Showing papers on "Filter design published in 1975"

A normalized digital filter structure

Abstract: A normalized digital filter structure is presented, based upon an orthonormal polynomial expansion. This structure is recursively designed, has several predictable stability properties in the presence of time-varying parameters, and appears to have roundoff noise properties which are superior to other known filter structures, particularly in the presence of clustered poles. Each section of the filter can be precisely implemented by one complex multiply.

The importance of phase in image processing filters

Abstract: We demonstrate that phase accuracy is extremely important in image processing filters and express the hope that more work will be done on the development of filter design techniques which use phase as well as magnitude specifications.

A novel implementation for narrow-band FIR digital filters

Abstract: In an earlier paper Crochiere and Rabiner [1] discuss the theory of using finite impulse response (FIR) digital filters for signal decimation, interpolation, and filtering. In this paper we expand on the ideas presented in the earlier paper on implementing narrow-band designs efficiently. It is shown how, using the techniques of decimation and interpolation, a desired narrow-band filter can be realized with a greatly reduced number of multiplications per second in the realization over standard direct form implementations. Further, it is shown that the proposed implementation can have less roundoff noise and less severe coefficient sensitivity problems than a standard direct form implementation. Several examples are presented to illustrate the applicability of this implementation to practical design problems.

Optical-waveguide filters: Synthesis

Abstract: Analytical formulas for the synthesis of optical-waveguide filters having arbitrary spectral-response characteristics are derived from coupled-mode formalism. Use of these general formulas is illustrated by design of several filters, one of which is a linear power discriminator. The synthesis yields the functional dependence of spatial-perturbation period on the distance along the direction of wave propagation in the waveguide filter. The coupled-mode equations for the functional perturbation forms as determined by the synthesis process were solved numerically to find the actual response characteristics of the filter designs. Excellent agreement was found between the desired characteristics and those of the synthesized filters.

Double layered optical low pass filter permitting improved image resolution

Abstract: A double layered optical low pass filter is provided for use in a color video system for monitoring an object scene. The optical filter is designed to optically complement a dichroic stripe filter to prevent the introduction of spurious signals by any interference between luminance and chrominance signals, while at the same time providing a transmittance of high spatial frequency luminance signal to permit an improved image resolution. The optical low pass filter includes a first phase retarding filter layer designed to provide a first spatial frequency response across the visual spectrum and a second phase retarding filter layer designed to provide a second spatial frequency response across the visual spectrum. The combined resultant optical transfer function of the filter layers disclose a cut off of the higher spatial signal components of the primary color design wavelengths of the respective first and second filter layers, while transmitting a higher spatial frequency for the luminance signal to provide the improved image resolution. The specific cut off spatial frequencies of the primary colors and the bandwidth of the luminance signals can be subjectively optimized for any applicable system.

On the stability of the forced response of digital filters with overflow nonlinearities

Abstract: The forced response of a general type of digital filter, implemented with overflow nonlinearities, is studied. A definition of stability of this forced response is given. It is indicated that this stability can be studied by means of the zero-input behavior of a system, with the same structure but with time-varying nonlinearities. A theorem is derived giving sufficient conditions for the stability of the forced response of a digital system. Some results of the application of this theorem to wave digital filters are given.

On multistage finite impulse response (FIR)filters with decimation

Abstract: Nonrecursive, finite impulse response (FIR) digital filters are known to require a reduced rate of arithmetic operations relative to recursive infinite impulse response filters (IIR) in many applications where the sample rate is to be diminished. FIR symmetry and the absence of state variables contribute to the efficiency. A significant further economy in both arithmetic and memory space can be achieved by realizing the desired filter as a cascade of FIR stages, with interstage decimation of sample rate. In this paper, an approach to optimally allocate (i.e., factor) the composite decimation among filter stages (and hence determine the filter state characteristics) is described.

A New Type of Digital Filter for Data Transmission

Abstract: A digital filter is introduced consisting of a transversal part and a simple recursive network. The coefficients of the transversal part are equal to integer powers of two or zero; thus complicated multipliers are avoided and instead a simple routing circuit is used. Use of several types of the recursive network makes the filter applicable in different frequency ranges. The coefficients of the transversal part can be interpreted as differences of successive values of the impulse response of the filter. It is shown that this difference routing digital filter (DRDF) is especially suited for application in data transmission.

A new type of wave digital filter

Abstract: This paper proposes a new wave digital filter derived from doubly terminated LC-ladder networks by replacing each series or shunt arm element of the ladder by its equivalent digital two-port. It is shown that such two-ports may be cascaded without the use of adapters defined by Fettweis ( 1 ). A number of realizations of the wave digital two-ports, which are canonic with respect to both multipliers and delays, have been obtained. Also a realization which is canonic with respect to multipliers only is given and an example considered using this realization. The sensitivity of this filter with respect to the multiplier coefficient changes due to finite word length is compared with the conventional cascaded digital filter and also the one proposed by Renner and Gupta. It is found that the proposed filter appears to be a more desirable form of implementation than the conventional cascade form and comparable to that of Renner and Gupta ( 2 ).

Noncollinear tunable acousto-optic filter

Abstract: An electronically tunable optical filter is described wherein an optical beam is diffracted by an acoustic beam the group velocity of which is noncollinear with the optical beam transmitted through the birefringent crystal medium. The frequency and the direction of the phase velocity of the acoustic wave are chosen for a given incidence direction of the optical beam so that the diffraction occurs for a band of optical frequencies centered about a predetermined optical frequency and a relatively large angular distribution of incident optical beams. The noncollinear acousto-optic filter has the following advantages: (i) it allows very simple coupling of optical and acoustic beams into the interaction medium, (ii) it allows the use of multiple transducers for increasing tuning range or varying resolution, (iii) it allows a broad class of crystals to be used as filter material, (iv) it has a large degree of freedom in the filter design for optimizing efficiency or angular aperture and throughput, realizing narrow or broad optical passband and (v) it provides the spatial separation of incident and transmitted light through the filter and thus can be operated without the use of polarizers.

Non-recursive digital filter employing simple coefficients

Abstract: A non-recursive digital filter is disclosed, composed of a cascaded plurality of basic sections, each of which is characterized by coefficient values of integer powers of two's. No hardware multipliers are required in the filter and the operating speed is several times faster than other filters which utilize multipliers.

Efficient Minimax Design of Networks without Using Derivatives

Abstract: A new minimax network optimization algorithm not requiring derivatives has been developed. It is based on successive linear approximations to the nonlinear functions defining the problem. Adequate modeling of distributed parameter circuits for optimization purposes often involves parasitic, etc., which makes the gradient computation by the adjoint network method or related methods rather complicated, and often numerical errors are introduced in the gradients. Consequently, the algorithm is found to be of particular relevance in optimum design of practical microwave networks. The relative advantages of the proposed algorithm are established by comparison with known gradient and nongradient algorithms based on optimization of cascaded transmission-line transformers. The relevance to microwave filter design is demonstrated by an example which represents an improvement of analytical filter design results. Finally, optimum broad-band design of a practical coaxial transferred-electron reflection-type amplilier is carried out by means of the proposed method. The results are supported by experimental verification.

Self-adjusting digital notch filter

Abstract: Digital filter auto-adjustment to variations in input signals is achieved by normalizing the output of the second of two substantially identical series-connected digital filter stages and subtracting the normalized output of the second filter stage from its input to provide a measure of filter signal residue power Deviations of the signal residue power from the preset limits of a signal residue power standard are detected and signals responsive thereto are used to appropriately adjust variable bandwidth constants in both filter stages

The application of an approximate non-linear filter to systems governed by coupled ordinary and partial differential equations

Abstract: An approximate non-linear filter is applied to systems described by coupled ordinary and partial differential equations with both volume and boundary disturbances and measurement errors. Measurements are assumed to be carried out at u discrete number of spatial positions in the distributed portion of the system. The filter is applicable to cases in which the data are available continuously in time and at discrete sampling instants. Both versions of the filter are applied to an ingot heating problem of some interest in steel-making. The numerical results show that the filter is rapidly convergent and quite robust.

Detection, location, and removal of delay-free loops in digital filter configurations

Abstract: Simple graph-theoretic methods for the detection and location of delay-free loops in a digital filter configuration are outlined along with a method to develop an equivalent realization without delay-free loops.

Nonrecursive digital filters for detecting multifrequency code signals

Abstract: It is shown that for nonrecursive digital filters in the first canonic form the hardware requirements are proportional to the output rate. Using a low output rate, a relatively simple receiver for multifrequency-code (MFC) signals can be achieved. Linear phase bandpass pairs with a 90° phase difference are used to eliminate the input signal phase dependence. Narrow linear phase bandpass filters with even and odd symmetric impulse response are compared with filters with no phase constraints. It turns out that linear phase filters are almost as good as filters without phase constraints and that 8 bits are enough for the filter coefficients. Some aspects of the design of a digital receiver for PCM-encoded MFC signals are discussed.

Two-dimensional recursive filter design using differential correction

Abstract: This paper is based upon a presentation given at the 1974 Arden House Digital Signal Processing Workshop. The presentation discussed a method for designing two-dimensional recursive digital filters using a differential correction algorithm. A brief description of the algorithm and some design examples will be presented here. This design approach is compared to a similar approach used by Bednar and Farmer [1].

Dynamical Model Compensation for Near-Earth Satellite Orbit Determination

Abstract: Operational requirements in modern space applications often demand orbit determination accuracies which are limited by fundamental mathematical and computational restrictions. It is shown in realistic computer simulation studies how these difficulties can be alleviated for typical near-Earth satellites by employing dynamical model compensation (DMC) and accurate observations in the extended Kalman filter. Unmodeled and unknown accelerations affecting the motion of the satellite are effectively compensated by treating them as a firstorder, Gauss-Markov stochastic process. Although conventional state noise compensation (SNC) can provide satisfactory results for many applications, the DMC method offers a significant increase in estimation accuracy. Numerical behavior of the DMC structure is summarized as a function of a priori statistical parameters to aid in filter design analyses for operational applications.

Design and Operation of High‐Rate Filters—Part 1

Abstract: coagulation and filtrationso much so, in fact, that under certain conditions, adequate treatment might be carried out in the filters alone, thus eliminating the need for coagulation, flocculation, and sedimentation tanks. This procedure often can produce a better quality of water at a considerably higher flow rate than can conventional sand filters where filtration is simply the last step in a four-step process. Even so, it is acknowledged that not all types of water and wastewater-treatment problems can be solved by the filtration process alone and that the conventional approach will probably always be required in certain applications. The primary objective in filter design is to provide a facility for steady production of high-quality water with minimum operational and maintenance costs. Therefore, engineers should be responsible for the design of an efficient, reliable, simple, and economical filter from both operational and construction aspects. Generally, most engineers rely on either their experience or available design criteria in filter design. Filters constructed on this basis usually provide satisfactory results. Engineers, however, should consider certain design variables such as size and thickness of media, filtration rate, and maximum head loss, depending upon the condition of raw water and the required degree of pretreatment. The filter designer should also choose a proper type of filter bottom, filter wash system, and size of filter. -

Noise suppression filter

Abstract: A two line power filter for digital equipment or the like comprising a pair of toroidal cores on both of which both wires are wound, on one in flux adding relationship and on the other in flux opposing or cancelling relationship in reference to the primary power supply, together with capacitive shunts between the wires and from the wires to ground to filter out both common mode and differential mode noise.

Speech analyzer/synthesizer using recursive filters

Abstract: A speech analyzer and synthesizer features a digital adaptive linear predictor, using a recursive (rather than transversal) filter in a negative feedback loop which develops both feedforward and feedback filter coefficients. An input circuit is responsive to an input speech signal and to a first synthesized speech signal for developing an error signal. An output circuit is responsive to the error signal and to first state signals for developing multiplexed speech data signals. The multiplexed speech data signals are fed back, demultiplexed and applied to a first recursive filter to control the development of the first synthesized speech signal and the first state signals by the first recursive filter. The multiplexed speech data signals from the output circuit are also transmitted to a receiver which demultiplexes and applies the demultiplexed received speech data signals to a second recursive filter to control the development of a second synthesized speech signal by the second recursive filter. This second synthesized speech signal is then converted into an output speech signal which substantially sounds like the input speech signal.

Phase filter for reducing the effects of the noise components altering discrete phase modulated signals

Abstract: A digital phase filter including two decision filters connected in cascade or parallel for minimizing the noise components in a received phase modulated digital signal. The first decision filter cancels the residual noise component representing the phase intercept component and the phase shift component introduced by frequency shift, and the second decision filter minimizes the random noise component representing phase jitter and white noise.

Programmable general purpose analog filter

Abstract: A programmable transversal filter employig two charge transfer delay lines one for sampling a reference function, the samples providing the weighting factors for the transversal filter; the other for sampling an input signal to the filter. Each sample of the input signal is then multiplied by the appropriate weighting factor obtained from the sampling of the reference signal and all products are summed to obtain a sample of the filter output.

Digital mean clutter doppler compensation system

Abstract: A digital technique is disclosed for filtering clutter in a pulse doppler I radar which allows a filter notch to be set for different doppler frequencies. In particular, the system includes two or more MTI filters cascaded to cancel different kinds of clutter signals that interfere with target signal reception. In each filter a digital detection circuit senses the mean-clutter-doppler frequency for a particular clutter type and automatically places a notch at that frequency. Each notch is fixed by treating the radar samples as complex numbers and replacing the conventional filter coefficients with complex coefficients indicating amplitude and phase as derived from the radar returns.

Hadamard filter design

Abstract: An improved Hadamard filter is disposed between the dispersive element and detector of a Hadamard transform spectrometer. The detector has a series of linear photo-active diode elements arranged in the focal plane of an array of optical lenses. The lenses may be a series of Fly's-Eye lenses with each series arranged to overlay the corresponding linear element of the detector, or may be a cylindrical lens overlying a corresponding detector element. Individual lenses of the Fly's-Eye lens or individual elements of the cylindrical lens are selectively masked according to the desired matrix so as to perform the Hadamard transform spectrometry.

Digital filter device for processing binary-coded signal samples

Abstract: A digital filter is described in which the filter coefficients which differ from zero are coded in unequal numbers of multipliers which are equal to 2 n and in which immediately after each multiplication of a given signal sample by the multipliers of the relevant filter coefficent a new signal sample is processed in the multiplying devices by the relevant multipliers. Thus the signal samples are applied to the multiplying device at instants which are mutually set independently of one another.

Design of a third-order single amplifier filter

Abstract: General design formulae are derived for the synthesis of a third-order low-pass filter with a finite transmission zero. The design procedure uses a unity gain positive feedback amplifier, with balanced twin-T RC networks. A design example is given for a fifth-order elliptic filter, whose performance is compared to the ideal computed response.

Envelope-Derived Timing Recovery in QAM and SQAM Systems

Abstract: The problem of continuously recovering timing phase and frequency in a high-rate carrier-modulated data communication device is addressed. The communication link is assumed narrowband with respect to the baud rate of the device, thus necessitating the use of some type of equalization in the receiver. The receiver demodulator is assumed coherent in nature. A timing recovery technique based on narrow-band filtering and squaring of the received demodulated waveform(s) is examined. Analysis of the first moment of the waveforms so generated reveals a convenient indicator of sampling instants. If the timing recovery filters are properly designed, then these sampling instants exactly satisfy a requirement previously derived, ensuring efficient operation of the equalizer. Design criteria for the filters are given. Two principal types of linear modulation techniques are considered. These are quardrature-carrier amplitude modulation (QAM) and staggered QAM (SQAM). A set of experimental waveforms observed in a QAM modem signaling at 2400 Bd is included for illustration. Several conclusions are drawn as a result of the study. First, the mean timing waveforms in QAM and SQAM systems bear fixed time relationships to suitable receiver sampling instants. Second, adequate timing filter design is attained with very narrowband filters centered on frequencies equal to half the baud rate and the full baud rate. Third, recovery circuits used in QAM systems may be designed so as to have negligible timing jitter. In SQAM systems and, consequently, in vestigial sideband-amplitude modulation (VSB-AM) systems, this advantage is generally not found.

Filter with a reduced number of shift register taps

Abstract: Disclosed is a discrete-time filter configuration employing a shift register having output taps at only preselected stages for developing an output signal that is responsive to the signal of each stage of a shift register. The desired output signal is computed in a filter processing network by multiplying each of the output taps signals by a plurality of filter coefficients, by appropriately delaying the multiplied signals and by summing the delayed signals to produce the desired output signal.