By Topic

Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 2 • Date April 1977

Filter Results

Displaying Results 1 - 23 of 23
  • [Front cover and table of contents]

    Publication Year: 1977 , Page(s): 0
    Save to Project icon | Request Permissions | PDF file iconPDF (170 KB)  
    Freely Available from IEEE
  • Correction to "Real-time adaptive linear prediction using the least mean square gradient algorithm"

    Publication Year: 1977 , Page(s): 205
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (113 KB)  

    First Page of the Article
    View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Corrections to "Letter-to-sound rules for automatic translation of english text to phonetics"

    Publication Year: 1977 , Page(s): 205
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (113 KB)  

    First Page of the Article
    View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • [Back cover]

    Publication Year: 1977 , Page(s): c4
    Save to Project icon | Request Permissions | PDF file iconPDF (716 KB)  
    Freely Available from IEEE
  • A direct approximation technique of log magnitude response for digital filters

    Publication Year: 1977 , Page(s): 127 - 133
    Cited by:  Papers (7)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (608 KB)  

    A new direct approximation technique of log magnitude response for digital filters is presented in this paper. The facts that the log magnitude response of digital filters can be expanded into Fourier series and a fairly accurate cosine type log magnitude response can be realized by the elemental digital filter presented in this paper are used in the present technique. The system functions obtained by this method provide the best mean-square approximation to an arbitrarily prescribed log magnitude response. The resulting digital filters are realized in the cascade form of the elemental digital filters, and they give relatively low coefficient sensitivity. The elemental filter is recursive but its form is very simple. Its coefficients are easily obtained by the cepstrum of the impulse response which is the Fourier transform of the desired log magnitude response. This method is very powerful in the realization of digital filters for speech synthesis filters with complicated log magnitude responses. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Speech recognition: Invited papers presented at the 1974 IEEE symposium

    Publication Year: 1977 , Page(s): 207
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (136 KB)  

    First Page of the Article
    View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A general form of continued-fraction expansion for two-dimensional recursive digital filters

    Publication Year: 1977 , Page(s): 198 - 200
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (296 KB)  

    A generalized continued-fraction (CF) expansion technique for two-dimensional (2-D) recursive digital filters is introduced. The conditions for realization and implementation of such an expansion are discussed. It is shown that all the structures so far known can be derived from such an expansion. Illustrative examples are also provided. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Normalization of vowels by vocal-tract length and its application to vowel identification

    Publication Year: 1977 , Page(s): 183 - 192
    Cited by:  Papers (49)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1008 KB)  

    A new approach to speech parameter normalization is presented in which no prior knowledge about the input speakers is required. The vocal-tract length and area function are first estimated from the acoustic speech waveform, and then the area function is normalized to an acoustic tube of the same shape having a certain reference length. The normalized formant frequencies are defined as the resonance frequencies of this acoustic tube. The distributions of unnormalized and normalized formant frequencies for 9 stationary American vowels were investigated with 14 male and 12 female speakers. Fairly compact distributions of the vowels in the normalized F1-F2-F3space were obtained. A preliminary identification test for stationary vowels based on this normalization method showed an expected average recognition rate of 84-96 percent for arbitrarily selected speakers, depending on the phonetic criteria adopted for defining "correct" identification. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Evaluating speech recognizers

    Publication Year: 1977 , Page(s): 178 - 183
    Cited by:  Papers (9)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (744 KB)  

    Although automatic word recognition systems have existed for some twenty-five years there is still no suitable standard for evaluating their relative performances. Currently, the merits of two systems cannot be meaningfully compared unless they have been tested with at least the same vocabulary or, preferably, with the same acoustic samples. This paper develops a standard for comparing the performance of different recognizers on arbitrary vocabularies based on a human word recognition model. This standard allows recognition results to be normalized for comparison according to two intuitively meaningful figures of merit: 1) the noise level necessary to achieve comparable human performance and 2) the deviation of the pattern of confusions from human performance. Examples are given of recognizers evaluated in this way, and the role of these performance measures in automatic speech recognition and other related areas is discussed. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Covariance-invariant digital filtering

    Publication Year: 1977 , Page(s): 143 - 151
    Cited by:  Papers (13)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (840 KB)  

    When discretizing continuous-time filters, one is often interested in preserving a property termed covariance-invariance. Techniques are outlined for synthesizing discrete-time filters which are covariance-invariant with corresponding continuous-time filters. The synthesis techniques involve straightforward matrix decompositions or polynomial root-finding algorithms that can easily be programmed on a digital computer. Applications of the technique to digital filter synthesis are outlined, with example designs presented for covariance-invariant Butterworth and Chebyshev digital filters. Based on the frequency response of these designs it is argued that the method of covariance-invariance is superior to the methods of impulse-invariance and bilinear-z as a response matching design technique for the synthesis of digital filters. This superiority is especially apparent at sampling rates that are marginal with respect to filter critical frequencies. Moreover, the covariance-invariant designs are stably invertible solutions to a so-called spectral factorization problem. This property may be important in inverse filtering applications. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • An introduction to programming the Winograd Fourier transform algorithm (WFTA)

    Publication Year: 1977 , Page(s): 152 - 165
    Cited by:  Papers (69)  |  Patents (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1176 KB)  

    Recently, Dr. Shmuel Winograd discovered a new approach to the computation of the discrete Fourier transform (DFT). Relative to fast Fourier transform (FFT), the Winograd Fourier transform algorithm (WFTA) significantly reduces the number of multiplication operations; it does not increase the number of addition operations in many cases. This paper introduces the new algorithm and discusses the operations comparison problem. A guide for programming is included, as are some preliminary running times. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Nine digital filters for decimation and interpolation

    Publication Year: 1977 , Page(s): 121 - 126
    Cited by:  Papers (55)  |  Patents (11)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (624 KB)  

    Filtering is necessary in decimation (decreasing the sampling rate of) or interpolation (increasing the sampling rate of a digital signal. If the rate change is substantial, the process is more efficient when the decimation or interpolation occurs in stages rather than in one step. Half-band filters are particularly efficient for effecting octave changes in sampling rate and nine digital filters are presented, eight of them half-band filters, to be used as components of multistage interpolators and decimators. Also presented is a procedure for combining the filters to produce multistage designs that meet a very wide range of accuracy requirements (stopband attenuation to 77 dB, passband ripple as low as 0.00014). The nine filters admit changes between sampling rates above 4W, where W is the nominal bandwidth of the signal. Established design techniques may be used to obtain efficient filters for conversion between 4W Hz sampling and 2W Hz, the "baseband sampling rate." With these multistage filters, the possible interpolation and decimation ratios are all integer multiples of powers of two. To overcome this restriction we present a simple resampling technique that extends the range of designs to conversions between any two rates. The interpolation or decimation ratio need not be an integer or even rational. In fact, it can vary slightly as in a practical situation where the input signal and output signal are under the control of autonomous clocks. We demonstrate the approach by means of several design examples and compare its results with those obtained from the optimization scheme of Crochiere and Rabiner. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Digital signal processing: Theory, design and implementation

    Publication Year: 1977 , Page(s): 206 - 207
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (272 KB)  

    First Page of the Article
    View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Investigation of the propagation stability of a doubly spread underwater acoustic channel

    Publication Year: 1977 , Page(s): 109 - 116
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (864 KB)  

    Investigation of acoustic propagation in an underwater channel is presented. Propagation between fixed transmitting and receiving sites is studied using a 420-Hz CW tone and a binary pulse sequence with 19 ms of time-delay resolution. The channel is shown to be underspread (BL < 1) so that unambiguous instantaneous measurements of the channel's approximate impulse response are obtained. The experiment is designed, in cooperation with the channel, to allow simultaneous and independent frequency spread, time spread, and broad-band noise power measurements. Some results and a preliminary model of the channel are presented. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A statistical, coarse-sampled data filter

    Publication Year: 1977 , Page(s): 195 - 196
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (208 KB)  

    A digital filter design may be implemented with only one analog summing device, two shift registers with respective lengths equal to the orders of numerator and denominator, a generator of uniformly distributed analog noise, and either a simple low-pass RC filter or a counter. The principle of operation is based on polarity correlation. It has an economic advantage due to lack of storage and digital multipliers, but is applicable only when the data sampling rate is fairly low. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Large amplitude, self-sustained oscillations in difference equations which describe digital filter sections using saturation arithmetic

    Publication Year: 1977 , Page(s): 134 - 143
    Cited by:  Papers (12)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1104 KB)  

    The possibility of oscillations due to adder overflow in digital filter sections of order exceeding two is investigated. Specifically, the following problem is posed: is the property that large amplitude limit cycles cannot be self-sustaining in saturation arithmetic special to second-order sections, or is it also true for some or all higher order sections? This issue is resolved and, beyond that, some new and rather interesting results are presented on the stability implications of approximating a nonlinear element in standard filter structures by a range of linear gains. For every order of the section beyond two we are able to identify a set of coefficients corresponding to an absolutely stable underlying linear system, and a set of initial conditions for which the solution (to the nonlinear recursion) is a limit cycle. As an aid to understanding this behavior, we show that a particular natural class of associated linear recursions is always stable if and only if their order does not exceed two. Finally, motivated by the above result and by the viewpoint of describing functions, we make a conjecture according to which these oscillations do not exist if every member of the above stated class of linear systems is absolutely stable. We show that even this conjecture is false for recursions of order three. Thus, to ensure that overflow oscillations do not occur in high-order sections in which saturation arithmetic is used particular care has to be exercised on a case by case basis. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A class of translation invariant transforms

    Publication Year: 1977 , Page(s): 203 - 205
    Cited by:  Papers (16)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (344 KB)  

    A class of translation invariant transforms containing the R-transform is defined, and it is shown that a particular member of this class is superior to the R-transform for pattern recognition applications. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Implementation of a new stability test for two-dimensional filters

    Publication Year: 1977 , Page(s): 117 - 120
    Cited by:  Papers (24)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (384 KB)  

    A recently introduced new stability test for single-dimensional digital filters is extended to apply to two-dimensional filters, and the procedure for the implementation of this test is discussed. Nontrivial examples for both discrete and continuous two-dimensional filters are used to illustrate the procedure. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Error spectrum shaping in narrow-band recursive filters

    Publication Year: 1977 , Page(s): 200 - 203
    Cited by:  Papers (40)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (384 KB)  

    The error introduced by zero memory quantizers can usually be modeled by an additive white noise component. By incorporating feedback and additional memory elements to these quantizers, it is possible to shape the error spectrum to advantage. Two simple error spectrum shaping quantizers are presented, one to be used with narrow-band low-pass filters and the other with high-pass filters. Simulation examples are presented where the reduction in roundoff errors is equivalent to three bits of data. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • An algorithm to evaluate the Lnorm for some common filters

    Publication Year: 1977 , Page(s): 193 - 194
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (184 KB)  

    The Lnorm is used in some algorithms to determine the ordering of the pole-zero sections of cascade digital filters. This note develops simple expressions that easily evaluate the Lnorm entirely in the Z-domain for filters having zeros on the unit circle (i.e., Butterworth, Chebyshev, and elliptic digital filters designed using the bilinear Z-transformation). View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A new phase unwrapping algorithm

    Publication Year: 1977 , Page(s): 170 - 177
    Cited by:  Papers (129)  |  Patents (14)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (488 KB)  

    A new phase unwrapping algorithm is proposed that combines the information contained in both the phase derivative and the principal value of the phase into an adaptive numerical integration scheme. This new algorithm has proven itself to be very reliable and it can be easily incorporated in standard homomorphic signal processors. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Application of the Rader-Brenner FFT algorithm to number-theoretic transforms

    Publication Year: 1977 , Page(s): 196 - 198
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (296 KB)  

    An FFT algorithm developed by Rader and Brenner, which eliminates complex multipliers, is shown to apply to complex number-theoretic transforms (NTT's). The major disadvantage of the algorithm, enhancement of quantization effects, is absent in the NTT application. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Computer-aided design of separable two-dimensional digital filters

    Publication Year: 1977 , Page(s): 165 - 169
    Cited by:  Papers (37)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (480 KB)  

    A procedure for the design of separable two-dimensional digital filters is presented. The computation involved in both the filter design and implementation is shown to be efficient. Several examples are presented which illustrate the application of the technique. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.

Aims & Scope

This Transactions ceased production in 1990. The current retitled publication is IEEE Transactions on Signal Processing.

Full Aims & Scope