Showing papers on "Sampling (signal processing) published in 1986"

Position measurement with a resolution and noise‐limited instrument

Abstract: A common problem in experimental data analysis is to locate the position of a signal to an accuracy which is substantially less than the actual signal width. By applying maximum likelihood estimation to this problem, this paper derives theoretical limits on the ability to locate signal position. The limiting error in position measurement is shown to be a simple function of the instrument resolution, the density of sample points, and the signal‐to‐noise ratio of the data. An interesting conclusion is that position information on a much finer scale than the minimum instrument sampling interval is contained in data of modest signal‐to‐noise ratio. The common procedure of excluding the portion of the data lying below an amplitude threshold to guard against background fluctuations is incorporated in the maximum likelihood analysis. It is shown that selection of the optimum amplitude threshold level depends on the type of noise present in the data, and can be an important factor in position accuracy. The analytical results exhibit close agreement with Monte Carlo simulations of position accuracy in the presence of noise.

Power spectra of fluid velocities measured by laser Doppler velocimetry

Abstract: The power spectrum and the correlation of the laser Doppler velocimeter velocity signal obtained by sampling and holding the velocity at each new Doppler burst are studied Theory valid for low fluctuation intensity flows shows that the measured spectrum is filtered at the mean sample rate and that it contains a filtered white noise spectrum caused by the steps in the sample and hold signal In the limit of high data density, the step noise vanishes and the sample and hold signal is statistically unbiased for any turbulence intensity

A Nonaliasing, Real-Time Spatial Transform Technique

Abstract: A two-pass spatial transform technique that does not exhibit the aliasing artifacts associated with techniques for spatial transform of discrete sampled images is possible through the use of a complete and continuous resampling interpolation algorithm. The algorithm is complete in the sense that all the pixels of the input image under the map of the output image fully contribute to the output image. It is continuous in the sense that no gaps or overlaps exist in the sampling of the input pixels and that the sampling can be performed with arbitrary precision. The technique is real time in the sense that it can be guaranteed to operate for any arbitrary transform within a given time limit. Because of the complete and continuous nature of the resampling algorithm, the resulting image is free of the classic sampling artifacts such as graininess, degradation, and edge aliasing.

Filtering by repeated integration

Abstract: Many applications of digital filtering require a space variant filter - one whose shape or size varies with position. The usual algorithm for such filters, direct convolution, is very costly for wide kernels. Image prefiltering provides an efficient alternative. We explore one prefiltering technique, repeated integration, which is a generalization of Crow's summed area table.We find that convolution of a signal with any piecewise polynomial kernel of degree n--1 can be computed by integrating the signal n times and point sampling it several times for each output sample. The use of second or higher order integration permits relatively high quality filtering. The advantage over direct convolution is that the cost of repeated integration filtering does not increase with filter width. Generalization to two-dimensional image filtering is straightforward. Implementations of the simple technique are presented in both preprocessing and stream processing styles.

Circuit breaker with solid-state trip unit with sampling and latching at the last signal peak

Abstract: A microprocessor-based digital solid-state trip-unit for processing digital signals derived from analog signals supplied by current sensors. In order to reduce measurement error of the peak value of the signals due to signal sampling, the microprocessor samples a signal latched at the last peak. A countdown timer determines the latching time to include two or three peak values.

Method of coding a video signal for transmission in a restricted bandwidth

Abstract: A video signal is coded for transmission in a restricted bandwidth by sub-dividing a frame of picture information into a set of constituent blocks, measuring the amount of picture activity in each block, sampling the information in each block at a rate related to the amount of picture activity in that block, and adding to the coded block a supplementary signal indicating the sampling rate used for the block. Thus a decision is made on a block-by-block basis as to whether the block is transmitted with full accuracy or whether it can be adequately reconstructed from the previous frames.

Tracking sample and hold phase detector

Abstract: A tracking sample and hold phase detector operating as an error sampled feedback loop which takes signal samples during successive periods of an input signal with a sampling gate, holds the signal sample in a storage gate, and feeds back the sample to the sampling gate to allow the bias of the sampling gate to track the input signal level, thereby maintaining balance of the tracking circuit. A fast comparator detects the zero crossing level of the downconverted signal which is comprised of successive signal sample steps and has the shape of the input signal. The sampling gate comprises a diode bridge enabled by a balanced strobe, the input signal and the strobe being close or equal in frequency, the frequency of the downconverted signal being the difference frequency.

Method and device for demodulating RF-modulated signals by means of digital filters and digital demodulators, and use of the method in a remote-control receiver

Abstract: The modulated baseband signal which has a continuous useful signal and noise signal spectrum is first of all band-restricted with a bandpass filter and then sampled at a sampling frequency (fa) which lies below the minimum sampling frequency prescribed by the sampling theorem. The sampling frequency (fa) is not less than the double value of the difference between the upper and the lower stop band frequency of the band-restricted useful signal spectrum, and the useful signal spectrum lies within one period segment ((1)/2fa). This undersampling produces a sampling frequency (fa) of around 3000 Hz which allows the use of an 8-bit microcomputer as a digital filter in a ripple-control receiver.

The Simple Rectification to Cartesian Space of Folded Radial Velocities from Doppler Radar Sampling

Abstract: Periodic sampling of the Doppler radar return signal at the pulse repetition frequency causes measured velocities to be ambiguous (folded) when true meteorological velocities along the radial direction exceed the Nyquist or folding value. Furthermore, mean radial velocity estimates become more uncertain as the spatial variability of velocity increases or the returned signal strength decreases. These data have conventionally been prepared for such uses as multiple-Doppler radar wind synthesis by unfolding and editing them in the sampling domain (range-azimuth-elevation spherical coordinates). An alternative method of locally (to the output grid point) unfolding the unedited radial velocities during their linear interpolation to a regular Cartesian grid is presented. The method preserves the spatial discontinuities of order twice the Nyquist value associated with velocity folding. A nondimensional velocity quality parameter is also computed which serves to identify interpolated values that contain ...

Bandwidth compressed transmission system

Abstract: In a bandwidth compressed transmission system in which a frequency band of a television signal of 2:1 interlace system is compressed, on an encoder side, by interfield offset subsampling at a first sampling frequency corresponding to a substantially upper limit frequency of the television signal and, after an output signal thus subsampled is applied to a LPF for passing a signal component having an aliased portion resulting from the interfield offset subsampling and having a cut-off frequency substantially equal to one half of the first sampling frequency, by interframe offset subsampling at a second sampling frequency lower than the first sampling frequency and higher than one half of the first sampling, so that a multiplexed subsampled transmission signal which does not include an aliased portion of reverse phase results from the interframe offset subsampling. When decoding, interframe interpolation processing of a multiplexed subsampled transmission signal is performed to obtain a signal for a still picture portion, while intrafield interpolation processing of the same transmission signal is performed to obtain a signal for a motion picture portion. An interframe difference of the transmission signal is detected to obtain a motion detection signal. The signals for still and motion picture are linearly mixed in accordance with the motion detection signal. Motion can be detected completely by utilizing an interframe difference. An arrangement of a receiver can be made simple and improves picture quality.

Phase-locked loop coefficient generator for a filter arrangement having a non-rational ratio between input and output sampling frequencies

Abstract: In a sample rate converter having a non-rational conversion factor the input samples coincide with low-rate clock pulses and the output samples coincide with high-rate clock pulses, or inversely. It comprises a filter coeffi­cient generator 3 which, based on the distance (deviation) between a low-rate clock pulse and the immediately preceding or immediately subsequent high-rate clock pulse, con­tinuously supplies a series of filter coefficients. To de­termine the deviation a phase-locked loop (30) is provided, with a phase detector (301) receiving the low-rate clock pulses as well as synthetic low-rate clock pulses and supplying a discrete-time phase difference signal u(.) which is applied to a processor circuit (302). This circuit supplies the desired deviation d(.) and a reference number N which is applied to a counter circuit (304). This circuit also re­ceives the high-rate clock pulses and each time after re­ceiving the number of clock pulses corresponding to the reference number it supplies a synthetic low-rate clock pulse. In the processor circuit 302 a filtering operation (3021) is first performed on the discrete-time phase dif­ference signal u(.) so that control signal samples H(.) are obtained. An auxiliary sample s(.) is subtracted from such a control signal sample and the difference is divided by a weighting factor incr. of the number P thus obtained those bits whose significance is less than 2° represent the deviation d(.), while the other bits represent the reference number N. By subsequently multiplying the deviation d(.) by the weighting factor incr, a new auxiliary sample s(. + 1) is obtained.

Apparatus useful in channel equalization adjustment

Abstract: In equalizing a digital channel, impulses are applied to the channel input at the Nyquist rate. The channel response is sampled to determine its amplitude values of specified sampling times, and the amplitude values are converted into corresponding pulse duration signals. The pulse duration signals are applied to a time interval analyzer whereby the signal input durations are displayed in the form of a histogram showing the number of pulses of a given duration as a function of pulse duration. The time dispersion of the displayed pulse durations measures the intersymbol interference present, and is a measure of the quality of the equalization.

Method and apparatus for calibrating an analog-to-digital conversion apparatus

Abstract: The phase relation of clock signals to be applied to an interleave type analog-to-digital conversion apparatus having N analog-to-digital converters is calibrated by applying a repetitive reference signal to the N analog-to-digital converters. The digital values from the N analog-to-digital converters are selected at corresponding sampling points of successive cycles of the reference signal. The clock phase relation is adjusted such that the selected digital values are made substantially equal to each other.

Random sampling: Distortion and reconstruction of velocity spectra from fast Fourier‐transform analysis of the analog signal of a laser Doppler processor

Abstract: We calculate the spectrum of a signal derived from a continuous signal by ‘‘sample and hold’’ random sampling. It is seen that an undistorted spectrum is obtained only if the mean sampling frequency is more than an order of magnitude greater than the highest frequency in the original signal. The cases of signal‐independent and signal‐dependent sampling probabilities are investigated. We show how to obtain a good estimate for the spectrum of the unsampled signal and determine the limitations of this form of processing.

Apparatus and method for high-speed determination of received RF signal strength indicator

Abstract: Rapid measurement of the received signal strength indicator (RSSI) of a radio frequency signal is performed by sampling received signal amplitudes and averaging only selected ones of the sampled amplitudes. In one embodiment, two sample values obtained successively in time are compared and the larger of the two is stored, this process continuing over a desired sampling interval. Sampled values which exceed both the value obtained from an immediately preceding sample time and the value obtained from an immediately succeeding sample time are stored twice, while sample values which are less than immediately preceding and succeeding sample values are never stored. The stored values are averaged to yield an indication of average received signal amplitude. The resulting average value is very close to true average signal amplitude, is substantially unaffected by Rayleigh fading phenomenon, and yet is sensitive to rapid changes in received signal amplitude caused, for exaple, by obstacles in the signal transmission path.

Motion-Compensating Field Interpolation From Interlaced And Non-Interlaced Grids

Abstract: Effects caused by sampling and motion compensating interpolation of television sequences containing translatory motion in the image plane are analyzed in the frequency domain. As a velocity-adapted lowpass filter possesses a "velocity passband", a precise motion-compensation is not required. Schemes for motion-compensating interpolation from non-interlaced and interlaced grids are proposed and investigated experimentally by means of moving zoneplate patterns. A novel scheme for interpolation from an interlaced grid preserves the full vertical resolution of the picture contents over a wide range of velocities.

Sampling clock generation circuit of video signal

Abstract: A recording apparatus receives a video signal containing a horizontal sync signal and a data signal comprised of a given number of image bit data arranged within each horizontal scanning period, and stores the number of image bit data in synchronization with the horizontal sync signal. A voltage-controlled oscillator produces in synchronization with the horizontal sync signal a frequency signal having a frequency higher than that of the sync signal. A divider frequency-divides the frequency signal by a given factor to produce a number of sampling pulses corresponding to the given number of image bit data. A data sampling circuit receives the data signal for sampling therefrom the number of image bit data in response to the corresponding sampling pulses to thereby write the image bit data into memory. Another divider frequency-divides the frequency signal by the product of the given factor and the given number to produce a feedback signal. A comparator compares the phase of the feedback and sync signals with each other to produce a control signal effective to maintain the VCO to continuously produce the frequency signal synchronized with the horizontal sync signal.

VLSI- A to D and D to A converters with multi-stage noise shaping modulators

Abstract: High resolution oversampling analog-to-digital and digital-to-analog converters are proposed. These converters utilize novel multi-stage noise shaping modulation techniques which can be implemented with fine pattern MOS technology. The modulators consist of multi-connected delta sigma modulation loops with single integration. The quantization noise in the first loop is re-quantized by the second loop. That is, the noise is canceled by adding the second loop output signals to the first loop output. The resolution of the modulators increases in proportion to the number of stages and there are no feedback-loop stability problems. An experimental prototype chip is fabricated using 1.5µm CMOS technology. This chip achieves a wide dynamic range of 88dB at a baseband of 3.4KHz and sampling rate of 2.048MHz. This dynamic range corresponds to a 14-bit equivalent resolution.

Method for compensating interference voltages in the electrode circuit in magnetic-inductive flow measurement

Abstract: In magnetic-inductive flow measurement with periodically reverse-poled DC magnetic field (with or without field-free phases) the signal voltage in each half period is sampled during a measuring signal sampling interval and the signal value obtained by the sampling is stored. For compensating an interference DC voltage superimposed on the measuring signal in a compensation interval following each measuring signal sampling interval within the same half period by sampling and storing the signal voltage a compensation voltage is generated which compensates the signal voltage within the compensation interval to the value zero. The compensation voltage is stored and superimposed on the signal voltage until the next compensation interval. During a correction sampling interval following each compensation interval within the same half period the signal voltage is again sampled and the signal value thereby obtained also stored. To recover a useful signal value firstly the difference is formed between the stored signal values obtained between two compensation intervals in different half periods and then the differences formed between two difference values obtained in this manner.

Bar code scanner for a video signal which has a shading waveform

Abstract: A bar code scanner has an image sensor for detecting a bar code and for generating an output signal corresponding to the detected bar code, a sample/hold circuit for sampling, holding, and smoothing an output signal from the image sensor, a slice level determination circuit for determining a slice level in accordance with an output signal from the sample/hold circuit, and a comparator for comparing the output signals from the sample/hold circuit and the slice level determination circuit and for generating a low or high output signal in accordance with a comparison result. The slice level determining circuit has a diode parallel circuit which is formed of a pair of diodes connected in parallel with each other in opposite directions and which has one end connected to the sample/hold circuit, a capacitor connected to the other end of the diode parallel circuit, a voltage transfer circuit for transferring the voltage charged in the capacitor to the comparator, and a discharging circuit for discharging the capacitor in response to the low output signal from the comparator.

Vibrating beam accelerometer with velocity change output

Abstract: A dual sensor, frequency output accelerometer that does not require either high sampling rates or mechanical matching of the sensors to achieve high levels of accuracy. In one embodiment, the accelerometer comprises a first sensor (12, 14, 16) that produces an output signal S1 having a frequency f1 related to acceleration along the sensitive axis, and a second sensor (18, 20, 22) that produces a second signal S2 having a frequency f2 related to acceleration along the sensitive axis, the sensors being arranged such that a given acceleration causes the frequency of one output signal to increase and the frequency of the other output signal to decrease. Velocity change DELTAV during time interval T is determined according to: DELTAV = A ADDELTA) + FT + BSIGMA) BD where A, F and B are constants, DELTA) is the difference between the phase changes of the output signals over time interval T, and SIGMA) is the sum of the phase changes of the output signals over time interval T. Higher order correction terms are also described for very high precision applications. Also disclosed is an accelerometer for measuring velocity change during a time interval that includes a subinterval during which electrical power is unavailable.

Optical communications transmitter and receiver

Abstract: An improved digital communications system is disclosed in which synchronization information is transmitted with the data. A start pulse, having a duration different than the other pulses in the transmitted digital data signal is used to mark the beginning of the frame of digital data. Preferably, a midpoint pulse is also transmitted with the start pulse to mark the midpoint of the frame to facilitate the generation of a local clock signal. Bit positions within the transmitted signal are sampled by sampling pulses which are generated by digital timers having time intervals keyed to the start and midpoint pulses.

Signal conditioner for electromagnetic flowmeter

Abstract: A voltage to frequency converter or conditioner adapted for use with electromagnetic flowmeters produces a signal whose output frequency is related to measured flow while compensating for power frequency interference and variations in magnet current or magnetic induction B. The driving signal is summed with a feedback signal and coupled to a sampling integrator which drives the detector. The output of the detector (which may be integrated) is provided as the input to a VCO for producing the desired output signal. The VCO output is also coupled into a feedback circuit to which is also coupled a reference voltage which is at least proportional to the magnetic induction B or magnet current I. In one embodiment of the invention, an up/down counter is up counted using the VCO's output as a clock for a fixed time and is then down counted by a clocking signal related to power main's frequency. The time required for the down count operation is used to meter the feedback voltage to the summing junction at the input of the integrating sampler. This metering of the feedback voltage may be eliminated by the use of a multiplying DAC which produces an analog voltage corresponding to the product of the reference voltage and the count attained in the up/down counter which is modified so as to eliminate the necessity for down counting and is rather reset at the beginning of each integration period.

Digital analyzing and synthesizing filter bank with maximum sampling rate reduction

Abstract: A digital analyzing and synthesizing filter bank in which a digital polyphase network in the analyzing filter bank provides sampling rate reduction by dividing an input signal into M complex sub-band signals, and the sub-band signals are subjected to a further staggered sampling rate reduction of the real and imaginary components thereof. This achieves maximum sampling reduction while permitting aliasing distortions to be compensated in the synthesizing filter bank.

Method and apparatus for detecting an edge position of a pattern and eliminating overlapping pattern signals

Abstract: Disclosed is a system for the signal processing of pattern profile information detected by optically or electronically scanning a pattern on an object such as a semiconductor wafer or a mask. The system includes a pattern information detector responsive to the scanning to generate an electric detection signal of a time series corresponding to the pattern along the scanning direction, position detecting means for generating a position signal discriminative for example of the direction of scanning each time the scanning advances a predetermined unit amount, and sampling extract means responsive to the position signals to successively sample the detection signal and obtain digital data of the pattern excluding any overlapping data of the pattern sampled at the same positions whereby even if the scanning involves minute oscillations of the scanning speed, the resulting noise in the detection signal is eliminated and an accurate pattern information signal corresponding to the absolute positions is extracted.

Clutter removal by polynomial compensation

Abstract: Apparatus and method for removing unwanted components of a signal received from a radar transmitter. The unwanted components lie in a clutter Doppler frequency band. Such components are removed by first sampling the received signal at a rate sufficient to cause components of the received signal in the clutter Doppler frequency band to be validly sampled and components of the received signal in a desired target Doppler frequency band to be invalidly sampled. A signal equivalent to the values of a polynomial fit to this sampled signal then provides an estimate of the unwanted clutter components. This estimate is subtracted from the original received signal to provide a signal with the unwanted clutter components removed. A significant improvement in receiver sensitivity as well as a reduction in dynamic range requirements of a post detection digital signal processor is achieved through the use of this invention.

Electronic musical instrument with digital filter

Abstract: A pitch synchronizing signal snychronized with the pitch of a digital tone signal to be filtered is generated by a pitch synchronizing signal generation circuit. A digital filter circuit which receives the digital tone signal and imparts it with a desired tone color by subjecting it to a proper filter operation executes this filter operation with a sampling period synchronized with the pitch synchronizing signal. A moving formant thereby is realized which is suitable for control of a tone. A pitch synchronization output circuit for sampling and outputting the output of the digital filter circuit in accordance with the pitch synchronizing signal may be provided and this will prevent occurrence of a sampling noise. A switching circuit is provided for enabling switching order of the digital filter circuit between an even number and an odd number. By this switching of order, a desired filter characteristic can be realized with high fidelity. The filter coefficient may be used commonly for two orders positioned at symmetrical positions. Filter coefficients which do not undergo timewise change and those which undergo timewise change may be selectively supplied in separate channels.

Tone signal generation device

Abstract: A note clock generation circuit (15) generates note clock pulses in correspondence to a note name of tone to be generated. An octave rate data generation circuit (14) generates rate data in correspondence to the octave range to which the tone to be generated belongs. By performing addition or subtraction of the rate data at the timing of generation of the note clock pulses, an address signal is generated. A tone generator (17) generates a tone : waveshape in the form of amplitude sampled values in response to an integer section of this address signal. An interpolation circuit (18) performs interpolation between adjacent amplitude sampled values thus generated in response to a decimal section of the address signal. The rate of change of the decimal section of the address signal is changed in accordance with the tone range so that a finer interpolation is made as the tone range becomes lower. By this arrangement, decrease in an effective sampling fequency in the lower tone range can be prevented. The interpolation circuit (18) receives the amplitude sampled values and the interpolation parameter (i.e., the decimal section of the address signal) respectively in a pitch synchronized state and performs the interpolation operation in the pitch synchronized state. The timing of the interpolation operation thereby is synchronized with the pitch of the generated tone so that inharmonic noise components are substantially removed.

Sampling effects in CdHgTe focal plane arrays

Abstract: Focal plane arrays (FPA) of CdHgTe detectors have been incorporated successfully into infrared imagers with sensitivities in either the 3-5 µm or 8-14 µm bands. Whilst it has been possible to obtain good imagery with minimal fixed pattern noise using digital non-uniformity correction electronics, these imagers suffer from problems caused by the sampling of the scene by the FPA; in particular, aliasing results in the obscuration of high frequency detail in the scene and its appearance at lower frequencies below the Nyquist frequency. It is demonstrated how microscanning may be used to reduce these sampling effects and in addition, ways of improving the ease of viewing of the final image by reducing the degree of pixelation are discussed.