Showing papers on "Noise (signal processing) published in 1977"

Optimum Reception in an Impulsive Interference Environment--Part II: Incoherent Reception

Abstract: Because communications systems are seldom significantly interfered with by classical white Gaussian noise, it is necessary to consider other, appropriate (and tractable) interference models, if realistic estimates of system performance are to be obtained for the general spectral-use environment. For this purpose, Middleton's recently developed canonical statistical-physical model of "impulsive" interference is applied to real-world communication channels. The principal features of this model are first summarized, including the statistical relations required for the solution of signal detection problems. [Excellent agreement of these model statistics with correspondingly measured statistics is also noted.] The model for narrow-band impulsive interference (Class A noise, a subset of the overall model) is next specifically applied to an important class of coherent signal detection problems. Algorithms for error probabilities in optimum detection are then obtained, along with performance bounds, for the same error probabilities. Since it is known that in order to gain significant improvement over current receivers, the number of (essentially) independent samples of the received interference waveform must be enlarged (i.e., large "processing gains"), the performance results here are given parametrically in the number of samples, or equivalently, in the time-bandwidth product. Performance of current suboptimum receivers is then obtained and compared to the optimum performance. It is shown that very substantial savings in signal power and/or spectrum space can usually be achieved by using the indicated optimal algorithms. Since physical realization of the completely optimum detection algorithms cannot, in general, be economically realized, the somewhat more conservative, corresponding locally optimum Bayes detection (LOBD) receivers are derived. In general, these LOBD structures require adaptive, highly non-linear filters, preceding the conventional correlation detector elements characteristic of optimum receivers for Gaussian interference. Performance for these non-linear, optimum threshold systems is then determined, specifically in Part I for coherent reception.

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.

On the Nature of Input Channels in Visual Processing.

Abstract: The research reported herein was designed to assess whether the presence of noise elements in a visual display affects the detection of target letters at the perceptual or feature extraction level of processing, as well as at the decision level, and more specifically, whether (a) input or processing channels operate in an independent or interactive fashion and (b) how the spatial relation between signal and noise items affects detection performance. In order to distinguish among current theories proposed to account for the influence of noise items on visual processing, a forced-choice detection task was modified to incorporate a cueing procedure, which permitted the independent variation of signal-noise similarity, confusability, and proximity. The results provide evidence for feature-specific inhibition at the perceptual level, and a theory is proposed that assumes hierarchically organized, limited-capacity feature detectors and feature-specific inhibitory channels. There is now considerable evidence that the detectability of a designated signal is impaired by the presence of noise elements in the same visual display. The degree of impairment has been shown to vary as a function of the confusability of noise elements with the set of alternative targets (Estes, 1972; Gardner, 1973; Mclntyre, Fox, & Neale, 1970) and the spatial proximity of target and noise (Strangert & Brannstrom, 1975; Wolford & Rollingsworth, 1974). However, whether the effects of signal-noise similarity on target detection might be different than the effects of signalnoise confusability and how either of these variables might interact with signal-noise proximity has not been established; nor is it clear at what level of processing noise items impair the visual detection of signals. Evidence Edward E. Smith acted as Guest Editor for this article. The research reported was supported by Grant

Multichannel complex maximum entropy (autoregressive) spectral analysis

Abstract: The Burg reflection-coefficient method for maximum entropy (antoregressive) spectral estimation is generalized to apply to multichannel complex signal. It is shown that all resulting power matrices are positive definite. Preliminary numerical results obtained for a monochromatic signal with noise indicate that the determinants of the power matrices are rapidly reduced as the number of filter coefficients is increased, and that superior spectral resolution can be expected.

Signal processing for the multistate myoelectric channel

Abstract: In the multistate myoelectric channel, a single myoelectric signal source is used to control a multifunction powered prosthesis. The selection of a prosthesis function requires a receiver to process the myoelectric signal, contaminated with noise, and to decide on the basis of the received signals which function is desired. Thus the channnel cleady presents a problem of choice of receiver and of decision strategy. Previous sotutions to this problem have been basically empirical. In this paper we seek the optimum receiver where optimum is in the minimum probability of error sense. First a model is developed for the bipolar myoelectric signal to provide information about the relevant signal parameters and statistics. Using this information the Bayes minimum probability of error receiver is derived for an orbitrary signal parameter set. The optimum signal parameter set is then found for the Bayes receiver, and the receiver performance calculated. The receiver performance is measured and compared with the calculated performance. A significant performance improvement is seen in the optimum receiver over a more conventional receiver.

Robust Wiener filters

Abstract: The performance of minimum mean-square-error estimation filters for signals in additive noise can deteriorate considerably for deviations of the actual signal and noise power spectral densities (PSD's) from assumed, nominal densities. We consider two classes of PSD's which are useful models for the signal and noise when their PSD's are not precisely known. For these classes, robust filters which are saddlepoints for mean-square- error performance are derived. The robust filters achieve their worst performance for pairs of least-favorable signal and noise PSD's for which they are the optimum filters. It is shown by a numerical example that the robust filter can be very useful in maintaining a reasonable error performance over the whole of the classes of PSD's.

A self-tuning filter for fixed-lag smoothing

Abstract: The problem of estimating a discrete-time stochastic signal which is corrupted by additive white measurement noise is discussed. How the stationary solution to the fixed-lag smoothing problem can be obtained is shown. The first step is to construct an innovation model for the process. It is then shown how the fixed-lag smoother can be determined from the polynomials in the transfer function of the innovation model. In many applications, the signal model and the characteristics of the noise process are unknown. It is shown that it is possible to derive an algorithm which on-line finds the optimal fixed-lag smoother, a self-tuning smoother. The self-tuning smoother consists of two parts: an on-line estimation of the parameters in the one-step ahead predictor of the measured signal, and a computation of the smoother coefficients by simple manipulation of the predictor parameters. The smoother has good transient, as well as good asymptotic, properties.

Image restoration by Wiener filtering in the presence of signal-dependent noise.

Abstract: An optimum filter to restore the degraded image due to blurring and the signal-dependent noise is obtained on the basis of the theory of Wiener filtering. Computer simulations of image restoration using signal-dependent noise models are carried out. It becomes clear that the optimum filter, which makes use of a priori information on the signal-dependent nature of the noise and the spectral density of the signal and the noise showing significant spatial correlation, is potentially advantageous.

Stimulus variability and auditory filter shape.

Abstract: We have investigated the way in which stimulus variability will affects the attenuation characteristic or auditory filter shape inferred from masking experiments. Stimulus variability was found to have a pronounced effect on filter shapes derived from bandlimiting experiments in which the signal is a tone, the masker is a band of noise centered on the tone, and the independent variable is the width of the noise band. But stimulus variability had virtually no effect on filters obtained from notched noise masking experiments. In an attempt to extend the utility of the filter‐shape concept we have replicated an earlier experiment in which the masker is two tones rather than noise. The tones, each 57 dB SPL, were used to mask a narrow band of noise centered midway between them and threshold for the noise signal was measured as a function of the frequency separation of the tonal maskers. The form of the data is in good agreement with the auditory filter shape derived using a notched‐noise masker and a tonal signal. In order to predict the absolute levels obtained in the noise‐masking experiments, it was necessary to assume that internal noise adds to the variability of the stimuli. The resulting model, which incorporates the filter shape from the notched noise experiment and the energy detection model, predicts not only the form but also the absolute levels obtained in the bandlimiting and notched noise experiments. In addition, it predicts the shape of the data from the two‐tone masking experiment but it does not predict the overall level.

Near-Orthogonal Basis Functions: A Real Time Fetal ECG Technique

Abstract: A general technique for the real-time extraction of a physiological signal in the presence of well-characterized noise is described using a noninvasive fetal ECG detection system as an example application.

Engine knock signal generating apparatus with noise channel inhibiting feedback

Abstract: A vibration sensor mounted on an internal combustion engine characterized by knock-induced vibrations at a characteristic frequency and by other vibrations is tuned to resonate at substantially the characteristic frequency. A bandpass filter tuned to the characteristic frequency is connected to the vibration sensor output; and the output of the filter is provided to one input of a comparator and to average detector circuitry for generating a unidirectional noise reference signal representing noise at the characteristic frequency, which signal is provided to the other input of the comparator. The knock signal, obtained from the output of the comparator and comprising pulses corresponding to knock-induced peaks of amplitude greater than the unidirectional noise reference signal, is fed back through a low-pass filter to the average detector circuitry in sense to oppose increases in the unidirectional noise reference signal during said knock-induced peaks. The connection of the average detector circuitry to the output of the bandpass filter provides adaptability for mistuned sensors; and the negative feedback to the average detector circuitry reduces the distorting effect, amplified by the bandpass filter, of said knock-induced peaks on the unidirectional noise reference signal, which might otherwise distort the output knock signal.

Maximum likelihood pitch estimation

Abstract: The problem of determining the pitch period of voiced speech is formulated as a maximum likelihood (ML) estimation problem for an unknown periodic signal in white Gaussian noise of unknown intensity. Modifications of the ML estimator for speech include removal of bias, derivation of a measure of confidence, and prefiltering for non-white noise. The pitch estimation procedure is evaluated in the frequency domain and related to comb filtering. An alternate derivation of the ML estimator, related to the procedure proposed by Lacoss for maximum likelihood spectral estimation is presented. The problem formulated is to find a filter which passes a periodic signal of period P without distortion while simultaneously suppressing, in an optimum manner, a noise signal having a known spectrum.

Electro-control system for data transmission

Abstract: An error-control system for a continuous ARQ system, wherein, when a transmitter detects a dummy signal different from a negative acknowledgment signal caused by a channel noise appearing on a transmission channel, the transmitter supplies a receiver with a block signal comprising a designating number of a given block, a block data thereof and a check code formed by reversing a prescribed error detection code, thereby controlling the occurrence of an error in data transmission between the transmitting and receiving sides.

Method and apparatus for cancelling room reverberation and noise pickup

Abstract: Room reverberation and other uncorrelated signal sources characteristic of monaural systems are removed, in accordance with the principles of this invention, by employing two microphones at the sound source and by manipulating the signals of the two microphones to develop a single nonreverberant signal. Both early echoes and late echoes in the signal received by each microphone are removed by manipulating the signals of the two microphones in the frequency domain. Corresponding frequency samples of the two signals are co-phased and added and the magnitude of each resulting frequency sample is modified in accordance with the computed cross-correlation between the corresponding frequency samples. The modified frequency samples are combined and transformed to form the nonreverberant or correlated signal portion.

Predictive encoder with a non-linear quantizing characteristic

Abstract: A predictive encoder is disclosed which has a non-linear quantizing characteristic The transmission gain of the quantizer is less than unity when the amplitude component of the predictive error signal is smaller than a predetermined level and is substantially equal to unity when the amplitude component of the predictive error signal is higher than the predetermined level As a result, undesired redundancy components are effectively suppressed without causing an increase in quantizing noise and a deterioration in information quality

Compander noise reduction method and apparatus

Abstract: An information signal is compressively encoded for application to a limited amplitude transmission channel, such as a cinemagraphic film, by generating a gain control signal which represents the information signal level, delaying the information signal until the gain control signal has been substantially generated, and using the gain control signal to control encoding of the delayed information signal. The encoded information signal may be transmitted by itself with a similar delay step employed to decode it after reception, or the gain control signal may be transmitted along with the information signal and employed in the decoding operation, thereby avoiding the need for a delay step in the decoding process. In the latter case the gain control signal may be encoded along with the information signal, and employed in a feedback control loop for the encoding process. The invention also includes apparatus designed to perform the disclosed method.

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.

Systolic pressure determining apparatus and process using integration to determine pulse amplitude

Abstract: OF THE DISCLOSURE Improved method and means for identifying and quantizing an effectively periodic, steeply-rising wavefront of an input signal in the possible presence of low amplitude interference, as for example in systolic pressure determin-ing means utilizing a pressure cuff A representation of the time-derivative of the steeply-rising wavefront is obtained The time-integral of the time-derivative is obtained over a certain interval in each repetition of the steep wave-front, as determined by the derivative exceeding a reference value A threshold level is established which is repre-sentative of the magnitude of substantially only the time-derivative corresponding with the steeply rising wavefront The time-derivative of the input signal is compared with the threshold level while the former exceeds the reference value to provide a control signal indicative of whether or not the time-derivative is representative of a valid steeply rising wavefront (as opposed to noise) Finally, the integral of a particular time-derivative is recognized as the quantized value of the steeply-rising wavefront only if the control signal indicates validity

Seismic signal processing machine and method for noise removal

Abstract: A machine and method processing seismic survey signals to reduce the effect of noise bursts by determining the power level of the signals recorded during a seismic survey and removing those portions of the signal which deviate from the remainder of the signals by unacceptable amounts.

Sparse array performance

Abstract: In a linear array with equispaced elements, any elements not forming a unique pair spacing with at least one other element are redundant and may be eliminated to form a sparse array. The average power output of a sparse array can be made the same as for the full array by substituting self‐ and cross‐power terms from the remaining elements. Such processing assures identical signal‐to‐noise gain and directivity (beam patterns) for the sparse and full arrays, but the output variability for any finite averaging time will generally be greater for the sparse array. Linear combinations of possible substitution terms for each missing term may be optimized for minimum output variance. This concept has been applied to both sparse and full arrays to develop an optimum processing technique that minimizes output variance. An analysis technique has been developed for evaluating the output variance of sparse arrays operating in sound fields with specified signal‐to‐noise ratio and noise fields with arbitrary portions of...

Synchronized compressor and expander voice processing system for radio telephone

Abstract: The voice processing system for radio telephone consists of a compressor in the transmitter and an expander in the receiver. The compressor divides the input voice signal into syllabic groups and applies a gain to each syllabic group depending on the peak amplitude of the signal of the respective group, resulting in a reduction or compression of the dynamic range of the voice. The expander on the other hand increases or expands the dynamic range of the signal restoring the voice to its original form. The compressor and expander are synchronized such that the amount of expansion or instantaneous gain in the receiver is exactly the reciprocal of the amount of compression in the transmitter, i.e. when the gain of the compressor is N, the gain of the expander is 1/N. In this way the total system gain is always unity and is therefore transparent to voice and other analog signals passing through it. A porton of the voice frequency band is pre-empted to carry a digital signal to synchronize the compressor to the expander. This band designated as the control channel is located between 2500 to 2700 Hz and is used to carry binary frequency shift keyed (FSK) data stream which defines the instantaneous gain of the expander denoted by N. The band between 300-2500 Hz is used to carry the compressed voice. The effect of noise encountered in the transmission path is reduced because the soft syllables are amplified to the same amplitude as the loud syllables thus achieving an improved signal-to-noise ratio.

Quasi-stationary noise cancellation system

Abstract: An electronic security system detects a resonant tag circuit even in the presence of a substantially identical spurious resonance. The spurious resonance is fixed or quasi-stationary in time, in relation to the time that a resonant tag circuit is present in a security system. A representation of the quasi-stationary noise is stored in an electronic memory and is subtracted from a received signal to automatically cancel the interfering noise.

System for reducing noise transients

Abstract: A system for limiting the instantaneous value of a noise transient, by means of an electronic circuit wherein the amplitude of the total input signal being monitored is compared with a threshold value which is varied continuously according to the peak carrier signal value, and the output signal is maintained below the threshold value.

Impulsive noise reducing system

Abstract: An impulsive noise reducing system comprises a peak level detector for producing a detection signal with specific rise and fall time constants, in response to positive and/or negative peak levels of an input signal. Any signal which relatively exceeds the level of the detection signal by positive and/or negative peak levels of impulsive noise admixed in the input signal causes a control signal to be generated. Responsive to the control signal, that part of the input signal wherein the impulsive noise exists is suppressed.

Detection and recognition of pure tones in noise

Abstract: We examine the predictions of a new theorem relating signal identification (specifying a signal as a particular member of a set of potential signals) to signal detection (discriminating the presence of a signal). The theorem, derived in the context of signal-detection theory, requires that the signals be equally detectable and orthogonal. Our sinusoidal signals are partially masked by noise and their intensities adjusted to produce equal-signal detectability; we do not examine this assumption of the theorem. The theorem generally provides a reasonably accurate description of recognition performance for two-signal and four-signal conditions and is equally accurate for both the Yes-No and category-rating procedures. In a preliminary investigation of the orthogonality assumption, we varied the frequency separation between two signals. When the frequency separation between two signals is small (20 Hz near 1 kHz), the theorem fails to provide a good description of performance.

Commutating autozero amplifier

Abstract: An active analog signal processing system preferably embodied as an integrated circuit includes a pair of like amplifiers which are alternately switched between a first signal processing mode and a second zeroing mode in which a capacitor in the amplifier circuit is reverse charged to a voltage level equivalent to the average noise voltage level of the system whereby the system automatically cancels the low frequency noise thereof to provide a simple low noise processing system.

Telephone system for diver communication

Abstract: A telephone system for use by divers employing demand breathing apparatus.ounds picked up by each diver's microphone are processed by selective filtering, amplification, rectification and comparison to a reference voltage level to provide logic signals representative of presence or absence of inhalation noise that could mask voice transmissions of other divers or of a tender. The microphone output to listening stations is switched through an attenuation path by the logic signal representing inhalation noise and through a by-pass or full strength path by the other logic signal.