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ICASSP '79. IEEE International Conference on Acoustics, Speech, and Signal Processing

2-4 April 1979

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Displaying Results 1 - 25 of 243
  • [Front cover and table of contents]

    Publication Year: 1979, Page(s): 0
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    Freely Available from IEEE
  • Coefficient inaccuracy in FIR filters

    Publication Year: 1979, Page(s):375 - 377
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (61 KB)

    Coefficient inaccuracy in FIR filters is known to degrade the frequency response particularly in stopband regions. The magnitude of the stopband degradation increases with the number of stages N, the length of the impulse response. A widely used formula for the error in frequency response is proportional tosqrt{N}. Recently, we have found that for random additive coefficient errors with... View full abstract»

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  • On state-space signal processing with application to image enhancement

    Publication Year: 1979, Page(s):646 - 649
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (230 KB)

    First Page of the Article
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  • New transformed variables for designing recursive digital filters

    Publication Year: 1979, Page(s):801 - 804
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (90 KB)

    In this paper we discuss the use of linear transformations on z+1/z in designing magnitude squared functionsH(z)H(frac{1}{z}). It is shown how the design of lowpass, highpass and nonsymmetric bandpass recursive digital filters with given zeroes is transformed to the problem of designing a magnitude squared function having its passband stretched onto the whole unit circle. This magnitude... View full abstract»

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  • Optimal design of digital Hilbert transformers with a concavity constraint

    Publication Year: 1979, Page(s):824 - 827
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (63 KB)

    A linear programming algorithm is described for designing FIR digital filters with the constraint that the magnitude response be concave over prescribed frequency bands. This is applied to odd-length Hilbert Transformers, and computational results are given. The concavity constraint avoids the ripple of the minimax design, and retains the advantage of maintaining half-band symmetry in the case of ... View full abstract»

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  • On the stability of a 1-bit-quantized feedback system

    Publication Year: 1979, Page(s):844 - 848
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (151 KB)

    The performance of a Delta(-Sigma)- Modulator can be greatly improved by increasing the effectiveness of the low-pass or band-pass filter which is placed in the feedback loop around the 1-bit quantizer. However, too steep an attenuation of the filter will cause the system to exhibit limit cycle oscillations, which have a detrimental effect on the performance. These are characterized by a periodic ... View full abstract»

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  • [Back cover]

    Publication Year: 1979, Page(s): c4
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    Freely Available from IEEE
  • Multichannel zero-crossing-interval pitch estimation

    Publication Year: 1979, Page(s):764 - 767
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (168 KB)

    A method for estimating the pitch of voiced speech is presented, based on the following operations: multichannel band-pass filtering; envelope extraction; low-frequency emphasis and DC-level removal; bin analysis of zero-crossing interval times of the yielded waveforms; smoothing of the resulting histogram to give the pitch-period estimator function for a single frame of speech. Since most of thes... View full abstract»

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  • A robust vocoder with pitch-adaptive spectral envelope estimation and an integrated maximum-likelihood pitch estimator

    Publication Year: 1979, Page(s):64 - 68
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (95 KB)

    This paper describes a low bit-rate vocoder designed for improved speech reproduction quality and robustness. The vocoder includes a new algorithm, the Spectral Envelope Estimator, which forms the nucleus of the spectral analyzer. In addition to estimating the speech spectrum, the spectral analyzer also allows determination of a continuous estimate of the background noise spectrum which may be use... View full abstract»

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  • The roles of integration time and acoustic multipaths in determining the structure of CW line spectra

    Publication Year: 1979, Page(s):298 - 301
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (128 KB)

    A recently developed normal mode theory of the acoustic field generated by a uniformly moving time harmonic point source [K. E. Hawker, "A Normal Mode Theory of Acoustic Doppler Effects in the Oceanic Waveguide," J. Acoust. Soc. Am., scheduled for publication in March 1979] is used to study the relative importance of multipath and integration time effects in determining the structure of cw line sp... View full abstract»

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  • Adaptive linear prediction filtering for airborne underwater acoustic signal processors

    Publication Year: 1979, Page(s):603 - 607
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (136 KB)

    The behavior of a particular implementation of a Widrow-Hoff, or iterative adaptive version of the classical Least Mean Square Wiener Filter, is evaluated by means of tests on the Adaptive Filter in conjunction with two different airborne under-water acoustic signal processors. The ability of the adaptive mechanism to enhance spectral estimates, detection and tracking of narrow-band signals in noi... View full abstract»

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  • Influence of the spatial coherence of the background noise on high resolution passive methods

    Publication Year: 1979, Page(s):306 - 309
    Cited by:  Papers (50)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (86 KB)

    Research directed towards the improvement of underwater passive listening technique has led to powerful methods but which require the knowledge of the spatial coherences of the sources and of the background noise. They are sensitive to the shape of the wavefront of the signals from the sources. But they are also sensitive to the spatial coherence of the background noise : this is shown in this pap... View full abstract»

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  • Locating a passive source with array measurements a summary of results

    Publication Year: 1979, Page(s):967 - 970
    Cited by:  Papers (13)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (82 KB)

    This paper deals with the localization of an acoustic source by means of array measurements. It gives lower bounds on bearing and range accuracy and discusses their implications for practical instrumentations. It also examines sensitivity to uncertainties in sensor location and ambiguities in delay measurement caused by very narrowband signals. View full abstract»

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  • A novel vocoder concept based on discrete time acoustic tubes

    Publication Year: 1979, Page(s):73 - 76
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (128 KB)

    A novel vocoder concept is presented which is based on discrete time equivalents of the uniform acoustic tube. Recently proposed models of lossy sections of different lengths are cascaded to reprensent the vocal tract. Results comparable to a standard 12-section LPC-vocoder are achieved cascading typically 5 tube sections of different lengths. Major savings are derived in terms of the hardware com... View full abstract»

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  • Narrowband LPC speech transmission over noisy channels

    Publication Year: 1979, Page(s):60 - 63
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (95 KB)

    Recently we described a variable-frame-rate LPC vocoder designed to transmit good quality speech over 2400 bps fixed-rate noisy channels with bit-error probabilities ranging up to 5% [3]. The basic idea was to lower the data rate by transmitting LPC parameters only when speech characteristics have changed sufficiently since the last transmission, and to employ the resulting bit-rate savings for pr... View full abstract»

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  • Design of 2-D recursive filters with separable denominator transfer functions

    Publication Year: 1979, Page(s):24 - 27
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (76 KB)

    The design of 2-D recursive filters with separable denominator transfer functions from the impulse response of a prototype filter is performed in two steps. First the poles of the transfer function of the recursive filter are found using an LMS criterion through an iterative scheme. According to the same criterion, the coefficients of the numerator may then be found by solving linear systems. For ... View full abstract»

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  • Considerations in applying clustering techniques to speaker independent word recognition

    Publication Year: 1979, Page(s):578 - 581
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (94 KB)

    Recent work at Bell Laboratories has demonstrated the utility of applying sophisticated pattern recognition techniques to obtain a set of speaker independent word templates for an isolated word recognition system [1,2]. In these studies, it was shown that a careful experimenter could guide the clustering algorithms to choose a small set of templates that were representative of a large number of re... View full abstract»

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  • A multidimensional Modelling approach to texture classification and segmentation

    Publication Year: 1979, Page(s):962 - 966
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (184 KB)

    The interest of an inverse filtering approach to picture modelling and recognition is emphasized. Following these lines, a multidimensional vector AR model is fitted to a reference region using a generalized Levinson procedure. The models of other regions are then used as inverse filters on the reference for classification. The approach is applied to natural pictures for recognition and segmentati... View full abstract»

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  • Reduction of computation in pole-zero modeling of speech signals

    Publication Year: 1979, Page(s):735 - 738
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (75 KB)

    The main problem with pole-zero modeling of speech signals is its computational complexity, as a highly nonlinear optimization problem has to be solved for each analysis frame. We propose a method to save computation. At each new analysis frame, it is first decided whether the parameter estimation routine needs to be started. The decision is made by using statistical hypothesis testing techniques.... View full abstract»

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  • Design of stable all-pass filters

    Publication Year: 1979, Page(s):813 - 817
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (89 KB)

    A method for converting the allpass filter design problem to a more tractable spectral factorization problem is presented. Spectral factorization by a new two-dimensional linear prediction technique is discussed, and an efficient lattice structure for implementing the resulting 2-D allpass filter is described. View full abstract»

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  • An analysis/Synthesis framework for transform coding of speech

    Publication Year: 1979, Page(s):81 - 84
    Cited by:  Papers (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (128 KB)

    Adaptive Transform coding techniques for speech communication have recently received considerable attention. The basic concept of these methods is to divide the speech into frequency components by a suitable transform and then encode them using adaptive PCM. Examples of designs and performance of adaptive transform coders will be presented for bit rates in the range of 8 to 16 kbs. View full abstract»

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  • Simplified error models for digital filters

    Publication Year: 1979, Page(s):363 - 366
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (89 KB)

    Statistical error models for digital filter roundoff noise estimation are usally based on a set of noise sources interconnected according to the filter structure. The location of the noise sources depends on the location of the quantizers within the filter network. Roundoff noise analysis by both, simulation of digital filters and analysis of noise models yields the result that the entire roundoff... View full abstract»

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  • Diphone synthesis for phonetic vocoding

    Publication Year: 1979, Page(s):891 - 894
    Cited by:  Papers (7)  |  Patents (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (112 KB)

    We report on the synthesis of speech in the context of a phonetic vocoder operating at 100 b/s. With each phoneme, the vocoder transmits the duration and a single pitch value. The synthesizer uses a large inventory of diphone "models" to synthesize a desired phoneme string. The diphone inventory has been selected to differentiate between prevocalic and postvocalic allophones of sonorants, to accou... View full abstract»

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  • Generation of two-dimensional digital functions without non-essential singularities of the second kind

    Publication Year: 1979, Page(s):13 - 19
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (192 KB)

    A class of two variable Hurwitz polynomials called very strict Hurwitz polynomials (VSHPs) are defined and their properties are studied. Their application in the generation of two variable functions without non-essential singularities of the second kind are indicated. Necessary and sufficient conditions on general and reactance one-variable to two-variable transformations so that they yield two va... View full abstract»

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  • Frequency warping for nonuniform talker normalization

    Publication Year: 1979, Page(s):566 - 569
    Cited by:  Papers (1)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (128 KB)

    This paper concerns a new approach to nonlinear spectral normalization to eliminate inter- speaker differences from frequency-band-limited speech. A frequency normalized distance between a test and a reference spectrum is defined on the basis of minimum mean square difference over all possible choices of frequency warping functions under certain constraints. This spectral distance is computed by m... View full abstract»

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