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

Date 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
  • [Back cover]

    Publication Year: 1979 , Page(s): c4
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    Freely Available from IEEE
  • Detection performance of an adaptive processor in non-stationary noise

    Publication Year: 1979 , Page(s): 136 - 139
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (94 KB)  

    New analytical and simulation results describing the performance of an adaptive detection processor for narrowband signals are given. The simulation results compare the detection performance of the adaptive processor with an incoherently-averaged, magnitude-squared FFT processor for a class of non-stationary input noise. An analytical derivation of the noise-only probability density function of the adaptive processor's output prior to post-detection integration is presented. View full abstract»

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  • Application of transposition to decimation and interpolation in digital signal processing systems

    Publication Year: 1979 , Page(s): 832 - 835
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (112 KB)  

    A method that was recently developed for describing linear time-varying digital systems in the frequency domain will be applied to decimators and interpolators. This method allows a generalization of transposition to these time-varying operations, giving both a structural and a functional relationship between a digital system and its transpose. It is shown that decimation and interpolation are related by transposition, and the consequences of this interconnection are discussed. View full abstract»

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  • A mixed-phase homomorphic vocoder

    Publication Year: 1979 , Page(s): 56 - 59
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (82 KB)  

    In this paper short-time homomorphic analysis and the harmonic representation of voiced speech are explored with the result of a mixed-phase homomorphic vocoder of somewhat higher quality than its minimum-phase counterpart. A theoretical framework is presented for unwrapped phase estimation from harmonic spectra through smoothing real and imaginary spectral components. The short-time harmonic model leads to pitch adaptive duration and alignment requirements on time-domain windowing. The underlying phase envelope is consequently preserved so that cepstral windowing can he applied. In addition, two alternative vocoders with mixed-phase are considered: the first is based on linear interpolation of complex harmonic peaks, and the second on Lim's spectral root deconvolution scheme. View full abstract»

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  • Statistical design of ARMA filters

    Publication Year: 1979 , Page(s): 818 - 821
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (90 KB)  

    Procedures are given for the systematic design of ARMA filters. All of the design algorithms are linear and of the Levinson type. Each design is initiated with a long AR approximation of an ideal spectrum. The long AR is used to generate consistent unit pulse and covariance sequences for use in the Levinson type algorithm of Mullis and Roberts. The latter algorithm allows one to approximate the unit pulse and covariance sequences and thereby obtain a low-order ARMA approximation. There is no need for the ARMA approximation to be proper. 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)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | 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 means of dynamic programming after adaptively eliminating the individual glottal characteristics. Applications to the identification of steady-state vowels and the detection of words in connected speech is discussed. View full abstract»

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  • Design of stable all-pass filters

    Publication Year: 1979 , Page(s): 813 - 817
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | 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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  • Techniques for recognition of spectrogram patterns based on dynamic modeling

    Publication Year: 1979 , Page(s): 266 - 268
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (112 KB)  

    A technique is described for recognition of contours in a spectrogram associated with a given string of events by means of dynamic models expressing relationships between frequency as the dependent variable and time as the independent variable. 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)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | 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 complexity during speech synthesis and in term of the data rate during speech transmission. The concept can also be used for modeling the nasal tract by interconnecting a discrete time acoustic tube system to the vocal tract model via a proper adaptor. Till now losses of the low-loss-type are considered. The analysis part of the vocoder is based ona standard autocorrelation LPC-analysis followed by an additional approximation stage yielding a reduction of the number of tube sections. The performance test of the proposed vocoder is done both by subjective evaluation and by comparison in the spectral and the acoustic tube domain. View full abstract»

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  • Performance of LPC vocoders in a noisy environment

    Publication Year: 1979 , Page(s): 216 - 219
    Cited by:  Papers (10)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (77 KB)  

    Although 2400 BPS vocoders based upon Linear Predictive Coding have produced speech intelligibility scores as high as 90% in a quiet laboratory setting, few actual system measurements have been made in noisy, stressful, military environments. This paper describes LPC vocoder performance in high acoustic noise environments and when the speaker is subjected to stress, vibrations and accelerations. Measurements were made on military platforms which included ships, conventional aircraft, helicopters, tracked vehicles and wheeled vehicles; acoustic noise levels varied from 70 to 125dB Sound Pressure Level. (1) View full abstract»

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  • Rate/Pitch modification of speech using the constant-Q transform

    Publication Year: 1979 , Page(s): 748 - 751
    Cited by:  Papers (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (112 KB)  

    Modification of the rate of occurrence of acoustic events without altering frequency content, and modification of pitch without changing time scale are presented as equivalent problems. While the short-time Fourier transform has been used to solve the rate modification problem, it is not a natural tool. It lacks the scaling property of the Fourier transform. The constant-Q transform, on the other hand, exhibits this property. A more natural rate/pitch modification system using the constant-Q transform is presented which performs well with rate/pitch changes by factors of between one-third and three. View full abstract»

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  • A broadband echo ranging system for measuring the frequency characteristics of fish schools

    Publication Year: 1979 , Page(s): 612 - 615
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (136 KB)  

    A broadband echo-ranging system has been employed to collect echoes from fish schools off the southern California coast. The system uses an implosive impulse source called Hydroshock(R)and a broad-band, constant beamwidth towed array. The equipment is capable of measuring target strength from 500 Hz to 50 kHz, with minimal frequency distortion due to transducer beamwidth effects. Averaged frequency spectra for echoes from several schools are presented, showing different cutoff and resonance characteristics. View full abstract»

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  • A frequency domain noise cancelling preprocessor for narrowband speech communications systems

    Publication Year: 1979 , Page(s): 212 - 215
    Cited by:  Papers (13)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (71 KB)  

    Performance of narrowband speech communications systems, such as Linear Predictive Coding (LPC), is often severely degraded by the presence of ambient acoustic noise in the input speech signal. Spectral subtraction techniques show promise in improving the overall performance of LPC in acoustic noise environments, but typically present annoying musical tones at the output. A spectral subtraction technique is described, which includes a biased estimate of the noise, that does not present musical tones at the output. In addition, an automatic speech activity detector is described and used to adapt the noise estimate to changing noise environments. View full abstract»

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  • Voice-excited LPC coders for 9.6 kbps speech transmission

    Publication Year: 1979 , Page(s): 558 - 561
    Cited by:  Papers (3)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (89 KB)  

    This paper considers the use of voice-excited linear predictive (LPC) coders for speech transmission at a bit-rate of 9.6 kbps. In our on-going work, we study in detail the various aspects of this class of speech coders, with the goal of maximizing the speech quality at the above rate. Important among these aspects are: baseband residual versus baseband speech transmission, coding of the baseband signal, and high-frequency regeneration from the baseband. We provide a discussion of these and other issues, and indicate a number of variables that have been included in our speech-quality optimization study. Experimental results obtained to date are summarized in the paper. More complete results and conclusions will be presented at the conference. View full abstract»

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  • A fixed-point iteration algorithm for adaptive linear estimation applied to spectral line enhancement

    Publication Year: 1979 , Page(s): 958 - 961
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (79 KB)  

    An Adaptive Linear Estimation Algorithm which can be applied to the Spectral Line Enhancement problem is proposed. Through computer simulation, the rate of convergence of the proposed method is found to be significantly better than that of the LMS Gradient Algorithm for the problem considered. Also, this increased performance is obtained with only a modest increase in computational complexity. View full abstract»

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  • Adaptive linear prediction filtering for airborne underwater acoustic signal processors

    Publication Year: 1979 , Page(s): 603 - 607
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | 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 noise is illustrated by initial test results which are presented as graphs of detection indices and photographs of airborne acoustic displays. A summary review of the adaptive filter theory is included. Certain peculiarities in the performance of the adaptive mechanism also are noted as they relate to integration of such equipment with existing airborne acoustic processors and displays. View full abstract»

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  • Computation of Fourier integral using polynomial interpolation

    Publication Year: 1979 , Page(s): 494 - 497
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (76 KB)  

    The Newton-Gregory interpolation formula is shown to lead to a sampling theorem for signals having Laplace transforms with pole frequencies less than a submultiple of the sampling rate. Plots of spectra computed by Fourier transforming polynomials fitted to short sample sequences are exhibited which accord with this sampling theorem. 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)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | 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 spectra. Widths are shown to result from an interplay between three effects, acoustic multi-path (Doppler) broadening, intrinsic width due to the finite integration time, and a "range rate" or FM slide effect due to the combination of finite integration time and a time variable instantaneous frequency. View full abstract»

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  • System identification using a maximum-likelihood spectral matching technique

    Publication Year: 1979 , Page(s): 405 - 408
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (76 KB)  

    The usefulness of the periodogram as a system identification tool is often underestimated because of the large variance in the estimates of the power spectral density function of the system output. If we approach the parameter estimation problem as an exercise in maximum likelihood estimation, with the measured periodogram as the input data, the result is a spectral matching technique that is rather simple to apply. A valuable by-product of this method is a value for the Fisher information matrix of the parameter estimates. Models of many forms such as AR, ARMA, with or without observation noise can be treated using the same algorithmic structure. The input data can be efficiently computed using the Fast Fourier Transform. Several examples illustrate the technique. View full abstract»

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  • An experimental system for acoustic-phonetic decoding of continuous speech

    Publication Year: 1979 , Page(s): 105 - 107
    Cited by:  Papers (1)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (84 KB)  

    In the framework of MYRTILLE II Speech Understanding System under development in our Laboratory we have realized a prototype for the acoustic-phonetic processing level. This prototype makes it possible to test various parameters and strategies for phonemic transcription of continuous speech. It can be considered as a metasystem in the sense that, given a hierarchy of recognition algorithms and a strategy, it can generate the optimal system for phoneme recognition. The system directly works on the digitized speech wave, which makes it possible to get the best accuracy on the parameters. The speech signal is segmented into phoneme-like units by a decision function which incorporates voicing, energy, zero-crossing rate and curve length. The segments thus obtained are then processed by the recognition system which can be viewed as a tree structure the nodes of which are algorithms. These algorithms take into account one or several features and their answers can be phoneme classes and/or other algorithms. Problems involved in the design of such a system are also presented in this paper together with a particular implementation. View full abstract»

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  • A high data rate, low power all-digital correlation circuit design

    Publication Year: 1979 , Page(s): 859 - 862
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    Correlation techniques are widely used in analog and digital signal processing; e.g. in optiman receiver design. In this paper, a new digital output correlator circuit design is proposed. The correlation results are presented as a binary count of the number of bits in agreement between 31-bit reference and input sequences. Correlation results could be presented at data rates up to about 50MHz. The proposed new circuit could be implemented in a high speed bipolar transistor technology as a monolithic large-scale-integrated circuit with a power dissipation of about .5 watts. A novel pipelined 31-bit digital summing circuit will be presented that employs new multivalued, multithresholded latching and counting circuits in the formation of its binary-coded output. Comparisons with an all-binary digital correlator circuit are made. View full abstract»

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  • Automatic recognition of continuous digits sequences by means of segmentation and dynamic programming

    Publication Year: 1979 , Page(s): 245 - 248
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (78 KB)  

    This paper describes an experiment of recognition of sequences of digits pronounced continuously. This experiment takes place in the design of a practical system for postal codes recognition. The speech signal is analyzed by a 15-channel spectral analyzer. The digit sequence is then recognized using a semi-global procedure which consists of two successive steps: - segmentation of the sequence into the digits which comprise the string, - recognition of the individual digits by a dynamic programming technique which was previously designed in the laboratory. Typical Results of segmentation and recognition will be given and discussed. View full abstract»

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  • An improved residual encoder for speech compression

    Publication Year: 1979 , Page(s): 542 - 545
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (76 KB)  

    An improved Residual Encoder is developed for digitization of speech for different languages such as: English, Chinese, Arabic, French, German and Russian, etc. Experiments show that the improved system achieves a 2-7dB increase in signal-to-noise ratio over the previous Residual Encoder (1). Effect of transmission bit error on the speech quality is also investigated, it is found that the variable input code of (1) suffers more degradation in speech quality than the fixed code used in this study with transmission errors. The implementation complexity is on the same order as other Residual Encoder Systems. View full abstract»

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  • 8 kbps voice transmission by SPAC

    Publication Year: 1979 , Page(s): 562 - 565
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (94 KB)  

    A speech Processing system named SPAC (Speech Processing system by use of AutoCorrelation function) can successfully compress or expand the speech spectrum and reduce the noise level superposed on speech signal. By employing these functions of SPAC, three types of low-bit-rate transmission system, which bit rate is 8 or 10 kbps, have been proposed. These are zero-crossing+SPAC, ADM+SPAC, and band-compression and -expansion by SPAC (compression ratio : 3). Evaluation by hearing tests is made on these sytems, zero-crossing and ADM. It is shown that ADM+SPAC with 8 kbps can transmit natural and acceptable speech. View full abstract»

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