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Acoustics, Speech, and Signal Processing, 1990. ICASSP-90., 1990 International Conference on

Date 3-6 April 1990

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  • ICASSP 90. 1990 International Conference on Acoustics, Speech and Signal Processing (Cat. No.90CH2847-2)

    Publication Year: 1990
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    Freely Available from IEEE
  • DSP specification using the Silage language

    Publication Year: 1990, Page(s):1056 - 1060 vol.2
    Cited by:  Papers (28)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (256 KB)

    The Silage language is presented, together with a number of extensions to the original design which result in a powerful specification language and design environment for complex digital signal processing (DSP) systems. Aspects of Silage discussed include the applicative language, timing information and time-domain operations, data typing, and functional language, pragmatic directives, and loops a... View full abstract»

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  • Results on the application of simulated annealing algorithm for the design of digital filters with powers-of-two coefficients

    Publication Year: 1990, Page(s):1301 - 1304 vol.3
    Cited by:  Papers (3)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (296 KB)

    Results on the application of the simulated annealing (SA) algorithm to the problem of finding the coefficients of a digital filter with very coarse coefficients values (namely, power-of-two) are presented. For a minimax criterion in the frequency response, the algorithm was found particularly useful for designing a cascade-form FIR (finite impulse response) filter whose performance is known to be... View full abstract»

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  • A dynamical systems approach to speech processing

    Publication Year: 1990, Page(s):365 - 368 vol.1
    Cited by:  Papers (20)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (344 KB)

    An approach to speech processing, based on nonlinear dynamical systems, is presented. It is shown that two fundamental problems in speech processing, dimensionality reduction and nonlinear temporal variability, can be addressed using geometrical methods from nonlinear dynamics. An effective dynamical system is extracted by training a nonlinear predictor of the signal samples. A variety of signal c... View full abstract»

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  • The Frazier-Jawerth transform

    Publication Year: 1990, Page(s):2483 - 2486 vol.5
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (276 KB)

    The Frazier-Jawerth transform (FJT), a simple yet rigorous method for the time-frequency analysis of nonstationary signals, is discussed. The FJT is related to the wavelet transform. Both the FJT and the wavelet methods improve upon the classical methods of time-frequency analysis due to Wigner and Gabor. A tutorial introduction to the FJT is given, and a brief discussion is presented of an exampl... View full abstract»

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  • On mask selection for time-varying filtering using the Wigner distribution

    Publication Year: 1990, Page(s):2487 - 2490 vol.5
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (288 KB)

    The application of a multiplicative modification function, or mask, to the Wigner distribution (WD) of a signal has been proposed as an implementation of time-varying filtering in the time-frequency plane. Because the least squares synthesis technique involves an eigenvalue/eigenvector decomposition, the effects of mask size, shape, and placement on the reconstructed system output are not well und... View full abstract»

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  • Robust pitch determination via SVD based cepstral methods

    Publication Year: 1990, Page(s):253 - 256 vol.1
    Cited by:  Papers (9)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (296 KB)

    A greatly enhanced cepstral-based pitch estimator which uses the MUSIC algorithm for estimating background noise characteristics is presented. This approach couples the signal enhancement capabilities of MUSIC, which is based on singular value decomposition (SVD) orthogonalization, with the harmonic spectrum estimation capabilities of the cepstrum. Marked improvements are demonstrated over standar... View full abstract»

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  • Effect of floating-point error reduction with recursive least square for parallel architecture

    Publication Year: 1990, Page(s):1487 - 1490 vol.3
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (248 KB)

    A finite wordlength floating-point error analysis for the RLS (recursive least squares) algorithm using UD factorization is presented. It is known that this algorithm is suitable for parallel architectures; however, it is not known how the wordlength of the algorithm should be chosen for hardware implementation. An investigation is conducted of the relation between the word length and the converge... View full abstract»

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  • Pitch predictors with high temporal resolution

    Publication Year: 1990, Page(s):661 - 664 vol.2
    Cited by:  Papers (43)  |  Patents (31)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (336 KB)

    A first-order pitch predictor is described whose delay is specified as an integer number of samples plus a fraction of a sample at the current sampling rate. This realization has a better performance than conventional multiple coefficient predictors and leads to more efficient coding of the predictor parameters. Also discussed is the application of noninteger delay pitch predictors to low-bit-rate... View full abstract»

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  • A novel multi-stage estimation of signal parameters

    Publication Year: 1990, Page(s):2491 - 2494 vol.5
    Cited by:  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (304 KB)

    A novel multistage estimation scheme is presented for estimating the parameters of a received carrier signal possibly phase-modulated by unknown data, and experiencing very high Doppler, Doppler rate, etc. Such a situation arises, for example, in the case of the Global Positioning System (GPS), where the signal parameters are directly related to the position, velocity, acceleration, and jerk of th... View full abstract»

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  • Transient behavior of the LMS algorithm: a deterministic approach

    Publication Year: 1990, Page(s):1491 - 1494 vol.3
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (296 KB)

    The dynamic behavior of the LMS (least mean square) algorithm is analyzed for an important class of nonstationary desired and input signals. Differential equation analysis is used to model the LMS algorithm and predicts in closed form the performance for signal sets that are members of the exponential family, such as damped sinusoids and phasors. The analysis provides new insight into the use of a... View full abstract»

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  • Direction finding using an interpolated array

    Publication Year: 1990, Page(s):2951 - 2954 vol.5
    Cited by:  Papers (27)  |  Patents (24)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (328 KB)

    A direction-finding technique is developed which uses the outputs of a virtual array computed from the real array using a linear interpolation procedure. The geometry of the virtual array is under the control of the designer. Using a linear virtual array, an extension of the root-MUSIC algorithm to arbitrary array geometries is developed View full abstract»

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  • Architecture and performance of a new arithmetic unit for the computation of elementary functions

    Publication Year: 1990, Page(s):1783 - 1786 vol.3
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (416 KB)

    The design of a recursive arithmetic unit for evaluation of elementary functions is discussed. Two distinct classes of binary algorithms are implemented. One class is based on a generalization of the compensated CORDIC method. The other class, which replaces the linear CORDIC case, is based on the convergence transformations in connection with multiple bit encoding techniques. The proposed hardwar... View full abstract»

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  • Finite state CELP for variable rate speech coding

    Publication Year: 1990, Page(s):37 - 40 vol.1
    Cited by:  Papers (13)  |  Patents (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (352 KB)

    A finite-state code excited linear prediction (CELP) system is proposed for variable-rate speech coding. The encoding system consists of a number of CELP coders with different linear predictive coding parameter quantization patterns, code book sizes, and population densities. The selection of the encoding state for each input vector depends on the input signal characteristics, the desired bit rate... View full abstract»

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  • DSP/C: a standard high level language for DSP and numeric processing

    Publication Year: 1990, Page(s):1065 - 1068 vol.2
    Cited by:  Papers (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (300 KB)

    A set of nonproprietary extensions is introduced to the American National Standards Institute (ANSI) C programming language (HLL) for digital signal processing (DSP) and numeric applications. The problems of using C as it exists today for DSP applications are defined, and solutions are proposed. This work is an interpretation of the work being done by the Numeric C Extensions Group (NCEG), a worki... View full abstract»

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  • A functional silicon compiler for high speed FIR digital filters

    Publication Year: 1990, Page(s):1329 - 1332
    Cited by:  Papers (12)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (216 KB)

    A functional compiler system for the implementation of high-speed finite impulse-response (FIR) digital filters on gate-array ICs is presented. The system is capable of implementing complex digital filters directly from frequency-domain specifications. Fast turnaround and sample rates in excess of 100 MHz are achieved by using a combination of architectural optimization and advanced 0.8-μm BiCM... View full abstract»

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  • A model block-training method for HMM-based speech recognition systems

    Publication Year: 1990, Page(s):541 - 544 vol.1
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (280 KB)

    A model block-training method for hidden Markov models (HMMs) is described. It combines several model estimations from several training sets, all of which are derived from utterances of the same word, into a new one. Although the recognition rate of the recognizer trained by the block-training method is lower than that of a recognizer trained by a batch-training method, the new one is much more fl... View full abstract»

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  • An efficient code structure and search strategy for stochastic coding at 8 kb/s

    Publication Year: 1990, Page(s):481 - 484 vol.1
    Cited by:  Papers (4)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (456 KB)

    Methods of increasing the performance of two components of stochastic coding, the codebook excitation search and the long-term synthesis filter, while maintaining or reducing complexity are discussed. A flexible tree-coding technique is introduced that significantly lowers the complexity of the codebook search with very little loss in quality. An efficient method of performing a three-tap long-ter... View full abstract»

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  • A new nesting scheme of PFA [prime factor algorithm]

    Publication Year: 1990, Page(s):1495 - 1498 vol.3
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (300 KB)

    A nesting scheme type of prime factor algorithm (PFA) is introduced. It takes advantage of both the PFA and the Winograd Fourier transform algorithm (WFTA) by developing a new nesting scheme and modifying the small-N discrete Fourier transform (DFT) algorithms. This new nesting scheme will not expand the data in the nesting multiplication part. It requires far fewer multiplications and fewer addit... View full abstract»

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  • Fast converging subband acoustic echo cancellation using RAP on the WE DSP16A

    Publication Year: 1990, Page(s):1141 - 1144 vol.2
    Cited by:  Papers (5)  |  Patents (19)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (324 KB)

    The application of row action projections (RAPs) to the problem of acoustic echo cancellation is discussed. It is shown how RAPs, a subset of projections onto convex sets (POCSs) and a generalization of LMS, can be used to further improve the tracking ability of subband acoustic echo canceller (SBAECs). In addition, the RAP algorithm has a useful complexity versus speed-of-convergence tradeoff. Th... View full abstract»

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  • Time-frequency concentrated basis functions

    Publication Year: 1990, Page(s):2459 - 2462 vol.5
    Cited by:  Papers (12)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (340 KB)

    The notion of concentration is discussed, and concentration with respect to an operator is defined. This is related to the idea of concentration based on second moments in time and frequency. A class of signals Bw(μ) concentrated with respect to an operator W is introduced. A best n-dimensional approximating subspace for this class is shown to be the space ... View full abstract»

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  • Pulsed noise in self-sustained oscillations of musical instruments

    Publication Year: 1990, Page(s):1157 - 1160 vol.2
    Cited by:  Papers (3)  |  Patents (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (252 KB)

    A theory of bow and breath noise generation has been tested by analyzing recorded cello tones and by simulation using physical models of the cello and clarinet. For the synthesis to be successful, the listener's perceptions of noise and sound must fuse. Evidence is presented that the noise must be pulse modulated in a period synchronous way, as is shown for voiced fricatives. A method is described... View full abstract»

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  • Adaptive least square complex lattice clutter rejection filters applied to the radar detection of low altitude windshear

    Publication Year: 1990, Page(s):1469 - 1472 vol.3
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (240 KB)

    Addresses the use of an adaptive complex clutter rejection filter for reducing the ground clutter bias in pulse-pair mean windspeed estimates of weather return data collected by low altitude airborne pulsed Doppler radar. A complex form of the square root normalized recursive-least-squares lattice estimation algorithm, which models the ground clutter as a low-order autoregressive process, is used ... View full abstract»

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  • ORION: a two pass hybrid system for isolated-words automatic speech recognition

    Publication Year: 1990, Page(s):41 - 44 vol.1
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (452 KB)

    A system, called ORION, which deals with speaker-independent automatic speech recognition (ASR) for isolated words is proposed. ORION is a two-pass hybrid system which uses several types of knowledge. This knowledge applies to psychoacoustics, physiology, and phonetics. During the first pass an auditory model, the perceptually based linear prediction analysis, combines static and dynamic features ... View full abstract»

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  • Narrow band analysis of a filter bank for the directional decomposition of images

    Publication Year: 1990, Page(s):1739 - 1742 vol.3
    Cited by:  Papers (5)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (340 KB)

    The case where there are a large number of directional components is investigated. In particular, the limiting case where the directional components are reduced to 1-D signals is studied. These 1-D signals are related to projections taken through the image at various angular orientations. The projection-slice theorem is invoked in order to develop the relationship between the directional filter ba... View full abstract»

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