems in the presence of quantization

    Publication Year: 1989, Page(s):1341 - 1344 vol.2
    Cited by:  Papers (6)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (236 KB)

    The authors have experimented with a variety of analysis/synthesis systems in the presence of quantization. The systems were block discrete Fourier transform (DFT), block DFT with trapezoidal windowing, block discrete cosine transform, Hanning windowed short-time Fourier transform (STFT) oversampled by a factor of two, Kaiser windowed STFT oversampled by a factor of four, quadrature mirror filter ... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Speech coding with time-varying bit allocations to excitation and LPC parameters

    Publication Year: 1989, Page(s):65 - 68 vol.1
    Cited by:  Papers (18)  |  Patents (8)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (208 KB)

    The authors explore the benefits of time-varying bit allocation to excitation and LPC (linear predictive coding) parameters for the case of codebook-excited LPC. The overall bit rate in the experiment was 4.8, 6.4, or 8.0 kb/s. In each case, permissible bit rates for the LPC component were 0, 24, 36, or 48 bits per frame, one of which was selected for each speech frame using a brute-force search m... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Image sequence coding using interframe VDPCM and motion compensation

    Publication Year: 1989, Page(s):1858 - 1861 vol.3
    Cited by:  Papers (1)  |  Patents (15)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (296 KB)

    The techniques of motion detection, interframe linear block prediction and vector quantization have been incorporated into a scheme for encoding monochrome image sequences displaying moderate motion. Data transmission rate reduction is accomplished by identifying and processing only those regions of the image that exhibit noticeable changes between successive frames, by estimating the magnitude of... View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.