Showing papers on "Linear phase published in 1995"

A new class of two-channel biorthogonal filter banks and wavelet bases

Abstract: Proposes a novel framework for a new class of two-channel biorthogonal filter banks. The framework covers two useful subclasses: i) causal stable IIR filter banks. ii) linear phase FIR filter banks. There exists a very efficient structurally perfect reconstruction implementation for such a class. Filter banks of high frequency selectivity can be achieved by using the proposed framework with low complexity. The properties of such a class are discussed in detail. The design of the analysis/synthesis systems reduces to the design of a single transfer function. Very simple design methods are given both for FIR and IIR cases. Zeros of arbitrary multiplicity at aliasing frequency can be easily imposed, for the purpose of generating wavelets with regularity property. In the IIR case, two new classes of IIR maximally flat filters different from Butterworth filters are introduced. The filter coefficients are given in closed form. The wavelet bases corresponding to the biorthogonal systems are generated. the authors also provide a novel mapping of the proposed 1-D framework into 2-D. The mapping preserves the following: i) perfect reconstruction; ii) stability in the IIR case; iii) linear phase in the FIR case; iv) zeros at aliasing frequency; v) frequency characteristic of the filters. >

Phase shifting for nonsinusoidal waveforms with phase-shift errors

Abstract: In phase measurement systems that use phase shifting techniques, phase errors that are due to nonsinusoidal waveforms can be minimized by applying synchronous phase shifting algorithms with more than four samples. However, when the phase shift calibration is inaccurate, these algorithms cannot eliminate the effects of nonsinusoidal characteristics. It is shown that, when a number of samples beyond one period of a waveform such as a fringe pattern are taken, phase errors that are due to the harmonic components of the waveform can be eliminated, even when there exists a constant error in the phase shift interval. A general procedure for constructing phase shifting algorithms that eliminate these errors is derived. It is shown that 2j + 3 samples are necessary for the elimination of the effects of higher harmonic components up to the jth order. As examples, three algorithms are derived, in which the effects of harmonic components of low orders can be eliminated in the presence of a constant error in the phase shift interval.

Digital filter bank design quadratic-constrained formulation

Abstract: Formulate the filter bank design problem as an quadratic-constrained least-squares minimization problem. The solution of the minimization problem converges very quickly since the cost function as well as the constraints are quadratic functions with respect to the unknown parameters. The formulations of the perfect-reconstruction cosine-modulated filter bank, of the near-perfect-reconstruction pseudo-QMF bank, and of the two-channel biorthogonal linear-phase filter bank are derived using the proposed approach. Compared with other design methods, the proposed technique yields PR filter banks with much higher stopband attenuation. The proposed technique can also be extended to design multidimensional filter banks. >

Voltage controlled phase shifter with unlimited range

Abstract: A voltage-controlled phase shift apparatus having an unlimited range for producing an output signal that varies in phase from an input signal by a predetermined phase difference. The phase shift apparatus includes a first delay circuit coupled to receive the input signal, the first delay circuit for outputting a first intermediate signal that is α degrees out of phase with the input signal, a second intermediate signal that is β degrees out of phase with the first intermediate signal, a third intermediate signal that is 180 degrees out of phase with the first intermediate signal, and a fourth intermediate signal that is 180 degrees out of phase with the second intermediate signal. The phase shift apparatus also includes a phase interpolator circuit coupled to receive a control voltage signal and the first, second, third and fourth intermediate signals, the phase interpolator for phase mixing a selected pair of the first, second, third and fourth intermediate signals in response to the control voltage signal, the phase interpolator for outputting the output signal. A phase selector circuit coupled to the phase interpolator circuit and coupled to receive a phase slope signal and the control voltage signal selects the selected pair in response to the phase slope signal and the control voltage signal such that the output signal varies in phase from the input signal by the predetermined phase difference.

Linear phase cosine modulated maximally decimated filter banks with perfect reconstruction

Abstract: We propose a novel way to design maximally decimated FIR cosine modulated filter banks, in which each analysis and synthesis filter has a linear phase. The system can be designed to have either the approximate reconstruction property (pseudo-QMF system) or perfect reconstruction property (PR system). In the PR case, the system is a paraunitary filter bank. As in earlier work on cosine modulated systems, all the analysis filters come from an FIR prototype filter. However, unlike in any of the previous designs, all but two of the analysis filters have a total bandwidth of 2/spl pi//M rather than /spl pi//M (where 2M is the number of channels in our notation). A simple interpretation is possible in terms of the complex (hypothetical) analytic signal corresponding to each bandpass subband. The coding gain of the new system is comparable with that of a traditional M-channel system (rather than a 2M-channel system). This is primarily because there are typically two bandpass filters with the same passband support. Correspondingly, the cost of the system (in terms of complexity of implementation) is also comparable with that of an M-channel system. We also demonstrate that very good attenuation characteristics can be obtained with the new system.

Design of efficient M-band coders with linear-phase and perfect-reconstruction properties

Abstract: This paper presents a new design technique for obtaining M-band orthogonal coders where M=2/sup i/. The structures obtained using the proposed technique have the perfect reconstruction property. Furthermore, all filters that constitute the subband coder are linear-phase FIR-type filters. In contrast with conventional design techniques that attempt to find a unitary alias-component matrix in the frequency domain, we carry out the design in the time domain, based on time-domain orthonormality constraints that the filters must satisfy. The M-band design problem is reduced to the problem of finding a suitable lowpass filter h/sub 0/(n). Once a suitable lowpass filter is found, the remaining (M-1) filters of the coder are obtained through the use of shuffling operators on the lowpass filter. This approach leads to a set of filters that use the same numerical coefficient values in different shift positions, allowing very efficient numerical implementation of the subband coder. In addition, by imposing further constraints on the lowpass branch impulse response h/sub 0/(n), we are able to construct continuous bases of M-channel wavelets with good regularity properties. Design examples are presented for four-, eight-, and 16-band coders, along with examples of continuous wavelet bases that they generate. >

High frequency receiving apparatus

Abstract: A high frequency television signal receiving apparatus providing excellent linear detection of output characteristics by improving the phase characteristic of the picture synchronous detector. A variable capacitive element is equivalently connected in parallel to a reference solid-state oscillation element. The reference solid-state oscillation element controls the frequency of a local oscillation device including a PLL circuit for feeding a local oscillation signal to a mixer for converting a high frequency signal into an intermediate frequency signal. A first low pass filter is connected between a phase comparator for detecting a phase difference of the intermediate frequency signal and the output of a detection oscillator for generating a detection oscillation signal with a specific phase difference. A second low pass filter having a larger time constant than the first low pass filter is connected to the variable capacitive element. The capacitance of the variable capacitive element is varied by the output voltage of the phase comparator. The signal frequency of the local oscillation device is shifted, and is controlled to converge the frequency of the intermediate frequency signal within the frequency range of the detection oscillation signal.

Preservation of subband symmetry in multirate signal coding

Abstract: The article derives necessary and sufficient conditions for a multirate linear phase FIR filter bank, acting on a symmetrically extended finite-length input signal, to produce either symmetric or antisymmetric downsampled subbands. Such techniques have gained popularity for image coding applications, and we provide the solution to a key technical problem in terms of the input and filter symmetries and the downsampling ratio.

Steered frequency phase locked loop

Abstract: A Steered Frequency Phase Lock Loop (SFPLL) comprises a phase loop that functions like a normal phase locked loop (PLL) and locks to the input signal, and a frequency loop that uses a reference frequency to influence the phase loop and effectively confines the output frequency of the phase loop and the SFPLL to be in a range of frequencies close to the reference frequency. The reference frequency is chosen to be very close to the input signal frequency that it is desired the SFPLL lock to. The SFPLL comprises a phase detector (10), a frequency detector (22), first and second gain components (12, 24), first, second and third filter components (14, 18, 26), a summer (16) and a voltage controlled oscillator (VCO) (20). By a judicious choice of the gains in the phase and frequency loops the SFPLL can be designed so that the range of frequencies to which the SFPLL will lock can be confined to an arbitrarily small region around the reference frequency ( omega 'r). Applications of the SFPLL include demodulation in CW modulation systems and timing recovery from NRZ data. Three advantages of the SFPLL are that the output frequency is equal or close to the reference frequency when no input signal is present, and the range of frequencies to which the SFPLL can lock is confined to a region around the reference frequency, and the phase and frequency instabilities of the VCO can be reduced.

Audio signal coding with a signal adaptive filterbank

Abstract: An audio coding technique is presented, which utilizes a time varying non-uniform filterbank in conjunction with a Perceptual Audio Coder The non-uniform filterbank is formed from a plurality of uniform filter bank sections, and a transition filter Each uniform filterbank section covers a portion of a predetermined frequency axis, and the transition filter is utilized to cover the remainder of the predetermined frequency axis Aliasing terms introduced at the band edges of the transition filter are cancelled to obtain an overall transfer function for the non-uniform filterbank, which has linear phase and is approximately flat Use of the non-uniform filterbank permits an increase in temporal resolution, which results in an improved coding technique over a wide range of frequencies

Design of nonuniform division multirate FIR filter banks

Abstract: In this paper, a new design for nonuniform division multirate filter banks is presented. Signals are divided into a number of nonuniform frequency bands using analysis filters having individual arbitrary center frequencies. Each bandwidth ratio is given by an integer. After band splitting, the signals are decimated and interpolated irregularly by the integer or fractional factor. The values of factors are selected to reduce the occurrence of aliasing on each channel. A division scheme of the frequency range is considered and the derivation of a condition for the perfect reconstruction of the original signal is explained for designing filter banks. A design procedure for linear phase FIR filter banks as well as some design examples are shown to demonstrate the effectiveness of the method. >

Symmetric extension methods for M-channel linear-phase perfect-reconstruction filter banks

Abstract: The symmetric extension method has been shown to he an efficient way for subband processing of finite-length sequences. This paper presents an extension of this method to general linear-phase perfect-reconstruction filter banks. We derive constraints on the length and symmetry polarity of the permissible filter banks and propose a new design algorithm. In the algorithm, different symmetric sequences are formulated in a unified form based on the circular-symmetry framework. The length constraints in symmetrically extending the input sequence and windowing the subband sequences are investigated. The effect of shifting the input sequence is included. When the algorithm is applied to equal-length filter banks, we explicitly show that symmetric extension methods can always be constructed to replace the circular convolution approach.

General analysis of two-band QMF banks

Abstract: The two-channel perfect-reconstruction quadrature-mirror-filter banks (PR QMF banks) are analyzed in detail by assuming arbitrary analysis and synthesis filters. Solutions where the filters are FIR or IIR correspond to the fact that a certain function is monomial or nonmonomial, respectively. For the monomial case, the design problem is formulated as a nonlinear constrained optimization problem. The formulation is quite robust and is able to design various two-channel filter banks such as orthogonal and biorthogonal, arbitrary delay, linear-phase filter banks, to name a few. Same formulation is used for causal and stable PR IIR filter bank solutions. >

Automatic frequency control method and apparatus therefor

Abstract: An automatic frequency control apparatus used in an MPSK communication system detects a frequency offset between a carrier and a local oscillation signal for adjustment of a local oscillation frequency. A phase difference detector generates a first phase difference detection signal having, as a phase value, a difference between the phases of various samples of the sampled signal. A phase altering unit generates a second phase difference detection signal having a phase value different from that of the first phase difference detection signal. A frequency offset signal generator estimates transmission phase information by using the phase value of the second phase difference detection signal and reference phase signals used for MPSK modulation, thereby generating a frequency offset signal which is determined by the transmission phase signal and the second phase difference detection signal. The result is that the number of the reference phases which are used for determination of the transmission phase information by altering the phase of the phase difference signal, is reduced. Accordingly, the hardware cost for implementing the apparatus can be lowered. The invention can be used for automatic frequency control in a modem which is used for all the types of MPSK modulation.

A frequency response function for linear, time-varying systems

Abstract: It is well known that the steady-state response of a linear, time-invariant, finite-dimensional, exponentially stable system to a periodic input signal results, after a phase shift, in a periodic output signal of the same period with amplitude equal to the rescaling of the input amplitude by the modulus of the value of the transfer function at the given frequency. Moreover, the phase shift of the output signal is equal to the phase of the value of the transfer function at the given frequency. For this reason the transfer function is also referred to as the “frequency response function.” We present an analogue of this idea for linear, finite-dimensional, time-varying systems, in both the continuous- and discrete-time settings. The problem of constructing a time-varying system which associates a given output signal to each complex exponential input signal in a prescribed set can be posed as a modeling question. This leads to a new modeling interpretation for some of the time-varying interpolation problems which have recently been studied in the literature and a new motivation for the study of point evaluation for triangular operators recently introduced by Alpay, Dewilde, and Dym and by the authors for the continuous-time case.

Complex, linear-phase filters for efficient image coding

Abstract: With the exception of the Haar basis, real-valued orthogonal wavelet filter banks with compact support lack symmetry and therefore do not possess linear phase This has led to the use of biorthogonal filters for coding of images and other multidimensional data There are, however, complex solutions permitting the construction of compactly supported, orthogonal linear phase QMF filter banks By explicitly seeking solutions in which the imaginary part of the filter coefficients is small enough to be approximated to zero, real symmetric filters can be obtained that achieve excellent compression performance >

Automatic frequency control method and circuit for use in a delay detection type demodulator

Abstract: A delay detection is performed to baseband signals subjected to an orthogonal demodulation to obtain a delay detection output and a frequency error is found from the delay detection output. The frequency error is converted into a control frequency which is accumulated to generate a phase rotation φ which is used to rotate the input baseband signals before being subjected to the delay detection, thereby to perform frequency correction. By providing filters downstream of a phase rotation circuit and upstream of a delay detection circuit, automatic frequency control can be realized without performance deterioration.

Two-channel IIR QMF banks with approximately linear-phase analysis and synthesis filters

Abstract: Perfect linear-phase two-channel QMF banks require the use of finite impulse response (FIR) analysis and synthesis filters. Although they are less expensive and yield superior stopband characteristics, perfect linear phase cannot be achieved with stable infinite impulse response (IIR) filters. Thus, IIR designs usually incorporate a postprocessing equalizer that is optimized to reduce the phase distortion of the entire filter bank. However, the analysis and synthesis filters of such an IIR filter bank are not linear phase. In this paper, a computationally simple method to obtain IIR analysis and synthesis filters that possess negligible phase distortion is presented. The method is based on first applying the balanced reduction procedure to obtain nearly allpass IIR polyphase components and then approximating these with perfect allpass IIR polyphase components. The resulting IIR designs already have only negligible phase distortion. However, if required, further improvement may be achieved through optimization of the filter parameters. For this purpose, a suitable objective function is presented. Bounds for the magnitude and phase errors of the designs are also derived. Design examples indicate that the derived IIR filter banks are more efficient in terms of computational complexity than the FIR prototypes and perfect reconstruction FIR filter banks. Although the PR FIR filter banks when implemented with the one-multiplier lattice structure and IIR filter banks are comparable in terms of computational complexity, the former is very sensitive to coefficient quantization effects. >

Design of 2-D multiplierless FIR filters using genetic algorithms

Abstract: This paper considers the design of reduced complexity two-dimensional FIR filters using genetic algorithms (GAs). Circularly symmetric and diamond shaped low-pass linear phase FIR filters are designed using coefficients comprising the sum or difference of two power-of-two terms. A minimax error criterion is adopted which leads to a minimisation of the weighted ripple extrema in both pass and stop bands. The results presented are compared with those obtained using simulated annealing, linear programming and simple rounding of an optimum (continuous) minimax solution. The filters designed using GAs are found to be as good or better than those designed using other methods, with less computational effort than the simulated annealing approach.

Method and apparatus for detecting current delay angle from motor terminal voltage

Abstract: A method and apparatus to be used with an inverter based motor controller for determining current phase angles for the current on a single phase of a three phase AC motor from an inverter generated high frequency phase voltage sequence. The invention detects turn on delay periods and compares phase voltages during consecutive turn on delay periods to determine phase current zero crossing times which can be used along with phase voltage zero crossing times to derive current phase angle information.

Programmable data acquisition system with a microprocessor for correcting magnitude and phase of quantized signals while providing a substantially linear phase response

Abstract: A programmable data acquisition system including a plurality of input signal channels for receiving a respective input signal during a normal mode of operation is provided. Individual test circuits are used for selecting respective ones of the plurality of channels to receive predetermined reference signals during a test mode of operation while uninterruptedly providing the normal mode of operation in any remaining unselected channels in the data acquisition system. An analog-to-digital (A/D) converter system allows for supplying quantized electrical signals at a predetermined rate. The A/D converter is responsive to any signals carried in the plurality of signal channels as selected by the individual test circuits. A control unit allows for supplying respective control signals to the test circuits and to the converter system. The data acquisition system is coupled to an external microprocessor having a magnitude and a phase corrector through a microprocessor interface that transfers a microprocessor-derived control word from the microprocessor to the control unit. The microprocessor interface further transfers the stream of quantized electrical signals supplied by the converter system to the microprocessor. The phase corrector provides a predetermined phase angle correction over a predetermined passband to the quantized signals transferred to the microprocessor from the converter system while the magnitude corrector provides a predetermined magnitude correction over the predetermined passband.

An efficient weighted least-squares design of linear-phase nonrecursive filters

Abstract: An accelerated procedure for the design of linear-phase nonrecursive filters using a weighted least-squares technique is described. This procedure is based on formulating the error reflecting the difference between the desired amplitude response and the amplitude response of the practical filter in a quadratic form. The coefficients of the filter are obtained by solving a system of linear equations involving a Toeplitz-plus-Hankel matrix. Such a system of linear equations can be solved by computationally efficient algorithms having only O(N/sup 2/) complexity. By choosing the appropriate frequency-dependent weighting function, a filter with either a least-squares or an equiripple error variation can be designed. >

Improved filter sharpening

Abstract: We propose a natural extension to Kaiser-Hamming (1977) filter sharpening methods to allow for a piecewise linear desired amplitude change function (ACF). The primary advantages of the proposed ACF over piecewise constant ACFs is that we obtain better control of selective improvement (or degradation) in either the passband or stopband or both, and we are not restricted to applying our methods to filters with piecewise constant pass and stopbands, since linear segments of slope 1 can be used to retain existing filter performance in either passband or stopband. The proposed ACF approximating polynomial (AP) is easy to compute, may be constrained to have simple (or integer) coefficients, and may be expressed as the AP of Kaiser and Hamming plus a correction polynomial. We also provide applications for motivation.

Phase locked loop (PLL) circuit having variable loop filter for shortened locking time

Abstract: Disclosed herein is A-phase-locked-loop PLL circuit including a voltage controlled oscillator (VCD) controlled in oscillation frequency by a control voltage, a divider dividing in frequency an oscillation signal of the VCO by a frequency division ratio to produce a frequency-divided signal, a phase comparator producing a phase comparison signal indicative of a difference in phase between a reference signal and the frequency-divided signal, and a filter converting the phase comparison signal into the control voltage to be used for controlling the VCO. The oscillation frequency of the VCO is thereby changed from a current frequency by variation of the frequency-division ratio and locked to a new frequency after a locking period of time elapses. There is further provided a control circuit which changes a time constant of the filter circuit a plurality of times during the locking period of time, this control circuit operating in response to the output of the phase comparator.

Minimum-phase filter design from linear-phase startpoint via balanced model truncation

Abstract: A new practical design approach for minimum-phase FIR or IIR filters, setting out from a high dimensionality FIR linear-phase prototype is described. The novelty of this technique lies in overcoming the inherent problem of finding the roots of a high order polynomial with repeated and/or very closely clustered roots.

Phase-locked loop circuit obtaining the phase difference between time series pulse strings and a reference frequency

Abstract: A phase-locked loop circuit includes a voltage-controlled oscillator, a pre-scaler, a main counter, a shift register, and a phase comparison section. The oscillation frequency of the voltage-controlled oscillator is controlled on the basis of phase different information. The pre-scaler frequency-divides an oscillation frequency output from the voltage-controlled oscillator by one of frequency division ratios of 1/j (j is a positive integer) and 1/(j+1) which is selected in accordance with an external control signal. The main counter frequency-divides a frequency division output from the pre-scaler by a frequency division ratio of n (n is a positive integer). The shift register generates α (α is an integer equal to or larger than two) time series pulse strings which are synchronized with the output from the pre-scaler and have phases sequentially delayed by one period on the basis of a frequency division output from the main counter. The phase comparison section detects the phase differences between the α time series pulse strings from the shift register and a predetermined reference frequency signal, adds/synthesizes the detected phase differences, and outputs the resultant information as phase difference information to the voltage-controlled oscillator.

High frequency fundamental resonators and filters fabricated by batch process using chemical etching

Abstract: The small sized 1st intermediate frequency (IF) filters at center frequency range of 70 MHz to 150 MHz and passband widths of /spl plusmn/5 to /spl plusmn/100 kHz with sharp selectivity are required in mobile communication systems such as mobile and portable cellular phone. Our solution to employ fundamental mode monolithic crystal filter (MCF) assembled in surface mountable package. We describe here in detail, the design approach including batch process etching technology. Through photolithography, 56 patterns are chemically etched on one wafer (25 mm/spl times/20 mm). Then, a similar etching process automatically adjusts the wafer thickness in accordance with frequency. For the MCF, the frequency of split electrodes and the degree of coupling between them are automatically adjusted by an accurately positioned mask evaporation process controlled by a computer. Further, we describe suppression of the spurious response, technology for realizing wide bandwidth and high stopband attenuation characteristics. By our above developed technology, we achieved 90 MHz miniaturized IF filter with 1/15 volume reduction of conventional 3rd overtone mode MCF, still possessing the same characteristic of conventional one. Also, we achieved 130 MHz of middle band with suppression of spurious response in wide stopband frequency range, and 71 MHz linear phase wide /spl plusmn/88 kHz band with the group delay distortion 0.9 /spl mu/s over f/sub 0//spl plusmn/80 kHz.

A design of linear-phased IIR Nyquist filters

Abstract: This paper discusses a new method of designing linear-phased IIR Nyquist filters with zero intersymbol interference. The filters designed by this method possess linear-phase characteristics and are lower in order than other Nyquist filters designed by existing methods. Expressions are derived for zero-phased IIR Nyquist filters and efficient design methods are examined for them. The opted design method is based on an iteration process, and in each iteration step a modified version of the Remez exchange algorithm is used. In addition, the implementation of the designed zero-phased IIR filters is considered. Finally, the proposed design method is demonstrated through various design examples. >

Novel Bragg grating distributed-strain sensor based on phase measurements

Abstract: We recently disclosed that the shape of the intensity spectrum reflected from a Bragg grating can serve as an indicator of strain gradient across this grating. However, in principle, to map the strain field distribution, both the intensity spectrum (Iota) ((lambda) ) and the phase spectrum (Psi) ((lambda) ) are needed. In this paper, we propose a method based on phase measurements. It relies on the correlation between the wavelength-dependent penetration depth of light in a nonuniform (chirped) grating and the phase delay of the reflected wave. If we know that the nonuniform strain field to be measured is a monotonic function of position along the grating (which is the most practical case), then the measured grating phase-spectrum (Psi) ((lambda) ) can be used to evaluate the penetration depth and consequently the distribution of the grating optical pitch-lengths, i.e. the strain distribution over the grating. Theory reveals that the phase slope spectrum d(Psi) ((lambda) )/d(lambda) induced by a grating can directly yield the strain field profile. For example, a linear phase slope spectrum indicates a linear chirp in the grating due to a constant strain gradient; and the strain gradient can be easily obtained from the dispersion constant d2(Psi) ((lambda) )/d(lambda) 2. The validity of this method is also discussed in this paper.© (1995) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Technique for stabilizing the phase of the reference signals in analog fiber-optic links.

Abstract: The effects of temperature and longitudinal stress on the phase delay of reference signals in a fiber-optic link are discussed. A feedback system that uses a fiber-optic phase modulator is used to compensate for the phase fluctuations of a reference signal in the link. The phase deviations of a 50-MHz reference frequency that are caused by temperature variations of the link is reduced by more than 95% on optimization of the correction system. The advantages of this technique are that the fiber-optic phase modulator has a greater stability compared with the electronic phase modulators, and signal conversions from electric to optic and optic to electric are avoided.