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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 3 • Date June 1974

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Displaying Results 1 - 14 of 14
  • [Front cover and table of contents]

    Publication Year: 1974 , Page(s): 0
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    Freely Available from IEEE
  • Editorial

    Publication Year: 1974 , Page(s): 163
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    Freely Available from IEEE
  • [Back cover]

    Publication Year: 1974 , Page(s): c4
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    Freely Available from IEEE
  • Time-domain design of frequency-sampling digital filters for pulse shaping using linear programming techniques

    Publication Year: 1974 , Page(s): 180 - 185
    Cited by:  Papers (12)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (664 KB)  

    The use of time-domain specifications is demonstrated for designing frequency-sampling digital filters that generate data pulses for transmission over an ideal band-limited channel. Desired characteristics of the transmitted pulse are used to formulate the set of constraint equations and objective function used in linear programming to obtain an optimum set of filter coefficients, i.e., frequency samples {| H(k) |} - The constraints are the amplitudes assigned to the set of regularily spaced samples taken from the pulse. The objective function is either to minimize the maximum absolute error between desired and generated pulse samples over the specified pulse duration or to provide zero crossings in the transmitted pulse with near-zero slope in order to protect against intersymbol interference (ISI) due to timing error (jitter). View full abstract»

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  • A technique for precision estimation in analog cross-spectrum measurements

    Publication Year: 1974 , Page(s): 186 - 188
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    The precision of an analog cross-spectrum measurement can be estimated if the product of the two signals is integrated by a voltage-to-frequency converter having separate pulse outputs for positive and negative inputs. The difference between the two pulse counts gives the spectrum reading, while their ratio is used in estimating the precision. Coherence may also be estimated in this way. View full abstract»

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  • A programming system for studies in speech synthesis

    Publication Year: 1974 , Page(s): 217 - 225
    Cited by:  Papers (3)
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    This paper describes a speech synthesis system which is particularly suitable for experimental investigations. The synthesis is accomplished in two stages. The concatenation stage generates a schematized spectrographic representation corresponding to the symbolic input. The second stage consists in generating the corresponding acoustic signal. The steady state characterization of each phoneme is supplied as data. Independent concatenation procedures incorporate context dependent effects such as format transitions, changes in the normal duration of vowels, etc. The parameter values for these procedures are obtained by a set of rules. Applicability of a rule is determined by attributes assigned to the phonemes. The phonemes are divided into classes and subclasses by the attribute assignment. The attribute STOP, for instance, defines the class of all stop consonants and BILABIAL STOP would define the set/p, b, m/. Thus, a rule specifies a parameter value when a subclass of phonemes occur, in the context of another subclass. Such a formulation considerably reduces the number of rules. The classification as well as the rules are supplied as data to the system, giving it considerable flexibility. The spectrographic output of the concatenation stage is used to actuate a simulated series terminal analog synthesizer. Rudimentary prosodics are incorporated which modify a monotonous pitch contour with stress markers and interrogative or declarative termination of a sentence. View full abstract»

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  • An improved fast Fourier transform

    Publication Year: 1974 , Page(s): 226 - 227
    Cited by:  Papers (1)
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    This correspondence presents an improved implementation of the fast Fourier transform without sorting. View full abstract»

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  • Statistical design of nonrecursive digital filters

    Publication Year: 1974 , Page(s): 188 - 196
    Cited by:  Papers (42)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1032 KB)  

    The problem of designing a finite duration impulse response (FIR) digital filter to approximate a desired spectral response is treated in this paper. The philosophy adopted is that for a given FIR filter structure, the filter coefficients can be designed to provide a minimum mean-squared error (MMSE) estimate of a random signal sequence (the design-signal) imbedded in a random noise sequence. By treating the signal and noise covariance functions as design parameters, one can design FIR filters with spectral responses that approximate the power spectral density of the design-signal. For signal processing applications that require some attention to signal fidelity, as well as noise rejection, the MMSE philosophy seems appropriate (as opposed to a maximum signal-to-noise ratio philosophy, for example). Several practical designs are presented that emphasize the simplicity of the design technique and illustrate the selection of design parameters. The designs show quite dramatically that the MMSE design technique can be competitive with existing low-pass and bandpass design techniques. Finally, considerable attention is given to an efficient Toeplitz matrix inversion algorithm that permits rapid inversion of the covariance matrices that arise in the MMSE design. The resulting computation times for the design of high-order filters (N = 128, e.g.) appear to be shorter than computation times for competing algorithms. View full abstract»

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  • Picture processing using one-dimensional implementations of discrete planar filters

    Publication Year: 1974 , Page(s): 164 - 173
    Cited by:  Papers (7)
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    The problem of implementing a two-dimensional recursive filter as a one-dimensional recursive filter is examined. The results of the present paper show that the exact one-dimensional implementation of a planar recursive filter is a time-varying filter. However, planar filters may be approximated by one-dimensional time-invariant recursive filters. The frequency response, stability, and storage requirements of the approximate filters are derived and illustrated. View full abstract»

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  • An on-line speech intelligibility measurement system

    Publication Year: 1974 , Page(s): 203 - 206
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (528 KB)  

    A speech intelligibility measurement system is described that uses a computer to administer the test, record the listener response, and automatically evaluate it on-line. This makes the intelligibility testing conditions uniform at all times and the test more efficient compared to conventional methods. The test words are presented in random scramblings by using a shuffling algorithm. The listener's response is entered via a graphic tablet used as a "software keyboard." The response evaluation is done based on the similarity of sounds and not of spellings. The system is being used in the development and evaluation of analysis-synthesis type of speech compression systems and for identifying perceptually important parameters from the linear prediction model of speech. Adaption of this system to various speech perception experiments is also discussed. View full abstract»

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  • An APL program for bilinear transformations

    Publication Year: 1974 , Page(s): 225 - 226
    Cited by:  Papers (5)
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    An efficient APL [1], [2] computer program is presented for the calculation of the bilinear transformations Z = (S + 1)/(S - 1) or S = (Z + 1)/(Z - 1) using the Q -matrix of Fielder [6]. The program utilizes the extremely efficient matrix manipulation and computational properties afforded by the APL language. View full abstract»

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  • Positivity and stability tests for multidimensional filters (discrete-continuous)

    Publication Year: 1974 , Page(s): 174 - 180
    Cited by:  Papers (42)
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    In this paper, a systematic procedure to test for stability of three-dimensional filters (discrete and continuous) is presented. The test is based on repeated applications of an extended Hermite or Schur-Cohn formulation, and use of Sturm's theorem to determine the content of a system of polynomial inequalities in a single indeterminate. The need for generating a constructive algorithm for stability tests for higher than three-dimensional filters using Tarski's generalization of Sturm's theorem is discussed. Application of certain combinatorial rules for transforming the multidimensional digital filter problem to the multidimensional continuous filter problem or vice versa) is made. View full abstract»

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  • Design method for stable second-order digital filters

    Publication Year: 1974 , Page(s): 196 - 202
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (744 KB)  

    The impulse-invariant equivalent digital versions of second-order bandpass analog filters are stable only for coefficients within specific ranges. The quantization of these coefficients may thus cause an originally stable filter to become unstable; the sensitivity of the digital filter to such changes is found to depend on the sampling frequency. Analytic and graphical design techniques are developed for determining the range of sampling frequency that, with a predetermined register length for the coefficients, renders the digital equivalent of a tuned circuit always stable. A comparison is also made between instability caused by quantization of coefficients and limit cycles resulting from the rounding of products. View full abstract»

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  • A spectral-flatness measure for studying the autocorrelation method of linear prediction of speech analysis

    Publication Year: 1974 , Page(s): 207 - 217
    Cited by:  Papers (37)  |  Patents (5)
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    The purpose of this paper is to introduce a spectral-flatness measure into the study of linear prediction analysis of speech. A spectral-flatness measure is introduced to give a quantitative measure of "whiteness," of a spectrum. It is shown that maximizing the spectral flatness of an inverse filter output or linear predictor error is equivalent to the autocorrelation method of linear prediction. Theoretical properties of the flatness measure are derived, and compared with experimental results. It is shown that possible ill-conditioning of the analysis problem is directly related to the spectral-flatness measure and that prewhitening by a simple first-order linear predictor to increase spectral flatness can greatly reduce the amount of ill-conditioning. 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