Showing papers on "Low-pass filter published in 1977"

Sharpening the response of a symmetric nonrecursive filter by multiple use of the same filter

Abstract: When processing data by filters, we often find it necessary to improve the performance of the filter, either by increasing the out-of-band rejection (loss) or by decreasing the error in the passband, or both. A first approach is to process the data by repeated passes through the same filter. Each pass, while increasing the out-of-band loss, also increases the passband error, often to an undesirable level. It also increases the length (order) of the equivalent filter. How can we do a better job of filtering by suitably combining the results of several passes through the same filter? By "better" we mean both less passband error and greater out-of-band, or stopband, loss. This process is called filter sharpening. A simple, powerful method for doing this is described in detail, and its computational efficiency is compared to the best possible filter designs meeting the same specifications. The design method, based on the idea of the amplitude change function, is restricted to symmetric nonrecursive (finite impulse response) filters with piecewise constant pass- and stopbands. Several illustrative examples are given.

Data smoothing using low‐pass digital filters

Abstract: With the increasing use of computer‐controlled data acquisition systems which record data in digital form, there has developed a need for techniques which perform a general smoothing process on digitized experimental data. This processing enables the experimentalist to eliminate or greatly reduce the amount of high‐frequency noise in order to obtain as accurate and clean representation of the true phenomenon as is consistent with his measurement accuracies. This filtering or smoothing process should be as simple and efficient (least amount of arithmetic per data sample) as is consistent with the experimental situation. The basic concepts of low‐pass filters are discussed and four different low‐pass filter design procedures are described, each with its own particular smoothing properties. These design procedures give directly the coefficients of a symmetrical weighting sequence having the desired passband width and the desired high‐frequency noise rejection. The uses of the filters are illustrated with examples and the fortran code for implementing each of the design procedures is given in an Appendix.

Sharpening the response of a symmetric nonrecursive filter by multiple use of the same filter

Abstract: When processing data by filters we often find it necessary to improve the performance of the filter, either by increasing the out-of-band rejection (loss) or by decreasing the error in the passband, or both. A first approach is to process the data by repeated passes through the same filter. Each pass, while increasing the out-of-band loss, also increases the passband error, often to an undesirable level. It also increases the length (order) of the equivalent filter. How can we do a better job of filtering by suitably combining the results of several passes through the same filter? By "better" we mean both less passband error and greater out-of-band, or stopband, loss. This process is called filter sharpening. A simple, powerful method for doing this is described in detail, and its computational efficiency is compared to the best possible filter designs meeting the same specifications. The design method, based on the idea of the amplitude change function, is restricted to symmetric nonrecursive (finite impulse response) filters with piecewise constant pass and stopbands. Several illustrative examples are given.

Method of and apparatus for measurement of blood flow using coherent light

Abstract: An apparatus for measuring the flow parameter of blood flowing in an organ includes a laser and associated optics which effect the illumination of tissues. The laser light, scattered by the tissues, emerges with a spectrum broadened by Doppler effect due to motion of red blood cells in the micro-circulation vessels. The light from the tissues is fed to a photomultiplier tube or photodiode via a pinhole mask and interference filter. The photomultiplier tube or photodiode, as a result of beating of various components of the light it receives, produces as its output signals a homodyne or heterodyne spectrum or both. These output signals are fed, via a low pass filter, to a differentiator, which differentiates the output signals from the multiplier tube or photodiode, which are passed by the filter, with respect to time. The output of the differentiator is fed to a root-mean-square (RMS) detector. The low pass filter passes, for example, signals having a frequency up to about 20 KHz. The output (R) from the detector, which represents the blood flow (average percolation) in the tissues plus shot noise, a constant (S), is fed to a digital voltmeter. The voltmeter produces a visible read-out indicative of the output from the detector. The output from the detector is also fed to a calculating circuit which also receives a signal corresponding to the mean current (I) produced by the photomultiplier or photodiode, the calculating circuit effecting a solution to the equation ##EQU1## Fnorm is a normalized output signal representation of blood flow parameter.

Optimum Performance of Suppressed Carrier Receivers with Costas Loop Tracking

Abstract: The performance of suppressed carrier receivers with Costas loop tracking is optimized by proper choice of loop arm filter bandwidth. In particular, it is shown that for a variety of passive arm filter types, there exists, for a given data rate and data signal-signal-to-noise ratio, an optimum filter bandwidth in the sense of minimizing the loop's squaring loss. For the linear theory case, this is equivalent to minimizing the loop's tracking jitter. When symbol synchronization is known, it is shown that by replacing the passive arm filters with active filters, i.e., integrate-and-dump circuits, one can achieve an improvement in carrier-to-noise ratio of as much as 4 to 6 dB depending on the passive arm filter type used for comparison and the value of data signal-to-noise ratio (SNR).

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 timesequenced 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.

Computer-aided design of separable two-dimensional digital filters

Abstract: A procedure for the design of separable two-dimensional digital filters is presented. The computation involved in both the filter design and implementation is shown to be efficient. Several examples are presented which illustrate the application of the technique.

A generalized |ω|-filter for 3-D reconstruction

Abstract: The 3-D reconstruction of a density function is based on a direct convolution algorithm developed first by Ramachandran and Lakshiminarayanan. Their method adopts a particular choice of weighting function or filter which is called here an |ω|-filter. In some cases this choice of filter had an undesirable oscillatory response. To remedy this problem Shepp and Logan found a weighting function which produced a better reconstruction of a head section. The filter functions of Ramachandran and Lakshminarayanan and Shepp and Logan are only two of many choices for an |ω|-filter. Shepp and Logan's filter was the best for the early tomographic machines. Their filter function provided both a damped response to the cut-off frequency and a low sensitivity to noise. For the new tomographic machines, however, it is desirable to find filters that are not sensitive to counting noise, sample size and sample spacing as the previous filters. Here a study and generalization is made of the previous |ω|-filters. It extends the important filters of Ramachandran and Lakshiminarayanan, and Shepp and Logan to a class of generalized |ω|-filters. A generalized |ω|-filter can be chosen to have both good accuracy and a flexibility to cope with noise. A detailed comparison is made among the different possible filter shapes with respect to their responses to simulated data and noise. Finally in this paper it is demonstrated that a substantial reduction in the x-ray exposure time can be accomplished by choosing the appropriate generalized |ω|-filter.

The display-equalized filter for frequency-time analysis

Abstract: Frequency-time analysis (FTAN) results are usually presented as an array of amplitude values associated with discrete intervals group velocity and log period. The display-equalized filter, a modified form of the Gaussian frequency-time filter, produces amplitude estimates for which the frequency-time “averaging region” is of approximately constant shape, although varying in size, throughout the display. The shape of the averaging region can be adjusted to permit greater frequency resolution, with a corresponding loss of time resolution, and vice versa. An improved time-variable filter effectively utilizes the optimum frequency-time resolution of the Gaussian filter.

On parasitic oscillations in digital filters under looped conditions

Abstract: If a filter is used in multiplex telephone equipment, it actually operates in a loop due to the presence of two-wire/four-wire terminating equipment. Due to this, the power fed back into the filter input can, under adverse conditions, nearly equal the power leaving the output. This can be a cause for parasitic oscillations in case of digital filters. If properly designed, however, wave digital filters and nonrecursive digital filters remain stable.

Optimum edge detection filter

Abstract: Edge detection and enhancement are required in a number of important image processing applications. In this paper we consider the problem of optimizing spatial frequency domain filters for detecting a class of edges in images. The filter is optimum in that it produces maximum energy in the vicinity of the location of the edge for a given spatial resolution I and the bandwidth Ω. We show that the filter transfer function can be specified in terms of the prolate spheroidal wavefunctions for a given space–bandwidth product IΩ. Further we show that for values of IΩ less than 2, the optimal filter represents the Laplacian operator in image space followed by a low pass filter with a cutoff frequency Ω.

Channel selection for a television receiver having low-gain high frequency RF-IF section

Abstract: A channel selector for a television receiver having a low gain, high-frequency radio frequency-intermediate frequency section as combined with a relatively high-gain, lower intermediate frequency in the intermediate frequency section to reduce noise, such as intermodulation distortion and cross-modulation distortion. The channel selector comprises a frequency spectrum filter in the RF section, a mixer which preferably includes a metal semiconductor field effect transistor, i.e. MESFET, and a channel selecting filter. The frequency spectrum filter is coupled to the antenna of the television receiver and filters a frequency spectrum of radio frequency signals as received from the antenna, such as a plurality of television channels. The mixer has an input coupled to the frequency spectrum filter and frequency shifts selected channels of the frequency spectrum to a predetermined relatively high intermediate frequency. The RF section has a low gain RF amplifier connected between the frequency spectrum filter and the mixer or may have no RF amplifier at all. The channel selecting filter has an input coupled to the mixer and filters the selected channel at the predetermined intermediate frequency. The total gain from the RF section through the channel selecting filter is no larger than necessary in order to obtain a desired system noise figure. Another mixer is provided in the IF section, this IF mixer being coupled to the output of the channel selecting filter. The IF mixer receives the filtered selected channel at the relatively high intermediate frequency from the channel selecting filter and frequency shifts the filtered selected channel to a substantially lower second intermediate frequency. An IF amplifier having a relatively high gain is connected to the output of the IF mixer and adds gain to the filtered selected channel at the lower second intermediate frequency, the added gain being substantially in excess of the total gain provided before the channel selecting filter.

Metal detector system with ground effect rejection

Abstract: A metal detector uses a transmitting search coil inductively coupled to a receiving coil for detecting the presence of metal objects near the surface of the ground within the field of the coils. An oscillator generates a signal transmitted by the transmit coil, and the signals detected by the receive coil are coupled to the signal inputs of two synchronous demodulators. The output of the oscillator also is applied at different phases to the reference signal inputs of the two synchronous demodulators. The outputs of these demodulators then are passed through low pass and bandpass filters having a low cutoff frequency which is higher than the highest frequency components generated in the synchronous demodulators due to ground effects. The signals passing through the bandpass filters then are applied respectively to the signal input and reference signal input of a third synchronous demodulator, the output signal of which has an amplitude representative of the presence of metal objects and the polarity of which is an indication of the type of metal being detected. Undesired signals produced by ground effects are reduced by a considerable amount, and the output of the third synchronous demodulator is applied to a suitable indicator circuit.

Digital-filter synthesis using concept of generalised-immittance convertor

Abstract: By using the concept of the generalised-immittance convertor, new 2nd-order digital-filter sections are developed. These are then used as building blocks in a cascade synthesis. The proposed synthesis yields lowpass, highpass and bandstop filters with improved inband signal/noise ratio relative to that in conventional cascade filters. In addition, it yields low-noise and economical digital equalisers.

A novel "Follow the master" filter

Abstract: Active-R filters have their pole-frequencies directly dependent on the gain-bandwidth (GB) product of the operational amplifiers used. A novel technique, by which the pole-frequencies of the system are made to depend on a stable reference-frequency times the ratio of the GB products, rather than their absolute values, is given.

Passive filter compensation network

Abstract: A compensation network for a passive filter which compensates for the variation of the loss attenuation of the filter dependent on ambient temperature variations consists of an L-section whose series arm is formed by a temperature dependent resistor and whose shunt arm is formed by another temperature dependent resistor in series with a resonant circuit. The resonance frequency value of the resonant circuit is chosen nearly equal to the band limit frequency of the filter.

Television horizontal oscillator frequency control arrangement for use with a tape recorder

Abstract: of the Disclosure A television horizontal oscillator for use with a video source having a step change in the phase of the horizontal synchronizing pulses occurring at the vertical rate uses a phase-lock loop including a low-pass filter having a controllable filter characteristic. The control input of the filter is coupled to a source of vertical deflection rate signals and the filter attenuation is varied at the vertical rate. A delay arrangement including a ramp generator and a comparator delays the variation of the filter characteristic relative to the vertical synchronizing signal.

Minimum-phase CCD transversal filters

Abstract: The calculated and measured response of a minimum-phase CCD low-pass transversal filter is compared to a linear-phase design with the same magnitude characteristics. It is shown that the minimum-phase design can offer up to an order of magnitude improvement in group delay, and is also less sensitive to transfer inefficiency and tap-weight error than the linear-phase design.

FM Feedback demodulator having threshold extension circuit with two-pole crystal filters

Abstract: Demodulation of FM signals employing an FM feedback loop of the present invention includes a first mixer circuit for mixing an IF input signal with a voltage controlled oscillator signal to down-convert the IF input signal to a predescribed center frequency IF signal. This signal is coupled through a tuned amplifier having an automatic gain control loop and is then filtered in a predetection filter and amplified. The predetection filter is composed of a two-pole crystal bandpass filter having a very narrow bandwidth. The filtered signal is then coupled to a frequency discriminator circuit comprised of a frequency detection second mixer and filter circuit, the frequency detection filter circuit being composed of a second two-pole crystal bandpass filter, the bandwidth of which is an order of magnitude larger than that of the predetection filter. The output of the frequency discriminator is coupled through a loop compensation filter to the voltage controlled oscillator so that changes in the output frequency of the voltage controlled oscillator reduce the deviation of the FM signal in the intermediate frequency and frequency discriminator stages. The use of two-pole crystal bandpass filters improves the frequency stability of the system and significantly reduces post detection noise levels, as compared with conventional limiter/discriminator techniques. The two-pole crystal bandpass filters have a Butterworth configuration. In addition to the FM feedback loop, a squelch circuit is also coupled to the output of the frequency discriminator.

Hearing aid with adjustable frequency response

Abstract: In the illustrated embodiments, a transistor amplifier stage has filter components in its input circuit providing a 12 dB/octave rise in response above a selected frequency. A bridging circuit of variable resistance may be provided for introducing a desired degree of low frequency response. A continuously controlled or switched transistor circuit may modify the resistance values in a base-emitter circuit and in a base-collector circuit of the filter network simultaneously to increase the cut-off frequency while retaining the 12 dB/octave rate of rise in response. Both the low frequency response in the attenuation range and the cut-off for the high frequency response range may be adjustable in the same amplifier stage, or a quantized selection of successive cut-off frequencies is feasible using e.g. a plurality of amplifier stages in cascade.

An Exact Solution for a Six-Cavity Dual-Mode Elliptic Bandpass Filter

Abstract: The paper treats the synthesis and realization of a six-cavity dual-TE/sub 101/-mode bandpass filter exhibiting all four theoretically possible attenuation poles at finite frequencies. A novel arrangement of the cavities in the dual-mode resonators effects the feasibility of an exact elliptic filter taking the place of the well known six-cavity pseudo-elliptic filter, which presents only two attenuation poles at finite frequencies. Measured curves of an example of implementation at 4 GHz show the results to be in good agreement with the theoretical responses.

A general second-order digital filter with controlled rounding to exclude limit cycles for constant input signals

Abstract: In this paper a new general second-order digital filter section is proposed. One of the two quantizers used in the filter applies controlled rounding which has the effect that limit cycles are completely eliminated under constant-input conditions. Furthermore, the configuration of the filter is such that multipliers can be saved in case the filter is used as an all-pass filter or as a filter with zeros of transmission on the unit circle in the z domain.

Short-Circuit Tuning Method for Singly Terminated Filters

Abstract: This paper describes a filter tuning method based upon the match of measured and computed input impedances for a short-circuited filter. Two singly terminated filters, an 8-pole Chebyshev filter, and a 6-pole pseudoelliptic function filter tuned by using this method have demonstrated excellent performance.

Self-contained charge-coupled split-electrode filters using a novel sensing technique

Abstract: Self-contained single-chip charge-coupled split-electrode filters with 55 taps and a novel channel structure have been built with a double-level polysilicon NMOS process. Operating at a sample rate of 32 kHz, these devices provide a low-pass filter function with a passband from 0 to 3.2 kHz and a stopband above 4 kHz. The image charge on the sense electrodes is detected with a novel sensing circuit employing two on-chip operational amplifiers, one of which suppresses the common-mode signal on the two sense buses while the other one integrates the difference signal. In addition, the chips carry antialiasing prefilters, a correlated double sample-and-hold circuit to minimize reset noise and to restore the output signal, and all the necessary peripheral logic and biasing circuitry so that the devices can be operated from a single master clock and two power supplies of +12 and -5 V, respectively.

A Versatile Three Phase Oscillator

Abstract: This paper presents the design and performance of a versatile three phase oscillator The oscillator is built using digital circuits It provides five distinct output waveforms: sine, triangle, square, trapezoidal and symmetrically clipped sine waves Independent signals control output frequency, amplitude and phase sequence The oscillator has been built and successfully used in a cycloconverter powering a linear synchronous motor

Digitally tuned circuit including switching of bank capacitors between plural tuned circuits

Abstract: An electrically tuned filter arrangement including several tuned filters which may alternatively be coupled to the input and output of the filter arrangement so as to provide the arrangement with different tuning ranges. Each tuned filter has a tuning input to which a capacitive impedance is to be connected for establishing the center tuning frequency of the filter. A capacitor bank is provided including plural individual capacitor elements, and means for coupling any selected combination of the capacitor elements to a capacitive impedance output. This capacitive impedance output is switchably connectable to the tuning inputs of any of the tuned filters. A filter control circuit is provided which responds to a frequency selection signal to automatically connect the appropriate tuned filter to the input and output of the filter arrangement, to couple the capacitive impedance output of the capacitor bank to the tuning input of the tuned filter thus selected, and to cause an appropriate combination of capacitor elements to be coupled to the capacitive impedance output. The filter arrangement is thus automatically tuned to the frequency indicated by the frequency selection signal.

Two-dimensional wave digital filters using doubly terminated two-variable LC-ladder configurations

Abstract: This paper proposes a method of obtaining a two-dimensional wave digital filter of the recursive type from a doubly terminated LC-ladder network in two variables by replacing each series or shunt arm element of the ladder by its equivalent digital two-port. A number of realizations of the wave digital two-ports, which are canonic in multipliers, have been obtained. An example of a circularly symmetric low-pass two-dimensional digital filter is considered using these realizations. The sensitivity of this filter with respect to the multiplier coefficient changes due to finite word length is compared with that of the direct realization. It is found that the wave digital filter appears to be a more desirable form of implementation than the conventional cascade form.

Digital filter circuit

Abstract: Apparatus in a digital filter circuit for substantially eliminating limit cycle noise comprises means for randomly inhibiting the rounding signal in at least one of the multipliers comprising the digital filter circuit.

Electrically tunable bandpass filter

Abstract: A bandpass filter is described for filtering a television signal. The filter has a passband which may be selectively centered for filtering signals in one of two channels of the television receiver. The television channel at which the filter passband is centered is selectable by applying a voltage to a control terminal. The filter frequency response for each channel selected may be independently controlled and a linear phase response is realized for the selected channel minimizing distortion.