Showing papers on "Sampling (signal processing) published in 1975"

Method and apparatus for error reduction in digital information transmission systems

Abstract: A method and apparatus for reducing idle channel noise, cross talk and quantizing error noise in a digital information transmission system designed for operation over a predetermined frequency spectrum and having an analog-to-digital converter, an information transmission medium, and a digital-to-analog converter all located between a transmitting station and a receiving station. A controlled noise signal having an amplitude lying in the range from about 1/4 to 1/2 of the magnitude of a quantizing interval of the ADC and DAC and a frequency content concentrated at (n+1/2) times the ADC and DAC sampling frequency but outside the system frequency spectrum is added to the analog information input signals prior to conversion to digital form. The injected controlled noise signal predominates whenever the amplitude of spurious noise signals is less than the amplitude of the controlled noise signal and the level of the input to the ADC is near a quantizing interval transition point so that the subsequent analog output signals from the DAC have frequencies concentrated outside the frequency spectrum of the system. The DAC output signals are subsequently filtered by a post sampling filter having a pass band coextensive with the system frequency spectrum so that the controlled noise signal components are removed before coupling to the receiving station.

The tokyo surface ship gravity meter: recent developments and results of comparison measurements

Abstract: The Tokyo Surface Ship Gravity Meter (TSSG) employs a single vibrating string accelerometer which is inherently nonlinear. This leads to errors unless its frequency is essentially constant over the sampling intervals. A new processor incorporating pulse train logic and employing a rapid sampling rate of about 50/sec has been designed to replace the digital computer which used about 2/sec.

Receiving pulse leading edge position detecting method for loran recei ver

Abstract: PURPOSE:A method for detecting the leading edge of the receiving pulse easily and surely by shifting the sampling pulses so that the difference of the outputs obtained through the phase-detection becomes the maximum, the minimum or zero.

Digital apparatus for waveform measurement

Abstract: Digital instrumentation for waveform measurement including method and apparatus for providing irregularly spaced sampling times to prevent misleading visual patterns of coordinate points due to the presence of signal frequencies above the Nyquist frequency. The method and apparatus includes means for plotting coordinate points having abscissa positions corrected according to actual irregular sampling times, method and apparatus whereby irregularities are repeated for each signal sweep, and method and apparatus utilizing a bit reversal system to achieve the irregularly spaced sampling times.

Digital pulse compression via fast convolution

Abstract: The mathematical structure of the digital ambiguity function for a matched filtered linear FM (LFM) waveform is derived as a function of time-bandwidth product, sampling rate, and arbitrary delay and frequency shifts. It is found to be well behaved for sampling rates equal to or greater than the swept signal bandwidth, provided that time sidelobes are controlled using standard frequency domain weighting techniques. A digital convolution processor comprised of cascaded pipeline fast Fourier transforms (FFT's) is presented as a viable architecture for real-time filtering of moderately high bandwidth LFM signals, and tradeoffs among radix, pipeline clock rate, and convolutional efficiency are discussed. It is found that a modified floating-point computational scheme performs well in such a context and is especially useful if a large signal dynamic range must be accommodated. A radix-4 4096-point design example is considered and the effects of quantization and finite register length arithmetic upon the digital ambiguity function are demonstrated via simulation. It is found that input data, FFT coefficients, reference filter coefficients, and intermediate results can be represented with mantissas of modest bit length.

Frequency spectrum analyzer

Abstract: A system is disclosed for analyzing in real time the frequency spectrum of an incoming signal of unknown frequency characteristics. This is accomplished by first generating a train of sampling pulses in repetitive frames, where each frame of pulses represents a series of cross-correlation functions in terms of a predetermined number of sampling intervals and a predetermined number of harmonics. These sampling pulses are then multiplied with the incoming signal to generate a train of discrete sampled outputs. The discrete sampled outputs from the multiplying means are accumulated in an accumulator and a frequency domain output signal is produced for each frame of sampling pulses.

Multiplex speech transmission system with speech analysis-synthesis

Abstract: A plurality of speech channels uses only one speech analyzer-synthesizer by Time-Multiplexing-Demultiplexing (sampling and processing) the speech channels sequentially. On the transmission side, speech signals of a plurality of channels are multiplexed by a pulse code modulation system, a partial autocorrelation coefficient and an excitation signal which constitute a feature parameter of the speech are extracted from the multiplexed signals by means of a speech analyzer for respective digital outputs corresponding to respective speech signals, and the extracted feature parameter is multiplexed again and then transmitted to the receiving side. On the receiving side, the received multiplexed signal of the feature parameter is applied to a speech synthesizer on a time division basis for reproducing a multiplex pulse code modulation signal of the speech wave, and the reproduced signal is distributed among respective channels.

Laser Doppler velocimeter signal processing using sampling spectrum analysis

Abstract: A new technique for processing laser Doppler velocimeter signals by sampling spectrum analysis is presented. A swept‐filter spectrum analyzer and pulse‐shaping and sample‐and‐hold circuits are employed. Sampling turbulence frequencies up to 4 kHz. In contrast to most tracking techniques, high dropout of signal cases and high frequency signals normally associated with directionally sensitive velocimeter systems can be processed.

Amplitude sorting of oscillatory burst signals by sampling

Abstract: A method and apparatus for amplitude sorting of oscillatory burst signals is described in which the burst signal is detected to produce a burst envelope signal and an intermediate or midportion of such envelope signal is sampled to provide a sample pulse output. The height of the sample pulse is proportional to the amplitude of the envelope signal and to the maximum burst signal amplitude. The sample pulses are fed to a pulse height analyzer for sorting. The present invention is used in an acoustic emission testing system to convert the amplitude of the acoustic emission burst signals into sample pulse heights which are measured by a pulse height analyzer for sorting the pulses in groups according to their height in order to identify the material anomalies in the test material which emit the acoustic signals.

Adaptive clutter velocity cancellation system for pulsed digital MTI system

Abstract: A digital MTI radar system for improved handling of "moving clutter" which employs a double phase detector scheme to provide both Doppler sine and Doppler cosine terms which are separately digitally encoded in a series of range increments throughout each pulse repetition interval. The discrete digitally encoded values thus represent instantaneous echo signal phase angle in each corresponding range increment. This angle data is compared by range increments with that of the last previous pulse repetition period, to provide net signal phase angle change in digital form (a velocity related term) by range increments successively. The remaining circuitry comprises a device for sampling this net phase angle change over a predetermined number of range increments to compute average clutter velocity with respect to the radar system location. A bona fide signal in the moving clutter may then be recognized on the basis of its exceeding the average clutter velocity. A quantized output is effected by means of a logic arrangement which detects the condition of signal presence in one or two adjacent ones of the range increments provided these increments are preceded and followed by a predetermined relatively small number of range increments exhibiting no signal above the said average clutter velocity.

A low-distortion oscillator with fast amplitude stabilization

Abstract: An improved technique is proposed for stabilizing the output of a variable-frequency RC sine-wave oscillator. Very low harmonic distortion figures and fast amplitude settling are achieved together by sampling the peaks of the output waveform to produce a d.c. voltage having virtually no ripple content. A simplified analysis of the stabilization loop is included, and the effects of non-ideal sampling on distortion are considered.

Modulated passband signal generator

Abstract: Disclosed is apparatus for generating carrier signals modulated by baseband symbol signals. The disclosed apparatus includes samplers for sampling input symbol signals, multipliers for multiplying the sampled signals by a precessing phase that is a function of the carrier signal's frequency, and modified filters for convolving the multiplied signals with a complex (real and imaginary) low pass impulse response and for selecting the real part of the convolved signals.

Quadrature sampling error formula

Abstract: It is known that the Grace−Pitt interpolation formula [O. D. Grace and S. P. tt, ’’Quadrature Sampling of High Frequency Waveforms,’’ J. Acoust. Soc. Am. 44, 1453−1454 (L) (1968)] allows an exact reconstruction of a high−frequency bandlimited waveform when the sampling rate is an integer submultiple of the center frequency. The problem addressed here is the determination of the ratio of the power in the error bands to the power in the signal band when the formula is used for any sampling frequency.Subject Classification: 60.XX.

Discrete control correction for synchronizing digital networks

Abstract: In a digital communications system having widely dispersed nodes transmitg between each other, a method and apparatus for synchronizing the processing of data at a node. Data transmitted between nodes is stored in speed buffers. The output of each buffer at a node is connected to the nodal processor and is controlled by the nodal clock. Synchronization is accomplished by performing a periodic adjustment of the frequency of the nodal clock. This adjustment is arrived at by sampling the buffer content of each node and extracting an error signal which represents the buffer position relative to the half full buffer position. The error signal is suitably weighted by a control and summed with all other weighted buffer error signals to produce a total error signal, which is added linearly to the original initial condition of the nodal clock and converted to an equivalent frequency.

A.m. signal generator having an r.f. output level control

Abstract: An A.M. signal generator having an r.f. level control arrangement includes a modulator whose output gain is controllable, a detector for detecting the A.M. wave produced by the modulator and means for sampling and storing the output of the detector at times occurring when the modulating wave passes through a predetermined amplitude level. The sampled and stored signal is then compared with a D.C. voltage and the resultant of the comparison is applied back to the modulator in opposite fashion to variation in the level of the A.M. wave.

Apparatus for use in an induced polarization system for measuring the deformation of a periodic signal concealed in noise

Abstract: An apparatus for measuring the deformation parameters of a periodic signal concealed in noise. The signal is transmitted by a transmitter controlled by a stable clock and received by a receiver also controlled by a stable clock of the same frequency as the one of the transmitter. The receiver comprises a detector for detecting the signal transmitted by the transmitter, means for resetting to a predetermined value the voltage detected by the detector at each integral number of cycles but at any particular time in the cycle, and an analog to digital converter connected to the detector for sampling the signal detected by the detector a predetermined number of times per cycle for converting the analog signal into a digital signal. A digit accumulator is connected to the output of the converter for accumulating the digital signals detected at corresponding sampling times in each cycle and a calculator is connected to the accumulator for determining the phase of the digital signal detected as compared to the signal transmitted by the transmitter and for evaluating therefrom the deformation parameters of the received signal.

Phase detection in phase lock loop circuit

Abstract: Phase detection in a phase lock loop circuit is performed by periodically sampling the A C carrier communication signal applied thereto in response to a sample timing signal having a periodic frequency twice that of the carrier and a polarity check signal which is indicative of the polarity of the communication signal at a predetermined interval prior to each sample timing signal, one polarity enabling a sample to be taken and the opposite polarity inhibiting it

Digital remote control system with signal verification

Abstract: A wireless remote control system has a transducer coupled to a sampling first counter for counting the frequency of incoming signals A portion of the count distinguishes among various control signal frequencies and is supplied to a memory and a comparator A second counter counts the number of identical comparisons of present and immediately preceding measurements, and is reset when a non-identical comparison is detected The memory supplies the stored portion of the count to a decoder which is disabled by the second counter until a predetermined number of identical frequency comparisons have been determined, which provides noise immunity The decoder is coupled to utilization means to activate the control function corresponding to the frequency of the control signal

Processing apparatus and method for color video signals

Abstract: An information processing method and apparatus, which is adapted to use, for example, in the recording of color image information, calls for sampling a set of basic colors (e.g., green, red, and blue) at rates which are proportioned according to the resolving power of the human visual system respective of color, and arranging the resulting samples to produce a sequential video signal. In one implementation, green, red, and blue color image information is sampled at respective rates decreasing progressively from green to red to blue in producing a sequential signal which is recorded, and on playback a luminance signal for, say, a television display is derived from the sequential signal based only on the relatively high frequency (preferably both green and red but not blue) sample information. Moreover, to improve image registration with such customized sampling, delays are selectively interposed in the individual color signals in accordance with their respective sampling frequencies.

Frequency measurement using phase continuous frequency switching

Abstract: An automatic transfer oscillator system for frequency measurement using a single sampler is described. Two signals, of frequencies f 1 and f 2 , are alternately used to drive the sampler. The two frequencies are sufficiently close by design so that the intermediate frequency signals generated arise from the same harmonic number when sampling a signal of unknown frequency f x . The resulting average IF output is accurately measured and used to compute the unknown frequency f x . Accurate measurement of the average IF output is made possible because the transition between f 1 driving the sampler and f 2 driving the sampler only occurs when the two signals are in phase. As a result, the IF signal transitions are also in phase and spurious signals are not produced. This technique is referred to as phase continuous switching.

Data acquisition in tomography

Abstract: Apparatus is so constructed as to employ a sampling beam of radiation of a form which at least approximates to a sinusoidal distribution of intensity between the extremities of its skirts, and is such that the range of sinusoidal variation has an extent of approximately four sampling intervals, whereby the sampling beams have themselves the effect of band limiting the required data to a spatial frequency of approximately half the sampling frequency.

Means and method for recording seismic signals

Abstract: An amplifying channel provides amplified signals corresponding to a received wide dynamic range analog input signal such as provided by a geophone group. A plurality of level comparators which number one less than the number of amplified signals compare the amplified signals with reference signals corresponding to a predetermined range of a desired output signal. The comparators provide outputs corresponding to the comparison to logic circuitry which develop sampling signals. The amplified signals from the amplified channel are also provided to a plurality of switches which are controlled by sampling pulses provided by the logic circuitry to sample an amplified signal within the range defined by the reference signals and provide it as the analog output signal. The analog output signal is converted to digital signals which are provided with other digital signals corresponding to the gain relationship between the analog output signal and the received analog input signal.

Musical-tone-waveform forming apparatus for an electronic musical instrument

Abstract: An apparatus for forming a musical-tone waveform for an electronic musical instrument characterized in that at least one cycle of a musical-tone waveform, which it is desired be produced, is divided in amplitude by dividing lines at equal intervals and sampling points on a time axis are determined from respective crossing points between the waveform and the dividing lines and the distance between the initial and the final sampling points is represented by an appropriate number of pulses so that each sampling point may be represented by a pulse number. There is further provided a memory circuit wherein each sampling point, that is, the pulse number thereof is set up in the form of a digital signal. Additionally, there is a set up in the form of a digital signal a tendency such as increase, decrease or equal at each sampling point with reference to the preceding sampling point. An envelope setting device is used wherein the envelope of a muscial-tone waveform, which it is desired be produced, is subjected to sampling and the analog amount of each sampling point is set up in the form of digital signal. An accumulatively adding device is employed whereby the output digital signal of the envelope setting device is accumulatively added to or subtracted from the output digital signal of the memory circuit. The accumulatively adding device is connected at its output terminal to a D-A converter for converting its output signal into an analog signal.

Multichannel digital modulator

Abstract: In a modulator of the type including a memory in which coded binary words representative of samples of a cycle of a sinusoidal signal are stored, which coded words are used to synthesize the modulated signal, the bits of each input digital signal S i are grouped in groups of P i bits (P i ≧1) and the sampling of the groups is so performed that several samples of each group appear in the multiplexed signal resulting from the sampling. Each sample of each group causes the reading of a coded word. This process gives a wide range of possibilities as concerns the transmission speeds of the input channels and the choice of the carrier frequency and of the number of modulation levels for each modulated signal.

Apparatus for sensing, transmitting and displaying signal states

Abstract: Apparatus for sensing, communicating and displaying signal states in a multiplicity of functional units is described. A signal multiplexer is assigned to each given functional unit; the multiplexer inputs are connected to signal lines carrying the signal states of the functional unit. The outputs of the signal multiplexers are connected by time division multiplex (TDM) trunks to a group multiplexer. A multiplexer control applies first encoded selection signals to the signal multiplexers for selection of signal lines at the signal multiplexers. The selected signaling lines are interrogated, and the signal states are transmitted timewise in parallel with the signal states of other functional units over the corresponding TDM trunks to the group multiplexer. The multiplexer control applies second encoded selection signals to the group multiplexer for connecting the TDM trunks to group TDM trunks throughout the duration of a sampling phase. At the display location, a shift register receives the signal states from the group TDM trunks. The clock pulse coupled to the shift register is interrupted upon acceptance of signal states sent from a functional unit for the duration of a display phase.

Analog-to-digital converter with electro-optical coding

Abstract: An EMP resistant A/D converter having no triggered circuits, the digital code word being formed by sampling the analog function on a periodic basis, selectively quantizing the levels sampled into channels providing the digits of the output code through a discrete fiber optic delay medium for each digit in the output. Electronic to light conversion and vice versa are included.

Direction determination in sonar systems

Abstract: In a procedure for determining the direction of echo signal pulses in a medium by means of a close range sonar system having at least one linear array of electroacoustic transducers and presenting a wide aperture angle and high azimuth and radial resolution, accurate determinations are made at a reduced component cost by: dividing the total aperture angle into a plurality of partial regions each having a respective median direction and an aperture angle ##EQU1## where c is the speed of sound in the medium, D is the length of the transducer array, and B is the bandwidth of the emitted signal pulses; subjecting the received signals associated with each partial region to a delay time compensation corresponding to its associated median direction; scanning the signals received by successive units of the transducers of said array at a selected frequency f s '; intermediately storing the scanned signals; sampling successive ones of the intermediately stored signals at a sampling frequency f s to form a serial signal; and phase compensating the resulting serial signal to produce a direction indication.

Reduction of aliasing distortion in sampled signals

Abstract: A method and apparatus for sampling continuous wave signals to reduce aliasing distortion. The signal is sampled at n times the Nyquist rate, where n is an integer greater than 1; and the n most recently produced sequential samples are combined at the Nyquist rate to produce a sequence of composite samples. This sequence of composite samples exhibits reduced aliasing distortion compared with a conventional sampling or sample-and-hold gate.