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Circuits and Systems II: Analog and Digital Signal Processing, IEEE Transactions on

Issue 5 • Date May 1997

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Displaying Results 1 - 13 of 13
  • Design of stable, causal 2-D digital filters using real coefficient 2-D all-pass building blocks

    Publication Year: 1997 , Page(s): 409 - 412
    Cited by:  Papers (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (176 KB)  

    Presented here is a method for the design of 2-D causal quarter-plane recursive digital filters with real coefficients and arbitrary magnitude with/without linear phase characteristics, by using all-pass building blocks. It is shown that in general, cascades of sum or difference of two 2-D all-pass filters with appropriate delay elements are required to guarantee the arbitrary shape of the cutoff boundary of the desired filters. To design a 2-D filter satisfying given specifications the binary parameters of the cascaded all-pass structure are adapted from the given table, and the coefficients of the 2-D all-pass filters are obtained via an iterative technique by using a nonlinear optimization method. Design examples are given to illustrate the usefulness of the proposed technique View full abstract»

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  • New operational amplifier using a positive feedback

    Publication Year: 1997 , Page(s): 412 - 417
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (160 KB)  

    A very high gain bipolar op amp was designed using controlled positive feedback. The circuit has a theoretical voltage gain of 160 dB and unity gain-bandwidth of 3.3 MHz. Despite using positive feedback, the op amp is compensated and a slew rate of 1 V/μs is achieved View full abstract»

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  • Macromodeling of hysteresis phenomena with SPICE

    Publication Year: 1997 , Page(s): 378 - 388
    Cited by:  Papers (15)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (328 KB)  

    A macromodel for nonlinear devices exhibiting hysteresis is presented. Although this model is based on existing ideas of ferromagnetic hysteresis, it can be applied to all hysteresis phenomena by choosing suitable parameters. It consists only of four algebro-differential equations with ten parameters and exhibits all main features of hysteresis, such as initial curve, saturation, minor loops, coercivity, and remanence. Using standard elements of the widespread simulator SPICE, the macromodel is applicable to all commonly used circuit simulators. It is difficult to extract parameters of nonlinear differential equations from measured values. Therefore, an algorithm for parameter extraction by optimization is presented which is successfully applied to this hysteresis model View full abstract»

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  • A Q-enhanced active-RLC bandpass filter

    Publication Year: 1997 , Page(s): 341 - 347
    Cited by:  Papers (19)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (232 KB)  

    A fully differential high-Q bandpass filter that uses lossy integrated inductors is presented. The circuit is implemented in a 0.8 μm BiCMOS technology and realizes a center frequency of 750 MHz with a Q-factor that is tunable from 10 to 490 while dissipating 80-100 mW from a single 5 V supply. Since the objective of the prototype was to explore the proposed Q-enhancement technique, the dynamic range is limited to 25 dB for Q=20 View full abstract»

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  • A novel technique for initializing digital IIR filters with a finite number of samples at a single frequency

    Publication Year: 1997 , Page(s): 417 - 420
    Cited by:  Papers (4)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (108 KB)  

    This brief deals with a new initialization technique for digital IIR filters. The technique is suitable for IIR filters processing a finite number of samples in phased array radars. The initialization processor is complex one and utilized the first received sample to force the filter to reach its steady-state value at a single frequency irrespective of the number of processed samples. This is done by initially loading the internal memories of the filter with their steady-state values. If the filter has a zero steady-state response at that particular frequency then this will ensure that the transient frequency responses of the filter will also have a zero response at that frequency. Different processors are derived for filters with real and complex coefficients View full abstract»

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  • Image filtering using hyperstable adaptive algorithms

    Publication Year: 1997 , Page(s): 358 - 370
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (492 KB)  

    In view of recent interest in applications of adaptive filtering of multidimensional (m-D) signals in practical problems such as video compression and image enhancement, implementation of a class of m-D infinite impulse response, adaptive, hyperstable filters is undertaken. While the theoretical results on convergence of such schemes have been made available in our recent work on the topic, the present paper reports our experiments with enhancement of noise-corrupted images for the first time. Aside from implementing the previously introduced m-D HARF algorithm, we also introduce and implement other variants of the algorithm that are conceptually more transparent, computationally less expensive, or converge faster. The new algorithms emerging from this study, namely, the modified m-D HARF and the m-D SHARF are compared with the earlier m-D HARF algorithm by explicitly deriving measures of computational complexities for both sequential and for parallel implementation, whenever appropriate. Performances of these new algorithms are also studied both theoretically and experimentally View full abstract»

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  • A robust echo canceler for acoustic environments

    Publication Year: 1997 , Page(s): 389 - 396
    Cited by:  Papers (19)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (232 KB)  

    This paper presents a novel acoustic echo canceler for teleconference and speakerphone systems based upon input orthogonalization and spread spectrum communication techniques. The new echo canceler represents a significant departure from traditional methods of removing acoustical echo and solves the open problem of echo cancellation during double-talk transmission. A decorrelation based adaptive filtering algorithm is introduced and its performance studied. A contraction mapping principle is applied to obtain a tight bound on the step size for the convergence of the adaptive algorithm. In conjunction, a direct sequence spread spectrum correlation technique is used to estimate the room impulse response during double-talk. Both components of the echo canceler rely on an innocuous wide-band training signal which is transmitted into the room along with the speech. The new echo canceler was implemented on a TMS320C30 DSP processor and experimental results are presented. The echo canceler was found to give excellent performance for all operating conditions, including during double-talk transmission View full abstract»

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  • Spline approximation using Kalman filter state estimation

    Publication Year: 1997 , Page(s): 421 - 424
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (104 KB)  

    Curves and surfaces in computer-aided design systems are often represented using B-spline basis functions. For a given set of locations corresponding to a subset of sampled points of the curve, there are efficient algorithms which solve a normal system of equations, to compute the basis function weights which are called control vertices. However, when these knot locations are not predetermined, there is no known efficient method for determining the optimal representation of the underlying curve. In this paper, we propose a suboptimal Kalman filter algorithm which “refines” an initial set of knot locations, to provide a “good” solution to the spline approximation problem View full abstract»

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  • Factorizing FSM's with modify and restore method

    Publication Year: 1997 , Page(s): 371 - 377
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (148 KB)  

    Implementation of large finite-state machines (FSMs) as smaller interacting machines, by factorizing them and interconnecting the factored and factoring FSM's in such a way so as to maintain the functionality of the original machine usually leads to an improvement in the performance (that is, reduction in delay) of the original machine. Exact factors, if present in an FSM, can result in the most effective way of factorization. However, it has been found that most of the FSMs are not exact factorizable. In this paper, we have presented a scheme called the modify and restore (MAR) method which attempts to make FSMs exact factorizable even if the original FSM is not directly exact factorizable. This is done by making minor changes in the next state space of the original FSM while maintaining the functionality of the FSM by a restoring logic. We have tested the effectiveness of our method of factorization followed by state assignment for both two-level and multilevel implementations. Experimental results on the MCNC benchmark examples have shown significant improvement in delays of the final realization View full abstract»

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  • Anchored blind equalization using the constant modulus algorithm

    Publication Year: 1997 , Page(s): 397 - 403
    Cited by:  Papers (7)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (224 KB)  

    Blind equalization is a technique of adapting an equalizer without the need of a training sequence. The constant modulus algorithm (CMA) is one of the first known blind equalization algorithms. The cost function of the CMA exhibits local minima, which are the primary cause of the ill-convergence of the CMA. Using the CMA with an anchored equalizer greatly improves the performance of the CMA in terms of ill-convergence. This technique is used in this paper with the linear and the decision feedback equalizers. Theory and simulation show that such an adaptive equalizer will always remove intersymbol interference (ISI) provided the main cursor's gain exceeds a certain critical value View full abstract»

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  • Design of two-dimensional FIR digital filters by a two-dimensional WLS technique

    Publication Year: 1997 , Page(s): 348 - 357
    Cited by:  Papers (21)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (328 KB)  

    For two-dimensional (2-D) FIR filter design, the conventional weighted least squares (WLS) technique rearranges the filter parameters of 2-D form into their corresponding one-dimensional (1-D) form, thus resulting in expensive computation. This paper presents a new computationally efficient WLS technique for the design of 2-D FIR filters. We introduce an updating desired frequency response which implicitly includes the weighting function such that the sum of weighted square errors to be minimized can be represented in a 2-D matrix form. This makes it possible to keep all filter parameters in their natural 2-D form, thereby reducing the computational complexity from O(NG ) to O(N3). It is confirmed through design examples that the new technique is computationally very efficient and leads to nearly optimal approximations. This technique is suitable for the design of 2-D real zero-phase FIR filters with quadrantal symmetric or antisymmetric frequency response and can also be applied to the design of 1-D FIR filters View full abstract»

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  • System identification with noisy input-output data using a cumulant-based Steiglitz-McBride algorithm

    Publication Year: 1997 , Page(s): 407 - 409
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (140 KB)  

    In this brief, we propose a cumulant-based, iterative method for identifying a linear time-invariant system from its noisy input/output data. The input and output are assumed to be non-Gaussian, while the input and output noises are assumed to be mutually correlated, colored, and Gaussian. At each iteration, the proposed method minimizes an objective function that asymptotically is equal to a scalar multiple of Steiglitz and McBride's (1965) (ensemble average version) objective function for noise-free data. Unlike Steiglitz and McBride's method, the proposed one is consistent for inputs that are persistently exciting of sufficient order View full abstract»

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  • Low-power BiCMOS continuous-time shaping filter

    Publication Year: 1997 , Page(s): 404 - 406
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (128 KB)  

    A biquadratic continuous-time filter designed to operate as signal shaper in the read-out electronics of elementary particles experiments has been implemented in 2 μm BiCMOS technology. The cell synthesizes a semi-Gaussian response with a shaping time adjustable in the range 18-30 ns. The power consumption is 1.25 mW from a single 5 V power supply. The integral nonlinearity is within 1% for an input signal amplitude up to 200 mV. The chip active area is 0.08 mm2. The measured input referred noise is 50 nV/√Hz View full abstract»

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Aims & Scope

This title ceased production in 2003. The current updated title is IEEE Transactions on Circuits and Systems II: Express Briefs.

Full Aims & Scope