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Showing papers on "Filter design published in 1988"


Proceedings ArticleDOI
C.W. Farrow1
07 Jun 1988
TL;DR: An FIR (finite-impulse-response) filter which synthesizes a controllable delay which has the ability to interpolate between samples in the data stream of a band-limited signal is described.
Abstract: The author describes an FIR (finite-impulse-response) filter which synthesizes a controllable delay. By changing the delay the filter has the ability to interpolate between samples in the data stream of a band-limited signal. Because high sampling rates are not required, the filter is especially suited for implementation in a digital signal processor (DSP), and has been implemented in a real-time DSP. The interpolator can be used as a practical way to reconstruct an original band limited signal from samples taken at the Nyquist rate. The variable delay filter can also be used as a more general computational element. Performance results are presented. >

853 citations


Journal ArticleDOI
TL;DR: In this article, a novel method is proposed for realizing exact inverse filtering of acoustic impulse responses in room, based on the principle called the multiple-input/output inverse theorem (MINT).
Abstract: A novel method is proposed for realizing exact inverse filtering of acoustic impulse responses in room. This method is based on the principle called the multiple-input/output inverse theorem (MINT). The inverse is constructed from multiple finite-impulse response (FIR) filters (transversal filters) by adding some extra acoustic signal-transmission channels produced by multiple loudspeakers or microphones. The coefficients of these FIR filters can be computed by the well-known rules of matrix algebra. Inverse filtering in a sound field is investigated experimentally. It is shown that the proposed method is greatly superior to previous methods that use only one acoustic signal-transmission channel. The results prove the possibility of sound reproduction and sound reception without any distortion caused by reflected sounds. >

734 citations


Journal ArticleDOI
TL;DR: This paper presents a 3rd order low-pass continuous-time filter with 4 MHz cut-off frequency, integrated in a 3 μm CMOS process, based on the direct simulation of a doubly-terminated LC ladder using capacitors and fully-balanced, current-controlled transconductance amplifiers with extended linear range.
Abstract: A third-order elliptic low-pass continuous-time filter with a 4-MHz cutoff frequency, integrated in a 3- mu m p-well CMOS process, is presented. The design procedure is based on the direct simulation of a doubly terminated LC ladder filter by capacitors and fully balanced, current-controlled transconductance amplifiers with extended linear range. The on-chip automatic tuning circuit uses a phase-locked loop implemented with an 8.5-MHz controlled oscillator that matches a specific two-integrator loop of the filter. The complete circuit features 70-dB dynamic range (THD >

652 citations


Journal ArticleDOI
TL;DR: A class of finite-impulse response (FIR) median hybrid (FMH) filters that contain linear FIR substructures to estimate the current signal value using forward and backward prediction is introduced and Predictors maximizing the signal-to-noise ratio on signal sections described by an lth-order polynominal are derived.
Abstract: A class of finite-impulse response (FIR) median hybrid (FMH) filters that contain linear FIR substructures to estimate the current signal value using forward and backward prediction is introduced. The output of the overall filter is the median of the predicted values and the actual signal value in the middle of the filter window. Predictors maximizing the signal-to-noise ratio on signal sections described by an lth-order polynominal are derived. The ramp enhancement filters are shown to attenuate the noise on a ramp signal better than the standard median (SM) filters. The new predictive FMH filters are shown to have root signals which do not exist for the SM filters, e.g. triangular waves. By combining the level and the ramp enhancement FMH filters, a filter is obtained which attenuates noise on constant and ramp signals. The noise attenuation on ramp signals is better than with the SM filter, and the predictive FMH filter has novel and meaningful root structures. The number of arithmetic operations needed to implement the predictive FMH filter grows linearly with the length of the filter. >

244 citations


PatentDOI
TL;DR: In this paper, the spectral energy analysis is carried out using pairs of high pass and low pass digital filters in cascade relation, with the output of each low pass filter being provided to the next pair of high-pass and low-pass filters.
Abstract: A hearing aid system utilizes digital signal processing to correct for the hearing deficit of a particular user and to maximize the intelligibility of the desired audio signal relative to noise. An analog signal from a microphone is converted to digital data which is operated on by a digital signal processor, with the output of the digital signal processor being converted back to an analog signal which is amplified and provided to the user. The digital signal processor includes a time varying spectral filter having filter coefficients which can be varied on a quasi-real time basis to spectrally shape the signal to match the hearing deficit of the user and to accommodate ambient signal and noise levels. The coefficients of the spectral filter are determined by estimating the energy in several frequency bands within the frequency range of the input signal, and using those energy estimates to calculate desired gains for the frequency bands and corresponding spectral filter coefficients. The spectral energy analysis may be carried out using pairs of high pass and low pass digital filters in cascade relation, with the output of each low pass filter being provided to the next pair of high pass and low pass filters. The rate at which output data is provided from the filters in each pair may be reduced from the sample rate of input data by one half for succeeding pairs of filters in the cascade to thereby reduce the computation time required.

228 citations


Journal ArticleDOI
02 Oct 1988
TL;DR: In this article, an active power filter using quad-series voltage source pulsewidth modulated (PWM) converters is presented, where instantaneous space vectors of voltage and current are used for the analysis.
Abstract: The modeling, analysis, and design of an active power filter using quad-series voltage source pulsewidth modulated (PWM) converters are presented. Some instantaneous space vectors of voltage and current are used for the analysis. A vector differential equation derived in this paper makes it easy to achieve the analysis and design of the active power filter. Experimental waveforms obtained from a prototype active power filter with a rating of 7 kVA, along with simulation waveforms, are included to verify the theory presented. >

175 citations


Journal ArticleDOI
TL;DR: In this paper, the design of Hilbert transformers and differentiators by eigenfilters is presented, where the filter coefficients are computed from an eigenvector of an appropriate real, symmetric, and positive-definite matrix.
Abstract: The design of Hilbert transformers and differentiators by eigenfilters is presented. The filter coefficients are computed from an eigenvector of an appropriate real, symmetric, and positive-definite matrix. The method is simple, and the performance is better than that of McClellan-Parks algorithm. The design time is fast and comparable for filter length less than one hundred. >

125 citations


Journal ArticleDOI
TL;DR: The authors show that by cascading two direct-form FIR filters, each with coefficients that are the sum or difference of two power-of-two terms, it is possible to achieve very small peak ripple.
Abstract: The authors show that by cascading two direct-form FIR filters, each with coefficients that are the sum or difference of two power-of-two terms, it is possible to achieve very small peak ripple. An iterative equalization strategy is used in the design of the cascaded filter. The success of the method depends on the initial prototype filter being used. Two equally effective methods are presented for selecting the prototype filter; each yields a final design with good roundoff noise property. >

111 citations


PatentDOI
TL;DR: In this article, a linear filter is used to adjust the coefficients of the digital filter in the feedback path to cancel out the acoustic feedback signal to reduce the build-up of feedback resonances.
Abstract: Acoustic feedback in digital signal processing hearing aids is suppressed by using signal processing techniques in the digital processor. A first processing technique causes the data to the main signal processing path in the digital signal processor to be delayed by varying amounts over time, preferably in a periodic manner, to disrupt the buildup of feedback resonances. In a second technique, a digital filter receives the input data and has its coefficients adjusted so that the output of the filter is substantially an optimal estimate of the current input sample based on past input samples. The output of the filter is then subtracted from the input signal data to provide difference signal data which substantially cancels out the resonant frequencies. In a third technique, the acoustic feedback path from the output to the input of the hearing aid is modeled in the digital signal processor as a delay and a linear filter. The output of the main signal processing path in the digital signal processor is delayed and the delayed data passed through the linear filter, with the output of the filter then being substracted from the input signal data to provide difference signal data which is provided to the main signal processing path. The coefficients of the digital filter in the feedback path are adjusted so that the signal passed through the feedback filter substantially corresponds to of the acoustic feedback signal to thereby cancel the same.

101 citations


Journal ArticleDOI
TL;DR: In this article, a suboptimal Kalman filter design method is presented for the problem of tracking a maneuvering target, which is essentially based on linear target dynamics and linear-like structured measurements called pseudomeasurements.
Abstract: A suboptimal Kalman filter design method is presented for the problem of tracking a maneuvering target. The design method is essentially based on linear target dynamics and linear-like structured measurements called pseudomeasurements. The pseudomeasurements are obtained by manipulating the original nonlinear measurements algebraically. The resulting filter has computational advantages over other filters with similar performance. Also, a variant of the Berg model is proposed as a target acceleration model under the assumption of a coordinated turn maneuver. The proposed model is consistent with the underlying assumption. Monte Carlo computer simulation results are included to demonstrate the effectiveness of the proposed suboptimal filter associated with the target acceleration model. >

99 citations


Proceedings ArticleDOI
25 May 1988
TL;DR: In this article, a construction technique is described that allows microstrip parallel-coupled filters to have greater passband symmetry while largely removing the parasitic passband at twice the center frequency.
Abstract: A construction technique is described that allows microstrip parallel-coupled filters to have greater passband symmetry while largely removing the parasitic passband at twice the center frequency. The filter design extends the odd mode phase length by allowing the coupled lines to overlap with lines outside the resonator proper, i.e. the coupling gap of the resonator is extended out onto the 50- Omega lines. The even-mode length has been set by the distance between the 50- Omega lines. Thus, the reference plane for the even mode has been moved into the resonator. This configuration makes the odd-mode length longer than the even mode and thus compensates for the phase velocity difference between modes. >

Journal ArticleDOI
TL;DR: This paper first develops a theory for optimal binary phase-only filters and then presents a numerical algorithm which designs the optimal binaryphase-only matched filter for a given image.
Abstract: Many spatial light modulators and computer-generated hologram techniques can very efficiently implement binary phase-only filters. At present, almost all binary phase-only filters are designed by first designing a matched spatial filter and then binarizing it. There is no theoretical basis to expect that such a filter will be optimal. This paper first develops a theory for optimal binary phase-only filters and then presents a numerical algorithm which designs the optimal binary phase-only matched filter for a given image. The filter is optimal in the matched filter sense of maximizing SNR at the output origin. Characteristics of the optimal filter compared to conventional binary phase-only filters are discussed.

Journal ArticleDOI
TL;DR: In this paper, an improved linear programming algorithm is required for the design of finite impulse response (FIR) digital filters, which avoids the numerical ill-conditioning problems which commonly occur due to the necessity to sample the frequency response on a very dense grid of points for high-order filters.
Abstract: An improved linear programming algorithm is required for the design of finite impulse response (FIR) digital filters. This algorithm avoids the numerical ill-conditioning problems which commonly occur due to the necessity to sample the frequency response on a very dense grid of points for high-order filters. This technique is applied to the design of equiripple FIR Nyquist filters and equiripple FIR transmit and receive matched-filters for data transmission applications. The use of linear programming insures that the designs are optimal in the sense that they achieve the maximum possible stopband attenuation for a given filter order and stopband edge frequency. The design algorithm for the matched filters consists of a two-stage process of linear programming to design a Nyquist filter with a nonnegative frequency response followed by standard spectral factorization techniques to extract the nonlinear-phase transmit and receive filters. The design software consists primarily of commonly available FORTRAN subroutine packages for linear programming and polynomial factorization, and is numerically well behaved and very accurate. >

Journal ArticleDOI
TL;DR: Structures are presented for the perfect-reconstruction quadrature mirror filter bank that are based on lossless building blocks that ensure that the frequency responses of the analysis (and synthesis) filters have pairwise symmetry with respect to pi /2.
Abstract: Structures are presented for the perfect-reconstruction quadrature mirror filter bank that are based on lossless building blocks. These structures ensure that the frequency responses of the analysis (and synthesis) filters have pairwise symmetry with respect to pi /2 and require fewer parameters than recently reported structures (also based on lossless building blocks). The design time for the proposed structures is correspondingly much less than for the earlier methods, which did not incorporate such symmetry. >

PatentDOI
TL;DR: In this paper, a digital echo cancellation with a good protection against double-talk is proposed, in which the first filter is a time-domain programmable filter (TDPF) and the second one is a frequency-domain block-adaptive (FDAF).
Abstract: A digital echo canceller [1] with a good protection against double-talk comprises a first filter [9] for the echo cancellation proper, a second adaptive filter [11] for making continuously adapted filter coefficients available to the first filter [9] and gate means [15] which will only apply the adapted filter coefficients to the first filter [9] if certain conditions are satisfied. In the echo canceller [1] a specific filter combination is used, in which the first filter [9] is a time-domain programmable filter (TDPF) and the second filter [11] is a frequency-domain block-adaptive (FDAF), and further means [20,20(1)] are provided that are arranged in cascade with the gate means [15] for transforming the frequency-domain filter coefficients [W(p;m)] of the FDAF [11] into time-domain filter coefficients [w(i;m)] for the TDPF [9]. These measures result in the advantages of a negligible delay in the echo cancellation by the TDPF [9], and of the possibility to improve in a simple manner the convergence behavior of the FDAF [11] and considerably reduce its computational complexity.

Journal ArticleDOI
TL;DR: In this article, the Wiener filter is formulated as a function of the basic image-gathering and image-reconstruction constraints, thereby providing a method for minimizing the mean-squared error between the radiance field and its restored (continuous-output) representation.
Abstract: The Wiener filter is formulated as a function of the basic image-gathering and image-reconstruction constraints, thereby providing a method for minimizing the mean-squared error between the (continuous-input) radiance field and its restored (continuous-output) representation. This formulation of the Wiener filter is further extended to the Wiener-characteristic filter, which provides a method for explicitly specifying the desired representation. Two specific examples of Wiener filters are presented.

Journal ArticleDOI
TL;DR: It is shown that the alpha -TM and STM filters perform better than the running mean and median filters in white noise suppression, while they can be designed to be comparable to the median filter in edge preservation in the presence of noise.
Abstract: The statistical properties of two classes of filters generalizing the median filter are considered. The two classes are the alpha-trimmed mean ( alpha -TM) filter and the standard type M(STM) filter, both of which are special cases of the L filter. Results are developed to quantify the white-noise suppression and edge-preservation characteristics of the filters by considering their output sequence error statistics. It is shown that the alpha -TM and STM filters perform better than the running mean and median filters in white noise suppression, while they can be designed to be comparable to the median filter in edge preservation in the presence of noise. >

Patent
20 Sep 1988
TL;DR: In this paper, a digital filter tree composed of a plurality of digital filter banks arranged in a tree structure one behind the other to branch out in stages with a separation into L.sub.ν individual signals taking place in each stage and the sampling rate being reduced each time by the factor M.Sub.ν =4 where ν = 1,2,... identifies the νth stage.
Abstract: The invention relates to a digital filter tree composed of a plurality of digital filter banks arranged in a tree structure one behind the other to branch out in stages with a separation into L.sub.ν individual signals taking place in each stage and the sampling rate being reduced each time by the factor M.sub.ν where ν=1,2, . . . identifies the νth stage. The filter tree employs a prototype filter with half-band functions for channel center frequencies fl =l·B+B/2, with a real frequency multiplex input signal being separated into L.sub.ν complex channel signals for further processing by means of a discrete Fourier transformation. For all stages M.sub.ν =2 and L.sub.ν =4 are fixed, with only two signals of the L.sub.ν =4 being utilized. The arrangement permits adaptation of a hierarchical multi-stage method also to numbers of channels which are not equal to a power of two without changing the input sampling frequency and without the causing channels, whose number is fixed by the difference from the next higher power of two, to idle.

Journal ArticleDOI
TL;DR: Improvements to the SELP algorithm are described which result in better speech quality and higher computational efficiency, and a new recursive algorithm which performs a very fast search through the adaptive codebook.

Proceedings ArticleDOI
11 Apr 1988
TL;DR: The contribution of this work is to quantify and justify the functional relationships between image features and filter parameters so that the design process can be easily modified for different conditions of noise and scale.
Abstract: A procedure for filter design is described for enhancing fingerprint images. Four steps of this procedure are described: user specification of appropriate image features, determination of local ridge orientations throughout the image, smoothing of this orientation image, and pixel-by-pixel image enhancement by application of oriented, matched filter masks. The contribution of this work is to quantify and justify the functional relationships between image features and filter parameters so that the design process can be easily modified for different conditions of noise and scale. Application of the filter shows good ridge separation, continuity, and background noise reduction. >

Journal ArticleDOI
TL;DR: The hyperbolic rotation algorithm is shown to be forward (weakly) stable and, in fact, comparable to an orthogonal downdating method showing to be backward stable by Stewart, and how the method's accuracy depends upon the conditioning is shown.

Journal ArticleDOI
TL;DR: A novel median filter using analog tapped delay lines is designed for real-time signal processing that requires only N+1 analog comparators for a window size of size N and can be realized as a general order-statistics filter with moderate increase of circuit complexity.
Abstract: A novel median filter using analog tapped delay lines is designed for real-time signal processing. It requires only N+1 analog comparators for a window size of size N. As an extension, a general order-statistics filter can also be realized with moderate increase of circuit complexity. For current available fabrication technology for MOS switched-capacitor circuits, the filter can work at a clock rate up to 10 MHz. This development opens up possibilities for practical applications of nonlinear filtering in real-time signal processing, since the analog circuitry is far less complex than the corresponding digital circuitry. >

Journal ArticleDOI
TL;DR: In this article, a multiplierless linear-phase FIR digital filter for compensating the sinc(x) frequency-response distortion resulting from D/A conversion is presented, which can be implemented using only a few adders and delay registers.
Abstract: Designs are presented for multiplierless linear-phase FIR digital filters for compensating the sinc(x) frequency-response distortion resulting from D/A conversion. The filter coefficients are represented by a simple canonic single-digit code, and thus the filter can be implemented using only a few adders and delay registers. A 7-tap design is given which compensates the D/A distortion to within +or-0.045 dB from DC through 0.41 f/sub s/ and an 11-tap design is given which compensates the D/A distortion to within +or-0.028 dB from DC through 0.045 f/sub s/. >

Journal ArticleDOI
TL;DR: It is demonstrated through analysis and simulation that BPOFs can be designed to perform well with respect to stochastic noise.
Abstract: A binary phase-only filter (BPOF) bandwidth and, correspondingly, the performance with respect to stochastic noise are introduced as filter design parameters. A BPOF figure of merit is defined which references the matched filter. Analytical bounds on the BPOF signal-to-noise ratio are derived. The noise performance is illustrated with simulation results. It is demonstrated through analysis and simulation that BPOFs can be designed to perform well with respect to stochastic noise.

Proceedings ArticleDOI
07 Jun 1988
TL;DR: The Lapped Orthogonal Transform (LOT) as discussed by the authors is a novel transform for block signal processing with overlapping basis functions, which can also be viewed as an efficient quadrature-mirror filter bank in which the analysis and synthesis filters have identical FIR (finite-impulse-response) responses.
Abstract: The lapped orthogonal transform (LOT) is a novel transform for block signal processing with overlapping basis functions. The LOT can also be viewed as an efficient quadrature-mirror-filter bank in which the analysis and synthesis filters have identical FIR (finite-impulse-response) responses. The author shows that the LOT filter bank leads to perfect signal reconstruction with a relative low level of spectral images. The main advantages of the LOT filter bank are its short filter lengths (twice the number of bands) and its fast algorithm, based on the fast discrete cosine transform. >

Journal ArticleDOI
TL;DR: In this article, small-signal characteristics of current-mode-controlled PWM converters with a second-stage LC filter are analyzed, and it is shown that a secondary filter can be designed to provide good attenuation of the switching ripple while maintaining adequate stability margins with capacitive loading.
Abstract: Small-signal characteristics of current-mode-controlled PWM converters with a second-stage LC filter are analyzed It is shown that a secondary filter can be designed to provide good attenuation of the switching ripple while maintaining adequate stability margins with capacitive loading The resonant frequencies and damping coefficients of the second filter are derived, and design guidelines are given It is shown that the current-loop gain of the buck converter is not affected by the addition of the second-stage filters when a small filter inductance is used Three design examples are presented to demonstrate the use of analysis results Two filter examples are designed for a buck converter One of the second filters shows the problems that arise with a poor design A third example is the design of a second-stage filter for a buck-boost converter In each of the design examples, the small-signal analysis was performed using EASY5 software and the circuits were simulated using the state-space simulation program COSMIR >

Journal ArticleDOI
TL;DR: The implementation of two recently introduced variable digital filter schemes using a TMS320-series digital signal processor is presented and the measured frequency responses compare well with the theory.
Abstract: The implementation of two recently introduced variable digital filter schemes using a TMS320-series digital signal processor is presented. One is a method for updating the coefficients of an FIR (finite-impulse response) filter in a simple manner such that the cutoff frequencies can be controlled through a single parameter. The other is a method for tuning the cutoff frequency of an IIR (infinite-impulse response) filter with one parameter using a series expansion of the low-pass-low-pass frequency transformation. The measured frequency responses compare well with the theory. >

Journal ArticleDOI
TL;DR: In this article, the Lagrange multipliers were used for the design of optimal finite-duration impulse-response (FIR) digital filters for multirate signal processing applications, where the filter passbands are constrained to be maximally flat, and the integral of the aliasing error is minimized.
Abstract: The method of Lagrange multipliers is presented for the design of optimal finite-duration impulse-response (FIR) digital filters for multirate signal processing applications. The filter passbands are constrained to be maximally flat, and the integral of the aliasing error is minimized. The use of these optimality criteria for practical applications is justified. It is shown that the overall frequency response can be represented as a linear combination of individual frequency responses corresponding to the separate design constraints, with the Lagrange multipliers indicating the sensitivity of the integrated aliasing error to changes in the constrained values. The constraints can be modified and the filter can be updated with only a small number of computations. In the case of maximally flat filters it is established that the constraint matrix is of the Vandermonde type, and, using this result, it is shown that a unique, optimal filter always exists. Its coefficients can be determined by solving a linear system of equations. >

Journal ArticleDOI
TL;DR: The proposed algorithm allows for the design of time-shared Doppler/imaging systems that carry out pulsed or continuous Dopplers measurements with essentially real-time imaging guidance.
Abstract: Methods for obtaining image-guided Doppler blood velocity measurements are briefly reviewed Preference is given to a time-sharing scheme in which the Doppler measurement is turned off during the data-acquisition period for a 2-D image frame (typically 20 ms) The signal dropout that occurs during the image-updating period is removed from the Doppler audio by inserting a synthetic signal segment The synthetic signal is generated by passing white noise through a discrete-time FIR (finite-impulse response) filter with filter coefficients that are a windowed version of the Doppler signal measured immediately prior to the imaging interrupt It is shown that the artificial signal has spectral properties (and thus audible sound) similar to those of the real Doppler signal segment on which it is based The time-sharing method is analyzed and evaluated experimentally, using dedicated hardware The proposed algorithm allows for the design of time-shared Doppler/imaging systems that carry out pulsed or continuous Doppler measurements with essentially real-time imaging guidance >

Proceedings ArticleDOI
07 Jun 1988
TL;DR: A necessary and sufficient condition for perfect reconstruction in multidimensional filter banks is derived, and an approach to filter-bank design using cascaded polyphase matrices and a statistical filter design method is presented as discussed by the authors.
Abstract: The theory and design of multidimensional perfect reconstruction filter banks are discussed. A necessary and sufficient condition for perfect reconstruction in multidimensional filter banks is derived, and an approach to filter-bank design using cascaded polyphase matrices and a statistical filter design method is presented. The approach is applicable to arbitrary (not necessarily rectangular) decimation and interpolation schemes and any number of channels. The results of a preliminary numerical design example are presented. >