Showing papers on "Noise (signal processing) published in 1981"

Algorithms for ambiguity function processing

Abstract: Calculation of the complex ambiguity function is viewed as the basis for joint estimation of the differential delay and differential frequency offset between two waveforms that contain a common component plus additive noise. In many applications, the required accuracy leads to a need for integration over long data sets that can become a challenge for near real-time digital processing. The nature of the ambiguity processing is interpreted, and an algorithm approach is shown that minimizes the processing burden over a broad category of applications without affecting performance.

Constrained iterative restoration algorithms

Abstract: This paper describes a rather broad class of iterative signal restoration techniques which can be applied to remove the effects of many different types of distortions. These techniques also allow for the incorporation of prior knowledge of the signal in terms of the specification of a constraint operator. Conditions for convergence of the iteration under various combinations of distortions and constraints are explored. Particular attention is given to the use of iterative restoration techniques for constrained deconvolution, when the distortion band-limits the signal and spectral extrapolation must be performed. It is shown that by predistorting the signal (and later removing this predistortion) it is possible to achieve spectral extrapolation, to broaden the class of signals for which these algorithms achieve convergence, and to improve their performance in the presence of broad-band noise.

The Correction of I and Q Errors in a Coherent Processor

Abstract: A method is presented for correcting the gain and phase imbalances and the bias errors of the in-phase and quadrature channels of a coherent signal processor [1] by means of coefficients which are derived from measurements of a test signal. The residual errors after correction depend upon the signal-to-noise ratio (S/N) of the test signal and the degree of filtering used in deriving the correction coefficients.

Application of Kalman Filtering in Computer Relaying

Abstract: During the first cycle following a power system fault, a high speed computer relay has to make a decision usually based on the 60 Hz information, which is badly corrupted by noise The noise in this case is the nonfundamental frequency components in the transient current or voltage, as the case may be For research and development purposes of computer relaying techniques, the precise nature of the noise signal is required The autocorrelation function and variance of the noise signal was obtained based on the frequency of occurrence of the different types of faults, and the probability distribution of fault location A new technique for modelling the signal and the measurements is developed based on Kalman Filtering theory for the optimal estimation of the 60 Hz information The results indicate that the technique converges to the true 60 Hz quanitities faster than other algorithms that have been used The new technique also has the lowest computer burden among recently published algorithms and appears to be within the state of the art of current microcomputer technology

Adaptive cancellation of intersymbol interference for data transmission

Abstract: In this paper, we analyze a technique for accurately detecting transmitted data symbols contained in a modulated signal that has been degraded by a linear dispersive channel and additive Gaussian noise. The approach uses an adaptive equalizer which provides preliminary decisions to an adaptive canceller. The canceller output is used to remove the interference from an adaptive matching filter, resulting in the desired signal. Channel equalization attempts to invert the channel transfer function, while avoiding excessive noise enhancement. However, cancellation (as used in echo cancellers), attempts to generate a replica of the interfering signal and subtract it from the actual received signal containing the sum of the desired signal and interference. The cancellation approach, unlike equalization, offers the possibility of removing interference without enhancing the level of noise already present in the received waveform. Simulation results for transmission over practical channels show significant improvement of linear cancellation over both linear forward and decision-feedback equalization.

Analysis of discrete implementation of generalized cross correlator

Abstract: A common discrete implementation of the cross correlator uses a parabolic fit to the peak when the delay is not an integral multiple of the sampling period. This correspondence analyzes and assesses the pitfalls of this approach. It is shown that this yields a biased estimate of the time delay, with both the bias and variance of the estimate dependent on the location of the delay between samples, SNR, signal and noise bandwidths, and the prefilter or window used in the generalized correlator.

Some statistical properties of median filters

Abstract: Some statistical properties of median filters are analyzed. It is shown that median filters can remove impulsive plus Gaussian white noise better than Hanning filters when the amplitude of the impulses is large or the energy of the Gaussian noise is relatively low. It is also shown that, unlike linear filters, median filters preserve sharp changes in signals when the noise energy is not too high.

SVD pseudoinversion image reconstruction

Abstract: The singular value decomposition (SVD) pseudoinversion method has been applied to image reconstruction from projections. In this paper, two SVD pseudoinversion methods are discussed in the search for optimum restoration; one uses Wiener filtering and the other uses truncated inverse filtering. These methods partly overcome the ill-conditioned nature of reconstruction problems by trading off between noise and signal quality. Using computer simulation, the present SVD method was compared with the conventional Fourier convolution method. Results are presented together with some limitations peculiar to the application of this method for image reconstruction and restoration.

Auditory filter shapes derived in simultaneous and forward masking.

Abstract: In this paper we describe a method for comparing frequency selectivity in simultaneous and forward masking. The method is designed to eliminate off‐frequency listening, which may have had a confounding influence in earlier studies. Thresholds for 1‐kHz sinusoidal signals were measured as a function of the width of a spectral notch, centered at 1 kHz, in a noise masker. In experiment I thresholds were measured in forward masking for signal durations of 5, 15, and 45 ms, with a noise spectrum level of 40 dB SPL/Hz. Thresholds decreased with increasing notch width for all signal durations. However, the change was more gradual at longer signal durations. This is consistent with a model in which the process of decay of masking follows the auditory filter. For each signal duration, threshold was also measured as a function of the level of a noise without a spectral notch. Each signal threshold for the notch‐noise condition was then expressed as the level of a flat‐spectrum noise which would give the same masking. When transformed in this way, the data for the three signal durations coincide, and can be interpreted in terms of the same auditory filter. In experiment II the 5‐ms signal, whose level was fixed, followed the masker with delays of 5, 15, or 25 ms. The noise spectrum level was varied to find threshold. The change in threshold with notch width was independent of signal delay, confirming that the process of decay of masking follows the auditory filter. Experiment III showed that short‐term spectral changes produced by differences in the shapes of the envelopes of signal and masker did not influence the results. In experiment IV thresholds for a 5‐ms signal were measured in simultaneous and forward masking, both for a fixed noise level and for a fixed signal level. The derived auditory filters are similar for the two types of measure for each type of masking. However, the auditory filters derived in forward masking have bandwidths 17% smaller and slopes 48% greater than those in simultaneous masking. The differences between simultaneous and forward masking are interpreted in terms of suppression.

Detection of weak signals in non-Gaussian noise

Abstract: A locally optimum detector structure is derived for the detection of weak signals in non-Gaussian environments. Optimum performance is obtained by employing a zero-memory nonlinearity prior to the matched filter that would be optimum for detecting the signal were the noise Gaussian. The asymptotic detection performance of the locally optimum detector under non-Gaussian conditions is derived and compared with that for the corresponding detector optimized for operations in Gaussian noise. Numerical results for the asymptotic detection performance are shown for signal detection in noise environments of practical interest.

A posteriori time-varying filtering of averaged evoked potentials. II. Mathematical and computational aspects.

Abstract: The problem of estimating an unknown transient signal, given an ensemble of waveforms, in which this signal appears as a nonrandom component in the presence of additive noise is considered. This problem is solved by generalizing the method of "a posteriori 'Wiener' filtering". In the new method, the ensemble average is filtered by a time-varying system which is based on estimated time-varying power spectra of signal and noise. The nature of this system, and the computational procedures involved, are discussed in detail. A software package for time-varying filtering is briefly described. Application of the method is illustrated by a simulation example, which also provides a comparison to time-invariant a posteriori "Wiener" filtering.

Power line communication system utilizing a local oscillator

Abstract: Disclosed is a power line transmission system which generates a narrow band multi-frequency shift keyed signal having a first predetermined frequency representing a first state of information (e.g., a "0") and a second predetermined frequency representing a second state of information (e.g., a "1"). The system includes a power line connected between a transmitter and a receiver and the FSK narrow band signal is transmitted from the transmitter over the power line to the receiver which recovers the transmitted information. The system provides improvement over noise problems present in a power line and in addition provides multi-channel capability.

Techniques For Speckle Noise Removal

Abstract: In this paper, several techniques to reduce speckle noise (more generally signal independent multiplicative noise) in images are studied. The techniques include gray scale modification, frame averaging, low-pass filtering in the intensity and density domains, and application of the short space spectral subtraction image restoration technique in the density domain. Some discussions on the theoretical basis of the techniques studied are given and their performances are illustrated by way of examples.

Multichannel adaptive filtering for signal enhancement

Abstract: An adaptive technique for enhancing a signal against additive noise is described. It makes use of two or more input channels containing correlated signal components but uncorrelated noise components. The various input signals need not be of the same waveshape, since the adaptive enhancer filters the inputs before summing them. The output is a best least squares estimate of the underlying signal in a chosen input channel. Adaptivity allows optimal performance even though the signal and noise characteristics differ from channel to channel and are unknown a priori. Formulas for signal distortion and output noise power are developed. The more input channels available containing correlated signal components, the better will be the system performance. Excellent performance is obtained when the sum of the filter input signal-to-noise ratios (SNR's), defined as functions of frequency, is large compared to unity at all frequencies of interest. In this case the output noise is small, the output signal distortion is small, and the output SNR is approximately equal to the sum of the filter input SNR's. As such, the multichannel adaptive signal enhancer is a generalization of the classic time-delay-and-sum beamforming antenna.

Physical and Technical Considerations of Computerized Fluoroscopy Difference Imaging

Abstract: Considerations for the optimum design and use of a computerized fluoroscopy apparatus for performing time dependent image subtraction are presented. The advantages of logarithmic processing are presented. Assuming such processing, the interrelationship of achievable signal to noise, dynamic range and the minimum number of grey levels needed to digitize each image is discussed, and a formula relating these three quantities is derived. Image quality limits imposed by noise sources not associated with the detected x-ray fluence are discussed and a criterion for choosing a maximum x-ray fluence which will not waste patient dose is presented. The limits to spatial resolution achievable with conventional image intensifiers are discussed and it is shown that the maximum one dimensional spatial resolution in the object plane is achieved when the magnification of the x-ray system is such that the image of the x-ray focal spot projected through a point in the object plane onto the detector plane just covers the width of two pixels. The effect of the detection of scattered radiation on the difference image is discussed and it is shown that a conventional scatter reduction grid will improve image quality only if the ratio of the number of detected scattered photons to the number of detected primary photons is greater than 0.8 when no grid is used.

Vibration/noise reduction device for electrical apparatus

Abstract: A vibration/noise reducing device for applying vibration/noise-reducing vibrations/sound waves of substantially opposite phase to vibrations/noises generated from an electrical apparatus is disclosed, in which the vibrations/noises from the electrical apparatus is sensed by a sensor (12, 70), the sensed analog time-domain signal (14) is A/D converted, the resulting digital time-domain signal (18) is then Fourier-transformed, the resulting Fourier-transformed digital frequency-domain signal (22) is modified in its amplitude/phase to produce a second digital frequency-domain signal (26) for generating a vibration/noise reducing control signal, the second digital frequency-domain signal (26) is inverse-Fourier-transformed, the resulting inverse-Fourier transformed second digital time-domain signal (30) is D/A converted, and the resulting analog time-domain signal (34) is used as the control signal to generate the vibration/noise-reducing vibrations/sound waves to be applied to the virbrations/noises.

Holographic system for the storage of audio, video and computer data

Abstract: A rotating disc-shaped carrier upon which have been recorded audio, video or computer data in holographic form, a method for forming such a carrier, and an apparatus for reading such a carrier is disclosed. A beam of coherent radiation is spatially modulated in a prescribed way, this beam then illuminating one radial of a rotating carrier upon which has been recorded a series of holograms in the form of a spiral track or series of concentric tracks. The holograms are elongated in a direction perpendicular to movement of the rotating carrier and overlap in directions both parallel and perpendicular to such movement, the amount of overlap in the perpendicular direction being inversely proportional to the amount of overlap in the parallel direction such that data storage density remains constant across the surface of the rotating carrier. The sequentially illuminated holograms generate a series of signal images which vary in amplitude, width, spatial frequency, azimuth or position relative to a centerline. Detection of the signal images and extraction of uncorrelated noise resulting from a coded reference beam multiplexing procedure are accomplished by means of a differential photodetection system. All virtual signal images generated by holograms occuring within one illuminated radial of the rotating carrier are confined to a specific distance from a point in space on the side of the rotating carrier opposite to that of the virtual signal images, at which point is placed a rotatable photodetection system such that alignment between the photodetection means and any signal image within one radial requires only rotary movement of the photodetection system. Toleration of lateral misalignment between signal images and the photodetection means is accomplished by elongation of signal images in a direction substantially perpendicular to the signal image path, or by employment of point-source-type signal images which are detected by a transversely-positioned array of differentially-operating pairs of photosensors.

Photodetectors for acousto-optic signal processors

Abstract: A description of devices, architecture, and performance criteria is presented for solid-state CCD and photodiode sensors useful in acousto-optic processing systems. Special attention is given to sensors used in both bulk and integrated optical acousto-optic instantaneous Fourier transform systems. The device physics of silicon photodetectors is presented in detail including discussion and analysis of quantum efficiency, responsivity, ctosstalk, charge handling capacity, data rate, optical to electrical transfer function, noise sources, dynamic range, and sensitivity. A description of new technologies, such as VHSIC and GaAs IC's which are applicable to sensor technology for AO signal processing systems is also presented.

A recursive algorithm for the Bayes solution of the smoothing problem

Abstract: The optimum fixed interval smoothing problem is solved using a Bayesian approach, assuming that the signal is Markov and is corrupted by independent noise (not necessarily additive). A recursive algorithm to compute the a posteriori smoothed density is obtained. Using this recursive algorithm, the smoothed estimate of a binary Markov signal corrupted by an independent noise in a nonlinear manner is determined demonstrating that the Bayesian approach presented in this paper is not restricted to the Gauss-Markov problem.

Image enhancer apparatus

Abstract: An image enhancer apparatus includes an aperture correction circuit for receiving an input video signal and producing emphasized edge sharpness information for addition to the input video signal as delayed in a signal delay line The edge sharpness information is generated employing a pair of further delay lines which provide the second derivative of the input signal A detail enhancing circuit detects large transitions in the video signal and clamps a differentiated delayed version of the video signal at the output of the signal delay line to suppress large transitions while retaining detail information Outputs of the aperture correction circuit and the detail enhancing circuit are adjustably combined at the input of a coring circuit which selectively removes undesired noise in order to isolate the desired information The cored sum is added to the delayed video signal

Transformation of image-signal-dependent noise into image-signal-independent noise

Abstract: A point transformation, the normalizing transform, is presented, which, when applied to a measured noisy image, renders its noise signal independent. The transform is suitable for arbitary noise-to-signal dependence. We demonstrate its applicability and its limitations by using, as an example, noisy signals that belong to a family of gamma-distributed random variables with power-law variance-to-mean dependence. Its normalizing and variance-stabilizing properties are studied.

A low-cost method for rapid transfer function measurements with direct application to biological impedance analysis

Abstract: The measurement of the linear transfer function of a biological system has found wide use in the characterization of the system's input/output properties, and in the separation and measurement of the properties of the different subelements that make up the system. Unfortunately sophisticated and expensive instrumentation has traditionally been required to make these measurements. In this paper, we present detailed design specifications of a low-cost instrument that is capable of yielding transfer function measurements with a high degree of accuracy and speed. The instrument is comprised of a pseudo-random binary sequence signal generator with precise date acquisition synchronization circuits, interfaced to a general-purpose mini-or microcomputer system common to many laboratory environments. The instrument is capable of measuring the transfer function of an arbitrary biological system up to a bandwidth of 8.3 KHz, with a frequency resolution of 425 points. In cases where the biological measurements are not contaminated with experimental noise, the transfer function can be determined in as little as 47 ms of data collection. In the case where experimental noise is present in the biological measurements, a simple averaging method is described which results in an effective increase in the signal-to-noise ratio, thereby yielding accurate transfer function estimates. The instrument is especially well suited to the measurement of transfer functions of biological systems, where experimental noise is a problem and where only limited time is available to acquire stable measurements.

Consistent estimation of continuous-time signals from nonlinear transformations of noisy samples

Abstract: A signal cannot in general be reconstructed from its sign, i.e., from its hard-limited version. However, by the deliberate addition of noise to samples of the signal prior to hard limiting, it is shown that the signal can be estimated consistently from its hard-limited noisy samples as the sampling rate tends to infinity. In fact, such estimates are shown to converge with probability one to the signal and to be asymptotically normal. Although the estimates are in general nonlinear, they can be made linear by a proper choice of the noise distribution. These rather unexpected results hold for all bounded and uniformly continuous signals. In addition to the hard-limiter, such results are also established for certain monotonic and nonmonotonic nonlinearities.

Reduction of harmonic distortion in vibratory source records

Abstract: Due to non-linear effects, the swept frequency signals (sweeps) transmitted into the subsurface by vibrators are contaminated by harmonics. Upon correlation of the recorded seismograms, these harmonics lead to noise trains which are particularly disturbing in the case of down-sweeps. The method described in this paper—which can be regarded as a generalization of Sorkin's approach to the suppression of even order harmonics—allows elimination, from the final vibratory source seismogram, of harmonics of the sweep up to any desired order. It requires that not one single signal but rather a series of M signals is employed where each signal has an initial phase differing from that of the previous one of the series by the phase angle 2πM. Prior to stacking, the seismograms generated with the different signals have to be brought into the form they would have if they had been generated with the same signal. The method seems also to be capable of reducing the correlation noise if sign-bit recording techniques are used.

Adaptive filtering of canine electrogastrographic signals. Part 1: system design.

Abstract: The study of the relation between gastric myo-electrical activities recorded from serosal and cutaneous electrodes is hindered by the poor quality of the cutaneous signal. This hindrance could be minimised by suitable filtering of the signal. Since it is not yet clear which aspects of the cutaneous signal constitute valuable information, the filter process should not affect phase, amplitude, frequency and waveform of the gastric component, while noise components should be suppressed strongly. The system design of a modified adaptive filter that meets these requirements is described. The filter was implemented on a digital Nova 2 minicomputer. the filter performance is described and tested.

Automatic threshold tracking system

Abstract: A magnetic recorder including a playback circuit for decoding a ternary waveform digital data signal is disclosed, wherein positive and negative threshold levels, used to distinguish between signal levels representative of data ones from signal levels representative of data zeros, are automatically adjusted to compensate for occasional reductions in overall signal level that can occur due to dropouts and modulation noise on the magnetic medium Two sample and hold units are used to generate an output signal whose amplitude reflects only those signal levels that are greater than some minimum absolute value This output signal is filtered and then used to generate the compensated positive and negative threshold levels