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IEEE Transactions on Acoustics, Speech, and Signal Processing

Issue 1 • February 1983

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Displaying Results 1 - 25 of 54
  • [Front cover and table of contents]

    Publication Year: 1983, Page(s): 0
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    Freely Available from IEEE
  • On the approximation of FIR by IIR digital filters

    Publication Year: 1983, Page(s):28 - 34
    Cited by:  Papers (6)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (690 KB)

    Kaiser's empirical formula for finite impulse response (FIR) digital filter length, as a function of transition width and rejection band loss (together with theI_{0}-sinhwindow function), provides a simple FIR filter design algorithm. Recent developments in time series analysis provide a theoretical procedure for selecting the numerator and denominator orders of an infinite impulse resp... View full abstract»

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  • Modeling of multichannel time series and extrapolation of matrix-valued autocorrelation sequences

    Publication Year: 1983, Page(s):45 - 55
    Cited by:  Papers (23)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (988 KB)

    This paper deals with multichannel time series, and extends the autocorrelation method for fitting the time series from the nondegenerate case to the general case. We shall show that the AR model obtained by solving the normal equations is always stable even in the degenerate case. The multichannel Levinson algorithm for solving the normal equations is extended from the nondegenerate case to the g... View full abstract»

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  • A highly concurrent algorithm and pipeleined architecture for solving Toeplitz systems

    Publication Year: 1983, Page(s):66 - 76
    Cited by:  Papers (104)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1403 KB)

    The design of VLSI parallel processors requires a fundamental understanding of the parallel computing algorithm and an appreciation of the implementational constraint on communications. Based on such consideration, this paper develops a highly concurrent Toeplitz system solver, featuring maximum parallelism and localized communication. More precisely, a highly parallel algorithm is proposed which ... View full abstract»

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  • Thinning digital filters: A piecewise-exponential approximation approach

    Publication Year: 1983, Page(s):105 - 113
    Cited by:  Papers (36)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (956 KB)

    An algorithm is presented for designing finite impulse response (FIR) recursive digital filters that require few multiplies to produce good frequency response, The process of reducing the number of multiplies needed to implement a digital filter is called thinning. This thinning algorithm uses dynamic programming techniques to find the best least-squares piecewise-exponential approximation to a de... View full abstract»

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  • Optimal estimation for band-limited signals including time domain considerations

    Publication Year: 1983, Page(s):113 - 122
    Cited by:  Papers (31)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1080 KB)

    The problem of estimation for band-limited signals is investigated. The desired estimate is based on a finite number of time samples taken from a measurement interval of length TM. With no additional information, the estimate must be based on these samples alone. When the signal is band-limited with normalized bandwidth β, this provides additional information which is used to impro... View full abstract»

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  • Causal invertibility and stability of asymmetric half-plane digital filters

    Publication Year: 1983, Page(s):195 - 201
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (819 KB)

    In this paper the following question is studied: supposeR in l_{1}(Z^{2}), and let Shdenote the semigroup generated by the support of h; when does h have an inverse in l1(Sh)? From an engineering viewpoint the question can be asked as follows: When does a causal stable filter have a causal stable inverse? In this paper we develop a mathematical framework... View full abstract»

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  • Comments on "Selection criteria for efficient implementation of FFT algorithms

    Publication Year: 1983, Page(s): 206
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (133 KB)

    First Page of the Article
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  • Foreword

    Publication Year: 1983, Page(s):250 - 251
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    Freely Available from IEEE
  • A compact, flexible LPC vocoder based on a commercial signal processing microcomputer

    Publication Year: 1983, Page(s):252 - 257
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (851 KB)

    A very small, flexible, high-quality, full-duplex 2.4-kbit/s linear predictive vocoder has been implemented with commercially available integrated circuits. This fully digital realization is based on a distributed signal processing architecture employing three Nippon Electric Company (NEC) μPD7720 signal processing interface (SPI) single-chip microcomputers. One SPI implements the LPC analyze... View full abstract»

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  • [Back cover]

    Publication Year: 1983, Page(s): c4
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    Freely Available from IEEE
  • An optimized weighting algorithm for variations in PCM energy levels

    Publication Year: 1983, Page(s):180 - 188
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1192 KB)

    The advantages of optimum weighted PCM for speech signals has been demonstrated by Sundberg [2]. Since the symbols in a PCM word have different sensitivities to digital transmission errors the relative signal energy used by each PCM word is matched accordingly. A near optimum performance may be achieved by transmitting groups of PCM symbols at the same energy level where the number of groups is le... View full abstract»

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  • A monolithic audio spectrum analyzer

    Publication Year: 1983, Page(s):288 - 293
    Cited by:  Papers (6)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (736 KB)

    A monolithic audio spectrum analyzer with 16 channels of bandpass filters, half-wave rectifiers, and postfiltering has been fabricated with double-poly NMOS technology. The chip was designed using switched-capacitor filter techniques. It performs a total of 84 poles of filtering and contains 100 operational amplifiers in an area of 225 × 280 mils. A system dynamic range of better than 43 dB, ... View full abstract»

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  • Improvements in isolated word recognition

    Publication Year: 1983, Page(s):157 - 167
    Cited by:  Papers (16)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1192 KB)

    For isolated word recognition, possibilities for improving the recognition accuracy are investigated for a given feature extraction, which is based on a short term spectrum analysis by means of band-pass filtering. A number of preprocessing steps are discussed, which are to be applied prior to time alignment via dynamic programming. These preprocessing steps include normalization of short term spe... View full abstract»

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  • Recursive least squares circular lattice and escalator estimation algorithms

    Publication Year: 1983, Page(s):228 - 231
    Cited by:  Papers (33)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (352 KB)

    This note supplements the paper of Sakai [1] about circular lattice filtering with the derivation of optimal recursive least squares circular lattice estimation algorithms. The derivation is based on the geometric method of Lee, Morf, and Friedlander [2]. The same method is also applied to obtain an optimal recursive estimation algorithm for escalator structure of Ahmed and Youn [3]. View full abstract»

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  • An auditory spectral analysis model using the chirp z-transform

    Publication Year: 1983, Page(s):148 - 156
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (856 KB)

    In this paper we discuss a new signal-processing approach to implementing an auditory processing model. The model is based on auditory physiology and psychophysics; it uses constant-bandwidth analysis below 500 Hz and constant-Q analysis above 500 Hz, with the analysis filters being implemented with multiple real poles. The constant-Q analysis is performed using the chirp z-transform (CZT). We rev... View full abstract»

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  • Recursive discrete Fourier transformation with unevenly spaced data

    Publication Year: 1983, Page(s):206 - 209
    Cited by:  Papers (8)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (408 KB)

    New recursive techniques for Fourier spectral analysis are reported, for which ongoing spectral estimates are generated from unevenly spaced data in real time. The algorithms are robust and computationally efficient, and are well suited to state variable form involving real number calculations. These methods are particularly attractive in filtering and signal processing applications where signals ... View full abstract»

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  • A generalization of the spectrogram for colored displays

    Publication Year: 1983, Page(s):222 - 225
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (496 KB)

    This paper defines a class of colored displays for analyzing the spectral content of nonstationary processes. The displays are based on a natural mapping that exists between colors and pairs of zero mean, equal variance complex random variables, and are capable of representing signal features not easily recognizable on a conventional spectrogram. Displays are constructed for both one and two chann... View full abstract»

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  • On qualifying windows for FIR filter design

    Publication Year: 1983, Page(s):237 - 240
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (312 KB)

    Although several measures of window performance have been defined, none of these directly characterizes the window's application to FIR filter design. But as the number of filter coefficients tends to infinity the peak ripple in the filter response tends to a value that is dependent only on the window. This value, the "Gibbs number," closely approximates the peak ripple for practical filters. View full abstract»

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  • Simplified tone detector for PCM channel

    Publication Year: 1983, Page(s):8 - 16
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (768 KB)

    A digital correlator for detecting multifrequency signaling tones is described. The theory is first presented for a continuous correlator using a fixed sine wave burst. A substitution, using a change in the time reference of the correlating sine wave, is then introduced. By introducing this change, a modified computational algorithm is derived which reduces the number of multiplications required a... View full abstract»

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  • A sequential processor for detecting Gaussian acoustic signals

    Publication Year: 1983, Page(s):143 - 147
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (480 KB)

    A sequential approach to detecting Gaussian acoustic signals is discussed. The optimum sequential processor is synthesized and its performance is evaluated. The performance obtained is compared with the performance of the optimum unsequential processor. It is shown that the sequential approach may considerably reduce the required time-bandwidth product when the desired signal is present in the inp... View full abstract»

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  • Normalized lattice algorithms for least-squares FIR system identification

    Publication Year: 1983, Page(s):122 - 128
    Cited by:  Papers (26)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (680 KB)

    Recently developed algorithms for least-squares identification of autoregressive models are extended in this paper so as to facilitate least-squares identification of finite impulse-response models. The algorithms belong to the class of square-root normalized lattice algorithms, hence they share the computational efficiency and good numerical behavior of the latter. Two versions are presented-one ... View full abstract»

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  • A single chip digital signal processor and its application to real-time speech analysis

    Publication Year: 1983, Page(s):339 - 346
    Cited by:  Papers (10)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1264 KB)

    A single chip high-performance digital signal processor (HSP) has been developed for speech, telecommunication, and other applications. The HSP uses 3 μm CMOS technology and its architecture features floating point arithmetic and pipeline structure. By adoption of floating point arithmetic, data covering a wide dynamic range (up to 32 bits) can be manipulated. The input clock frequency is 16 ... View full abstract»

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  • A theoretical and experimental comparison of output signal-to-noise ratios of three types of cross correlators

    Publication Year: 1983, Page(s):202 - 205
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (440 KB)

    The output signal-to-noise ratio as a function of input signal-to-noise ratios is compared for three different types of cross-correlators: 1) the standard analog cross correlator consisting of bandpass filters, multiplier, and low-pass filter; 2) the polarity coincidence correlator (PCC) which incorporates hard limiters in each analog input channel; and 3) a modified type of PCC wherein only one o... View full abstract»

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  • A single chip speech synthesizer using a switched-capacitor multiplier

    Publication Year: 1983, Page(s):313 - 323
    Cited by:  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1248 KB)

    A single chip speech synthesizer was designed using a switched-capacitor multiplier to implement the LPC algorithm. The chip contains the LPC-10 filter, 20 kbit ROM, all control logic, a three-pole switched-capacitor low-pass filter, and an audio amplifier capable of driving a speaker directly. The chip was fabricated in 5 μm CMOS technology and is 218 mils on the side. 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