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Audio and Electroacoustics, IEEE Transactions on

Issue 2 • Date June 1970

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Displaying Results 1 - 20 of 20
  • [Front cover and table of contents]

    Publication Year: 1970 , Page(s): 0
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    Freely Available from IEEE
  • Editorial

    Publication Year: 1970 , Page(s): 81 - 82
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    Freely Available from IEEE
  • [Back cover]

    Publication Year: 1970 , Page(s): c4
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    Freely Available from IEEE
  • IEEE Proposed standard for measurement of loudspeaker electrical impedance

    Publication Year: 1970 , Page(s): 213
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  • An approach to the approximation problem for nonrecursive digital filters

    Publication Year: 1970 , Page(s): 83 - 106
    Cited by:  Papers (68)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (2208 KB)  

    A direct design procedure for nonrecursive digital filters, based primarily on the frequency-response characteristic of the desired filters, is presented. An optimization technique is used to minimize the maximum deviation of the synthesized filter from the ideal filter over some frequence range. Using this frequency-sampling technique, a wide variety of low-pass and bandpass filters have been designed, as well as several wide-band differentiators. Some experimental results on truncation of the filter coefficients are also presented. A brief discussion of the technique of nonuniform sampling is also included. View full abstract»

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  • Time domain design of recursive digital filters

    Publication Year: 1970 , Page(s): 137 - 141
    Cited by:  Papers (74)  |  Patents (3)
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    The problem of synthesis of recursive digital filters to give a desired pulse response over a specified interval is studied. Realizability conditions are stated and a linear design method is developed. Several design procedures that require only linear calculations are given for approximate realization of recursive filters. Finally, an error analysis of the techniques is made. View full abstract»

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  • Synthesis of recursive digital filters using the FFT

    Publication Year: 1970 , Page(s): 211 - 212
    Cited by:  Papers (4)
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    It has been shown that recursive digital filters can be synthesized using the fast Fourier transform. An algorithm for computer implementation has been developed and used in comparing the computation times and noise figures of filters synthesized in this manner with the computation times and noise figures of filters synthesized by recursion. A model has been proposed for analysis of noise in the two-pole filter. Predictions of this model have been found to be in good agreement with noise measurements. View full abstract»

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  • Adaptive digital filters for equalization of telephone channels

    Publication Year: 1970 , Page(s): 195 - 200
    Cited by:  Papers (1)  |  Patents (4)
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    An application of digital filtering to communication over a channel having amplitude and phase distortion in the frequency band occupied by the transmitted signal is presented. Described are a nonrecursive and a recursive digital filter which can serve as adaptive equalizers in compensating for the amplitude and phase distortion caused by the channel. The filter coefficients are adjusted automatically by the use of a test signal which is transmitted over the channel. The adjustment of the coefficients is carried out in the presence of noise with the aid of a steepest descent algorithm. The recursive filter, consisting of a comb filter in cascade with a bank of parallel two-pole filters, is shown to be especially suited for performing equalization. By choosing as a test signal sinusoids whose frequencies coincide with the poles of the two-pole filters, the coefficients of the recursive filter are easily adjusted. View full abstract»

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  • The design of wide-band recursive and nonrecursive digital differentiators

    Publication Year: 1970 , Page(s): 204 - 209
    Cited by:  Papers (30)
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    Designs for recursive and nonrecursive wide-band differentiators are presented. The coefficients for the recursive differentiators were optimally chosen to minimize a square-error criterion based on the magnitude of the frequency response. The coefficients for the nonrecursive differentiators were chosen using a frequency sampling technique. One or more of the coefficients were optimally selected to minimize the peak absolute error between the obtained frequency response and the response of an ideal differentiator. The frequency response characteristics of the recursive differentiators had small magnitude errors but significant phase errors. The nonrecursive differentiators required on the order of 16 to 32 terms for the magnitude error of the frequency response to be as small as the magnitude errors for the recursive differentiators; however, there were no phase errors for the nonrecursive case. The delay of the recursive differentiators was small compared to the delay of the nonrecursive differentiators. View full abstract»

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  • Computer-aided design of recursive digital filters

    Publication Year: 1970 , Page(s): 123 - 129
    Cited by:  Papers (78)
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    A practical method is described for designing recursive digital filters with arbitrary, prescribed magnitude characteristics. The method uses the Fletcher-Powell optimization algorithm to minimize a square-error criterion in the frequency domain. A strategy is described whereby stability and minimum-phase constraints are observed, while still using the unconstrained optimization algorithm. The cascade canonic form is used, so that the resultant filters can be realized accurately and simply. Design examples are given of low-pass, wide-band differentiator, linear discriminator, and vowel formant filters. View full abstract»

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  • An algorithm for the numerical spectral estimation of a band -limited continuous signal

    Publication Year: 1970 , Page(s): 210 - 211
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  • Nonrecursive digital filtering using cascade fast Fourier transformers

    Publication Year: 1970 , Page(s): 177 - 183
    Cited by:  Papers (8)
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    The organization of a special-purpose digital processor for performing nonrecursive digital filtering is described. The processor uses two complementary cascade fast Fourier transformers. Each transformer can simultaneously transform two independent data blocks of length N words using \log _{2} N arithmetic units and 3/2 N complex words of digital storage. Continuous filtering is achieved by sectioning the input signal, performing a fast transform on each section, multiplying by the frequency characteristics of the desired filter, and inverse transforming. The cascade organization of the processor allows processing at very high speeds. Word rates in excess of 3 MHz are possible with currently available hardware. View full abstract»

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  • Audio frequency spectrum analysis using MOS memory elements

    Publication Year: 1970 , Page(s): 201 - 203
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    The speed of present-day logic elements is such that it is possible to effect real-time spectrum analysis of audio frequency signals by means of digital hardware. This paper describes a structure that computes the discrete Fourier transform of a 400-point time sequence in 10 ms. This allows the real-time analysis of signals having components in the range from dc to 5 kHz. A high-speed bipolar arithmetic section is used in conjunction with relatively slower MOS shift registers and READ-ONLY memory to provide an efficient and economical design. View full abstract»

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  • A special-purpose online processor for bandpass analysis

    Publication Year: 1970 , Page(s): 188 - 194
    Cited by:  Papers (1)  |  Patents (2)
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    The design of a special-purpose on-line processor for bandpass analysis with application to spectral estimation is described. The processor utilizes a digital filtering procedure based on the frequency shift of the sampled signal spectrum. A peculiarity of the processor is that it performs a bandpass analysis by using a unique set of digital filter coefficients for all the analyzed bands. A small and simple memory is therefore used as coefficient memory, while a special, but not complex, organization with many sections is required for the sample memory. The processing of an audio signal to obtain a type of digital vocoder is also considered. From the point of view of the actual implementation, only MSI and LSI circuits are used for the memory and arithmetic units. View full abstract»

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  • Digital filtering via block recursion

    Publication Year: 1970 , Page(s): 169 - 176
    Cited by:  Papers (55)
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    Digital filters which have poles in their transfer functions are usually implemented by various direct feedback arrangements. Such filters can also be synthesized in a form amenable to implementation via the FFT. The key feature of this procedure is block feedback through a special finite-response filter. Although the procedure appears to offer few immediate practical advantages, its asymptotic properties are interesting. It also provides a useful conceptual link between recursive and nonrecursive filtering techniques. View full abstract»

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  • Pulse-transfer-function identification using discrete orthonormal sequences

    Publication Year: 1970 , Page(s): 184 - 187
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    A method for estimating the pulse-transfer function of a linear time-invariant sampled-data system from on-line input and output records is presented. The impulse response of the system is approximated by a finite linear combination of a set of discrete orthonormal sequences. These sequences can be generated by a chain of digital filters. The method is based on the fact that the system output at a given instant is the sum of the products of the system Fourier coefficients and the up-to-date input Fourier coefficients generated by passing the input through the chain of digital filters. The system model coefficients are chosen to give a least-squares fit between the actual system outputs and the model outputs. A recursive form of the estimation equations is derived. This allows the estimates to be updated easily with additional observations. View full abstract»

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  • Realization of digital filters using block-floating-point arithmetic

    Publication Year: 1970 , Page(s): 130 - 136
    Cited by:  Papers (35)  |  Patents (3)
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    Recently, statistical models for the effects of roundoff noise in fixed-point and floating-point realizations of digital filters have been proposed and verified, and a comparison between these realizations presented. In this paper a structure for implementing digital filters using block-floating-point arithmetic is proposed and a statistical analysis of the effects of roundoff noise is carried out. On the basis of this analysis, block-floating-point is compared to fixed-point and floating-point arithmetic with regard to roundoff noise effects. View full abstract»

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  • FFT organizations for high-speed digital filtering

    Publication Year: 1970 , Page(s): 159 - 168
    Cited by:  Papers (5)
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    The subject of this paper concerns the analysis of high-radix FFT algorithms appropriate to hardware implementation of nonrecursive filters and the error propagation properties associated with these algorithms. Variations in the structure of high-radix fast Fourier transforms are illustrated by means of their Kronecker product expansions. The implications in hardware design of the different structures are discussed. These implications include tradeoffs between memory size and throughput rate as shown by Comparison of the logic organizations of serial and cascade fast Fourier processors. It is shown that in a high-radix cascade processor, significant reduction in memory can often be obtained with a modest increase in complexity of the arithmetic units. The relationship between the permutation matrices specified in a Kronecker factorization, and the structure and addressing of memory is brought out. As an example, the implementation of a nonrecursive digital filter is included using a radix-4 factorization which provides for an extremely simple memory organization. The permutation matrix is such that both data storage and addressing are provided using serial MOS shift registers. In order to compare the accuracy of high-radix algorithms relative to the base-2 algorithm, a fixed-point simulation study of the error propagation properties was conducted of a full radix-4 and a radix-16 FFT for N= 1024 and 256, respectively. The computational errors were compared with corresponding results obtained from the base-2 algorithm. The results of the simulations were also compared with a model which is a modification of that reported by P. D. Welch. The modification consists of the representation of roundoff error buildup in the radix-2 and radix-4 FFT in closed form by a nonhomogeneous difference equation. The illustrative example chosen is the lower bound discussed by Welch. View full abstract»

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  • Designing simple, effective digital filters

    Publication Year: 1970 , Page(s): 142 - 158
    Cited by:  Papers (27)
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    Models for a class of narrow-band digital filters and a class of wide-band digital filters are developed and analyzed The analysis forms a basis for a computer-aided design technique for such filters. Our aim is to show that useful performance can be achieved within the constraints of very simple realizations. View full abstract»

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  • Roundoff-noise analysis for fixed-point digital filters realized in cascade or parallel form

    Publication Year: 1970 , Page(s): 107 - 122
    Cited by:  Papers (115)
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    The roundoff-noise outputs from two transpose configurations, each for the cascade and parallel forms of a digital filter, are analyzed for the case of uncorrelated roundoff noise and fixed dynamic range. Corresponding transpose configurations are compared on the basis of the variance, or total average power, and the peak spectral density of the output roundoff noise. In addition to providing general computational techniques to be employed in choosing an appropriate configuration for the digital filter, these results also indicate useful "rules of thumb" relating to this choice of configuration. Included are indications of good (although not necessarily optimum) sequential orderings and pole-zero pairings for the second-order sections comprising the cascade form. Computational results are presented which indicate that the analysis is quite accurate and useful. View full abstract»

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Aims & Scope

This Transactions ceased production in 1973. The current retitled publication is IEEE Transactions on Signal Processing.

Full Aims & Scope