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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 4 • Date August 1974

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Displaying Results 1 - 15 of 15
  • [Front cover and table of contents]

    Page(s): 0
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    Freely Available from IEEE
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  • [Back cover]

    Page(s): c4
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    Freely Available from IEEE
  • A class of generalized continuous orthogonal transforms

    Page(s): 245 - 254
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    This paper presents a class of generalized continuous transforms for the orthogonal decomposition of signals. Base functions for the continuous transform range from Walsh functions of order two to stair-like functions which resemble approximations to sinusoids and which are distinct from the generalized Walsh functions. Standard desirable properties which are shown to hold for the generalized continuous transform operator include orthogonality of the base functions, linearity of the transform operator, inverse transformability, and admissibility to fast transform representation. The transform class is governed by a definition of time translation in terms of signed-bit dyadic time shift. Mathematical properties leading to this definition are discussed and the impact of the definition is assessed. Properties of the continuous class of generalized transforms make feasible analysis which could be extremely tedious using matrix representations of the operations actually mechanized in a sampled-data system. Analysis techniques are illustrated with a target detection system which is conceptually designed using the generalized continuous transform and implemented using fast transform algorithms to perform correlation operations. Since the correlation operations are valid for inputs which include signals represented in terms of Walsh functions, the example illustrates one instance in which the binary Fourier representation (BIFORE) transform can be used for practical pattern recognition. View full abstract»

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  • Sequential least-squares Fourier estimation with fading memory

    Page(s): 300 - 303
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    Necessary and sufficient conditions are developed which insure finite fading memory least-square Fourier coefficient estimates from time and frequency domain data are equal. The implementation of such filters are also presented. View full abstract»

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  • Interpolation, extrapolation, and reduction of computation speed in digital filters

    Page(s): 231 - 235
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    Any digital filter can be decomposed into two basic subsets, an extrapolator the output of which is sampled at a frequency depending only on the filter bandwidth and an interpolator delivering the filtered signal at the imposed output sampling rate. Redundancy in extrapolator and interpolator is removed by introducing half-band nonrecursive filtering elements for which definition, performance figures and efficient implementation are supplied. They reduce significantly the necessary computation and storage at the cost of a slight group delay increase. A formula is given for the amount of multiplications to be carried out every second in a filter; it depends on the filter bandwidth, signal to distortion ratio, and input-output sampling rate. The method is extended to recursive filters and a comparison is made with existing techniques of implementing digital filters for the needs in computation and storage hardware: a specific example of design underlines the reduction in computation speed achieved in practice through this method, which brings digital filters in a most favorable position for their competition against analog filters in many application fields. View full abstract»

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  • Consonant durations in clusters

    Page(s): 282 - 295
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    The durations of initial and final consonant clusters in monosyllabic and bisyllabic words within a frame sentence were studied from spectrograms of readings by 3 speakers. The durations of consonants within a cluster varied with the features of the consonant and its phonetic environment, such as voicing and point and manner of articulation. A durational model was proposed based on two mechanisms. An articulatory mechanism was attributed to effects involving coarticulation and restrictions in the motion of the tongue and lips during a cluster. Shortening of consonants in clusters seemed to arise from the shorter distances that the articulators travel in clusters. Lengthening of consonants before fricatives and voiced consonants and aspiration effects were noted. The other factor was a phonological mechanism, related to the use of duration as an acoustic cue in consonant perception. Single consonants varying only in the voicing characteristic had substantial durational differences which could aid in distinguishing them. However, phonological restrictions, such as common voicing among stops and fricatives, arise in the clusters, and the redundancies allow the durational differences to become less. View full abstract»

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  • A parallel arithmetic hardware structure for recursive digital filtering

    Page(s): 255 - 258
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    A flexible, low-cost hardware structure is described for implementing recursive digital filters. Either a parallel or cascade arrangement of any number of second-order filter sections may be implemented for each of the input channels. Although the internal arithmetic is fixed-point, filter coefficients may be represented in floating-point form to reduce coefficient roundoff error. View full abstract»

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  • Fixed-point truncation arithmetic implementation of a linear prediction autocorrelation vocoder

    Page(s): 273 - 282
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    The implementation of the Linear Prediction Auto-correlation Vocoder (LPAV) with fixed-point truncation arithmetic is presented. For a speech bandwidth of up to 3.3 kHz, it is possible to implement LPAV with a maximum of 16-bit fixed-point arithmetic, thus allowing for very efficient software simulation on minicomputers or economical hardware implementation on a special purpose computer. The resultant degradation with respect to a floating-point implementation is negligible. The three most critical system components are discussed in detail along with various tradeoffs such as effects on system stability as a function of preemphasis and sampling frequency. The results are also applicable to general finite word length digital filter implementation problems. View full abstract»

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  • On piecewise-linear basis functions and piecewise-linear signal expansions

    Page(s): 263 - 268
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    A set of piecewise-linear (PL) basis functions for signal or functional decomposition is introduced. These basis functions provide a PL approximation to the signal and an a priori determination of the required number for a finite term expansion to achieve a certain pointwise approximation error. Moreover, the expansion coefficients are linear combinations of samples of the function to be expanded and are virtually trivial to determine. View full abstract»

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  • Unequal bandwidth spectral analysis using digital frequency warping

    Page(s): 236 - 244
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    The application to unequal bandwidth and vernier spectrum analysis of a technique referred to as digital frequency warping is discussed. In this technique a sequence is transformed in such a way that the Fourier transforms of the original and transformed sequences are related by a nonlinear transformation of the frequency axis. An equal bandwidth analysis carried out on the transformed sequence then corresponds to an unequal bandwidth analysis of the original sequence. A comparison is presented between the bandwidth as a function of frequency for the digital warping technique and proportional bandwidth analysis. An analysis of the effects of finite register length in implementing digital frequency warping is also presented. View full abstract»

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  • On a signal detection problem and the Hadamard transform

    Page(s): 296 - 297
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    A signal detection scheme of moderate complexity is implemented using the Hadamard transform. The performance of the scheme is between those of a data editing system and a matched filter. The scheme is very simple to implement, requires much less computation than a matched filter and could be of considerable value when large amounts of input data are being processed. View full abstract»

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  • The thermo-acoustic properties of fibrous material

    Page(s): 297 - 300
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    The compressibility of air inside a loudspeaker enclosure can be increased if the enclosure can be made isothermal. Stuffing fibrous material into an enclosure tends to make it isothermal since the material acts as a heat sink. In order to evaluate the effectiveness of filling material in loudspeaker enclosures, the thermal time response of fibrous materials is calculated. The time response is found to depend on the density of the material and the radius of the fibers. The frequency response of a typical fiberglass material is presented. View full abstract»

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  • Waveform synthesis by means of random searching

    Page(s): 259 - 262
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    This paper presents a random search computer design technique that is shown to be suitable to the solution of a class of waveform design problems. The class of problems considered is that class in which only the phase angles associated with a fixed power spectrum ensemble of sinusoids are varied so that the sum of the sinusoids approximates some desired waveform. Two illustrative peak-factor design problems are considered in order to demonstrate use of the random search, which is seen to have the capability for both adaptive search radius and adaptive direction. View full abstract»

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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