Showing papers on "Low-pass filter published in 1981"

Real-Time Adaptive Contrast Enhancement

Abstract: A recursive filter approach is introduced to simplify real-time implementation of an adaptive contrast enhancement scheme for imaging sensors. With this scheme, even scenes possessing large global dynamic ranges (>40 dB) can be accommodated by the limited dynamic range (20 dB) of a display without losing the local contrast essential for image interpretation. This paper describes the recursive filter implementation of the local area contrast enhancement scheme using charge-coupled devices and the resultant real-time hardware capable of processing standard 525 and 875 line TV compatible video (from vidicons, videotape recorders, etc). Several examples from video imagery are included to demonstrate its effectiveness.

Sampling frequency converter

Abstract: A sampling frequency converter for converting a first signal sampled at a first sampling frequency f1 into a second signal sampled at a second sampling frequency f2 comprising an interpolation device supplied with the first signal, for inserting L-1 zeros (L is an integer) for every sampling time, a filter circuit for attenuating a frequency component over a frequency f/2 (f is a frequency) within an output signal of said interpolation device, where the filter circuit has a series circuit consisting of a finite impulse response digital filter and an infinite impulse response digital filter, and the frequency f is equal to the first sampling frequency f1 when f1 f2, and a decimation device for extracting every M-th (M is an integer) output signal of the filter circuit, to produce said second signal.

Optimum Input and Output Filters for a Single-Phase Rectifier Power Supply

Abstract: The ``optimum'' output filter inductance Lf and the input filter capacitor Ci for a single-phase uncontrolled bridge rectifier employed for low power de-to-dc converters or inverters is established. The filter Ci is optimized to obtain maximum input power factor, minimum filter inductance, and minimum output dc voltage regulation. A design example is provided and theoretical results have been verified on an experimental model.

Properties of Digital Smoothing Polynomial (DISPO) Filters

Abstract: Digital filters for spectrometric applications are compared with the classical RC filter. Properties discussed include noise reduction, line shift, and conservation of line moments. For Gaussian and Lorentzian lines, signal deformation and change of half-width as a function of time constant and line width are calculated for several filter types. Using accuracy, sensitivity, and scan speed as criteria, it is shown that a fourth-order digital smoothing polynomial (DISPO) filter is better by typically 1 or even 2 orders of magnitude than the RC filter. Since a real time implementation of these filters is possible, they can directly replace RC filters in all spectrometric applications.

PLL Detection circuit

Abstract: A time constant resistor in a phase lock loop filter is short-circuited by a plurality of diodes which are activated in response to a detection output indicating that the filter output is within a desired locking range.

The time-sequenced adaptive filter

Abstract: A new form of adaptive filter is proposed which is especially suited for the estimation of a class of nonstationary signals. This new filter, called the time-sequenced adaptive filter, is an extension of the least mean-square error (LMS) adaptive filter. Both the LMS and time-sequenced adaptive filters are digital filters composed of a tapped delay line and adjustable weights, whose impulse response is controlled by an adaptive algorithm. For stationary stochastic inputs the mean-square error, which is the expected value of the squared difference between the filter output and an externally supplied "desired response," is a quadratic function of the weights-a paraboloid with a single fixed minimum point which can be sought by gradient techniques, such as the LMS algorithm. For nonstationary inputs however the minimum point, curvature, and orientation of the error surface could be changing over time. The time-sequenced adaptive filter is applicable to the estimation of that subset of nonstationary signals having a recurring (but not necessarily periodic) statistical character, e.g., recurring pulses in noise. In this case there are a finite number of different paraboloidal error surfaces, also recurring in time. The time-sequenced adaptive filter uses multiple sets of adjustable weights. At each point in time, one and only one set of weights is selected to form the filter output and to be adapted using the LMS algorithm. The index of the set of weights chosen is synchronized with the recurring statistical character of the filter input so that each set of weights is associated with a single error surface. After many adaptations of each set of weights, the minimum point of each error surface is reached resulting in an optimal time-varying filter. For this procedure, some a priori knowledge of the filter input is required to synchronize the selection of the set of weights with the recurring statistics of the filter input. For pulse-type signals, this a priori knowledge could be the location of the pulses in time; for signals with periodic statistics, knowledge of the period is sufficient. Possible applications of the time-sequenced adaptive filter include electrocardiogram enhancement and electric load prediction.

Digital adaptive finite impulse response filter with large number of coefficients

Abstract: A digital adaptive finite impulse response (AFIR) filter having a large number of coefficients is composed of two or more separate filter units. The first filter unit computes response samples h₀ - h n+1 in response to input signal samples to provide a partial estimated response during each sampling period. After the signal samples are fully processed by the first unit, they are transferred to a second filter unit which produces response samples h n+2 - h p+1 in response thereto to provide a second partial response. The sum of the two partial responses is computed to provide the total estimated system response. The two independent filter units thus act simultaneously to provide twice as many coeffi­ cients as prior art AFIR filters in the same amount of time.

Diplexer for television tuning systems

Abstract: A diplexer comprises a first frequency selective filter coupled between its first input terminal and a circuit point, a second frequency selective filter coupled between the circuit point and its output terminal, and third frequency selective filter coupled between its second input terminal and the circuit point. The first filter selects frequencies higher than a first frequency and the second filter selects frequencies lower than a second frequency which is higher than the first frequency. The third filter selects frequencies lower than a third frequency which is lower than the first frequency. In a television receiver, the first and second filters serve as a bandpass filter for the UHF band, and the third filter passes the VHF and CATV bands.

Noise reduction system having series connected low and high frequency emphasis and de-emphasis filters

Abstract: In a noise reduction system including a compressor and an expander having a complementary frequency response characteristic to the frequency response characteristic of the compressor, the compressor includes a high frequency emphasis variable frequency filter and a low frequency emphasis variable gain filter which are connected in a series circuit between input and output terminals and which respectively vary their parameters in response to the respective high and low frequency components of the compressor output. The expander includes a high frequency de-emphasis variable frequency filter and a low frequency de-emphasis variable gain filter which are connected in a series circuit between input and output terminals and which respectively vary their parameters in response to the respective high and low frequency components of the expander input.

MFIR filters: Properties and applications

Abstract: The class of multiplicative FIR (MFIR) filters is considered and some properties and applications of MFIR filters are investigated. MFIR filter approximation of a given IIR filter is shown to reduce the number of multiplications and additions logarithmically, in comparison to the corresponding FIR filter in direct form. The pure multiplicity property is introduced and is shown to apply to a class of MFIR filters. This property results in a criterion for optimal ordering and expressions for roundoff noise when no scaling is used and also results in the invariance of roundoff noise output under l 2 -scaling. Linear phase MFIR filter realization of a desired low-pass frequency response magnitude |H_{d}| with centered transition band is shown to require 0(n \log_{2} N) each multipliers and adders. N is the order of the min-max FIR filter design of |H_{d}| and n is the order of the elliptic IIR filter design of |H_{d}|^{1/2} . Several design examples of linear phase low-pass filters are used to compare MFIR filter designs versus those of min-max FIR filters in direct form. Comb filters of order N are shown to have an exact MFIR realization that requires fewer than 2 \log_{2} N additions. Suggestions for further research and applications conclude the paper.

A note on the use of windows for two-dimensional FIR filter design

Abstract: Using a one-dimensional window as a prototype, a two-dimensional window may be formulated having either a square region of support or a circular one. In this paper we compare the effects of using each of these window formulations for 2-D FIR filter design and present formulas for estimating filter order in terms of design specifications, using a Kaiser window as a prototype.

Arrangement for the attitude stabilization of flexible vehicles with weakly-dampened structural vibrations and discontinuous control intervention

Abstract: An arrangement for the attitude stabilization of flexible vehicles, such asircraft and spacecraft. Such vehicles, due to their lightweight construction and/or large spatial extension or high degree of slenderness, have structurally weakly-dampened bending vibrations and/or torsional vibrations. For generating the forces and moments required for stabilizing such vehicle, discontinuously operating units are used. In one embodiment, for each vehicle axis, an observer or Kalman filter is employed for obtaining estimated values of the state variables of the vehicle system to be controlled. The observer has a first transfer function of at most third order. A state controller device is responsive to the observer for controlling the state variables. A modulator network is responsive to the state controller device wherein the modulator network has a relay characteristic. In this embodiment, a feedback path is arranged from the output of the modulator network to the observer which eliminates the need for a special modulator network having a second transfer function and also eliminates internal modulator feedback. In another form of the invention, the feedback is arranged from the input of the modulator to the observer and a low pass filter is included between the state controller device and the modulator network. The filter has either at least two low pass filter sections of first order or a single low pass filter section of at least second order. The break-point frequencies of the filter section or sections lies between the natural frequency of the rigid body and the lower structural resident frequency of the respective vehicle axis. Equivalent circuits are disclosed for replacing the observer and state controller device as described above. Adaptive techniques as well as digital realizations of the various embodiments are also disclosed.

Image gradient detectors operating in a partitioned low-pass channel

Abstract: An electronic image processing system, for image enhancement and noise suppression, from signals representing an array of picture elements, or pels. The system is of the kind providing a feature-enhanced output by the addition of outputs of a high-pass filter acting as image-feature detector and a complementary low-pass filter. The low-pass filter also acts as an image-feature detector and includes a prefilter (130 and FIG. 22) and a sub-sampling filter (32) based on a set of weighting patterns in the form of sparse matrices (FIG. 23). The sub-sampling filter in a bandpass channel of the low-pass filter (206 and FIG. 28) comprises a pair of gradient detectors (210, 220, 230, 240 and FIGS. 32, 33, 34, 35) arranged back to back.

Signal processing for digital optical disc players

Abstract: The dc coupled signal processor of the invention provides a method for accurately converting an analog input signal from a digital optical disc which signal has a finite rise time to a digital signal with accurately positioned edges by preserving the dc content of the input signal. A peak-to-peak detector detects the average of the peak-to-peak analog signals and feeds it to a low pass filter which operate above the periodic drift-like analog input signal. The output of this filter is employed as a reference voltage by a comparator which is used as a slicer or zero crossing detector. The comparator output is a digital signal with minimum bias distortion. The output of the filter is also fed to a differential amplifier and a second low pass filter which operates at a frequency below the cutoff of the periodic drift analog input signal. The resulting signal is negatively fed back to the input to be summed and amplified with the input analog signal.

A universal, digital sampling frequency converter for digital audio

Abstract: Digital Audio requires high-quality signal conversion between a variety of sampling frequencies, often in non-trivial integer ratios. In such applications, conventional methods based on analog processing or classical FIR filter rate-changing are not adequate. A novel approach to digital sampling frequency conversion based on a single, multistage filter and adapted to conversion between arbitrary, a priori unknown sampling frequency ratios is presented. Its design, implementation and control are discussed in some detail.

Side lock avoidance network for PSK demodulator

Abstract: A sidelock avoidance scheme for preventing sidelock in a PSK demodulator's carrier recovery loop contains augmenting sweep control circuitry, including a frequency discriminator and an associated window comparator. The output of the frequency discriminator, which is low pass filtered to remove noise, is applied to the window comparator which compares any differential between the true carrier and the output of a carrier recovery loop to a preset reference threshold representative of a frequency error condition that may approach sidelock. When the output of the frequency discriminator is greater that this preset reference threshold, an augmented frequency control voltage is applied to the voltage control oscillator of the loop to drive the oscillator away from a possible sidelock condition and toward the true carrier. The augmented frequency control voltage may be derived from a frequency sweep generator or from the output of the frequency discriminator, depending upon a selected strapping option.

Simplified MOS switched capacitor ladder filter structures

Abstract: A general technique for time-sharing amplifiers to reduce die area in switched capacitor ladder filters is described and illustrated with a fifth-order elliptic low-pass ladder filter requiring only three operational amplifiers. Techniques for synthesizing filters with maximum passband accuracy in the presence of parasitic capacitances are presented, and verified with two versions of the same fifth-order design integrated in a standard NMOS process. Passband accuracies of better than 0.1 dB have been achieved using only 0.3 pF unit-sized capacitors. The dynamic range is 75 dB.

Sampling filter for reducing aliasing distortion

Abstract: Sampling filter which employs fractional period displacement of samples to force additional nulls in the filter characteristic response. This sampling method and cascading antialiasing filters (501, 502, 503) provides a filter structure which minimizes the complexity of the circuit yet which provides great flexibility to implement a desired filter characteristic.

Simultaneous signal detection for IFM receivers by detecting intermodulation products

Abstract: An apparatus for use in conjunction with an instantaneous frequency measurement (IFM) receiver, for detecting the presence of two or more RF pulse signals, differing in frequency, between the onset of the first RF signal pulse and the completion of the frequency encode strobe. A low pass filter branching off from the output of the limiting amplifier of the IFM receiver will allow passage of only the low frequency intermodulation products formed from the presence of simultaneous signals. These intermodulation products are then detected and sampled during the time period of interest with a flag set if simultaneous pulses are present. This flag, which indicates to the receiver that the data associated with the received pulse may contain erroneous information, is maintained until reset by the onset of a succeeding RF pulse.

Electronic circuit using digital techniques for vortex shedding flowmeter signal processing

Abstract: A circuit for processing the vortex shedding frequency signal of a vortex shedding flowmeter comprises a phase detector, switch and low pass filter connected in series. The phase detector receives the vortex shedding frequency and applies it over the normally closed switch to the low pass filter which produces an analog signal corresponding to the frequency of the vortex shedding frequency signal. A voltage controlled oscillator is connected between an output of the low pass filter and an input of the phase detector for tracking the vortex shedding frequency and producing a tracking frequency signal which is maintained when the vortex shedding frequency signal disappears. A range or gain code circuit portion is connected for establishing a set time period during which the frequency signal is accumulated in a counter. The accumulated signal from the counter is utilized as a digital signal corresponding to the vortex shedding frequency signal.

Switchable filter circuits

Abstract: A switchable filter circuit for use in a receiver to select a first or a second frequency band from a received signal and for providing predetermined minimum stop band attenuation at frequencies outside the selected frequency band, the second frequency band lying wholly within the first. A first filter element is provided which defines a primary pass band corresponding to the first frequency band, and a second filter element is provided which defines a secondary pass band corresponding to the second frequency band, the first filter element having an input to which the received signal is applied, and the second filter element have an input for receiving the output of the first filter element. The stop band attenuation of the first filter element is at least as great as the predetermined minimum stop band attenuation required, and the stop band attenuation of the secondary filter element within the primary pass band being that which is required by the application. Thus the first filter element significantly determines the stop band characteristics of the filter circuit as a whole outside the primary pass band.

Controlled antenna tuner

Abstract: A controlled antenna tuner comprises a power amplifier and a microprocessor, an attenuator connected to the power amplifier for selectively controlling the power output, a low pass filter connected to the power amplifier and microprocessor for channel frequency selection, a vertical standing wave ratio (VSWR) circuit inductively connected to the low pass filters output for determining the VSWR of the antenna, an impedance (magnitude) bridge and a phase detector bridge selectively connected to the low pass filter RF output and to the microprocessor for sensing the tuning satus of an antenna, and an LC tuner network connected to the phase detector bridge and microprocessor, the microprocessor processing outputs of the impedance and phase detector bridges into tuning signals for the LC tuner to tune the antenna to the selected channel and storing the tuning signals forthe selected channel forfuture use.

A synthesis of frequency domain filters for time delay estimation

Abstract: A simple model for two-channel delay estimation filtering is presented. The problem is subdivided into three classes based on initial assumptions. General filters described in the frequency domain are presented as solutions to these specific classes. It is shown that many of these filters, which include the "Wiener" least-squares estimation filter and classical, matched detection filter, can be derived as specific cases of a very general ideal filter form. We call this general ideal filter the weighted distortion balance filter. Relationships between a standard set of ideal filters and some filters previously proposed in the literature for delay estimation are discussed. An illustrative example is presented to compare the delay estimated from the use of various filters.

Linear phase filter with self-equalized group delay

Abstract: In modern communication systems, it has become important to provide filters, and in particular bandpass filters, which can provide substantially uniform group delay across the bandwidth of the filter while still achieving good amplitude response. In this regard, it is particularly desirable to substantially reduce the large variations in the phase characteristics found in conventional bandpass filters at the nominal band edge of the filter. To accomplish this, a filter is provided having at least one pair of lattice arms coupled in parallel to one another between the input and output of the filter. Each of the lattice arms includes a plurality of resonant LC resonators, each of the resonators having a different resonant frequency than the center frequency of the filter. In particular, within the bandwidth of the filter, the exponential damping coefficients for the resonators in each arm are set to decay at the same rate. This desired decay can be accomplished by exponential sizing of the components.

A prototype switched-capacitor voltage-wave filter realized in NMOS technology

Abstract: Presents a novel approach to the realization of monolithic filters. The method is based on using sampled analog signals and is related to the wave digital filter in its design techniques. The eventual monolithic realization in NMOS technology is in the form of a switched-capacitor structure. The design is exact and there is no requirement for a high relative clock frequency. Only unity-gain buffers are required, as opposed to high-gain differential-input operational amplifiers, and so the technique is well suited to CMOS technology. Performance is determined by capacitance ratios and the design is optimally insensitive to parameter variations. Capacitance ratios are moderate relative to those encountered in existing switched-capacitor filters. Results are presented for a prototype integrated circuit design containing fifth- and seventh-order low-pass Chebyshev filters with a designed cutoff at one-eighth clock frequency. The responses achieved for the prototype design show excellent agreement with the theory.

Digital signal separation filters

Abstract: A digital finite impulse response (FIR) filter is provided in which a plurality of weighted signal taps are symmetrically located in time about a weighted center tap. Weighted signals from the symmetrically located taps are summed at a first point in the filter, which sum is then combined with signals from the center tap in one sense, that is, either additively or subtractively, to produce signals at a first output. The summed signals at the first point are also combined with signals from the center tap in an opposite sense to produce signals at a second output. The two outputs will exhibit bandpass and lowpass filter response characteristics, with the outputs at which the respective responses are produced being determined by the respective senses of signal combination. The FIR filter may be of either the input tap-weighted or output tap-weighted variety.

Frequency domain analysis of wraparound error in fast convolution algorithms

Abstract: Fast algorithms exist for computing cyclic convolutions. To obtain the linear convolution required for an FIR filter, the data records must be overlapped by at least L - 1 points, where L is the length of the filter impulse response. If the overlap is too small, wraparound error occurs. This error transforms a linear time-invariant filter into a periodic time-varying filter, whose output is periodically nonstationary for a wide-sense stationary input. The first part of this paper contains a review of the frequency domain theory of periodic filters and processes, in the second part of the paper the theory is applied to the specific periodic filter that results from wraparound error in fast convolution algorithms.