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Acoustics, Speech, and Signal Processing, IEEE International Conference on ICASSP '84.

Date 19-21 March 1984

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Displaying Results 1 - 25 of 538
  • Proceedingss ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing [front cover and table of contents]

    Publication Year: 1984 , Page(s): 0
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    Freely Available from IEEE
  • Synthesis by rule of english intonation patterns

    Publication Year: 1984 , Page(s): 77 - 80
    Cited by:  Papers (5)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (125 KB)  

    This papet reports work on synthesizing English F0 contours. One motivation for this work is to improve the naturalness and liveliness of the prosody in speech synthesis systems. However, our main goal is to develop a theory of the dimensions of variation controlling intonation, and of their interaction. View full abstract»

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  • A new digital voice summing technique for teleconferencing

    Publication Year: 1984 , Page(s): 116 - 119
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (58 KB)  

    The objectives of this paper are to investigate various speech coding techniques to determine their applicability to voice conferencing, to present a new technique for summing directly from the encoded signals, and to demonstrate the audio results and effectiveness of the proposed voice summing technique. View full abstract»

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  • A speech direction finder

    Publication Year: 1984 , Page(s): 128 - 131
    Cited by:  Papers (3)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (71 KB)  

    The speech direction finder described here is a relatively simple device based on an off the shelf microcomputer. It can provide the direction to a talker to within 3 degrees of azimuth angle on a single spoken syllable, will only respond to speech, and when used with Wallace linear array microphones can provide this at distances of 50 feet or more. There are numerous applications for the device which may enhance the quality of audio and video teleconferences. View full abstract»

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  • A convergence analysis of an adaptive underwater passive tracking system

    Publication Year: 1984 , Page(s): 206 - 209
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (83 KB)  

    The ability of the adaptive filtering system to converge to an unbiased estimate cf those target parameters of interest such as range and depth is examined. Passive target measurements make use of difference in signal arrival time between geometrically separated sensor systems such as those described in Knapp and Carter (1976), Hassab and Boucher (1976), Hassab (1976), McCabe and Moose (1981). While generally good results of different simulated tracking scenarios have been reported upon in Moose (1983), and Moose and Dailey (1983) these results are valid only for the geometries that were specifically simulated. Thus a theoretical investigation is necessary to examine filter convergence after an initial target detection or target maneuver has occurred. Due to the complexity of the nonlinear data generation and tracking system shown for the vertical plane, and not shown, though very similar for range and bearing in the horizontal ocean plane the convergence analysis is part analytic and part computer analysis. Preliminary results show that the tracking systems converge, but converge with a small bias that is both geometry and signal to noise ratio dependent. View full abstract»

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  • Square root normalized feedback ladder algorithm for the identification of moving average systems

    Publication Year: 1984 , Page(s): 236 - 239
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (107 KB)  

    We have presented a square root normalized version of the feedback ladder algorithm for the identification of the parameters of a moving average model. The number of equations needed is reduced from eight in the unnormalized case to just five. The complexity of the equations increases but the procedure is justified because it seems to lead to a more convenient hardware realization. Moreover, this realization would be completely similar (for the backward and forward residuals lines) to the CORDIC processors implementation already proposed for the feedlorward ladder algorithms (FFLA). A possible disadvantage is that three of the variables used may have magnitudes greater than one. However the essential feature of the FBLA, that of being able to read out directly the estimated coefficients of the -monic-polynomial model is not modified. View full abstract»

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  • An information theoretic approach to the automatic determination of phonemic baseforms

    Publication Year: 1984 , Page(s): 304 - 307
    Cited by:  Papers (27)  |  Patents (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (344 KB)  

    We present a method for predicting the phonemic baseform of a word given its spelling and a sample utterance. We have developed a procedure, based on information theoretic principles, for discovering a set of spelling-to-sound rules automatically. We have also constructed a simple phoneme recognizer. The spelling-to-sound rules predict the correct phoneme 93.7% of the time. When coupled with the phoneme recognizer, we are able to predict the correct phoneme 97.1% of the time. View full abstract»

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  • Some properties of a family of generalized time-limited window functions

    Publication Year: 1984 , Page(s): 436
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (21 KB)  

    For use in the design of finite impulse response (FIR) digital filters via the window function method the first integral of the window is required in order to relate transition width, filter order, and maximum passband and stopband error values. Again this relationship for the generalized family is found to be nearly linear if maximum error is measured in logarithmic units. Approximate empirical expressions are given for these relationships. Thus one can now design FIR filters with controlled error concentrated to any prescribed degree near to the band edges. Convenient computation methods for the generalized window functions are also described as well as the location of zeros and maxima and minima of their transforms. View full abstract»

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  • Modern, active sonar AGC design considerations

    Publication Year: 1984 , Page(s): 530 - 533
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (67 KB)  

    The following identifies key elements of the AGC design problem for active sonars. Because the character of the background noise and of the echo are highly dependent on a sonar environment that varies from place to place and from hour to hour, the signal statistics are unknown. Researchers have left the area of AGC design to practitioners; such designs are guided usually by heuristics. This paper follows that tradition. It postulates a microprocessor-controlled gain controller that adapts its parameters to the sonar environment. Since the sonar for which this AGC was designed is still in development, the performance of the AGC is evaluated with the aid of a simulation. View full abstract»

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  • On redefining the optimal least squares filter under floating point operations

    Publication Year: 1984 , Page(s): 594 - 597
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (75 KB)  

    A novel solution on approximation to the least squares filter problem under floating point arithmetic is presented for a linear stochastic model. View full abstract»

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  • VLSI Architecture for signal processing with alternate low-level primitive structures (ALPS)

    Publication Year: 1984 , Page(s): 688 - 691
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (108 KB)  

    A set of Alternate Low-Level Primitive Structures (ALPS) has been considered in this context. It is envisaged that each standalone structure consists of an input queue, an output queue, the processing primitive, and mechanisms for control and synchronization. Some of these primitives and a new system architecture, which allows orderly VLSI/VHSIC transition are described. View full abstract»

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  • ICASSP '84. IEEE International Conference on Acoustics, Speech, and Signal Processing [Back cover]

    Publication Year: 1984 , Page(s): c4
    Save to Project icon | Request Permissions | PDF file iconPDF (419 KB)  
    Freely Available from IEEE
  • Applications of singular value decomposition to system modeling in signal processing

    Publication Year: 1984 , Page(s): 208 - 211
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (112 KB)  

    We consider the evaluation of the order of a linear system transfer function represented as an AR or an ARMA model based on the use of the singular value decomposition technique for the efficient determination of the rank of a matrix. Inputs to the system are modeled as binary-valued random data and outputs of the system are observed in the presence of uncorrelated noises. Results are obtained for the case when the relevant statistics given in autocorrelation and crosscorrelation values are assumed to be available as well as the case when the required statistics are computed explicitly from the sequence sample values. Various numerical examples are considered and shown to be more efficient than the Woodside determinant ratio approach. View full abstract»

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  • The analysis of non-oscillating transients using the covariance method with many different orders

    Publication Year: 1984 , Page(s): 212 - 215
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (89 KB)  

    We discuss the use of the covariance method for the analysis of transients, in particular to determine the number and time constants of decaying exponentials. We propose a new technique, involving the repeated use of the covariance method with many different orders, all larger than the the actual number of terms in the transient, and a comparison the results as a good means for the estimation of the true values of the parameters of interest. View full abstract»

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  • A new spectrum analysis approach using autocorrelation technique and MEM

    Publication Year: 1984 , Page(s): 216 - 219
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (78 KB)  

    This paper describes a new technique for estimating the frequency of a sinusoidal signal immersed in broad-band noise. The technique is eminently suitable for identifying a single spectral peak in broad-band noise when the signal-to-noise ratio is very poor. This situation is a characteristic of a phased detected pulse doppler radar return in an ECM environment, or a very weak target signal immersed in broad-band receiver noise. Computer simulation results are included. The results show that for a single sinusoidal signal in a background of additive broad-band noise, this new technique is supperior in performance when compared with the conventional MEM doppler processor. View full abstract»

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  • Coherent recombination of sediment borne and water path acoustic signals

    Publication Year: 1984 , Page(s): 307 - 310
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (91 KB)  

    Both the water layer and bottom sediment act as transmission media for acoustic signals. In April 1981, the Marine Physical Laboratory (MPL) conducted an experiment investigating acoustic signal propagation in thick sediments. In this paper, the coherence between sediment borne and water path acoustic signals will be discussed. Data from the Monterey Fan experiment will be presented along with comments regarding the viability of coherently recombining the total acoustic field which is available at the receiving array. View full abstract»

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  • Optimal Lloyd-Max quantization of LPC speech parameters

    Publication Year: 1984 , Page(s): 29 - 32
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (176 KB)  

    This paper introduces an approach to scalar quantization of LPC reflection coefficients which outperforms typical scalar quantization schemes using a squared-error distortion measure. As in vector quantization, an iterative algorithm is used to generate the source codebook. Scalar quantization produces a greater distortion, for a given number of bits- than vector quantization but reduced computation time and storage requirements make the Lloyd-Max approach economically attractive for implementation in real-time hardware. The algorithm is developed and a comparison with another scalar quantization scheme is made. View full abstract»

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  • The influence of Doppler effect on information transmission in sound channel

    Publication Year: 1984 , Page(s): 311 - 314
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (65 KB)  

    The received signal average power P' = αP is established for the sound channel with Doppler effect. The capacity for the AWGN Shannon channel with Doppler effect and the achievable rate region for the AWGN Multiple-Access Channel with Doppler effect are studied. The asymmetry in information transmission efficiency for two-way sound communication with Doppler effect is discussed. View full abstract»

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  • Spectra using data distribution and covariance modelling

    Publication Year: 1984 , Page(s): 642 - 645
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (128 KB)  

    The autoregressive and Prony methods for spectral estimation do not make full use of the statistics of the additive noise. These statistics are included in the following as constraints on the solution. The finite number of data samples imposes a limit on the resolution thus making it possible to approximate random signals by a set of deterministic signals which can be modelled in terms of a finite set of time-invariant parameters, θ. The non-Toeplitz covariance calculated from the data is modelled here using the θ parameters. Statistical constraints and covariance modelling are combined to produce non-linear methods for spectral estimation. The resulting higher resolution requires high computational complexity; this can often be substantially reduced by using knowledge-based techniques. View full abstract»

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  • Use of computational psychoacoustical models in speech processing: Coding and objective performance evaluation

    Publication Year: 1984 , Page(s): 33 - 36
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    The primary objective of this study was to determine what benefit could be gained in speech coding by using a psychoacoustical frequency scale instead of a linear scale. To partially overcome the well known difficulties in objective speech quality measurements, a computational performance criterium based on psychoacoustical models was developed. Several Finnish phonemes were then coded using regular LPC and LPC computed on a psychoacoustically correct frequency scale (Bark scale)and the coding performance of these both methods was tested via computational performance tests. The results indicate a significant improvement in speech quality for the same bit-rate, when applying LPC on psycho-acoustical frequency scale. Preliminary listening tests support both the better coding capability of the Bark LPC compared to the regular LPC and the reliability of the developed speech quality criterium as an objective performance evaluation method. View full abstract»

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  • Adaptive image smoothing algorithms for edge and texture preservation

    Publication Year: 1984 , Page(s): 287 - 290
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (136 KB)  

    A class of reasonably simple and efficient adaptive algorithms are developed to enhance noise degraded images while preserving the edge and texture information. Techniques from one-dimensional adaptive signal processing and systems identification are extended and applied to two-dimensional image smoothing through proper modelling of the image. Both the AR and ARMA models are treated. The conceptual separation of "image causality" and "processing causality" is advocated. Advanced topics that are covered include: faster computation algorithms, simultaneous contrast stretching and smoothing, etc. View full abstract»

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  • Image segmentation using spatial linear prediction

    Publication Year: 1984 , Page(s): 670 - 673
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (184 KB)  

    This paper discusses the use of spatial linear prediction to detect region borders in aerial photographs and thus segment the image into regions of differing natural terrain. The algorithm is motivated by a significance test which under certain assumptions and approximations can be expressed as the weighted sum of error residuals of two-dimensional (2-D) linear prediction. Examples are presented to illustrate the performance of the algorithm. View full abstract»

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  • Median filters: Analysis for 2 dimensional recursively filtered signals

    Publication Year: 1984 , Page(s): 187 - 190
    Cited by:  Papers (5)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (184 KB)  

    Median filtering is a nonlinear technique for smoothing signals. In this paper we find the output distribution of recursively median filtered two dimensional signals with additive impulsive noise. We study the edge jitter effect that recursive median filtering introduces. Finally some examples are included illustrating these results. View full abstract»

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  • A new method for wideband sensor array processing

    Publication Year: 1984 , Page(s): 181 - 184
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (176 KB)  

    A computationally efficient sensor array processing method has been developed for detection and bearing estimation of wideband planewave sources above or below a planar array of sensors. The method uses the time-averaged covariances of undelayed sensor signals to estimate the wavenumber spectrum in the array plane. This spectrum contains directional information as well as wideband temporal frequency characteristics of the planewave sources. The usefulness of this method for an acoustic aircraft tracking problem is demonstrated with experiments on real data. View full abstract»

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  • Quantization reduction for evaluating laser gyro performance using a moving average filter

    Publication Year: 1984 , Page(s): 614 - 617
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (94 KB)  

    Ring Laser gyro performance is typically evaluated by measuring the random walk in angle associated with each instrument over a range of operating and environmental conditions. However, the ring laser gyro, being basically a digital sensor, outputs accumulated angle in quantized pulses. This quantization in the presence of dither has a similar effect on gyro test data as white noise in angle. To reduce the effect of quantization and dither on gyro test data, so as to enable the random walk coefficient of the instrument to be determined to high precision in short times, a high speed Moving Average Filter (1 kHz) was developed which outputs accumulated gyro data samples once per second. The quantization noise power is reduced by a factor greater than 512 and the dither is reduced by a factor greater than 200 db while the random walk characteristic is virtually unaffected. The following describes the operation of this Moving Average Filter, analyses its effect on gyro test data and presents test results. View full abstract»

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