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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 2 • Date April 1978

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Displaying Results 1 - 17 of 17
  • [Front cover and table of contents]

    Page(s): 0
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  • [Back cover]

    Page(s): c4
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  • Hardware modifications in radix-2 cascade FFT processors

    Page(s): 171 - 172
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    In this correspondence, some techniques are presented, which reduce coefficient storage and hardware cost of cascade FFT processors. Computer simulation results are presented which give hardware-error tradeoffs and also show the effect of coefficient accuracy. A reduced hardware multiplier is suggested which gives only insignificant loss in accuracy. View full abstract»

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  • A generalized hamming window

    Page(s): 176 - 177
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    A formal generalization of the Hamming window is given in terms of cosines raised to an arbitrary power. View full abstract»

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  • Speed enhancement of digital signal processing software via microprogramming a general purpose minicomputer

    Page(s): 135 - 140
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    Execution time of digital signal processing (DSP) software can be substantially reduced using automatically generated code which incorporates precomputed, data-independent control and data access parameters. In this paper, further speed enhancement of such software via microprogramming of computation kernels is discussed and demonstrated. It is shown that, in addition to elimination or reduction of instruction fetch and decode overhead, and reduction of data access time via use of high-speed scratchpad registers, microprogramming may permit fundamental operations to be recoded to yield substantially reduced execution times. An example is given whereby Booth's multiplication algorithm is implemented in microcode to yield a fixed-point multiply/add time several times faster than an implementation using the machine-level multiply/add instructions. Moreover, the systematic nature of the computational microcode allows it to be generated by programs using the technique described previously. Finally, program execution times approaching an order of magnitude less than that of conventional assembler language implementations are shown to be feasible by incorporation of all techniques discussed. View full abstract»

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  • Digital image restoration

    Page(s): 178 - 179
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    First Page of the Article
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  • Improved approximation of bias in squared coherence estimates for weakly smooth spectra

    Page(s): 172 - 174
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    Bias in squared coherence estimates for normal processes is approximated by a function of coherence and second derivatives divided by values of spectral densities which may vary slightly over the window width. The result approximates more accurately than a known formula results from nonrefuted literature concerning very smooth spectra. View full abstract»

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  • Notes on linear image restoration by maximizing the a posteriori probability

    Page(s): 174 - 176
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    A distinction is made between the minimum mean-square error estimate derived for the linear case of maximum a posteriori restoration and the common Wiener estimate. The linear MAP filter is shown to produce good results with less a priori knowledge than is required for the Wiener filter. View full abstract»

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  • Sectioned methods for image restoration

    Page(s): 157 - 164
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    Locally adaptive image processing methods are constructed by sectioning the image and applying a modified MAP restoration algorithm. These local algorithms are shown to be effective in processing nonstationary images. The algorithms can work in both signal-independent and signal-dependent noise. The gains achieved by local and signal-dependent processing are analyzed. View full abstract»

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  • Design of recursive digital filters with optimum magnitude and attenuation poles on the unit circle

    Page(s): 150 - 156
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    A class of infinite impulse response (IIR) digital filters with optimum magnitude in the Chebyshev sense, arbitrary attenuation in the passband and stopband, all zeros on the unit circle, and different order numerator and denominator is discussed. Several properties of low-pass filters of this type are described, such as the effect of an extra ripple in the passband and the minimum attainable passband ripple for a given order. An algorithm for the design of these filters is presented, which given the order of the filter, passband edge, stopband edge and passband ripple minimizes the stopband ripple. Alternatively, the stopband ripple can be fixed and the passband ripple minimized. This is done by working with the numerator and denominator separately. This algorithm is fast compared to other existing design procedures. Several examples are presented and compared with the classical elliptic filters. Filters are described which meet the same tolerance scheme as an elliptic filter with fewer multiplications. View full abstract»

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  • Convolution using a conjugate symmetry property for the generalized discrete Fourier transform

    Page(s): 165 - 170
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    Often, signals which lie in a ring S are convolved using a generalized discrete Fourier transform (DFT) over an extension ring R in order to allow longer sequence lengths. In this paper, a conjugate symmetry property which generalizes the well known property of the complex DFT for real data is presented for this situation. This property is used to obtain a technique for computing the DFT of μ sequences with values in a ring S using a single DFT in an extension ring R of degree μ over S. From this result, a method to compute the convolution of length μn S-sequences using a length n DFT in R is derived. Example of the application to the complex DFT and to a number theoretic transform are presented to illustrate the theory. View full abstract»

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  • Fast inversion of banded Toeplitz matrices by circular decompositions

    Page(s): 121 - 126
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    Banded Toeplitz matrices of large size occur in many practical problems [1]-[6]. Here the problem of inversion as well as the problem of solving simultaneous equations of the type Hx = y, when H is a large banded Toeplitz matrix, are considered. It is shown via certain circular decompositions of H that such equations may be exactly solved inO(N log_{2} N)rather than in O(N2) computations as in Levinson-Trench algorithms. Furthermore, the algorithms of this paper are nonrecursive (as compared to the Levinson-Trench algorithms), and afford parallel processor architectures and others such as transversal filters [17] where the computation time becomes proportional to N rather than toN log N. Finally, a principle of matrix decomposition for fast inversion of matrices is introduced as a generalization of the philosophy of this paper. View full abstract»

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  • A comparative study of time efficient FFT and WFTA programs for general purpose computers

    Page(s): 141 - 150
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    Time efficient autogen software implementations of the fast Fourier transform (FFT) and the Winograd Fourier transform algorithm (WFTA) are examined and compared in detail. Both high-level language (optimized Fortran IV) and assembler implementations are considered on two general purpose computers, the DEC PDP-11/55 and the IBM- 370/168, both having floating-point multiply/add time ratios of about 1.17. It is shown that although the WFTA reduces the number of multiplications relative to the FFT, a substantial increase in data transfer, both memory/register and register/register, together with a smaller increase in additions and data reordering overhead, combine to give WFTA execution times about 20-40 percent longer than those for the FFT. These results are explained by examining the internal computational kernel structure for both algorithms and relating the arithmetic operation sequencing to the computer instructions necessary to implement the software. It is concluded that for floating-point software implementations on the class of general purpose computers considered, the WFTA offers neither time nor space advantages over the radix-4 FFT. View full abstract»

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  • Control of limit cycle oscillations in second-order recursive digital filters using constrained random quantization

    Page(s): 127 - 134
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    Limit cycles occur in recursive digital filters due to quantization of fixed point products. A common method of preventing these limit cycles from degrading the performance of the filter is to increase the internal wordlength of the filter beyond the input encoding accuracy, increasing the cost of the filter. This paper proposes two techniques which suppress limit cycles. One involves constrained random quantization of the output of the B2multiplier of a standard second-order digital filter section. A second technique detects and eliminates dc and half sampling rate limit cycles. These techniques control all kinds of limit cycles produced by recursive filters with zero input, at the cost of a modest increase in circuit complexity. 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