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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 2 • Date April 1976

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Displaying Results 1 - 18 of 18
  • [Front cover and table of contents]

    Publication Year: 1976 , Page(s): 0
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    Freely Available from IEEE
  • Structures for evaluating the discrete Fourier transform on staggered blocks

    Publication Year: 1976 , Page(s): 128 - 131
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (347 KB)  

    In this work three structures are presented for evaluating in real time the discrete Fourier transform (DFT) on successive sections of a sampled signal. Each section consists of N elements and two successive sections are staggered of M \leq N elements. Each structure evaluates N/M DFT coefficients at every step and requires about (N/(2M)) \log _{2} 4M multipliers. View full abstract»

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  • [Back cover]

    Publication Year: 1976 , Page(s): c4
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    Freely Available from IEEE
  • Automatic generation of voiceless excitation in a vocal cord-vocal tract speech synthesizer

    Publication Year: 1976 , Page(s): 163 - 170
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1896 KB)  

    A speech synthesis technique is described which incorporates acoustic models for sound propagation in a tube with yielding walls, turbulent noise generation at locations of constricted volume flow in the vocal tract, and the self-oscillatory properties of the vocal cord source. This formulation frees the experimenter from a traditional limitation, namely, the assumption of linear separability of sound source and resonant system. As a consequence, new opportunities accrue for building realistic physiological characteristics into the synthesizer. These built-in characteristics represent information that need not be overtly supplied to control the synthesizer. The system is used to synthesize test syllables from controls which are stylized models of articulation and connected speech from controls automatically derived from printed text. The synthesis technique demonstrates the feasibility of generating all speech sounds (voiced, unvoiced, nasal) from a common set of physiologically based control parameters, as the human does. View full abstract»

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  • A general language-operated decision implementation system (GLODIS): Its application to continuous-speech segmentation

    Publication Year: 1976 , Page(s): 137 - 162
    Cited by:  Papers (10)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (2096 KB)  

    The general language-operated decision implementation system (GLODIS) represents a flexible, operating-system approach to the generation and implementation of complex rules for decision making in pattern recognition. GLODIS is briefly described from a general mathematical and philosophical point of view; a current implementation is described in the context of a phonemic-level segmenter for continuous speech. This segmenter is presented in sufficient detail for duplication by others, not only for speech segmentation but also for alternate applications of a similar nature. Performance data are given for a large amount ( 8frac{1}{2} min) of continuous speech, for a currently running version of the speech segmentation GLODIS. Recent results from a total continuous speech recognition system, which incorporates the above, are also given. View full abstract»

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  • Procedures for computing the discrete Fourier transform on staggered blocks

    Publication Year: 1976 , Page(s): 132 - 137
    Cited by:  Papers (4)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (528 KB)  

    In this work the problem of evaluating successively the discrete Fourier transform (DFT) on ordered sets of N elements staggered of M is considered. Three procedures for solving such a problem are given, of which two are recursive and one nonrecursive. The complexity of each procedure, in number of complex multiplications, is about (N/2) \log _{2} 4M . View full abstract»

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  • Speech recognition experiments with linear predication, bandpass filtering, and dynamic programming

    Publication Year: 1976 , Page(s): 183 - 188
    Cited by:  Papers (64)  |  Patents (14)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (648 KB)  

    Automatic speech recognition experiments are described in which several popular preprocessing and classification strategies are compared. Preprocessing is done either by linear predictive analysis or by bandpass filtering. The two approaches are shown to produce similar recognition scores. The classifier uses either linear time stretching or dynamic programming to achieve time alignment. It is shown that dynamic programming is of major importance for recognition of polysyllabic words. The speech is compressed into a quasi-phoneme character string or preserved uncompressed. Best results are obtained with uncompressed data, using nonlinear time registration for multisyllabic words. View full abstract»

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  • Some preliminary experiments in the recognition of connected digits

    Publication Year: 1976 , Page(s): 170 - 182
    Cited by:  Papers (13)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1304 KB)  

    This paper describes an implementation of a speaker independent system which can recognize connected digits. The overall recognition system consists of two separate but interrelated parts. The function of the first part of the system is to segment the digit string into the individual digits which comprise the string; the second part of the system then recognizes the individual digits based on the results of the segmentation. The segmentation of the digits is based on a voiced-unvoiced analysis of the digit string, as well as information about the location and amplitude of minima in the energy contour of the utterance. The digit recognition strategy is similar to the algorithm used by Sambur and Rabiner [1] for isolated digits, but with several important modifications due to the impreciseness with which the exact digit boundaries can be located. To evaluate the accuracy of the system in segmenting and recognizing digit strings a series of experiments was conducted. Using high-quality recordings from a soundproof booth the segmentation accuracy was found to be about 99 percent, and the recognition accuracy was about 91 percent across ten speakers (five male, five female). With recordings made in a noisy computer room the segmentation accuracy remained close to 99 percent, and the recognition accuracy was about 87 percent across another group of ten speakers (five male, five female). View full abstract»

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  • An approach towards a synthesis-based speech recognition system

    Publication Year: 1976 , Page(s): 194 - 196
    Cited by:  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (360 KB)  

    Preliminary results are presented of experiments with a recognition scheme intended for continuous speech. The scheme utilizes information about interphoneme contextual effects contained in formant transitions and employs internal trial synthesis and feedback comparison as a means for recognition. The aim is to achieve minimal sensitivity to the appreciable variability which occurs in the speech signal, even for utterances of a single speaker. While the approach outlined here is quite general, it has initially been tried out on vowel-stop-vowel utterances. Recognition scores obtained are encouraging and demonstrate the viability of the approach. View full abstract»

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  • Digital filtering by polyphase network:Application to sample-rate alteration and filter banks

    Publication Year: 1976 , Page(s): 109 - 114
    Cited by:  Papers (186)  |  Patents (24)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (592 KB)  

    The digital filtering process can be achieved by a set of phase shifters with suitable characteristics. A particular set, named polyphase network, is defined and analyzed. It permits the use of recursive devices for efficient sample-rate alteration. The comparison with conventional filters shows that, with the same active memory, a reduction of computation rate approaching a factor of 2 can be achieved when the alteration factor increases. A more substantial gain can be obtained in the direct realization of a uniform bank of recursive filters through combination of the polyphase network with a discrete Fourier transform (DFT) computer; savings in hardware also result from the low sensitivity of the structure to coefficient word lengths. View full abstract»

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  • Modifications to formant tracking algorithm of april 1974

    Publication Year: 1976 , Page(s): 192 - 193
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (208 KB)  

    This correspondence describes an improved version of the linear prediction formant tracking algorithm which was described in [1]. The new algorithm, like the original one, applies continuity constraints and branches out from an anchor point in the middle of each vowel. The changes are that the initial estimates for the formant frequencies at the anchor are determined more carefully, and that the option of choosing a new anchor point near the original one is allowed if the original one caused problems. View full abstract»

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  • Two-dimensional spectral factorization with applications in recursive digital filtering

    Publication Year: 1976 , Page(s): 115 - 128
    Cited by:  Papers (132)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1464 KB)  

    The concept of spectral factorization is extended to two dimensions in such a way as to preserve the analytic characteristics of the factors. The factorization makes use of a homomorphic transform procedure due to Wiener. The resulting factors are shown to be recursively computable and stable in agreement with one-dimensional (1-D) spectral factorization. The factors are not generally two-dimensional (2-D) polynomials, but can be approximated as such. These results are applied to 2-D recursive filtering, filter design, and a computationally attractive stability test for recursive filters. View full abstract»

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  • Bounds on zero-input limit cycles in all-pole digital filters

    Publication Year: 1976 , Page(s): 189 - 192
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    New bounds on the limit-cycle oscillations of a fixed-point second-order digital filter are found. The new bounds are smaller, for some values of the coefficients, than the presently available bounds. View full abstract»

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  • Pruning the decimation in-time FFT algorithm

    Publication Year: 1976 , Page(s): 193 - 194
    Cited by:  Papers (50)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (240 KB)  

    Significant time-saving can be achieved by a simple modification to the radix-2 decimation in-time fast Fourier transform (FFT) algorithm when the data sequence to be transformed contains a large number of zero-valued samples. The time-saving is accomplished by replacing M - L stages of the FFT computation with a simple recopying procedure where 2Mis the total number of points to be transformed of which only 2Lare nonzero. View full abstract»

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  • Benchmark papers in electrical engineering--Volume 12, digital filters and the fast Fourier transform

    Publication Year: 1976 , Page(s): 198
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    First Page of the Article
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  • Utterance classification confidence in automatic speech recognition

    Publication Year: 1976 , Page(s): 188 - 189
    Cited by:  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (224 KB)  

    A variety of automatic speech recognition experiments have been executed that support a measure of confidence for utterance classification. The confidence measure tested was the ratio of the two best "Hamming distance" scores obtained in matching utterance templates with an unknown utterance. The results show that it is possible to reliably predict when the utterance classifier has made the correct decision. View full abstract»

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  • Signal processing: Discrete spectral analysis, detection, and estimation

    Publication Year: 1976 , Page(s): 197
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    First Page of the Article
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  • Applications of Walsh functions and sequency theory

    Publication Year: 1976 , Page(s): 196 - 197
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    First Page of the Article
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Aims & Scope

This Transactions ceased production in 1990. The current retitled publication is IEEE Transactions on Signal Processing.

Full Aims & Scope