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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 1 • Date February 1980

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Displaying Results 1 - 25 of 25
  • [Front cover and table of contents]

    Publication Year: 1980 , Page(s): 0
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    Freely Available from IEEE
  • Correction to "Two-dimensional digital filters with no overflow oscillations"

    Publication Year: 1980 , Page(s): 117
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  • [Back cover]

    Publication Year: 1980 , Page(s): c4
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  • Time-frequency representation of digital signals and systems based on short-time Fourier analysis

    Publication Year: 1980 , Page(s): 55 - 69
    Cited by:  Papers (170)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (2504 KB)  

    This paper develops a representation for discrete-time signals and systems based on short-time Fourier analysis. The short-time Fourier transform and the time-varying frequency response are reviewed as representations for signals and linear time-varying systems. The problems of representing a signal by its short-time Fourier transform and synthesizing a signal from its transform are considered. A new synthesis equation is introduced that is sufficiently general to describe apparently different synthesis methods reported in the literature. It is shown that a class of linear-filtering problems can be represented as the product of the time-varying frequency response of the filter multiplied by the short-time Fourier transform of the input signal. The representation of a signal by samples of its short-time Fourier transform is applied to the linear filtering problem. This representation is of practical significance because there exists a computationally efficient algorithm for implementing such systems. Finally, the methods of fast convolution age considered as special cases of this representation. View full abstract»

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  • On the implementation of a short-time spectral analysis method for system identification

    Publication Year: 1980 , Page(s): 69 - 78
    Cited by:  Papers (10)  |  Patents (2)
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    Recent work has demonstrated the utility of a short-time spectral analysis approach to the problems of spectral estimation and system identification. In this paper several important aspects of the implementation are discussed. Included is a discussion of the computational effects (e.g., storage, running time) of the various analysis parameters. A computer program is included which illustrates one implementation of the method. View full abstract»

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  • Reduction of quantization noise in PCM speech coding

    Publication Year: 1980 , Page(s): 107 - 110
    Cited by:  Papers (3)
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    A new technique to reduce the effect of quantization noise in PCM speech coding is proposed. The procedure consists of using dither noise to ensure that the quantization errors can be modeled as additive signal-independent noise, and then reducing this noise through the use of a noise reduction system. The procedure is illustrated with examples. View full abstract»

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  • Bias in magnitude-squared coherence estimation due to misalignment

    Publication Year: 1980 , Page(s): 97 - 99
    Cited by:  Papers (13)
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    This correspondence reviews the interpretation of the coherence function and provides a derivation for bias of coherence estimates due to misalignment. The derived expression is compared with previous results. It is shown that subsequent processing of underwater acoustic signals from a recent experiment corroborates the derived results for coherence estimation bias. View full abstract»

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  • A note on variable recursive digital filters

    Publication Year: 1980 , Page(s): 111 - 112
    Cited by:  Papers (12)
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  • Two-dimensional recursive filter design--A spectral factorization approach

    Publication Year: 1980 , Page(s): 16 - 26
    Cited by:  Papers (30)
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    This paper concerns development of an efficient method for the design of two-dimensional (2-D) recursive digital filters. The specific design problem addressed is that of obtaining half-plane recursive filters which satisfy prescribed frequency response characteristics. A novel design procedure is presented which incorporates a spectral factorization algorithm into a constrained, nonlinear optimization approach. A computational implementation of the design algorithm is described and its design capabilities demonstrated with several examples. View full abstract»

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  • A parameter estimation approach to time-delay estimation and signal detection

    Publication Year: 1980 , Page(s): 8 - 16
    Cited by:  Papers (73)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1504 KB)  

    Present techniques that estimate the difference in arrival time between two signals corrupted by noise, received at two separate sensors, are based on the determination of the peak of the generalized cross correlation between the signals. To achieve good resolution and stability in the estimates, the input sequences are first weighted. Invariably, the weights are dependent on input spectra which are generally unknown and hence have to be estimated. By approximating the time shift as a finite impulse response filter, estimation of time delay becomes one of determination of the filter coefficients. With this formulation, a host of techniques in the well-developed area of parameter estimation is available to the time-delay estimation problem-with the possibilities of reduced computation time as compared with present methods. In particular, it is shown that the least squares estimation of the filter coefficients is equivalent to estimating the Roth processor. However, the parameter estimation approach is expected to have a smaller variance since it avoids the need for spectra estimation. Indeed, experimental results from two examples show that the Roth processor, found by least squares parameter estimation, has a smaller variance than the approximate maximum likelihood estimator of Hannan-Thomson where spectral estimation is required. A detector that uses the sum of the estimated parameters as a test statistic is also given, together with its receiver operating characteristics. View full abstract»

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  • On the use of the I0-sinh window for spectrum analysis

    Publication Year: 1980 , Page(s): 105 - 107
    Cited by:  Papers (23)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (624 KB)  

    Closed form expressions for main-lobe width, modified main-lobe width, and relative sidelobe amplitude are given for the I_{0}- \sinh window function. These formulas facilitate exploring the tradeoff between record length, spectral resolution, and leakage in digital spectrum analysis. An especially simple empirical approximation relating main-lobe width and relative sidelobe amplitude is given. View full abstract»

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  • Audibility of transient intermodulation distortion

    Publication Year: 1980 , Page(s): 91 - 96
    Cited by:  Papers (1)
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    The threshold of audibility of TIM distortion was determined for 68 subjects representing all categories of listeners, from musicians and sound engineers to the average man on the street. Three different music samples were used, and controlled amounts of distortion were produced with a digital stereophonic TIM generator. Two basic experimental methods were used to obtain the approximate threshold, after which the reliability of the detection of the distorted passages was verified with a time localization test. The results show that the audibility varies very much depending on the music sample, listening media, and person. The most sensitive group of listeners could reliably perceive 0.5 percent of momentary TIM. Low values of TIM were generally perceived only as slight changes in the tonal character of the sound, and not as distortion. In a number of cases, a preference was found for the slightly distorted sound. View full abstract»

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  • Generating covariance sequences and the calculation of quantization and rounding error variances in digital filters

    Publication Year: 1980 , Page(s): 102 - 104
    Cited by:  Papers (25)
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    A linear algorithm is given for the generation of covariance sequences for rational digital filters using numerator and denominator coefficients directly. There is no need to solve a Lyapunov equation or to solve for the residues of a spectrum, as in other methods. By appealing to certain results from the theory of inners, we show that the algorithm provides a unique solution, provided only that the filter is stable. Our results may be used to compute error variances due to product rounding and signal quantization, and to generate covariance strings {r_{k}}\min{0}\max {K} used in other studies involving second-order properties of digital filters. View full abstract»

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  • Low-noise realizations for narrow-band recursive digital filters

    Publication Year: 1980 , Page(s): 41 - 54
    Cited by:  Papers (10)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1936 KB)  

    Agarwal and Burrus have proposed two all-pole digital filter structures which exhibit very small error for poles near the unit circle. ROM-accumulator (ROM/ACC) implementations of these structures have also been shown to have small error. We extend these resuits by including zeros in both the Agarwal-Burrus structures and their ROM/ACC counterparts. A modified approach toward scaling and selecting various wordlengths is required. Expressions for MSE and bounds on zero input limit cycles are derived for the new structures and shown to be small. As a consequence, the new pole-zero structures can offer significant hardware and speed advantages as illustrated by example. View full abstract»

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  • Acoustic imaging with electronic circuits

    Publication Year: 1980 , Page(s): 119 - 120
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  • A fast cosine transform in one and two dimensions

    Publication Year: 1980 , Page(s): 27 - 34
    Cited by:  Papers (103)  |  Patents (5)
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    The discrete cosine transform (DCT) of an N-point real signal is derived by taking the discrete Fourier transform (DFT) of a 2N-point even extension of the signal. It is shown that the same result may be obtained using only an N-point DFT of a reordered version of the original signal, with a resulting saving of 1/2. If the fast Fourier transform (FFT) is used to compute the DFT, the result is a fast cosine transform (FCT) that can be computed using on the order of N \log _{2} N real multiplications. The method is then extended to two dimensions, with a saving of 1/4 over the traditional method that uses the DFT. View full abstract»

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  • A weighted overlap-add method of short-time Fourier analysis/Synthesis

    Publication Year: 1980 , Page(s): 99 - 102
    Cited by:  Papers (66)  |  Patents (21)
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    In this correspondence we present a new structure and a simplified interpretation of short-time Fourier synthesis using synthesis windows. We show that this approach can be interpreted as a modification of the overlap-add method where we inverse the Fourier transform and window by the synthesis window prior to overlap-adding. This simplified interpretation results in a more efficient structure for short-time synthesis when a synthesis window is desired. In addition, we show how this structure can be used for analysis/synthesis applications which require different analysis and synthesis rates, such as time compression or expansion. View full abstract»

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  • Recognition of unaspirated plosives--A statistical approach

    Publication Year: 1980 , Page(s): 85 - 91
    Cited by:  Papers (6)
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    In this paper the results of a study of the computer recognition of unaspirated plosives in commonly used polysyllabic words uttered by three different informants are presented. The onglide transitions of the first two formants and their durations have been found to be an effective set of features for the recognition of unaspirated plosives. The rates of transition of these two formants as a feature set have been found to be significantly inferior to the features mentioned earlier. The maximum likelihood method, under the assumption of a normal distribution for the feature set, provides an adequate tool for classification. The assumption of both intergroup and intragroup independence of the features reduces recognition scores. A prior knowledge of target vowels is found necessary for attaining reasonable efficiency. A prior knowledge of voicing manner improves classification efficiency to some extent. The physiological factors responsible for the variation of the recognition score for the various plosives are discussed. For labials and velars the recognition score is very high, nearly 90 percent. An attempt to correlate the dynamics of tongue-body motion with the variations in recognition scores has been made. Back vowels as targets have been found to give improved classification of the preceding consonants. A comparison of the result of machine recognition with those of published results on perception tests has been included. The results are found to be of the same order. View full abstract»

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  • An efficient processor for multistage decimation and filtering

    Publication Year: 1980 , Page(s): 1 - 7
    Cited by:  Papers (1)
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    An efficient processor for multistage decimation and filtering, utilizing the same IIR digital filter coefficients for all stages, but having a different cutoff frequency at each decimation stage, is presented along with its algorithmic implementation. The upper bound on the decimation ratio per stage is shown to be inversely proportional to the sum of two parameters governing the magnitude response of the IIR low-pass digital filter, while the upper bound on the overall decimation ratio is shown to be proportional to a decimation index. The lower bound on the number of stages is also presented. An example utilizing the proposed algorithm to decimate and filter human colonic electrical control activity is presented. View full abstract»

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  • Signal-to-noise losses associated with hardware-efficient designs of moving target indicators

    Publication Year: 1980 , Page(s): 35 - 40
    Cited by:  Papers (1)
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    In many radar systems the signal processor contains a moving target indicator which is implemented by either an infinite or finite impulse response digital filter. The finite impulse response configuration is most often used, and there are two major implementation alternatives: the moving window (or sliding window) and the fixed window (or block window) methods. The hardware requirements and the signal-to-noise characteristics for various implementations of moving and fixed window moving target indicators are determined and compared. It is demonstrated that the moving window configuration produces correlation of noise samples which is not present in fixed window systems. Equations are given that relate the effective number of outputs from these systems when the same number of input samples are used. It is shown that a square-law detector, or a linear detector, does not alter the relative performance of the moving and fixed window moving target indicator systems. A tabulated result is given that compares hardware (i.e., memories, adders, multiplexers, multipliers, multiplication speed) for the moving and fixed window systems. One significance of this effort is that it allows a tradeoff to be made between the better signal-to-noise improvement for the moving window system and the less complex hardware requirements of the fixed window system. View full abstract»

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  • Digital processing of speech signals

    Publication Year: 1980 , Page(s): 118 - 119
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  • On the design of recursive low-pass digital filters

    Publication Year: 1980 , Page(s): 79 - 84
    Cited by:  Papers (9)
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    The magnitude-squared characteristic of a low-pass filter is approximated, over the finite interval [-1, 1 ], by the ratio φ(x)/[φ(x) + P(x)] of two polynomials. For elliptic filter design, a special case, the polynomials φ(x) and P(x) (of the same order) are chosen such that the ratios P(x)/φ(x) and φ(x)/P(x) approximate, in the Chebyshev sense, the zero function over the passband [xp, 1] and stopband [-1, xs], respectively. The passband and stopband form two disjoint intervals. The polynominals are determined by repeated applications of Darlington's technique for obtaining a rational function generalization of Chebyshev polynominals. View full abstract»

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  • A new delay line for discrete analog-signal-processing devices based on IC monostable multivibrators

    Publication Year: 1980 , Page(s): 112 - 114
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    This correspondence proposes a new delay line for the implementation of discrete analog-signal processing. One stage of the delay line can be constructed by two IC monostable multivibrators (MM's) and one IC AND gate without using analog memory elements. The delay line is suitable especially for the applications in the low-frequency range, where charge transfer devices (CTD's) cannot operate. View full abstract»

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  • The relationship between image restoration by the maximum a posteriori method and a maximum entropy method

    Publication Year: 1980 , Page(s): 114 - 117
    Cited by:  Papers (9)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1384 KB)  

    For the linear imaging model g = Hf + n, the maximum a posteriori (MAP) restoration method is compared to the maximum entropy (ME) method defined by maximizing f_{T} \ln f subject to \parallel g - Hf\parallel^{2} = \parallel n\parallel^{2} . It is shown that the ME solution is a member of the set of MAP solutions defined by a set of a priori probability densities. The numerical methods developed for MAP restoration can be applied to ME restoration. The importance of the a priori probability distribution for the MAP restoration is demonstrated. Examples of ME restoration with the new method are shown and compared to previous results. View full abstract»

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  • An adaptive lattice algorithm for recursive filters

    Publication Year: 1980 , Page(s): 110 - 111
    Cited by:  Papers (52)  |  Patents (2)
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    The purpose of this correspondence is to introduce an adaptive algorithm for recursive filters, which are implemented via a lattice structure. The motivation for doing so is that stability can be achieved during the adaptation process. For convenience, the corresponding algorithm is referred to as an "adaptive lattice algorithm" for recursive filters. Results pertaining to using this algorithm in a system-identification experiment are also included. 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