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Showing papers on "Noise (signal processing) published in 1980"


Journal ArticleDOI
TL;DR: Spectrogram correlation can be used for classification as well as for estimation and detection, and for maximum likelihood parameter estimation, e.g., estimation of delay or center frequency of a signal.
Abstract: A locally optimum detector correlates the data spectrogram with a reference spectrogram in order to detect (i) a known signal with unknown delay and Doppler parameters, (ii) a random signal with known covariance function, or (iii) the output of a random, time‐varying channel with known scattering function. Spectrogram correlation can also be used for maximum likelihood parameter estimation, e.g., estimation of delay or center frequency of a signal. To estimate an analog input signal from its spectrogram, a modified deconvolution operation can be used together with a predictive noise canceler. If no noise is added to the spectrogram, the mean‐square error of this signal estimate is independent of the window function that is used to construct the spectrogram. When estimates of specific signal parameters are obtained directly from the spectrogram, these estimates have mean‐square errors that depend upon both signal and window waveforms. Spectrogram correlation can be used for classification as well as for estimation and detection. Parameter estimators and detectors are, in fact, specialized kinds of classifiers.

248 citations


Proceedings ArticleDOI
09 Apr 1980
TL;DR: An analysis of this technique is extended to the case when a linear filter appears in the auxiliary signal path and a general solution to this problem is obtained.
Abstract: A technique known as a "multiple correlation cancellation loop" and also as the "LMS algorithm" is widely used in adaptive arrays for radar, sonar, and communications, as well as in many other signal processing applications. In this paper, an analysis of this technique is extended to the case when a linear filter appears in the auxiliary signal path. A general solution to this problem is obtained and several examples for narrowband and broad-band signals are presented.

219 citations


Journal ArticleDOI
TL;DR: This article demonstrates the application of least squares for the estimation of system parameters and solutions are discussed for the case of white noise and correlated noise corrupting the useful output signal of the system.

123 citations


01 Sep 1980
TL;DR: In this paper, the importance of phase algorithms for reconstructing a signal from its phase potential applications is discussed. But, the authors do not discuss the role of phase in phase algorithms.
Abstract: : Experimental results supporting the importance of phase attempts at explaining the importance of phase algorithms for reconstructing a signal from its phase potential applications. (Author)

115 citations


Journal ArticleDOI
TL;DR: In this article, a digital deconvolution of ultrasonic echo signals was proposed to improve the resolution and quality of the lateral resolution of B-scan images by constructing a filter from the measured signal amplitudes across the transmitter beam.
Abstract: Abstrocr-Digital deconvolution of ultrasonic echo signals improves resolution and quality of ultrasonic images. Filtering usual B-scan images with a special digital fdter increases the lateral resolution by 50 percent. The filter is constructed from the measured signal amplitudes across the transmitter beam. By an empirical approach, it is optimized for low noise and high resolution. This method for lateral filtering produces good results for a great variety of test objects. The quality of the filter is maintained over a great range of depth, where the original beamwidth varies about 30 percent. This filtering method has high stability, such that application on measured profdes that are heavily disturbed does not increase noise nor produce disturbances.

87 citations


Journal ArticleDOI
TL;DR: In this paper, the transient behavior of the LMS adaptive filter was studied when configured as an adaptive line enhancer operating in the presence of a fixed or variable complex frequency sine-wave signal buried in white noise.
Abstract: The transient behavior of the LMS adaptive filter is studied when configured as an adaptive line enhancer operating in the presence of a fixed or variable complex frequency sine-wave signal buried in white noise. For a fixed frequency signal, the mean weights are shown to respond to signal more rapidly than to noise alone. For a chirped signal, a fixed parameter matrix first-order difference equation is derived for the mean weights and a closed-form steady-state solution obtained. The transient response is obtained as a function of the eigenvectors and eigenvalues of the input covariance matrix. Sufficient conditions for the stability of the transient response are derived and an upper bound on the eigenvalues obtained. Finally, the mean-square error is evaluated when responding to a chirped signal. A gain coefficient of the LMS algorithm is determined which minimizes the mean-square error for chirped signals as a function of chirp rate and signal and noise powers.

77 citations


Journal ArticleDOI
TL;DR: In this paper, the authors consider an observed stochastic process consisting of a signal with additive noise and assume that the signal has finite energy and that both the signal and noise are independent.
Abstract: Consider an observed stochastic process consisting of a signal with additive noise. Assume that the signal has finite energy and that the signal and noise are independent. In this paper we show tha...

67 citations


Patent
14 Jul 1980
TL;DR: In this paper, a low phase noise signal source is presented, which incorporates a voltage controlled oscillator (VCO) and a feedback network that, in effect, demodulates the VCO output signal and supplies negative feedback representative of the VOC signal noise to the VCA frequency control terminal.
Abstract: A low phase noise signal source is disclosed which incorporates a voltage controlled oscillator (VCO) and a feedback network that, in effect, demodulates the VCO output signal and supplies negative feedback representative of the VCO signal noise to the VCO frequency control terminal. The feedback network includes a frequency discriminator of the type wherein a time delay network is connected to one input port of a phase detector, with the VCO output signal being supplied to the time delay network and the second input port of the phase detector. A variable phase shifter, responsive to the signal supplied by the phase detector, is included in one of the phase detector input paths to cause a zero crossover of the frequency discriminator transfer characteristic to occur at the frequency to which the VCO is tuned. By establishing predetermined relationships between the transmission poles and zeroes of the feedback network, normal VCO tuning characteristics are preserved while simultaneously decreasing VCO phase noise. Various arrangements for operating the signal source over a wide band of output frequencies are disclosed, including a low-noise phase-locked loop frequency synthesizer.

67 citations


PatentDOI
TL;DR: In this article, a serial analog delay bucket brigade device is used to compensate for the angle between the listener and the speakers, and a continuously variable delay is provided in the 100 microsecond -1 millisecond range by the utilization of a SADB device.
Abstract: A stereo enhancement system utilizes a difference signal derived from the left and right stereo channels in which the difference signal is delayed, amplified, and then added into the appropriate channels to cancel left/right speaker mixing at the listener's ears, thereby to improve stereo separation without center region distortion. Depending on the amplification level of the difference signal, an increase in the perimeter sound over that produced at the central region gives a "wrap around" effect with only two speakers by increasing the volume of only the left/right directional sound components relative to centrally located sounds which have no left/right directionality. In order to compensate for the angle between the listener and the speakers, a continuously variable delay is provided in the 100 microsecond -1 millisecond range by the utilization of a serial analog delay bucket brigade device. Ultrashort delays are created, in one embodiment, by utilizing equal delays in the left and right stereo channels. The difference signal is then delayed by an amount equal to either of the two initial delays plus the amount of desired delay. When serial analog delay devices are utilized, the clock rate of the delay line used for the difference signal is made slightly lower than the clock rate of the delay lines utilized in initially delaying the left/right signals, thereby to provide for ultrashort delayed difference signals. Off-center listener positions are accommodated by a further continuously variable delay line in one or the other of the left and right channels. In alternative embodiments to minimize the noise characteristic of analog delay devices, the input signal to the device is compressed, followed by expansion of the output signal from the device.

66 citations


PatentDOI
TL;DR: In this article, a method of reducing the adaption time needed to adapt a synthetically generated secondary waveform so that it nulls a primary repetitive waveform (e.g. noise) from a source of vibration was proposed.
Abstract: A method of reducing the adaption time needed to adapt a synthetically generated secondary waveform so that it nulls a primary repetitive waveform (e.g. noise) from a source (1) of vibration, the secondary waveform generation (5) being synchronized by the source (1). The clarity and/or the amplitude of at least a part of the signal resulting from interaction between the primary and secondary waveforms is sensed (6) and used (7, 8) to reshape the secondary waveform. The invention finds application in the cancelling of repetitive vibrations (e.g. in quietening the driving cab of a vehicle).

61 citations


Journal ArticleDOI
01 Mar 1980
TL;DR: In this paper, two new methods are presented for the estimation of the frequencies of closely spaced complex valued sinusoidal signals in the presence of noise, and the most effective method is a computationally efficient method for realization of maximum likelihood or maximum posterior probability estimates of the frequency.
Abstract: Two new methods are presented for the estimation of the frequencies of closely spaced complex valued sinusoidal signals in the presence of noise. The most effective method is a computationally efficient method for realization of maximum likelihood or maximum posterior probability estimates of the frequencies. The second method is a class of algorithms for removing some of the deficiencies of present adaptive filtering and correlation-estimation approaches to estimation of frequencies, such as the forward-backward linear prediction method. In both of these new methods one is fitting a signal model to data. In method 1 the data are the observed samples of two complex sinusoids plus noise. In the second method the data are elements of an estimated correlation matrix, or of some of its eigenvectors, obtained from the observed samples.

PatentDOI
TL;DR: In this paper, a method and apparatus for detecting and identifying one or more excessively vibrating blades of the rotating portion of a turbomachine utilizing analysis of the characteristic Doppler waveform that results as the rotating, vibrating blade passes a fixed sensor.
Abstract: A method and apparatus for detecting and identifying one or more excessively vibrating blades of the rotating portion of a turbomachine utilizing analysis of the characteristic Doppler waveform that results as the rotating, vibrating blade passes a fixed sensor. The acoustic energy in the vicinity of the rotating portion of the turbomachine is sensed to generate a composite electrical signal representative of the broadband acoustic spectrum. Then, through both time domain and frequency domain signal manipulations, the undesirable noise components of the composite signal are removed. The resulting signal is then displayed to reveal the characteristic Doppler waveform of the blade vibrations, which may be analyzed to indicate the location of the excessively vibrating blade as well as its relative vibration amplitude. Changes in the latter with time indicate the initiation or propagation of a blade crack.

Proceedings ArticleDOI
01 Dec 1980
TL;DR: In this paper, principal eigenvalues and eigenvectors of a sample correlation matrix are used to improve the signal to noise ratio (SNR) in the data and to increase the resolution capability of nonlinear least squares at low SNR and linear prediction based frequency estimation methods.
Abstract: Principal component (eigenvalue-eigenvector) analysis is applied to processing of narrow band signals in noise. The amount of data available is assumed to be limited. Principal eigenvalues and eigenvectors of a sample correlation matrix are used to improve the signal to noise ratio (SNR) in the data and to increase the resolution capability of nonlinear least squares at low SNR and linear prediction based frequency estimation methods. Relation to Pronylike methods is explored. Performance of different methods is compared experimentally among themselves and to the Cramer-Rao (CR) bound.

Journal ArticleDOI
TL;DR: This method makes it possible to test the fitting of a signal to a model, but chiefly, to detect small shape differences between two signals without needing any modeling of the signals or the systems from which they come.
Abstract: A method for the detection of very small differences between the shapes of two signals is presented. This method uses the normalized integrals of the signals to be compared; hence it filters the noise. After a general presentation giving an especially detailed account of the definition of equality of two shapes, the possible application fields with respect to the intrinsic properties of the method is analyzed. It makes it possible to test the fitting of a signal to a model, but chiefly, to detect small shape differences between two signals without needing any modeling of the signals or the systems from which they come. Finally it is shown how the method was applied to the particular problem of detecting twocomponent chromatographic peaks visually confounded and with low resolution.

Patent
Fouad Daaboul1, Tiu Le Van1
16 Sep 1980
TL;DR: Speech signal presence is decided based on the input signal exceeding either of two thresholds: one, a fixed threshold (TF) set at an arbitrary level relatively high above anticipated noise; the other, an adaptive threshold (TL) which idles slightly above noise as mentioned in this paper.
Abstract: Speech signal presence is decided based on the input signal exceeding either of two thresholds: one, a fixed threshold (TF) set at an arbitrary level relatively high above anticipated noise; the other, an adaptive threshold (TL) which idles slightly above noise. If the input signal rises above the idling threshold of TL, speech presence is indicated. If the input signal continues to rise (i.e., amplitude-time slope positive), the presence indication continues. If the input signal level falls, the adaptive threshold is adjusted (TL=BT+D, where for example B=1, D=5 and T=the current signal sample average value). Hangover is controlled by the amount of time the input signal exceeds the threshold TL. Speech presence is also indicated by the input signal exceeds a third threshold (TH) which is also adaptive, and idles at a relatively high level above noise.

Journal ArticleDOI
TL;DR: Evaluation of dynamic system descriptors in PWRs using the multivariate analysis is presented for the first time in this paper and applications of mini computer oriented algorithms are evaluated using test data from operating power reactors.

Patent
15 Jul 1980
TL;DR: In this article, the average value of a voltage was established during a burst formed by an alternation of 1s and 0s, this value being stored until the following burst appears.
Abstract: This process enables the average value of a voltage to be established during a burst formed by an alternation of 1s and 0s, this value being stored until the following burst appears. According to the invention, the value considered is brought to an initial level less than or equal to half the smallest foreseeable burst amplitude, before appearance of the burst, then this level is increased depending on the signal received, in a time interval preceding the earliest probable time of appearance of the burst, e.g. by detecting the peak value of the received signal. A threshold signal taking into account the level of the noise and the disturbances is thus fixed. The average value is then taken during a definite period, counted from the first 1 of the burst.

Journal ArticleDOI
TL;DR: The Wiener theory for optimum linear filtering, as formulated by Walter and Doyle, was implemented in four different versions of the Wiener filter¿two a posteriori filters and two recursive filters.
Abstract: In order to obtain better estimates of the true evoked potential signal with fewer stimuli than is possible with classical signal averaging, the Wiener theory [16] for optimum linear filtering, as formulated by Walter [15] and Doyle [5], was implemented in four different versions of the Wiener filter?two a posteriori filters and two recursive filters. The effectiveness of the filters in separating signal from noise was tested in two simulations with known signal, and both simulated noise and real spontaneous brain wave activity. Correlation coefficients between the known test signals and filtered and unfiltered averages of test signal plus noise showed no significant difference between filtered and unfiltered averages.

DOI
01 Sep 1980
TL;DR: In this paper, a model for assigning the prior probability of an image in Bayes' theorem is proposed, which leads to a very general algorithm for image enhancement, and examples of sharpening blurred photographs show how the success of deconvolution depends on the signal/noise ratio in the degraded images.
Abstract: A model is suggested for assigning the prior probability of an image in Bayes' theorem which leads to a very general algorithm for image enhancement. Examples of sharpening blurred photographs show how the success of deconvolution depends on the signal/noise ratio in the degraded images.

Patent
19 Feb 1980
TL;DR: In this paper, the magnetic compass is used to calibrate a heading reference system mounted in a vehicle, e.g., an aircraft or a ta, using a microprocessor.
Abstract: A heading reference system mounted in a vehicle, e.g. an aircraft or a ta includes a magnetic compass and a gyroscope. The magnetic compass is subject to deviation in the magnetic field, hence the system must be calibrated. If the vehicle is oriented in a starting direction, and the output of the magnetic compass compared to the output of the gyroscope, an error signal is developed. This error signal is filtered to reduce noise, then stored. The vehicle is then re-oriented and the procedure repeated until sufficient information is available to calibrate the system. A microprocessor may be used for the computations and filtering of noise.

Patent
21 Mar 1980
TL;DR: In this paper, a high pass filter is used to accentuate the high frequency components of the video signal by partially differentiating the substantially square wave input video signal and feeding forward a sample of the undifferentiated signal.
Abstract: In a pattern recognition system the video output signal is a signal representative of either black or white. The shading effect of the video camera is minimized by using a high pass filter to accentuate the high frequency components of the signal by partially differentiating the substantially square wave input video signal and feeding forward a sample of the undifferentiated signal thereby retaining some low frequency components of the video signal and thereby minimizing the shading effect of the video camera without increasing detected noise components. Positive and negative-going zero crossings with respect to ground are detected through a capacitive coupled comparator circuit. In the absence of zero crossings the discharge circuit and charge circuit of the capacitor is interrupted thereby maintaining the value of the voltage on the capacitor with respect to ground. Changing the charge and discharge rate of the capacitor reduces the probability that noise on the signal will cause a false zero crossing.

Patent
21 Apr 1980
TL;DR: In this article, a noise suppressor for an automotive vehicle control system including a digital computer and a power unit for supplying power to the digital computer is described, and the inductor and capacitors are enclosed in a metal casing.
Abstract: A noise suppressor is disclosed for an automotive vehicle control system including a digital computer and a power unit for supplying power to the digital computer. The noise suppressor comprises an inductor connected in series with a power line extending from the power unit and capacitors each connecting each of signal lines extending from the digital computer with ground. The inductor and capacitors are enclosed in a metal casing.

Journal ArticleDOI
TL;DR: In this article, the theoretical basis for the development of a power-spectrum centroid detector is presented, and known and new applications to centroid detection are described, based on these applications, the requirements are formulated for a versatile centroid detectors which can be employed in various types of ultrasonic doppler systems.

Journal ArticleDOI
TL;DR: In this article, an analytic expression for the output spectrum is derived for the more general case of a reference signal containing a sinusoid in white noise, and it is shown that although the notch depth is a decreasing function of the reference signal-to-noise ratio, the exact relationship also depends on the primary input.
Abstract: The performance of the adaptive noise canceller is typically analyzed in terms of Wiener filter theory. However, it has recently been shown by Glover that a narrow-band signal in the reference at a frequency for which there is no correlated signal in the primary gives rise to a notch in the output spectrum, a solution which lies outside the scope of the classical theory. In this paper an analytic expression for the output spectrum is derived for the more general case of a reference signal containing a sinusoid in white noise. It is found that although the notch depth is a decreasing function of the reference signal-to-noise ratio, the exact relationship also depends on the primary input; thus, with noise present, there is no longer a time-invariant relationship between the output and primary spectra. These results are shown to reduce to those of Glover and the classical Wiener filter in the appropriate contexts.

PatentDOI
TL;DR: Time compression/expansion audio reproduction system of the type that provides pitch correction by repetitive variable time delay achieves improved performance by separating the reproduced signal from a recording into components which are separately delayed.
Abstract: A time compression/expansion audio reproduction system of the type that provides pitch correction by repetitive variable time delay achieves improved performance by separating the reproduced signal from a recording into components which are separately delayed. For studio quality reproduction the signal is separated into contiguous frequency bands which are each delayed synchronously and the processing noise in each band is eliminated by filtering each band signal after delay to eliminate high frequency components. Bandpass filtering prior to recombination as well as blanking and amplitude compression and expansion are also disclosed.

PatentDOI
TL;DR: In this article, a hearing aid in which input signals picked up through a microphone are divided to those of two or more frequency bands by means of a frequency division means, signal outputs of frequency bands in which noise signals are included are saturated or reduced by a compensating means such as compression or saturation amplifier, and mixed with signals of other signal of frequency band(s) which are not compensated.
Abstract: A hearing aid in which input signals picked up through a microphone are divided to those of two or more frequency bands by means of a frequency division means, signal outputs of frequency band(s) in which noise signals are included are saturated or reduced by a compensating means such as compression or saturation amplifier, and mixed with signals of other signal of frequency band(s) which are not compensated, whereby ambient noises can be removed from voice sounds.

Journal ArticleDOI
TL;DR: A novel two-stage adaptive signal extractor for intermittent signal applications that will adapt only when the signal is present and thereby effect a reduction in the distortion caused by the gust stage is presented.
Abstract: A novel two-stage adaptive signal extractor for intermittent signal applications is presented. If the presence and absence of the signal can be detected, the first stage will adapt only while the signal is absent and thereby effect a reduction in noise, whereas the second stage will adapt only when the signal is present and thereby effect a reduction in the distortion caused by the gust stage. Bounds on performance are derived, and performance improvement relative to a conventional one-stage adaptive noise canceller is assessed.

Journal ArticleDOI
TL;DR: In this article, the authors considered the problem of matching and Wiener filters for signal processing applications when the a priori information about signal and noise characteristics is not completely specified, and designed saddle-point or max-min solutions for the criterion functional (mean-square error or signal-to-noise ratio) over the classes of allowable signal shapes and signal spectral densities.
Abstract: Matched and Wiener filters are considered for signal processing applications when the a priori information about signal and noise characteristics is not completely specified. The approach is to design filters which are saddle-point or max-min solutions for the criterion functional (mean-square-error or signal-to-noise ratio) over the classes of allowable signal shapes and signal and noise spectral densities. Two-dimensional discrete-parameter processes are considered, and a numerical example is presented.

Patent
Sadao Kondo1, Hiroyasu Kishi1
25 Apr 1980
TL;DR: In this paper, a noise removing apparatus is interposed between an FM detecting circuit and a stereo demodulating circuit, which is responsive to a pulsive noise included in the output from the detecting circuit to enable a first gate to interrupt transmission of the output.
Abstract: A noise removing apparatus is interposed between an FM detecting circuit and a stereo demodulating circuit. The noise removing apparatus is responsive to a pulsive noise included in the output from the detecting circuit to enable a first gate to interrupt transmission of the output. Although the stereo pilot signal is disturbed during the interruption period, the disturbed stereo pilot signal is canceled by a pseudo-stereo pilot signal (a signal similar to the pilot signal) separately generated, whereby the disturbed stereo pilot signal is prevented from being applied to the stereo demodulating circuit.

Patent
Dressler Roger Wallace1
26 Mar 1980
TL;DR: In this article, a controllable gate receives an input signal and selectively passes and blocks the input signal in response to received noise blanking pulses which are generated by high peak magnitude noise impulses.
Abstract: A noise blanker which has circuitry that enables the threshold level of noise blanking to track the average background noise level is disclosed. In general, a controllable gate receives an input signal and selectively passes and blocks the input signal in response to received noise blanking pulses which are generated in response to high peak magnitude noise impulses. A signal related to background and impulse noise is extracted from an input signal. A controllable gain noise amplifier is utilized to amplify the separated background and impulse noise and negative feedback circuitry is utilized to maintain the average peak output of the noise amplifier substantially constant except for occasional large magnitude noise impulses which do not substantially change the average peak magnitude of the background and impulse noise. The output of the controllable noise amplifier is applied to a threshold switch means which produces blanking pulses in response to the amplified noise signal having a peak exceeding a magnitude which is greater than the substantially constant peak output level of the noise amplifier. This results in a blanker circuit in which the threshold level of blanking pulses closely tracks the average peak value of the background and impulse noise.