Showing papers on "Linear phase published in 1994"

Method and circuitry for clock synchronization

Abstract: Circuitry for performing fine phase adjustment within a phase locked loop is described. The phase selector selects an even phase signal and an odd phase signal from the twelve phase signals output by the VCO. The even and odd phase signals are selected by an even select signal and an odd select signal, respectively. The phase interpolator interpolates between the even phase signal and the odd phase signal to generate an output signal. The affect of the even phase signal and the odd phase signal on the output signal is determined by an even weighting signal and an odd weighting signal, respectively. The weighting signals prevent glitches from appearing on the output signal when either the even phase signal or the odd phase signal is switching. A method of performing fine phase adjustment in a phase locked loop is also described.

Subaperture autofocus for synthetic aperture radar

Abstract: A subaperture autofocus algorithm for synthetic aperture radar (SAR) partitions range-compressed phase-history data collected over a full aperture into equal-width subapertures. Application of a one-dimensional Fourier transform to each range bin converts each subaperture data set into a full-scene image (map). Any linear phase difference, or phase ramp, between a pair of subapertures expresses itself as cross-range drift in their maps. A traditional autofocus algorithm fits a polynomial to inferred equal-width phase ramps. If the true phase error function contains significant high-order components, then polynomial regression generates a poor estimate of the phase error function. Instead of filling a polynomial, we fit a sinusoidal function through the inferred phase ramps. An example with a degraded SAR image shows how a sinusoidal correction improves image quality. We compare lower bounds on mean squared error (MSE) for polynomial and sinusoidal parameterizations. Sinusoidal parameterization reduces MSE significantly for model orders greater than five. >

Simple and robust method for the design of allpass filters using least-squares phase error criterion

Abstract: We consider a simple scheme for the design of allpass filters for approximation (or equalization) of a given phase function using a least-squares error criterion. Assuming that the desired phase response is prescribed at a discrete set of frequency points, we formulate a general least-squares equation-error solution with a possible weight function. Based on the general formulation and detailed analysis of the introduced error, we construct a new algorithm for phase approximation. In addition to iterative weighting of the equation error, the nominal value of the desired group delay is also adjusted iteratively to minimize the total phase error measure in equalizer applications. This new feature essentially eliminates the difficult choice of the nominal group delay which is known to have a profound effect on the stability of the designed allpass filter. The proposed method can be used for highpass and bandpass equalization as well, where the total phase error can be further reduced by introducing an adjustable-phase offset in the optimization. The performance of the algorithm is analyzed in detail with examples. First we examine the approximation of a given phase function. Then we study the equalization of the nonlinear phase of various lowpass filters. Also, a bandpass example is included. Finally we demonstrate the use of the algorithm for the design of approximately linear-phase recursive filters as a parallel connection of a delay line and an allpass filter. >

A simplified continuous phase modulator technique

Abstract: A low cost GMSK (Gaussian minimum shift keying) modulation technique is presented. GMSK is chosen as an example although the technique is suitable for any continuous phase constant envelope modulation with discriminator, differential or coherent detection in the targeted receiver. In this technique, a fractional-N synthesizer is used to control the instantaneous frequency and phase of the phase locked synthesizer output. As a result of this approach, no in-phase and quadrature mixer or D/A converters are required. Furthermore, the desired signal can be generated directly at RF with no IF conversion stages. The look-up table to generate the modulation is only one bit wide. /spl Delta/-/spl Sigma/ techniques are used to obtain high accuracy through the long term average of a sequence of the single bits. The narrow band filtering of the transmit data is accomplished in two parts, a precise digital linear phase band-reject part using the one bit stored ROM look-up table and a less precise low pass analog part inherent to the PLL of the synthesizer. >

Multiplier-less FIR filter design using a genetic algorithm

Abstract: A new synthesis technique is described for multiplier-less FIR digital filters consisting of a cascade of primitive linear phase sections. For medium-order filters the search space for an optimal cascade is typically of the order of 1030 and this can be examined in a computation efficient way using the parallel-search capability of a genetic algorithm (GA). A particular form of GA based upon multilevel or structured chromosomes has been developed for the primitive cascade problem. Initial results suggest that, for the cost of increased filter delay, a typical 2:1 advantage can be achieved in both VLSI chip area and clock rate compared to filters designed using the usual equiripple method.

Broadband operation of a Ka-band tapered gyro-traveling wave amplifier

Abstract: A wideband low-voltage millimeter-wave gyro-traveling wave tube (gyro-TWT) amplifier operating in the TE/sub 10/ rectangular waveguide mode at the fundamental cyclotron frequency is under investigation, The device incorporates precise axial tapering of both the magnetic field and the interaction circuit for broadband operation. Experimental results of a wide (33%) instantaneous bandwidth with a small signal gain in excess of 20 dB and saturated efficiency of /spl sim/10% were achieved and shown to be in good agreement with the theory. Reflective instability due to multi-pass effects by mismatches was observed and characterized. Gain and efficiency have been limited by this reflective instability rather than by absolute instabilities which limit the performance of gyro-TWT's with uniform cross-section. The start-oscillation current in terms of the relevant experimental parameters such as the beam velocity ratio (/spl alpha/), magnetic field detuning and reflection coefficient has been measured and compared with theory. Measurements of the phase variation in terms of the RF frequency have shown that the phase varies /spl plusmn/30/spl deg/ from fitted linear phase line. >

Phase lock loop with selectable frequency switching time

Abstract: A phase lock loop (PLL) circuit for controlling an oscillator includes a phase comparator, a loop filter, a reference converter and a feedback converter whose performance characteristics are dynamically controlled so as to provide a phase-locked output signal with both high frequency stepping resolution and low phase locking time The phase comparator compares the relative phases of the reference and feedback signals, and outputs a phase difference signal representing such phase comparison The loop filter, in accordance with a filter bandwidth dynamically selected by a filter control signal, filters the phase difference signal to provide a frequency control signal for a voltage controlled oscillator (VCO) The reference converter is a programmable frequency divider which, in accordance with a reference proportionality factor dynamically selected by a reference control signal, reduces the frequency of the PLL reference signal frequency used by the phase comparator The feedback converter is another programmable frequency divider which, in accordance with a feedback proportionality factor dynamically determined by a feedback control signal, reduces the frequency of the VCO feedback signal frequency used by the phase comparator Each combination of a selected filter bandwidth, a reference proportionality factor and a feedback proportionality factor corresponds to a different time interval within which phase lock is achieved

Generalized symmetric extension for size-limited multirate filter banks

Abstract: A multirate (MR) filter bank is called size-limited if the total number of output samples equals the number of input samples. A method called symmetric extension improved performance in subband image compression systems compared to the earlier method of circular convolution. However, the symmetric extension method was developed only for two-band uniform filter banks, and required even-length linear phase analysis filters. The authors generalize the symmetric extension method to the M-band, possibly nonuniform filter banks, where M/spl ges/2. The length restriction on the analysis filters is relaxed. >

Vector field analysis for oriented patterns

Abstract: Presents a method, based on the properties of vector fields, for the estimation of a set of symbolic descriptors (node, saddle, star-node, improper-node, center, and spiral) from linear orientation fields. Planar first-order phase portraits are used to model the linear orientation fields. A weighted linear estimator is developed to estimate linear phase portraits, using only the flow orientation. A classification scheme for planar first-order phase portraits, based on their local properties: curl, divergence, and deformation is developed. The authors present results of experiments on noise-added synthetic flow patterns and real oriented textures. >

Multi-pass and multi-channel interferometric synthetic aperture radars

Abstract: A method of providing accurate elevation data from three or more complex SAR images. A SAR vehicle is operated to produce a number of SAR images at different grazing angles. The present method provides precise elevation maps derived from the SAR images. In a multiple channel variant of the present invention, more than two antennas collect SAR data from various grazing angles, increasing sensitivity while resolving the ambiguity problem in a mathematically optimum manner. A multiple pass variant of the present invention collects data from more than two passes, and from various grazing angles (interferometric "baselines"). The multiple pass approach and associated processing preserves the high sensitivity of the long-baseline available from dual pass IFSAR while resolving the ambiguity problem in a mathematically optimum manner. Accuracy and ambiguity resolution are improved with each additional pass or channel. The method processes raw SAR data to produce a plurality of complex images therefrom. The complex images are processed to produce a plurality of linear phase matched complex images and predetermined processing parameters. The plurality of linear phase matched complex images are then processed to produce a elevation map. The plurality of linear phase matched complex images are produced by matching noise power contained in each of the complex images to normalize the images to compensate for differences in gain of the systems from scan to scan, matching linear phase values of each of the complex images to correct for phase differences by estimating phase matching parameters for each of the complex images, estimating a phase-difference scale factor for each of the complex images, and ordering each of the complex images in terms of increasing phase difference scale factors. The linear phase values of each of the complex images are matched by computing a pairwise interferogram for adjacent complex images, summing sample-pair products of the interferogram to estimate the linear phase thereof, and computing the phase of the summed sample-pair products. The elevation map is computed by computing an initial elevation map using conventional IFSAR techniques applied to a pair of complex images, estimating additive phase correction values for each of the linear phase matched complex images, and estimating elevation values by maximizing the magnitude of the complex weighted sum of M complex images or complex interferograms, where the weights include the phase attributable to the estimated elevation to produce the elevation map.

Weighted Chebyshev approximation for the design of broadband beamformers using quadratic programming

Abstract: A method to solve a general broadband beamformer design problem is formulated as a quadratic program. As a special case, the minimax near-field design problem of a broadband beamformer is solved as a quadratic programming formulation of the weighted Chebyshev approximation problem. The method can also be applied to the design of multidimensional digital FIR filters with an arbitrarily specified amplitude and phase. For linear phase multidimensional digital FIR filters, the quadratic program becomes a linear program. Examples are given that demonstrate the minimax near-field behavior of the beamformers designed. >

Linear phase cosine modulated maximally decimated filter banks with perfect reconstruction

Abstract: In this paper a new type of maximally decimated FIR cosine modulated filter bank is proposed. Each analysis and synthesis filter in this filter bank has linear phase. We can design the system to have approximate reconstruction property (pseudo-QMF system) or perfect reconstruction property (PR system). The filter bank is paraunitary in the PR case. Although there are 2M channels in the new system, the cost (in terms of design and implementation complexity) is comparable to that of an M channel system. Correspondingly, the coding gain of the new system is also comparable to that of a traditional M channel system (rather than a 2M channel system). Examples will be given to demonstrate that very good attenuation characteristics can be obtained with the new system. >

A multifrequency AM-based ultrasonic system for accuracy distance measurement

Abstract: With conventional time-of-flight sonar ranging systems, it is difficult to obtain a high ranging accuracy due to the finite bandwidth of the transducer used and the serious acoustic attenuation in the air for the high acoustic frequencies. In this paper a multifrequency amplitude modulation (AM)-based sonar system is exploited to obtain information about the high-resolution distance measurement for robotic ranging applications. The target distance is obtained by measuring the linear phase shift of the reflected acoustic waves with respect to the reference signal. In order to analyze the ranging error two theoretical formulations are presented for characterization of the noisy phase measurement. The error effects on the phase measurement of the distorted input waveform due to the acoustic cross coupling are detailed, leading to the development of a multitransmitter sensing configuration. Since multifrequency is used, the nature of the target surface may bring about a certain ranging ambiguity to the system. The error effect of the rough surface is also analyzed at the end of the paper. >

Wiener kernel analysis of inner ear function in the American bullfrog

Abstract: The response of 17 primary auditory nerve fibers in the American bullfrog (Rana catesbeiana) to acoustic noise stimulation of the tympanic membrane was recorded. For each fiber, the first‐ and second‐order Wiener kernels, k1(τ1) and k2(τ1,τ2), were computed by cross correlation of the stimulus and the response. The kernels revealed amplitude and phase characteristics of auditory filters of both phase‐locking and non‐phase‐locking fibers. Wiener kernels of high‐ and midfrequency fibers (best frequency, BF≳500 Hz), implied a simple sandwich model, consisting of a cascade of a linear bandpass filter, a static nonlinearity, a linear low‐pass filter, and a spike generator. The bandpass filter was at least of order 7, and had a linear phase response, for both the high‐ and the midfrequency fibers. Averaged across fibers, filter order 2, and cutoff frequency 451 Hz for the second filter in the model was observed. The responses of low‐frequency fibers (BF<500 Hz) could not be fit with the sandwich model, because ...

A digital filter chip for ECG signal processing

Abstract: A VLSI implementation of a linear-phase digital filter for ECG signal processing has been designed. With a sampling rate of 100 Hz, the passband is from 0.5 Hz to 49.5 Hz with 0.5-dB ripple. The filter architecture is based on the use of recursive running-sum blocks, resulting in a very low computational complexity. Module generators have been used in the layout design for high integration density. The circuit has been designed for a 2.0-/spl mu/m double-metal CMOS technology, having about 34000 transistors and a 15.43-mm/sup 2/ chip area. >

Digital controlled xtal osc

Abstract: According to the present invention, an unlimited number of steps or increments of given size are obtained in a delay line for phase alignment of, for instance, the signal from a crystal oscillator (XO) by momentarily switching between two parallel delay lines. One delay line operates as an active or enabled delay line while the other line is disabled or inactive. It is ensured at the same time that the inactive delay line produces a signal which has the same relative phase as the active delay line, this absolute phase differing by N x 2π, where N is a positive or a negative integer other than zero. The inventive method and the inventive device enable the active delay line to operate constantly within its regulation range and the phase of the local oscillator can be kept continuously locked to the phase of the reference signal. The inventive device also includes an oscillator (2), a phase comparator (5) and counter logic (4) and a further phase comparator (7) and a selection circuit (6) for signal selection from that one of the two parallel delay lines (10, 11) that has been placed in its active mode.

McClellan transform based design techniques for two-dimensional linear-phase FIR filters

Abstract: This paper considers the design problem of two-dimensional (2-D) linear-phase FIR digital filters through the use of the McClellan transform method. We present two methods for determining the coefficients of the McClellan transform when designing 2-D linear-phase FIR digital filters with continuous and powers-of-two coefficients, respectively. Based on the proposed methods, the contour approximation for 1-D (one-dimensional) to 2-D digital filter transformation and finding the band-edge frequencies of the corresponding 1-D FIR digital filter can be achieved simultaneously. This results in that the required complexity of the designed 2-D digital filter satisfying the design specifications can be minimized. Moreover, the proposed methods allow us to employ the McClellan transform with order more than one for enhancing the capability of the original McClellan transform. Considering the determination of the McClellan transform coefficients and the band-edge frequencies for the design, we formulate the design problem as a linear programming optimization problem. Then, an efficient design procedure is presented to avoid the use of time consuming simplex algorithms. For the powers-of-two coefficient design, a discrete design procedure is presented to avoid the use of time-consuming mixed integer linear programming algorithms. Computer simulations for showing the effectiveness of the proposed design techniques are also presented. >

Frequency translation of true time delay signals

Abstract: An apparatus and method for frequency translation of true time delay signals (50) in a phased array radar system is provided. A local oscillator signal (52) is true time delayed (54) or modulo 2π phase shifted and then mixed (56) with a true time delay beamsteering signal (50) from a true time delay circuit (48) to produce a frequency translated transmit signal (58) which is supplied to an antenna element. A receive signal (58) received by an antenna element is mixed (56) with a true time delayed (54) or modulo 2π phase shifted local oscillator signal (52) to provide a frequency translated receive signal (50). The frequency translated receive signal (50) can then be passed through a true time delay circuit (48) and other subsequent processing circuitry. The frequency translation permits higher frequency transmit and receive signals (58) to be used with lower frequency true time delay devices (48).

Linear phase IIR filters composed of two parallel allpass sections

Abstract: A class of approximately linear phase recursive digital filters composed of two allpass sections is introduced. The passband response for these filters is equiripple with the maximum number of alternations as for elliptic filters. By slightly widening the passband region and transferring some zeros close to the poles, the poles are forced to move to locations generating an approximately linear phase in the specified passband. For this class of filters, there exists an analytic formula relating the squared-magnitude response to the passband ripple and the zero locations, making the filter optimization very fast. Several examples illustrate that, especially in narrowband applications, these filters are superior to linear-phase non-recursive filters and phase equalized elliptic filters. >

Linear phase compressive filter

Abstract: A phase linear filter for soliton suppression is in the form of a laddered series of stages of non-commensurate low pass filters with each low pass filter having a series coupled inductance (L) and a reverse biased, voltage dependent varactor diode, to ground which acts as a variable capacitance (C). L and C values are set to levels which correspond to a linear or conventional phase linear filter. Inductance is mapped directly from that of an equivalent nonlinear transmission line and capacitance is mapped from the linear case using a large signal equivalent of a nonlinear transmission line.

Phase-compensated servo pattern and position error-sensing detector

Abstract: A demodulator for a servo control system receives a servo signal that is produced from reading a servo pattern that comprises a repeating sequence of trigonometric-compensated, phase-encoded magnetic flux transitions that extend continuously across the width of a servo track and includes a burst cosine phase detector that samples the amplitude of the servo signal at regular intervals and produces burst sample data, from which the demodulator produces a position error signal that is a substantially linear function of the servo head position relative to the servo track width. Because a plurality of amplitude samples is taken from each cycle of the servo signal, an increased signal-to-noise ratio can be obtained for the servo signal. The superior signal characteristics permit smaller and less complex demodulator elements to be used, thereby reducing the overall servo control system complexity. The burst cosine phase detector multiplies the servo signal samples with a predetermined set of reference coefficients to produce a linear phase difference that indicates the servo head radial position.

EMI filter and method

Abstract: An improved EMI filter design is provided for an inverter operated dynamoelectric machine. A lossy balun wound choke is connected between a power supply in the form of a bridge rectifier and the inverter. The choke acts as a source resistance. Using the choke as the source resistance, an equiripple approximation to linear phase filter is derived. The filter is placed on the input side of the bridge rectifier. A method of designing an EMI filter for use in conjunction with an inverter operated dynamoelectric machine.

Wide band, low noise, fine step tuning, phase locked loop frequency synthesizer

Abstract: A wide bandwidth, low noise, fine frequency step phase locked loop frequency synthesizer (10) and processing method. The synthesizer (10) includes a first divider (16) for dividing a first signal provided by a reference frequency source (12). A first phase locked loop (20) is provided for receiving a divided first signal from the first divider (16) and providing an intermediate frequency signal in response thereto. The first phase locked loop (20) includes a first phase detector (22) for comparing the phase of the first signal to the phase of a first feedback signal and providing a first phase difference signal in response thereto. The first phase locked loop (20) includes a first voltage controlled oscillator (30) for providing the intermediate signal in response to the first phase difference signal. The first phase locked loop (20) includes a first mixer (32) for mixing the intermediate signal with a second signal at a second frequency F2 and providing the first feedback signal in response thereto. A second divider (38) is included in the first phase locked loop (20) for dividing the first feedback signal prior to its application to the first phase detector (22). A third divider (39) is provided for shifting the frequency of the intermediate frequency signal, and a second phase locked loop (40) is coupled to the third divider for processing the frequency shifted intermediate frequency signal and providing an output frequency signal from the synthesizer. The second phase locked loop (40) includes a second phase detector (42) for comparing the phase of the shifted intermediate signal to the phase of a second feedback signal and providing a second phase difference signal in response thereto. A second voltage controlled oscillator (50) is included for providing the output frequency signal froth the synthesizer (10).

An improvement to the Powell and Chau linear phase IIR filters

Abstract: A technique using Jacobian elliptic functions is given that, by removing a previous method's double-zero constraint, yields improved designs of linear phase IIR filters. >

On fast FIR filters implemented as tail-canceling IIR filters

Abstract: We have developed an algorithm based on synthetic division for deriving the transfer function that cancels the tail of a given arbitrary rational (IIR) transfer function after a desired number of time steps. Our method applies to transfer functions with repeated poles, whereas previous methods of tail-subtraction cannot. We use a parallel state-variable technique with periodic refreshing to induce finite memory in order to prevent accumulation of quantization error in cases where the given transfer function has unstable modes. We present two methods for designing linear-phase truncated IIR (TIIR) filters based on antiphase filters. We explore finite-register effects for unstable modes and provide bounds on the maximum TIIR filter length. In particular, we show that for unstable systems, the available dynamic range of the registers must be three times that of the data. Considerable computational savings over conventional FIR filters are attainable for a given specification of linear-phase filter. We provide examples of filter design. We show how to generate finite-length polynomial impulse responses using TIIR filters. We list some applications of TIIR filters, including uses in digital audio and an algorithm for efficiently implementing Kay's optimal high-resolution frequency estimator.

Phase detector which eliminates frequency ripple

Abstract: A phase detector is disclosed that eliminates frequency ripple in a phase-locked loop circuit. The detector includes first and second circuits for providing UP and DOWN signals respectively. It also includes a delay element for setting the duration of the DOWN signal so as to eliminate phase jitter and static phase offset.

Generalized linear-phase lapped orthogonal transforms

Abstract: The general factorization of a linear-phase paraunitary filter bank (LPPUFB) is revisited and we introduce a class of lapped orthogonal transforms with extended overlap (GenLOT). In this formulation, the discrete cosine transform (DCT) is the order-1 GenLOT, the lapped orthogonal transform is the order-2 GenLOT, and so on, for any filter length which is an integer multiple of the block size. All GenLOTs are based on the DCT and have fast implementation algorithms. The degrees of freedom in the design of GenLOTs are described and design examples are presented along with some practical applications. >

Design of linear-phase quadrature mirror filters with powers-of-two coefficients

Abstract: This paper considers the design of linear-phase quadrature mirror filters (QMF's) with powers-of-two coefficients. A design method based on a recently developed weighted least-squares (WLS) algorithm is presented. First, we design QMF's with continuous coefficients using the WLS algorithm. This design process provides two favorable design results that the prototype analysis filter has quasi-equiripple stopband response and the resulting QMF bank shows quasi-equiripple reconstruction error behavior. To avoid the effect of nonuniformly distributed coefficient grid due to discretization of the filter coefficients, a procedure is then performed for obtaining an appropriate filter gain factor. Finally, utilizing the resulting error weighting function obtained by the WLS algorithm, we propose an efficient discrete optimization process to obtain a discrete solution in the minimax optimal sense. Design examples demonstrating the effectiveness of the proposed method are included. >